|
[-]
[+]
|
Changed |
live.spec
|
|
|
|
[-]
[+]
|
Changed |
live-CHANGELOG
^
|
| @@ -1,3 +1,43 @@
+2008.12.20:
+- Removed - from "MultiFramedRTPSource.cpp" - the historical sanity check for bizarrely delayed packets.
+ Accepting such bizarro packets causes problems in the code; it's best not to accept them at all.
+- Added to "FramedFilter" a new public function "detachInputSource()" that sets the filter's input source
+ to NULL. This is useful if you want to close the filter independently of closing its input source.
+ (Thanks to Alex Cichowski for this suggestion.)
+- Modified the header file for "MultiFramedRTPSink" to make its implementation of the "stopPlaying()" virtual
+ function public. (Thanks to Guy Bonneau for this suggestion.)
+- Changed "MediaSession::initiate()" to make the port number selection mechanism (for unicast streams) more robust.
+ (Thanks to Pierre Ynard for this suggestion.0
+
+2008.11.13:
+- Fixed an obscure bug in "BasicTaskScheduler::SingleStep()" in which the call to "handleAlarm()" would
+ (on rare occasions) modify the set of read handlers in such a way as to be inconsistent with the result of
+ the earlier call to "select()". We fixed the bug by moving the call to "handleAlarm()" after the socket
+ handler call. (Thanks to Sigismondo Boschi for tracking down this bug.)
+- Fixed a bug in "parseRTSPRequestString()" that could cause a crash when parsing certain weird, non-compliant
+ requests. (Thanks to Sebastien Escudier for reporting this.)
+
+2008.11.04:
+- Fixed a bug in "RTPInterface" that was sometimes causing sockets not to be reclaimed properly by a
+ RTSP server when streaming RTP/RTCP-over-TCP. (Thanks to Ralf Globisch for tracking this down.)
+- Changed the RTSP 'start time' parameter from "float" to "double", to allow for
+ accurate 'start times' that are very large - e.g., Unix 'epoch' times.
+ (Thanks to Morgan Toervolt for the suggestion and patch.)
+- Made some simplifications to the 'reception stats' code in "RTPSource".
+ (Thanks to Guy Bonneau for this suggestion.)
+- Fixed a minor bug in "QuickTimeFileSink". (Thanks to Pramod Bhagwat.)
+
+2008.10.07:
+- Added an optional "timeout" parameter (in seconds) to the 'connecting' RTSP
+ client commands. (Thanks to Sebastien Escudier for this patch.)
+- Increased the size of a "DummySink"s buffer in "MPEG1or2FileServerDemux", to make
+ it large enough for any PES packet.
+- Fixed a minor memory leak in the "createNew()" functions for
+ "AVIFileSink" and "QuickTimeFileSink" if the output fid did not get created.
+ (Thanks to Mehmet Ozgul for noting this.)
+- Fixed a couple of spelling erros, removed some unnecessary whitespace,
+ and updated the address of the FSF in the LGPL preamble (thanks to Erik Hovland).
+
2008.09.02:
- Fixed a minor bug in "RTSPServer". (Thanks to Renato Mauro.)
- Minor improvements fo "FileSink" and "MP3FileSource". (Thanks to Pramod Bhagwat.)
|
|
[-]
[+]
|
Deleted |
live.2007.11.01.tar.bz2/liveMedia/foo.patch
^
|
| @@ -1,33 +0,0 @@
-diff -U7 -p -r1.1 OnDemandServerMediaSubsession.cpp
---- OnDemandServerMediaSubsession.cpp 29 May 2007 11:35:10 -0000 1.1
-+++ OnDemandServerMediaSubsession.cpp 7 Jun 2007 10:17:31 -0000
-@@ -295,24 +295,25 @@ void OnDemandServerMediaSubsession::setS
- if (streamState != NULL && streamState->mediaSource() != NULL) {
- setStreamSourceScale(streamState->mediaSource(), scale);
- }
- }
-
- void OnDemandServerMediaSubsession::deleteStream(unsigned clientSessionId,
- void*& streamToken) {
-+ StreamState* streamState = (StreamState*)streamToken;
-+
- // Look up (and remove) the destinations for this client session:
- Destinations* destinations
- = (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));
- if (destinations != NULL) {
- fDestinationsHashTable->Remove((char const*)clientSessionId);
-- }
-
-- // Stop streaming to these destinations:
-- StreamState* streamState = (StreamState*)streamToken;
-- if (streamState != NULL) streamState->endPlaying(destinations);
-+ // Stop streaming to these destinations:
-+ if (streamState != NULL) streamState->endPlaying(destinations);
-+ }
-
- // Delete the "StreamState" structure if it's no longer being used:
- if (streamState != NULL && streamState->referenceCount() >= 0) {
- if (streamState->referenceCount() > 0) --streamState->referenceCount();
- if (streamState->referenceCount() == 0) {
- delete streamState;
- if (fLastStreamToken == streamToken) fLastStreamToken = NULL;
|
|
|
Deleted |
live.2007.12.27.tar.bz2
^
|
|
|
Deleted |
live.2008.01.18.tar.bz2
^
|
|
|
Deleted |
live.2008.07.25.tar.bz2
^
|
|
|
Deleted |
live.2008.09.02.tar.bz2
^
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/BasicHashTable.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Basic Hash Table implementation
// Implementation
@@ -108,7 +108,7 @@
BasicHashTable::TableEntry* entry = fNextEntry;
fNextEntry = entry->fNext;
-
+
key = entry->key;
return entry->value;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/BasicTaskScheduler.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Basic Usage Environment: for a simple, non-scripted, console application
// Implementation
@@ -47,7 +47,7 @@
void BasicTaskScheduler::SingleStep(unsigned maxDelayTime) {
fd_set readSet = fReadSet; // make a copy for this select() call
-
+
DelayInterval const& timeToDelay = fDelayQueue.timeToNextAlarm();
struct timeval tv_timeToDelay;
tv_timeToDelay.tv_sec = timeToDelay.seconds();
@@ -66,7 +66,7 @@
tv_timeToDelay.tv_sec = maxDelayTime/MILLION;
tv_timeToDelay.tv_usec = maxDelayTime%MILLION;
}
-
+
int selectResult = select(fMaxNumSockets, &readSet, NULL, NULL,
&tv_timeToDelay);
if (selectResult < 0) {
@@ -91,10 +91,7 @@
exit(0);
}
}
-
- // Handle any delayed event that may have come due:
- fDelayQueue.handleAlarm();
-
+
// Call the handler function for one readable socket:
HandlerIterator iter(*fReadHandlers);
HandlerDescriptor* handler;
@@ -137,6 +134,10 @@
}
if (handler == NULL) fLastHandledSocketNum = -1;//because we didn't call a handler
}
+
+ // Also handle any delayed event that may have come due. (Note that we do this *after* calling a socket
+ // handler, in case the delayed event handler modifies the set of readable socket.)
+ fDelayQueue.handleAlarm();
}
void BasicTaskScheduler::turnOnBackgroundReadHandling(int socketNum,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/BasicTaskScheduler0.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Basic Usage Environment: for a simple, non-scripted, console application
// Implementation
@@ -28,13 +28,13 @@
AlarmHandler(TaskFunc* proc, void* clientData, DelayInterval timeToDelay)
: DelayQueueEntry(timeToDelay), fProc(proc), fClientData(clientData) {
}
-
+
private: // redefined virtual functions
virtual void handleTimeout() {
(*fProc)(fClientData);
DelayQueueEntry::handleTimeout();
}
-
+
private:
TaskFunc* fProc;
void* fClientData;
@@ -60,7 +60,7 @@
AlarmHandler* alarmHandler = new AlarmHandler(proc, clientData, timeToDelay);
fDelayQueue.addEntry(alarmHandler);
- return (void*)(alarmHandler->token());
+ return (void*)(alarmHandler->token());
}
void BasicTaskScheduler0::unscheduleDelayedTask(TaskToken& prevTask) {
@@ -124,7 +124,7 @@
handler = new HandlerDescriptor(fHandlers.fNextHandler);
handler->socketNum = socketNum;
}
-
+
handler->handlerProc = handlerProc;
handler->clientData = clientData;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/BasicUsageEnvironment.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Basic Usage Environment: for a simple, non-scripted, console application
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/BasicUsageEnvironment0.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Basic Usage Environment: for a simple, non-scripted, console application
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/DelayQueue.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// Help by Carlo Bonamico to get working for Windows
// Delay queue
// Implementation
@@ -52,7 +52,7 @@
DelayInterval operator-(const Timeval& arg1, const Timeval& arg2) {
time_base_seconds secs = arg1.seconds() - arg2.seconds();
time_base_seconds usecs = arg1.useconds() - arg2.useconds();
-
+
if (usecs < 0) {
usecs += MILLION;
--secs;
@@ -69,11 +69,11 @@
DelayInterval operator*(short arg1, const DelayInterval& arg2) {
time_base_seconds result_seconds = arg1*arg2.seconds();
time_base_seconds result_useconds = arg1*arg2.useconds();
-
+
time_base_seconds carry = result_useconds/MILLION;
result_useconds -= carry*MILLION;
result_seconds += carry;
-
+
return DelayInterval(result_seconds, result_useconds);
}
@@ -123,9 +123,9 @@
newEntry->fDeltaTimeRemaining -= cur->fDeltaTimeRemaining;
cur = cur->fNext;
}
-
+
cur->fDeltaTimeRemaining -= newEntry->fDeltaTimeRemaining;
-
+
// Add "newEntry" to the queue, just before "cur":
newEntry->fNext = cur;
newEntry->fPrev = cur->fPrev;
@@ -134,7 +134,7 @@
void DelayQueue::updateEntry(DelayQueueEntry* entry, DelayInterval newDelay) {
if (entry == NULL) return;
-
+
removeEntry(entry);
entry->fDeltaTimeRemaining = newDelay;
addEntry(entry);
@@ -147,7 +147,7 @@
void DelayQueue::removeEntry(DelayQueueEntry* entry) {
if (entry == NULL || entry->fNext == NULL) return;
-
+
entry->fNext->fDeltaTimeRemaining += entry->fDeltaTimeRemaining;
entry->fPrev->fNext = entry->fNext;
entry->fNext->fPrev = entry->fPrev;
@@ -175,7 +175,7 @@
// This event is due to be handled:
DelayQueueEntry* toRemove = head();
removeEntry(toRemove); // do this first, in case handler accesses queue
-
+
toRemove->handleTimeout();
}
}
@@ -195,7 +195,7 @@
EventTime timeNow = TimeNow();
DelayInterval timeSinceLastSync = timeNow - fLastSyncTime;
fLastSyncTime = timeNow;
-
+
// Then, adjust the delay queue for any entries whose time is up:
DelayQueueEntry* curEntry = head();
while (timeSinceLastSync >= curEntry->fDeltaTimeRemaining) {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/include/BasicHashTable.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Basic Hash Table implementation
// C++ header
@@ -25,7 +25,7 @@
#endif
// A simple hash table implementation, inspired by the hash table
-// implementation used in Tcl 7.6: <http://www.tcl.tk/>
+// implementation used in Tcl 7.6: <http://www.tcl.tk/>
#define SMALL_HASH_TABLE_SIZE 4
@@ -36,22 +36,22 @@
public:
BasicHashTable(int keyType);
virtual ~BasicHashTable();
-
+
// Used to iterate through the members of the table:
class Iterator; friend class Iterator; // to make Sun's C++ compiler happy
class Iterator: public HashTable::Iterator {
public:
Iterator(BasicHashTable& table);
-
+
private: // implementation of inherited pure virtual functions
void* next(char const*& key); // returns 0 if none
-
+
private:
BasicHashTable& fTable;
unsigned fNextIndex; // index of next bucket to be enumerated after this
TableEntry* fNextEntry; // next entry in the current bucket
};
-
+
private: // implementation of inherited pure virtual functions
virtual void* Add(char const* key, void* value);
// Returns the old value if different, otherwise 0
@@ -59,7 +59,7 @@
virtual void* Lookup(char const* key) const;
// Returns 0 if not found
virtual unsigned numEntries() const;
-
+
private:
class TableEntry {
public:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/include/BasicUsageEnvironment.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Basic Usage Environment: for a simple, non-scripted, console application
// C++ header
@@ -27,7 +27,7 @@
class BasicUsageEnvironment: public BasicUsageEnvironment0 {
public:
static BasicUsageEnvironment* createNew(TaskScheduler& taskScheduler);
-
+
// redefined virtual functions:
virtual int getErrno() const;
@@ -61,7 +61,7 @@
BackgroundHandlerProc* handlerProc,
void* clientData);
virtual void turnOffBackgroundReadHandling(int socketNum);
-
+
protected:
// To implement background reads:
int fMaxNumSockets;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/include/BasicUsageEnvironment0.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Basic Usage Environment: for a simple, non-scripted, console application
// C++ header
@@ -40,7 +40,7 @@
public:
// redefined virtual functions:
virtual MsgString getResultMsg() const;
-
+
virtual void setResultMsg(MsgString msg);
virtual void setResultMsg(MsgString msg1,
MsgString msg2);
@@ -48,18 +48,18 @@
MsgString msg2,
MsgString msg3);
virtual void setResultErrMsg(MsgString msg);
-
+
virtual void appendToResultMsg(MsgString msg);
-
+
virtual void reportBackgroundError();
-
+
protected:
BasicUsageEnvironment0(TaskScheduler& taskScheduler);
virtual ~BasicUsageEnvironment0();
private:
void reset();
-
+
char fResultMsgBuffer[RESULT_MSG_BUFFER_MAX];
unsigned fCurBufferSize;
unsigned fBufferMaxSize;
@@ -77,13 +77,13 @@
// "maxDelayTime" is in microseconds. It allows a subclass to impose a limit
// on how long "select()" can delay, in case it wants to also do polling.
// 0 (the default value) means: There's no maximum; just look at the delay queue
-
+
public:
// Redefined virtual functions:
virtual TaskToken scheduleDelayedTask(int64_t microseconds, TaskFunc* proc,
void* clientData);
virtual void unscheduleDelayedTask(TaskToken& prevTask);
-
+
virtual void doEventLoop(char* watchVariable);
protected:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/include/BasicUsageEnvironment_version.hh
^
|
| @@ -1,10 +1,10 @@
// Version information for the "BasicUsageEnvironment" library
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
#ifndef _BASICUSAGEENVIRONMENT_VERSION_HH
#define _BASICUSAGEENVIRONMENT_VERSION_HH
-#define BASICUSAGEENVIRONMENT_LIBRARY_VERSION_STRING "2007.10.31"
-#define BASICUSAGEENVIRONMENT_LIBRARY_VERSION_INT 1193788800
+#define BASICUSAGEENVIRONMENT_LIBRARY_VERSION_STRING "2008.12.19"
+#define BASICUSAGEENVIRONMENT_LIBRARY_VERSION_INT 1229644800
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/include/DelayQueue.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
- // Copyright (c) 1996-2000, Live Networks, Inc. All rights reserved
+ // Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// Delay queue
// C++ header
@@ -46,7 +46,7 @@
time_base_seconds useconds() {
return fTv.tv_usec;
}
-
+
int operator>=(Timeval const& arg2) const;
int operator<=(Timeval const& arg2) const {
return arg2 >= *this;
@@ -63,16 +63,16 @@
int operator!=(Timeval const& arg2) const {
return !(*this == arg2);
}
-
+
void operator+=(class DelayInterval const& arg2);
void operator-=(class DelayInterval const& arg2);
// returns ZERO iff arg2 >= arg1
-
+
protected:
Timeval(time_base_seconds seconds, time_base_seconds useconds) {
fTv.tv_sec = seconds; fTv.tv_usec = useconds;
}
-
+
private:
time_base_seconds& secs() {
return (time_base_seconds&)fTv.tv_sec;
@@ -80,7 +80,7 @@
time_base_seconds& usecs() {
return (time_base_seconds&)fTv.tv_usec;
}
-
+
struct timeval fTv;
};
@@ -145,9 +145,9 @@
protected: // abstract base class
DelayQueueEntry(DelayInterval delay);
-
+
virtual void handleTimeout();
-
+
private:
friend class DelayQueue;
DelayQueueEntry* fNext;
@@ -164,13 +164,13 @@
public:
DelayQueue();
virtual ~DelayQueue();
-
+
void addEntry(DelayQueueEntry* newEntry); // returns a token for the entry
void updateEntry(DelayQueueEntry* entry, DelayInterval newDelay);
void updateEntry(long tokenToFind, DelayInterval newDelay);
void removeEntry(DelayQueueEntry* entry); // but doesn't delete it
DelayQueueEntry* removeEntry(long tokenToFind); // but doesn't delete it
-
+
DelayInterval const& timeToNextAlarm();
void handleAlarm();
@@ -178,7 +178,7 @@
DelayQueueEntry* head() { return fNext; }
DelayQueueEntry* findEntryByToken(long token);
void synchronize(); // bring the 'time remaining' fields up-to-date
-
+
EventTime fLastSyncTime;
};
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/BasicUsageEnvironment/include/HandlerSet.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Basic Usage Environment: for a simple, non-scripted, console application
// C++ header
@@ -53,7 +53,7 @@
friend class HandlerIterator;
HandlerDescriptor fHandlers;
};
-
+
class HandlerIterator {
public:
HandlerIterator(HandlerSet& handlerSet);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/UsageEnvironment/HashTable.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2003 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Generic Hash Table
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/UsageEnvironment/UsageEnvironment.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Usage Environment
// Implementation
@@ -36,7 +36,7 @@
TaskScheduler::~TaskScheduler() {
}
-
+
void TaskScheduler::rescheduleDelayedTask(TaskToken& task,
int64_t microseconds, TaskFunc* proc,
void* clientData) {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/UsageEnvironment/include/Boolean.hh
^
|
| @@ -11,7 +11,7 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
#ifndef _BOOLEAN_HH
#define _BOOLEAN_HH
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/UsageEnvironment/include/HashTable.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2003 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Generic Hash Table
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/UsageEnvironment/include/UsageEnvironment.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Usage Environment
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/UsageEnvironment/include/UsageEnvironment_version.hh
^
|
| @@ -1,10 +1,10 @@
// Version information for the "UsageEnvironment" library
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
#ifndef _USAGEENVIRONMENT_VERSION_HH
#define _USAGEENVIRONMENT_VERSION_HH
-#define USAGEENVIRONMENT_LIBRARY_VERSION_STRING "2007.10.31"
-#define USAGEENVIRONMENT_LIBRARY_VERSION_INT 1193788800
+#define USAGEENVIRONMENT_LIBRARY_VERSION_STRING "2008.12.19"
+#define USAGEENVIRONMENT_LIBRARY_VERSION_INT 1229644800
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/UsageEnvironment/include/strDup.hh
^
|
| @@ -11,13 +11,13 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
#ifndef _STRDUP_HH
#define _STRDUP_HH
-// Copyright (c) 1996-2003 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A C++ equivalent to the standard C routine "strdup()".
// This generates a char* that can be deleted using "delete[]"
// Header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/UsageEnvironment/strDup.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2003 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A C++ equivalent to the standard C routine "strdup()".
// This generates a char* that can be deleted using "delete[]"
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/WindowsAudioInputDevice/WindowsAudioInputDevice_common.cpp
^
|
| @@ -11,7 +11,7 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// Copyright (c) 2001-2004 Live Networks, Inc. All rights reserved.
// Windows implementation of a generic audio input device
@@ -79,13 +79,13 @@
short* samplePtr = (short*)(curHdr->lpData);
unsigned numSamples = blockSize/2;
totNumSamples += numSamples;
-
+
while (numSamples-- > 0) {
short sample = *samplePtr++;
if (sample < 0) sample = -sample;
levelTotal += (unsigned short)sample;
}
-
+
if (curHdr == readTail) break;
curHdr = curHdr->lpNext;
}
@@ -109,7 +109,7 @@
handleClosure(this);
return;
}
-
+
// Try again after a short delay:
unsigned const uSecondsToDelay = fGranularityInMS*1000;
fTotalPollingDelay += uSecondsToDelay;
@@ -126,7 +126,7 @@
return;
}
fDurationInMicroseconds = 1000000/fSamplingFrequency;
-
+
// Call our own 'after getting' function. Because we sometimes get here
// after returning from a delay, we can call this directly, without risking
// infinite recursion
@@ -145,7 +145,7 @@
Boolean WindowsAudioInputDevice_common::openWavInPort(int index, unsigned numChannels, unsigned samplingFrequency, unsigned granularityInMS) {
uSecsPerByte = (8*1e6)/(_bitsPerSample*numChannels*samplingFrequency);
-
+
// Configure the port, based on the specified parameters:
WAVEFORMATEX wfx;
wfx.wFormatTag = WAVE_FORMAT_PCM;
@@ -155,16 +155,16 @@
wfx.nBlockAlign = (numChannels*_bitsPerSample)/8;
wfx.nAvgBytesPerSec = samplingFrequency*wfx.nBlockAlign;
wfx.cbSize = 0;
-
+
blockSize = (wfx.nAvgBytesPerSec*granularityInMS)/1000;
-
+
// Use a 10-second input buffer, to allow for CPU competition from video, etc.,
// and also for some audio cards that buffer as much as 5 seconds of audio.
unsigned const bufferSeconds = 10;
numBlocks = (bufferSeconds*1000)/granularityInMS;
-
+
if (!waveIn_open(index, wfx)) return False;
-
+
// Set this process's priority high. I'm not sure how much this is really needed,
// but the "rat" code does this:
SetPriorityClass(GetCurrentProcess(), HIGH_PRIORITY_CLASS);
@@ -173,23 +173,23 @@
Boolean WindowsAudioInputDevice_common::waveIn_open(unsigned uid, WAVEFORMATEX& wfx) {
if (shWaveIn != NULL) return True; // already open
-
+
do {
waveIn_reset();
if (waveInOpen(&shWaveIn, uid, &wfx,
(DWORD)waveInCallback, 0, CALLBACK_FUNCTION) != MMSYSERR_NOERROR) break;
-
+
// Allocate read buffers, and headers:
readData = new unsigned char[numBlocks*blockSize];
if (readData == NULL) break;
-
+
readHdrs = new WAVEHDR[numBlocks];
if (readHdrs == NULL) break;
readHead = readTail = NULL;
-
+
readTimes = new struct timeval[numBlocks];
if (readTimes == NULL) break;
-
+
// Initialize headers:
for (unsigned i = 0; i < numBlocks; ++i) {
readHdrs[i].lpData = (char*)&readData[i*blockSize];
@@ -198,44 +198,44 @@
if (waveInPrepareHeader(shWaveIn, &readHdrs[i], sizeof (WAVEHDR)) != MMSYSERR_NOERROR) break;
if (waveInAddBuffer(shWaveIn, &readHdrs[i], sizeof (WAVEHDR)) != MMSYSERR_NOERROR) break;
}
-
+
if (waveInStart(shWaveIn) != MMSYSERR_NOERROR) break;
-
+
hAudioReady = CreateEvent(NULL, TRUE, FALSE, "waveIn Audio Ready");
return True;
} while (0);
-
+
waveIn_reset();
return False;
}
void WindowsAudioInputDevice_common::waveIn_close() {
if (shWaveIn == NULL) return; // already closed
-
+
waveInStop(shWaveIn);
waveInReset(shWaveIn);
-
+
for (unsigned i = 0; i < numBlocks; ++i) {
if (readHdrs[i].dwFlags & WHDR_PREPARED) {
waveInUnprepareHeader(shWaveIn, &readHdrs[i], sizeof (WAVEHDR));
}
}
-
+
waveInClose(shWaveIn);
waveIn_reset();
}
void WindowsAudioInputDevice_common::waveIn_reset() {
shWaveIn = NULL;
-
+
delete[] readData; readData = NULL;
bytesUsedAtReadHead = 0;
-
+
delete[] readHdrs; readHdrs = NULL;
readHead = readTail = NULL;
-
+
delete[] readTimes; readTimes = NULL;
-
+
hAudioReady = NULL;
}
@@ -244,7 +244,7 @@
if (readHead != NULL) {
int hdrIndex = readHead - readHdrs;
creationTime = readTimes[hdrIndex];
-
+
// Adjust this time to allow for any data that's already been read from this buffer:
if (bytesUsedAtReadHead > 0) {
creationTime.tv_usec += (unsigned)(uSecsPerByte*bytesUsedAtReadHead);
@@ -252,7 +252,7 @@
creationTime.tv_usec %= 1000000;
}
}
-
+
// Then, read from each available buffer, until we have the data that we want:
unsigned numBytesRead = 0;
while (readHead != NULL && numBytesRead < numBytesWanted) {
@@ -265,31 +265,31 @@
releaseHeadBuffer();
}
}
-
+
return numBytesRead;
}
void WindowsAudioInputDevice_common::releaseHeadBuffer() {
WAVEHDR* toRelease = readHead;
if (readHead == NULL) return;
-
+
readHead = readHead->lpNext;
if (readHead == NULL) readTail = NULL;
-
- toRelease->lpNext = NULL;
- toRelease->dwBytesRecorded = 0;
+
+ toRelease->lpNext = NULL;
+ toRelease->dwBytesRecorded = 0;
toRelease->dwFlags &= ~WHDR_DONE;
- waveInAddBuffer(shWaveIn, toRelease, sizeof (WAVEHDR));
+ waveInAddBuffer(shWaveIn, toRelease, sizeof (WAVEHDR));
bytesUsedAtReadHead = 0;
}
void WindowsAudioInputDevice_common::waveInProc(WAVEHDR* hdr) {
unsigned hdrIndex = hdr - readHdrs;
-
+
// Record the time that the data arrived:
int dontCare;
gettimeofday(&readTimes[hdrIndex], &dontCare);
-
+
// Add the block to the tail of the queue:
hdr->lpNext = NULL;
if (readTail != NULL) {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/WindowsAudioInputDevice/WindowsAudioInputDevice_common.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2005 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Windows implementation of a generic audio input device
// Base class for both library versions:
// One that uses Windows' built-in software mixer; another that doesn't.
@@ -30,7 +30,7 @@
public:
static Boolean openWavInPort(int index, unsigned numChannels, unsigned samplingFrequency, unsigned granularityInMS);
static void waveIn_close();
- static void waveInProc(WAVEHDR* hdr); // Windows audio callback function
+ static void waveInProc(WAVEHDR* hdr); // Windows audio callback function
protected:
WindowsAudioInputDevice_common(UsageEnvironment& env, int inputPortNumber,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/WindowsAudioInputDevice/WindowsAudioInputDevice_mixer.cpp
^
|
| @@ -11,7 +11,7 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// Copyright (c) 2001-2004 Live Networks, Inc. All rights reserved.
// Windows implementation of a generic audio input device
@@ -33,13 +33,13 @@
public:
Mixer();
virtual ~Mixer();
-
+
void open(unsigned numChannels, unsigned samplingFrequency, unsigned granularityInMS);
void open(); // open with default parameters
void getPortsInfo();
Boolean enableInputPort(unsigned portIndex, char const*& errReason, MMRESULT& errCode);
void close();
-
+
unsigned index;
HMIXER hMixer; // valid when open
DWORD dwRecLineID; // valid when open
@@ -64,17 +64,17 @@
samplingFrequency, granularityInMS,
success);
if (!success) {delete newSource; newSource = NULL;}
-
+
return newSource;
}
AudioPortNames* AudioInputDevice::getPortNames() {
WindowsAudioInputDevice::initializeIfNecessary();
-
+
AudioPortNames* portNames = new AudioPortNames;
portNames->numPorts = WindowsAudioInputDevice::numInputPortsTotal;
portNames->portName = new char*[WindowsAudioInputDevice::numInputPortsTotal];
-
+
// If there's more than one mixer, print only the port name.
// If there's two or more mixers, also include the mixer name
// (to disambiguate port names that may be the same name in different mixers)
@@ -86,11 +86,11 @@
} else {
portNameFmt = "%s (%s)";
}
-
+
unsigned curPortNum = 0;
for (unsigned i = 0; i < WindowsAudioInputDevice::numMixers; ++i) {
Mixer& mixer = WindowsAudioInputDevice::ourMixers[i];
-
+
if (WindowsAudioInputDevice::numMixers <= 1) {
mixerNameBuffer[0] = '\0';
} else {
@@ -105,13 +105,13 @@
}
#endif
}
-
+
for (unsigned j = 0; j < mixer.numPorts; ++j) {
sprintf(portNameBuffer, portNameFmt, mixer.ports[j].name, mixerNameBuffer);
portNames->portName[curPortNum++] = strDup(portNameBuffer);
}
}
-
+
return portNames;
}
@@ -133,7 +133,7 @@
WindowsAudioInputDevice::~WindowsAudioInputDevice() {
if (fCurMixerId >= 0) ourMixers[fCurMixerId].close();
-
+
delete[] ourMixers; ourMixers = NULL;
numMixers = numInputPortsTotal = 0;
}
@@ -142,7 +142,7 @@
if (ourMixers != NULL) return; // we've already been initialized
numMixers = mixerGetNumDevs();
ourMixers = new Mixer[numMixers];
-
+
// Initialize each mixer:
numInputPortsTotal = 0;
for (unsigned i = 0; i < numMixers; ++i) {
@@ -153,9 +153,9 @@
// This device has a valid mixer. Get information about its ports:
mixer.getPortsInfo();
mixer.close();
-
+
if (mixer.numPorts == 0) continue;
-
+
numInputPortsTotal += mixer.numPorts;
} else {
mixer.ports = NULL;
@@ -166,12 +166,12 @@
Boolean WindowsAudioInputDevice::setInputPort(int portIndex) {
initializeIfNecessary();
-
+
if (portIndex < 0 || portIndex >= (int)numInputPortsTotal) { // bad index
envir().setResultMsg("Bad input port index\n");
return False;
}
-
+
// Find the mixer and port that corresponds to "portIndex":
int newMixerId, portWithinMixer, portIndexCount = 0;
for (newMixerId = 0; newMixerId < (int)numMixers; ++newMixerId) {
@@ -182,7 +182,7 @@
break;
}
}
-
+
// Check that this mixer is allowed:
if (allowedDeviceNames != NULL) {
int i;
@@ -198,8 +198,8 @@
return False;
}
}
-
- if (newMixerId != fCurMixerId) {
+
+ if (newMixerId != fCurMixerId) {
// The mixer has changed, so close the old one and open the new one:
if (fCurMixerId >= 0) ourMixers[fCurMixerId].close();
fCurMixerId = newMixerId;
@@ -243,17 +243,17 @@
do {
MIXERCAPS mc;
if (mixerGetDevCaps(index, &mc, sizeof mc) != MMSYSERR_NOERROR) break;
-
+
// Copy the mixer name:
strncpy(name, mc.szPname, MAXPNAMELEN);
-
+
// Find the correct line for this mixer:
unsigned i, uWavIn;
unsigned nWavIn = waveInGetNumDevs();
for (i = 0; i < nWavIn; ++i) {
WAVEINCAPS wic;
if (waveInGetDevCaps(i, &wic, sizeof wic) != MMSYSERR_NOERROR) continue;
-
+
MIXERLINE ml;
ml.cbStruct = sizeof ml;
ml.Target.dwType = MIXERLINE_TARGETTYPE_WAVEIN;
@@ -261,7 +261,7 @@
ml.Target.vDriverVersion = wic.vDriverVersion;
ml.Target.wMid = wic.wMid;
ml.Target.wPid = wic.wPid;
-
+
if (mixerGetLineInfo((HMIXEROBJ)index, &ml, MIXER_GETLINEINFOF_TARGETTYPE/*|MIXER_OBJECTF_MIXER*/) == MMSYSERR_NOERROR) {
// this is the right line
uWavIn = i;
@@ -270,23 +270,23 @@
}
}
if (i >= nWavIn) break; // error: we couldn't find the right line
-
+
if (mixerOpen(&newHMixer, index, (unsigned long)NULL, (unsigned long)NULL, MIXER_OBJECTF_MIXER) != MMSYSERR_NOERROR) break;
if (newHMixer == NULL) break;
-
+
// Sanity check: re-call "mixerGetDevCaps()" using the mixer device handle:
if (mixerGetDevCaps((UINT)newHMixer, &mc, sizeof mc) != MMSYSERR_NOERROR) break;
- if (mc.cDestinations < 1) break; // error: this mixer has no destinations
-
+ if (mc.cDestinations < 1) break; // error: this mixer has no destinations
+
if (!WindowsAudioInputDevice_common::openWavInPort(uWavIn, numChannels, samplingFrequency, granularityInMS)) break;
hMixer = newHMixer;
return;
} while (0);
-
+
// An error occurred:
- close();
-}
+ close();
+}
void Mixer::open() {
open(1, 8000, 20);
@@ -295,7 +295,7 @@
void Mixer::getPortsInfo() {
MIXERCAPS mc;
mixerGetDevCaps((UINT)hMixer, &mc, sizeof mc);
-
+
MIXERLINE mlt;
unsigned i;
for (i = 0; i < mc.cDestinations; ++i) {
@@ -306,7 +306,7 @@
if (mlt.dwLineID == dwRecLineID) break; // this is the destination we're interested in
}
ports = new AudioInputPort[mlt.cConnections];
-
+
numPorts = mlt.cConnections;
for (i = 0; i < numPorts; ++i) {
MIXERLINE mlc;
@@ -317,7 +317,7 @@
ports[i].dwComponentType = mlc.dwComponentType;
strncpy(ports[i].name, mlc.szName, MIXER_LONG_NAME_CHARS);
}
-
+
// Make the microphone the first port in the list:
for (i = 1; i < numPorts; ++i) {
#ifdef OLD_MICROPHONE_TESTING_CODE
@@ -336,7 +336,7 @@
Boolean Mixer::enableInputPort(unsigned portIndex, char const*& errReason, MMRESULT& errCode) {
errReason = NULL; // unless there's an error
AudioInputPort& port = ports[portIndex];
-
+
MIXERCONTROL mc;
mc.cMultipleItems = 1; // in case it doesn't get set below
MIXERLINECONTROLS mlc;
@@ -351,7 +351,7 @@
return False;
}
#endif
-
+
MIXERLINE ml;
memset(&ml, 0, sizeof (MIXERLINE));
ml.cbStruct = sizeof (MIXERLINE);
@@ -360,10 +360,10 @@
errReason = "mixerGetLineInfo()1";
return False;
}
-
+
char portname[MIXER_LONG_NAME_CHARS+1];
strncpy(portname, ml.szName, MIXER_LONG_NAME_CHARS);
-
+
memset(&ml, 0, sizeof (MIXERLINE));
ml.cbStruct = sizeof (MIXERLINE);
ml.dwLineID = dwRecLineID;
@@ -371,7 +371,7 @@
errReason = "mixerGetLineInfo()2";
return False;
}
-
+
// Get Mixer/MUX control information (need control id to set and get control details)
mlc.cbStruct = sizeof mlc;
mlc.dwLineID = ml.dwLineID;
@@ -385,7 +385,7 @@
mlc.dwControlType = MIXERCONTROL_CONTROLTYPE_MIXER; // Multiple Select
mixerGetLineControls((HMIXEROBJ)hMixer, &mlc, MIXER_GETLINECONTROLSF_ONEBYTYPE/*|MIXER_OBJECTF_HMIXER*/);
}
-
+
unsigned matchLine = 0;
if (mc.cMultipleItems > 1) {
// Before getting control, we need to know which line to grab.
@@ -394,10 +394,10 @@
mcd.cbStruct = sizeof mcd;
mcd.cChannels = ml.cChannels;
mcd.cMultipleItems = mc.cMultipleItems;
- MIXERCONTROLDETAILS_LISTTEXT* mcdlText = new MIXERCONTROLDETAILS_LISTTEXT[mc.cMultipleItems];
+ MIXERCONTROLDETAILS_LISTTEXT* mcdlText = new MIXERCONTROLDETAILS_LISTTEXT[mc.cMultipleItems];
mcd.cbDetails = sizeof (MIXERCONTROLDETAILS_LISTTEXT);
mcd.paDetails = mcdlText;
-
+
if (mc.dwControlID != 0xDEADBEEF) { // we know the control id for real
mcd.dwControlID = mc.dwControlID;
if ((errCode = mixerGetControlDetails((HMIXEROBJ)hMixer, &mcd, MIXER_GETCONTROLDETAILSF_LISTTEXT/*|MIXER_OBJECTF_HMIXER*/)) != MMSYSERR_NOERROR) {
@@ -417,7 +417,7 @@
return False;
}
}
-
+
for (unsigned i = 0; i < mcd.cMultipleItems; ++i) {
if (strcmp(mcdlText[i].szName, portname) == 0) {
matchLine = i;
@@ -426,34 +426,34 @@
}
delete[] mcdlText;
}
-
+
// Now get control itself:
MIXERCONTROLDETAILS mcd;
mcd.cbStruct = sizeof mcd;
mcd.dwControlID = mc.dwControlID;
mcd.cChannels = ml.cChannels;
mcd.cMultipleItems = mc.cMultipleItems;
- MIXERCONTROLDETAILS_BOOLEAN* mcdbState = new MIXERCONTROLDETAILS_BOOLEAN[mc.cMultipleItems];
+ MIXERCONTROLDETAILS_BOOLEAN* mcdbState = new MIXERCONTROLDETAILS_BOOLEAN[mc.cMultipleItems];
mcd.paDetails = mcdbState;
mcd.cbDetails = sizeof (MIXERCONTROLDETAILS_BOOLEAN);
-
+
if ((errCode = mixerGetControlDetails((HMIXEROBJ)hMixer, &mcd, MIXER_GETCONTROLDETAILSF_VALUE/*|MIXER_OBJECTF_HMIXER*/)) != MMSYSERR_NOERROR) {
delete[] mcdbState;
errReason = "mixerGetControlDetails()3";
return False;
}
-
+
for (unsigned j = 0; j < mcd.cMultipleItems; ++j) {
mcdbState[j].fValue = (j == matchLine);
}
-
+
if ((errCode = mixerSetControlDetails((HMIXEROBJ)hMixer, &mcd, MIXER_OBJECTF_HMIXER)) != MMSYSERR_NOERROR) {
delete[] mcdbState;
errReason = "mixerSetControlDetails()";
return False;
}
delete[] mcdbState;
-
+
return True;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/WindowsAudioInputDevice/WindowsAudioInputDevice_mixer.hh
^
|
| @@ -11,16 +11,16 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2005 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Windows implementation of a generic audio input device
// This version uses Windows' built-in software mixer.
// C++ header
//
// To use this, call "AudioInputDevice::createNew()".
// You can also call "AudioInputDevice::getPortNames()" to get a list
-// of port names.
+// of port names.
#ifndef _WINDOWS_AUDIO_INPUT_DEVICE_MIXER_HH
#define _WINDOWS_AUDIO_INPUT_DEVICE_MIXER_HH
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/WindowsAudioInputDevice/WindowsAudioInputDevice_noMixer.cpp
^
|
| @@ -11,7 +11,7 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// Copyright (c) 2001-2004 Live Networks, Inc. All rights reserved.
// Windows implementation of a generic audio input device
@@ -49,23 +49,23 @@
samplingFrequency, granularityInMS,
success);
if (!success) {delete newSource; newSource = NULL;}
-
+
return newSource;
}
AudioPortNames* AudioInputDevice::getPortNames() {
WindowsAudioInputDevice::initializeIfNecessary();
-
+
AudioPortNames* portNames = new AudioPortNames;
portNames->numPorts = WindowsAudioInputDevice::numAudioInputPorts;
portNames->portName = new char*[WindowsAudioInputDevice::numAudioInputPorts];
-
+
for (unsigned i = 0; i < WindowsAudioInputDevice::numAudioInputPorts; ++i) {
AudioInputPort& audioInputPort = WindowsAudioInputDevice::ourAudioInputPorts[i];
-
+
portNames->portName[i] = strDup(audioInputPort.name);
}
-
+
return portNames;
}
@@ -86,7 +86,7 @@
WindowsAudioInputDevice::~WindowsAudioInputDevice() {
if (fCurPortIndex >= 0) ourAudioInputPorts[fCurPortIndex].close();
-
+
delete[] ourAudioInputPorts; ourAudioInputPorts = NULL;
numAudioInputPorts = 0;
}
@@ -95,7 +95,7 @@
if (ourAudioInputPorts != NULL) return; // we've already been initialized
numAudioInputPorts = waveInGetNumDevs();
ourAudioInputPorts = new AudioInputPort[numAudioInputPorts];
-
+
// Initialize each audio input port
for (unsigned i = 0; i < numAudioInputPorts; ++i) {
AudioInputPort& port = ourAudioInputPorts[i];
@@ -107,7 +107,7 @@
Boolean WindowsAudioInputDevice::setInputPort(int portIndex) {
initializeIfNecessary();
-
+
if (portIndex < 0 || portIndex >= (int)numAudioInputPorts) { // bad index
envir().setResultMsg("Bad input port index\n");
return False;
@@ -128,8 +128,8 @@
return False;
}
}
-
- if (portIndex != fCurPortIndex) {
+
+ if (portIndex != fCurPortIndex) {
// The port has changed, so close the old one and open the new one:
if (fCurPortIndex >= 0) ourAudioInputPorts[fCurPortIndex].close();;
fCurPortIndex = portIndex;
@@ -160,10 +160,10 @@
return;
} while (0);
-
+
// An error occurred:
- close();
-}
+ close();
+}
void AudioInputPort::open() {
open(1, 8000, 20);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/WindowsAudioInputDevice/WindowsAudioInputDevice_noMixer.hh
^
|
| @@ -11,16 +11,16 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2005 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Windows implementation of a generic audio input device
// This version does not use Windows' built-in software mixer.
// C++ header
//
// To use this, call "AudioInputDevice::createNew()".
// You can also call "AudioInputDevice::getPortNames()" to get a list
-// of port names.
+// of port names.
#ifndef _WINDOWS_AUDIO_INPUT_DEVICE_NOMIXER_HH
#define _WINDOWS_AUDIO_INPUT_DEVICE_NOMIXER_HH
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/WindowsAudioInputDevice/showAudioInputPorts.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2003, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A program that prints out this computer's audio input ports
#include "AudioInputDevice.hh"
|
|
[-]
[+]
|
Added |
live.2008.12.20.tar.bz2/config.armeb-uclibc
^
|
| @@ -0,0 +1,19 @@
+CROSS_COMPILE= armeb-linux-uclibc-
+COMPILE_OPTS = $(INCLUDES) -I. -Os -DSOCKLEN_T=socklen_t -DNO_STRSTREAM=1 -D
+LARGEFILE_SOURCE=1 -D_FILE_OFFSET_BITS=64
+C = c
+C_COMPILER = $(CROSS_COMPILE)gcc
+C_FLAGS = $(COMPILE_OPTS)
+CPP = cpp
+CPLUSPLUS_COMPILER = $(CROSS_COMPILE)g++
+CPLUSPLUS_FLAGS = $(COMPILE_OPTS) -Wall -DBSD=1
+OBJ = o
+LINK = $(CROSS_COMPILE)gcc -o
+LINK_OPTS = -L.
+CONSOLE_LINK_OPTS = $(LINK_OPTS)
+LIBRARY_LINK = $(CROSS_COMPILE)ld -o
+LIBRARY_LINK_OPTS = $(LINK_OPTS) -r -Bstatic
+LIB_SUFFIX = a
+LIBS_FOR_CONSOLE_APPLICATION =
+LIBS_FOR_GUI_APPLICATION =
+EXE =
|
|
[-]
[+]
|
Added |
live.2008.12.20.tar.bz2/config.bfin-linux-uclibc
^
|
| @@ -0,0 +1,18 @@
+CROSS_COMPILER = bfin-linux-uclibc-
+COMPILE_OPTS = $(INCLUDES) -I. -DSOCKLEN_T=socklen_t -D_LARGEFILE_SOURCE=1 -DUCLINUX -D_FILE_OFFSET_BITS=64
+C = c
+C_COMPILER = $(CROSS_COMPILER)gcc
+C_FLAGS = $(COMPILE_OPTS) -Wall
+CPP = cpp
+CPLUSPLUS_COMPILER = $(CROSS_COMPILER)g++
+CPLUSPLUS_FLAGS = $(COMPILE_OPTS) -Wall
+OBJ = o
+LINK = $(CROSS_COMPILER)g++ -o
+LINK_OPTS = -L.
+CONSOLE_LINK_OPTS = $(LINK_OPTS)
+LIBRARY_LINK = $(CROSS_COMPILER)ld -o
+LIBRARY_LINK_OPTS = $(LINK_OPTS) -r -Bstatic -m elf32bfinfd
+LIB_SUFFIX = a
+LIBS_FOR_CONSOLE_APPLICATION =
+LIBS_FOR_GUI_APPLICATION =
+EXE =
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/config.bfin-uclinux
^
|
| (renamed to config.bfin-uclinux)
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/config.bfin-uclinux
^
|
| (renamed to config.bfin-uclinux)
|
|
[-]
[+]
|
Added |
live.2008.12.20.tar.bz2/config.cygwin-for-vlc
^
|
| @@ -0,0 +1,17 @@
+COMPILE_OPTS =Â Â Â Â $(INCLUDES) -I. -O -DSOCKLEN_T=socklen_t
+C =Â Â c
+C_COMPILER =Â Â gcc
+C_FLAGS =Â Â $(COMPILE_OPTS) -DUSE_OUR_BZERO=1 -D_WIN32 -mno-cygwin
+CPP =Â Â Â cpp
+CPLUSPLUS_COMPILER =Â Â c++
+CPLUSPLUS_FLAGS =Â Â $(COMPILE_OPTS) -Wall -DBSD=1 -D_WIN32 -Wno-deprecated -mno-cygwin
+OBJ =Â Â Â o
+LINK =Â Â c++ -o
+LINK_OPTS =Â Â -L.
+CONSOLE_LINK_OPTS =Â $(LINK_OPTS)
+LIBRARY_LINK =Â Â Â ld -o
+LIBRARY_LINK_OPTS =Â Â $(LINK_OPTS) -r -Bstatic
+LIB_SUFFIX =Â Â a
+LIBS_FOR_CONSOLE_APPLICATION =
+LIBS_FOR_GUI_APPLICATION =
+EXE =
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/config.macosx
^
|
| @@ -1,4 +1,4 @@
-COMPILE_OPTS = $(INCLUDES) -I. -DBSD=1 -O -DSOCKLEN_T=socklen_t -DHAVE_SOCKADDR_LEN=1
+COMPILE_OPTS = $(INCLUDES) -I. $(EXTRA_LDFLAGS) -DBSD=1 -O -DSOCKLEN_T=socklen_t -DHAVE_SOCKADDR_LEN=1
C = c
C_COMPILER = cc
C_FLAGS = $(COMPILE_OPTS)
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/config.mingw
^
|
| @@ -1,15 +1,15 @@
COMPILE_OPTS = $(INCLUDES) -I. -O -DSOCKLEN_T=int
C = c
-C_COMPILER = gcc
+C_COMPILER = $(CC)
C_FLAGS = $(COMPILE_OPTS) -DUSE_OUR_BZERO=1 -D__MINGW32__
CPP = cpp
-CPLUSPLUS_COMPILER = c++
+CPLUSPLUS_COMPILER = $(CXX)
CPLUSPLUS_FLAGS = $(COMPILE_OPTS) -D__MINGW32__ -Wall -Wno-deprecated
OBJ = o
-LINK = c++ -o
+LINK = $(CXX) -o
LINK_OPTS = -L.
CONSOLE_LINK_OPTS = $(LINK_OPTS)
-LIBRARY_LINK = ld -o
+LIBRARY_LINK = $(LD) -o
LIBRARY_LINK_OPTS = $(LINK_OPTS) -r -Bstatic
LIB_SUFFIX = a
LIBS_FOR_CONSOLE_APPLICATION = -lws2_32
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/genWindowsMakefiles.cmd
^
|
| @@ -4,28 +4,21 @@
for %%I in (%0) do cd "%%~pI"
cd liveMedia
del /Q liveMedia.mak
-del /Q Makefile
type Makefile.head ..\win32config Makefile.tail > liveMedia.mak
cd ../groupsock
del /Q groupsock.mak
-del /Q Makefile
type Makefile.head ..\win32config Makefile.tail > groupsock.mak
cd ../UsageEnvironment
del /Q UsageEnvironment.mak
-del /Q Makefile
type Makefile.head ..\win32config Makefile.tail > UsageEnvironment.mak
cd ../BasicUsageEnvironment
del /Q BasicUsageEnvironment.mak
-del /Q Makefile
type Makefile.head ..\win32config Makefile.tail > BasicUsageEnvironment.mak
cd ../testProgs
del /Q testProgs.mak
-del /Q Makefile
type Makefile.head ..\win32config Makefile.tail > testProgs.mak
cd ../mediaServer
del /Q mediaServer.mak
-del /Q Makefile
type Makefile.head ..\win32config Makefile.tail > mediaServer.mak
-
ENDLOCAL
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/GroupEId.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-1997, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// "Group Endpoint Id"
// Implementation
@@ -60,7 +60,7 @@
Scope::~Scope() {
clean();
}
-
+
unsigned Scope::publicKeySize() const {
return fPublicKey == NULL ? 0 : strlen(fPublicKey);
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/Groupsock.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// 'Group sockets'
// Implementation
@@ -62,7 +62,7 @@
if (!writeSocket(env(), socketNum(), destAddr, port, ttl,
buffer, bufferSize))
return False;
-
+
if (sourcePortNum() == 0) {
// Now that we've sent a packet, we can find out what the
// kernel chose as our ephemeral source port number:
@@ -74,7 +74,7 @@
return False;
}
}
-
+
return True;
}
@@ -120,15 +120,15 @@
<< env.getResultMsg() << "\n";
}
}
-
+
// Make sure we can get our source address:
- if (ourSourceAddressForMulticast(env) == 0) {
+ if (ourIPAddress(env) == 0) {
if (DebugLevel >= 0) { // this is a fatal error
env << "Unable to determine our source address: "
<< env.getResultMsg() << "\n";
}
}
-
+
if (DebugLevel >= 2) env << *this << ": created\n";
}
@@ -157,7 +157,7 @@
}
}
}
-
+
if (DebugLevel >= 2) env << *this << ": created\n";
}
@@ -170,7 +170,7 @@
} else {
socketLeaveGroup(env(), socketNum(), groupAddress().s_addr);
}
-
+
delete fDests;
if (DebugLevel >= 2) env() << *this << ": deleting\n";
@@ -267,17 +267,17 @@
if (!writeSuccess) break;
statsOutgoing.countPacket(bufferSize);
statsGroupOutgoing.countPacket(bufferSize);
-
+
// Then, forward to our members:
int numMembers = 0;
if (!members().IsEmpty()) {
numMembers =
outputToAllMembersExcept(interfaceNotToFwdBackTo,
ttlToSend, buffer, bufferSize,
- ourSourceAddressForMulticast(env));
+ ourIPAddress(env));
if (numMembers < 0) break;
}
-
+
if (DebugLevel >= 3) {
env << *this << ": wrote " << bufferSize << " bytes, ttl "
<< (unsigned)ttlToSend;
@@ -288,7 +288,7 @@
}
return True;
} while (0);
-
+
if (DebugLevel >= 0) { // this is a fatal error
env.setResultMsg("Groupsock write failed: ", env.getResultMsg());
}
@@ -300,9 +300,9 @@
struct sockaddr_in& fromAddress) {
// Read data from the socket, and relay it across any attached tunnels
//##### later make this code more general - independent of tunnels
-
+
bytesRead = 0;
-
+
int maxBytesToRead = bufferMaxSize - TunnelEncapsulationTrailerMaxSize;
int numBytes = readSocket(env(), socketNum(),
buffer, maxBytesToRead, fromAddress);
@@ -313,18 +313,18 @@
}
return False;
}
-
+
// If we're a SSM group, make sure the source address matches:
if (isSSM()
&& fromAddress.sin_addr.s_addr != sourceFilterAddress().s_addr) {
return True;
}
-
+
// We'll handle this data.
// Also write it (with the encapsulation trailer) to each member,
// unless the packet was originally sent by us to begin with.
bytesRead = numBytes;
-
+
int numMembers = 0;
if (!wasLoopedBackFromUs(env(), fromAddress)) {
statsIncoming.countPacket(numBytes);
@@ -346,14 +346,14 @@
}
env() << "\n";
}
-
+
return True;
}
Boolean Groupsock::wasLoopedBackFromUs(UsageEnvironment& env,
struct sockaddr_in& fromAddress) {
if (fromAddress.sin_addr.s_addr
- == ourSourceAddressForMulticast(env)) {
+ == ourIPAddress(env)) {
if (fromAddress.sin_port == sourcePortNum()) {
#ifdef DEBUG_LOOPBACK_CHECKING
if (DebugLevel >= 3) {
@@ -363,7 +363,7 @@
return True;
}
}
-
+
return False;
}
@@ -373,7 +373,7 @@
netAddressBits sourceAddr) {
// Don't forward TTL-0 packets
if (ttlToFwd == 0) return 0;
-
+
DirectedNetInterfaceSet::Iterator iter(members());
unsigned numMembers = 0;
DirectedNetInterface* interf;
@@ -381,7 +381,7 @@
// Check whether we've asked to exclude this interface:
if (interf == exceptInterface)
continue;
-
+
// Check that the packet's source address makes it OK to
// be relayed across this interface:
UsageEnvironment& saveEnv = env();
@@ -394,7 +394,7 @@
continue;
}
}
-
+
if (numMembers == 0) {
// We know that we're going to forward to at least one
// member, so fill in the tunnel encapsulation trailer.
@@ -402,7 +402,7 @@
TunnelEncapsulationTrailer* trailerInPacket
= (TunnelEncapsulationTrailer*)&data[size];
TunnelEncapsulationTrailer* trailer;
-
+
Boolean misaligned = ((unsigned long)trailerInPacket & 3) != 0;
unsigned trailerOffset;
u_int8_t tunnelCmd;
@@ -422,29 +422,29 @@
trailer = trailerInPacket;
}
trailer += trailerOffset;
-
+
if (fDests != NULL) {
trailer->address() = fDests->fGroupEId.groupAddress().s_addr;
trailer->port() = fDests->fPort; // structure copy, outputs in network order
}
trailer->ttl() = ttlToFwd;
trailer->command() = tunnelCmd;
-
+
if (isSSM()) {
trailer->auxAddress() = sourceFilterAddress().s_addr;
}
-
+
if (misaligned) {
memmove(trailerInPacket, trailer-trailerOffset, trailerSize);
}
-
+
size += trailerSize;
}
-
+
interf->write(data, size);
++numMembers;
}
-
+
return numMembers;
}
@@ -477,18 +477,18 @@
static Boolean unsetGroupsockBySocket(Groupsock const* groupsock) {
do {
if (groupsock == NULL) break;
-
+
int sock = groupsock->socketNum();
// Make sure "sock" is in bounds:
if (sock < 0) break;
-
+
HashTable* sockets = getSocketTable(groupsock->env());
if (sockets == NULL) break;
-
+
Groupsock* gs = (Groupsock*)sockets->Lookup((char*)(long)sock);
if (gs == NULL || gs != groupsock) break;
sockets->Remove((char*)(long)sock);
-
+
if (sockets->IsEmpty()) {
// We can also delete the table (to reclaim space):
delete sockets;
@@ -497,7 +497,7 @@
return True;
} while (0);
-
+
return False;
}
@@ -511,10 +511,10 @@
env.setResultMsg(buf);
break;
}
-
+
HashTable* sockets = getSocketTable(env);
if (sockets == NULL) break;
-
+
// Make sure we're not replacing an existing Groupsock
// That shouldn't happen
Boolean alreadyExists
@@ -527,11 +527,11 @@
env.setResultMsg(buf);
break;
}
-
+
sockets->Add((char*)(long)sock, groupsock);
return True;
} while (0);
-
+
return False;
}
@@ -539,13 +539,13 @@
do {
// Make sure the "sock" parameter is in bounds:
if (sock < 0) break;
-
+
HashTable* sockets = getSocketTable(env);
if (sockets == NULL) break;
-
+
return (Groupsock*)sockets->Lookup((char*)(long)sock);
} while (0);
-
+
return NULL;
}
@@ -564,7 +564,7 @@
isNew = True;
}
} while (0);
-
+
return groupsock;
}
@@ -584,7 +584,7 @@
isNew = True;
}
} while (0);
-
+
return groupsock;
}
@@ -626,14 +626,14 @@
sourceFilterAddr.s_addr = sourceFilterAddress;
groupsock = new Groupsock(env, groupAddr, sourceFilterAddr, port);
}
-
+
if (groupsock == NULL || groupsock->socketNum() < 0) break;
-
+
if (!setGroupsockBySocket(env, groupsock->socketNum(), groupsock)) break;
-
+
fTable.Add(groupAddress, sourceFilterAddress, port, (void*)groupsock);
} while (0);
-
+
return groupsock;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/GroupsockHelper.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Helper routines to implement 'group sockets'
// Implementation
@@ -60,20 +60,20 @@
socketErr(env, "Failed to initialize 'winsock': ");
return -1;
}
-
+
int newSocket = socket(AF_INET, SOCK_DGRAM, 0);
if (newSocket < 0) {
socketErr(env, "unable to create datagram socket: ");
return newSocket;
}
-
+
if (setsockopt(newSocket, SOL_SOCKET, SO_REUSEADDR,
(const char*)&reuseFlag, sizeof reuseFlag) < 0) {
socketErr(env, "setsockopt(SO_REUSEADDR) error: ");
closeSocket(newSocket);
return -1;
}
-
+
#if defined(__WIN32__) || defined(_WIN32)
// Windoze doesn't properly handle SO_REUSEPORT or IP_MULTICAST_LOOP
#else
@@ -85,7 +85,7 @@
return -1;
}
#endif
-
+
#ifdef IP_MULTICAST_LOOP
const u_int8_t loop = (u_int8_t)setLoopback;
if (setsockopt(newSocket, IPPROTO_IP, IP_MULTICAST_LOOP,
@@ -96,7 +96,7 @@
}
#endif
#endif
-
+
// Note: Windoze requires binding, even if the port number is 0
netAddressBits addr = INADDR_ANY;
#if defined(__WIN32__) || defined(_WIN32)
@@ -117,12 +117,12 @@
#else
}
#endif
-
+
// Set the sending interface for multicasts, if it's not the default:
if (SendingInterfaceAddr != INADDR_ANY) {
struct in_addr addr;
addr.s_addr = SendingInterfaceAddr;
-
+
if (setsockopt(newSocket, IPPROTO_IP, IP_MULTICAST_IF,
(const char*)&addr, sizeof addr) < 0) {
socketErr(env, "error setting outgoing multicast interface: ");
@@ -130,7 +130,7 @@
return -1;
}
}
-
+
return newSocket;
}
@@ -153,20 +153,20 @@
socketErr(env, "Failed to initialize 'winsock': ");
return -1;
}
-
+
int newSocket = socket(AF_INET, SOCK_STREAM, 0);
if (newSocket < 0) {
socketErr(env, "unable to create stream socket: ");
return newSocket;
}
-
+
if (setsockopt(newSocket, SOL_SOCKET, SO_REUSEADDR,
(const char*)&reuseFlag, sizeof reuseFlag) < 0) {
socketErr(env, "setsockopt(SO_REUSEADDR) error: ");
closeSocket(newSocket);
return -1;
}
-
+
// SO_REUSEPORT doesn't really make sense for TCP sockets, so we
// normally don't set them. However, if you really want to do this
// #define REUSE_FOR_TCP
@@ -225,7 +225,7 @@
if (socket < 0) break;
FD_SET((unsigned) socket, &rd_set);
const unsigned numFds = socket+1;
-
+
result = select(numFds, &rd_set, NULL, NULL, timeout);
if (timeout != NULL && result == 0) {
break; // this is OK - timeout occurred
@@ -237,7 +237,7 @@
socketErr(env, "select() error: ");
break;
}
-
+
if (!FD_ISSET(socket, &rd_set)) {
socketErr(env, "select() error - !FD_ISSET");
break;
@@ -264,7 +264,7 @@
} else if (result <= 0) {
break;
}
-
+
SOCKLEN_T addressSize = sizeof fromAddress;
bytesRead = recvfrom(socket, (char*)buffer, bufferSize, 0,
(struct sockaddr*)&fromAddress,
@@ -294,7 +294,7 @@
break;
}
} while (0);
-
+
return bytesRead;
}
@@ -305,7 +305,7 @@
struct timeval* timeout) {
/* read EXACTLY bufferSize bytes from the socket into the buffer.
fromaddress is address of last read.
- return the number of bytes acually read when an error occurs
+ return the number of bytes actually read when an error occurs
*/
int bsize = bufferSize;
int bytesRead = 0;
@@ -442,7 +442,7 @@
}
#endif
}
-
+
return True;
}
@@ -507,7 +507,7 @@
socketErr(env, "setsockopt(IP_ADD_SOURCE_MEMBERSHIP) error: ");
return False;
}
-
+
return True;
}
@@ -524,7 +524,7 @@
(const char*)&imr, sizeof (struct ip_mreq_source)) < 0) {
return False;
}
-
+
return True;
}
@@ -532,7 +532,7 @@
sockaddr_in test; test.sin_port = 0;
SOCKLEN_T len = sizeof test;
if (getsockname(socket, (struct sockaddr*)&test, &len) < 0) return False;
-
+
resultPortNum = ntohs(test.sin_port);
return True;
}
@@ -549,7 +549,7 @@
return False;
}
}
-
+
port = Port(portNum);
return True;
}
@@ -564,16 +564,16 @@
Boolean loopbackWorks = 1;
-netAddressBits ourSourceAddressForMulticast(UsageEnvironment& env) {
+netAddressBits ourIPAddress(UsageEnvironment& env) {
static netAddressBits ourAddress = 0;
int sock = -1;
struct in_addr testAddr;
-
+
if (ourAddress == 0) {
// We need to find our source address
struct sockaddr_in fromAddr;
fromAddr.sin_addr.s_addr = 0;
-
+
// Get our address by sending a (0-TTL) multicast packet,
// receiving it, and looking at the source address used.
// (This is kinda bogus, but it provides the best guarantee
@@ -583,18 +583,18 @@
testAddr.s_addr = our_inet_addr("228.67.43.91"); // arbitrary
Port testPort(15947); // ditto
-
+
sock = setupDatagramSocket(env, testPort);
if (sock < 0) break;
-
+
if (!socketJoinGroup(env, sock, testAddr.s_addr)) break;
-
+
unsigned char testString[] = "hostIdTest";
unsigned testStringLength = sizeof testString;
-
+
if (!writeSocket(env, sock, testAddr, testPort, 0,
testString, testStringLength)) break;
-
+
unsigned char readBuffer[20];
struct timeval timeout;
timeout.tv_sec = 5;
@@ -624,7 +624,7 @@
env.setResultErrMsg("initial gethostname() failed");
break;
}
-
+
#if defined(VXWORKS)
#include <hostLib.h>
if (ERROR == (ourAddress = hostGetByName( hostname ))) break;
@@ -641,7 +641,7 @@
for (unsigned i = 0; ; ++i) {
char* addrPtr = hstent->h_addr_list[i];
if (addrPtr == NULL) break;
-
+
netAddressBits a = *(netAddressBits*)addrPtr;
if (!badAddress(a)) {
addr = a;
@@ -655,7 +655,7 @@
break;
}
} while (0);
-
+
// Make sure we have a good address:
netAddressBits from = fromAddr.sin_addr.s_addr;
if (badAddress(from)) {
@@ -666,15 +666,15 @@
env.setResultMsg(tmp);
from = 0;
}
-
+
ourAddress = from;
#endif
-
+
if (sock >= 0) {
socketLeaveGroup(env, sock, testAddr.s_addr);
closeSocket(sock);
}
-
+
// Use our newly-discovered IP address, and the current time,
// to initialize the random number generator's seed:
struct timeval timeNow;
@@ -687,7 +687,7 @@
netAddressBits chooseRandomIPv4SSMAddress(UsageEnvironment& env) {
// First, a hack to ensure that our random number generator is seeded:
- (void) ourSourceAddressForMulticast(env);
+ (void) ourIPAddress(env);
// Choose a random address in the range [232.0.1.0, 232.255.255.255)
// i.e., [0xE8000100, 0xE8FFFFFF)
@@ -700,7 +700,7 @@
char const* timestampString() {
struct timeval tvNow;
gettimeofday(&tvNow, NULL);
-
+
#if !defined(_WIN32_WCE)
static char timeString[9]; // holds hh:mm:ss plus trailing '\0'
char const* ctimeResult = ctime((time_t*)&tvNow.tv_sec);
@@ -717,7 +717,7 @@
static char timeString[50];
sprintf(timeString, "%lu.%06ld", tvNow.tv_sec, tvNow.tv_usec);
#endif
-
+
return (char const*)&timeString;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/IOHandlers.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-1998 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// IO event handlers
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/NetAddress.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-1998 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Network Addresses
// Implementation
@@ -90,14 +90,14 @@
if (addr != INADDR_NONE) { // yes it was an IP address string
//##### host = gethostbyaddr((char*)&addr, sizeof (netAddressBits), AF_INET);
host = NULL; // don't bother calling gethostbyaddr(); we only want 1 addr
-
+
if (host == NULL) {
// For some unknown reason, gethostbyaddr() failed, so just
// return a 1-element list with the address we were given:
fNumAddresses = 1;
fAddressArray = new NetAddress*[fNumAddresses];
if (fAddressArray == NULL) return;
-
+
fAddressArray[0] = new NetAddress((u_int8_t*)&addr,
sizeof (netAddressBits));
return;
@@ -126,11 +126,11 @@
++fNumAddresses;
++hAddrPtr1;
}
-
+
// Next, set up the list:
fAddressArray = new NetAddress*[fNumAddresses];
if (fAddressArray == NULL) return;
-
+
for (unsigned i = 0; i < fNumAddresses; ++i) {
fAddressArray[i]
= new NetAddress(hAddrPtr[i], host->h_length);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/NetInterface.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-1998 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Network Interfaces
// Implementation
@@ -130,14 +130,14 @@
if (sock == NULL) { // we need to create one:
sock = CreateNew(env, port);
if (sock == NULL || sock->socketNum() < 0) break;
-
+
fTable->Add((char*)(long)(port.num()), (void*)sock);
isNew = True;
}
-
+
return sock;
} while (0);
-
+
delete sock;
return NULL;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/include/GroupEId.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "multikit" Multicast Application Shell
-// Copyright (c) 1996-1997, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// "Group Endpoint Id"
// C++ header
@@ -64,7 +64,7 @@
portNumBits portNum,
unsigned numSuccessiveGroupAddrs = 1);
// used for a 'source-specific multicast' group
- GroupEId(); // used only as a temp constructor prior to initialization
+ GroupEId(); // used only as a temp constructor prior to initialization
struct in_addr const& groupAddress() const { return fGroupAddress; }
struct in_addr const& sourceFilterAddress() const { return fSourceFilterAddress; }
@@ -75,7 +75,7 @@
// could be >1 for hier encoding
return fNumSuccessiveGroupAddrs;
}
-
+
portNumBits portNum() const { return fPortNum; }
const Scope& scope() const { return fScope; }
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/include/Groupsock.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-1998 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// 'Group sockets'
// C++ header
@@ -40,7 +40,7 @@
public:
OutputSocket(UsageEnvironment& env);
virtual ~OutputSocket();
-
+
Boolean write(netAddressBits address, Port port, u_int8_t ttl,
unsigned char* buffer, unsigned bufferSize);
@@ -107,21 +107,21 @@
struct in_addr const& sourceFilterAddress() const {
return fIncomingGroupEId.sourceFilterAddress();
}
-
+
Boolean isSSM() const {
return fIncomingGroupEId.isSSM();
}
-
+
u_int8_t ttl() const { return fTTL; }
-
+
void multicastSendOnly(); // send, but don't receive any multicast packets
Boolean output(UsageEnvironment& env, u_int8_t ttl,
unsigned char* buffer, unsigned bufferSize,
DirectedNetInterface* interfaceNotToFwdBackTo = NULL);
-
+
DirectedNetInterfaceSet& members() { return fMembers; }
-
+
Boolean deleteIfNoMembers;
Boolean isSlave; // for tunneling
@@ -133,21 +133,21 @@
NetInterfaceTrafficStats statsGroupOutgoing; // *not* static
NetInterfaceTrafficStats statsGroupRelayedIncoming; // *not* static
NetInterfaceTrafficStats statsGroupRelayedOutgoing; // *not* static
-
+
Boolean wasLoopedBackFromUs(UsageEnvironment& env,
struct sockaddr_in& fromAddress);
-
+
public: // redefined virtual functions
virtual Boolean handleRead(unsigned char* buffer, unsigned bufferMaxSize,
unsigned& bytesRead,
struct sockaddr_in& fromAddress);
-
+
private:
int outputToAllMembersExcept(DirectedNetInterface* exceptInterface,
u_int8_t ttlToFwd,
unsigned char* data, unsigned size,
netAddressBits sourceAddr);
-
+
private:
GroupEId fIncomingGroupEId;
destRecord* fDests;
@@ -182,19 +182,19 @@
class Iterator {
public:
Iterator(GroupsockLookupTable& groupsocks);
-
+
Groupsock* next(); // NULL iff none
-
+
private:
AddressPortLookupTable::Iterator fIter;
};
-
+
private:
Groupsock* AddNew(UsageEnvironment& env,
netAddressBits groupAddress,
netAddressBits sourceFilterAddress,
Port port, u_int8_t ttl);
-
+
private:
friend class Iterator;
AddressPortLookupTable fTable;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/include/GroupsockHelper.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Helper routines to implement 'group sockets'
// C++ header
@@ -75,7 +75,7 @@
Boolean getSourcePort(UsageEnvironment& env, int socket, Port& port);
-netAddressBits ourSourceAddressForMulticast(UsageEnvironment& env); // in network order
+netAddressBits ourIPAddress(UsageEnvironment& env); // in network order
// IP addresses of our sending and receiving interfaces. (By default, these
// are INADDR_ANY (i.e., 0), specifying the default interface.)
@@ -93,7 +93,7 @@
#define SET_SOCKADDR_SIN_LEN(var) var.sin_len = sizeof var
#else
#define SET_SOCKADDR_SIN_LEN(var)
-#endif
+#endif
#define MAKE_SOCKADDR_IN(var,adr,prt) /*adr,prt must be in network order*/\
struct sockaddr_in var;\
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/include/IOHandlers.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-1997 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// IO event handlers
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/include/NetAddress.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Network Addresses
// C++ header
@@ -86,7 +86,7 @@
friend class Iterator;
unsigned fNumAddresses;
- NetAddress** fAddressArray;
+ NetAddress** fAddressArray;
};
typedef u_int16_t portNumBits;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/include/NetCommon.h
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
/* "groupsock" interface
- * Copyright (c) 1996-2003 Live Networks, Inc. All rights reserved.
+ * Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
* Common include files, typically used for networking
*/
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/include/NetInterface.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Network Interfaces
// C++ header
@@ -28,10 +28,10 @@
class NetInterface {
public:
virtual ~NetInterface();
-
+
static UsageEnvironment* DefaultUsageEnvironment;
// if non-NULL, used for each new interfaces
-
+
protected:
NetInterface(); // virtual base class
};
@@ -39,12 +39,12 @@
class DirectedNetInterface: public NetInterface {
public:
virtual ~DirectedNetInterface();
-
+
virtual Boolean write(unsigned char* data, unsigned numBytes) = 0;
-
+
virtual Boolean SourceAddrOKForRelaying(UsageEnvironment& env,
unsigned addr) = 0;
-
+
protected:
DirectedNetInterface(); // virtual base class
};
@@ -53,25 +53,25 @@
public:
DirectedNetInterfaceSet();
virtual ~DirectedNetInterfaceSet();
-
+
DirectedNetInterface* Add(DirectedNetInterface const* interf);
// Returns the old value if different, otherwise 0
Boolean Remove(DirectedNetInterface const* interf);
-
+
Boolean IsEmpty() { return fTable->IsEmpty(); }
-
+
// Used to iterate through the interfaces in the set
class Iterator {
public:
Iterator(DirectedNetInterfaceSet& interfaces);
virtual ~Iterator();
-
+
DirectedNetInterface* next(); // NULL iff none
-
+
private:
HashTable::Iterator* fIter;
};
-
+
private:
friend class Iterator;
HashTable* fTable;
@@ -80,26 +80,26 @@
class Socket: public NetInterface {
public:
virtual ~Socket();
-
+
virtual Boolean handleRead(unsigned char* buffer, unsigned bufferMaxSize,
unsigned& bytesRead,
struct sockaddr_in& fromAddress) = 0;
// Returns False on error; resultData == NULL if data ignored
-
+
int socketNum() const { return fSocketNum; }
-
+
Port port() const {
return fPort;
}
-
+
UsageEnvironment& env() const { return fEnv; }
-
+
static int DebugLevel;
-
+
protected:
Socket(UsageEnvironment& env, Port port,
Boolean setLoopback = True); // virtual base class
-
+
Boolean changePort(Port newPort); // will also cause socketNum() to change
private:
@@ -116,15 +116,15 @@
class SocketLookupTable {
public:
virtual ~SocketLookupTable();
-
+
Socket* Fetch(UsageEnvironment& env, Port port, Boolean& isNew);
// Creates a new Socket if none already exists
Boolean Remove(Socket const* sock);
-
+
protected:
SocketLookupTable(); // abstract base class
virtual Socket* CreateNew(UsageEnvironment& env, Port port) = 0;
-
+
private:
HashTable* fTable;
};
@@ -134,14 +134,14 @@
class NetInterfaceTrafficStats {
public:
NetInterfaceTrafficStats();
-
+
void countPacket(unsigned packetSize);
-
+
float totNumPackets() const {return fTotNumPackets;}
float totNumBytes() const {return fTotNumBytes;}
-
+
Boolean haveSeenTraffic() const;
-
+
private:
float fTotNumPackets;
float fTotNumBytes;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/include/TunnelEncaps.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "mTunnel" multicast access service
-// Copyright (c) 1996-1998 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Encapsulation trailer for tunnels
// C++ header
@@ -42,9 +42,9 @@
public:
Cookie& srcCookie()
- { return *(Cookie*)byteOffset(0); }
+ { return *(Cookie*)byteOffset(0); }
Cookie& dstCookie()
- { return *(Cookie*)byteOffset(2); }
+ { return *(Cookie*)byteOffset(2); }
u_int32_t& address()
{ return *(u_int32_t*)byteOffset(4); }
Port& port()
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/include/groupsock_version.hh
^
|
| @@ -1,10 +1,10 @@
// Version information for the "groupsock" library
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
#ifndef _GROUPSOCK_VERSION_HH
#define _GROUPSOCK_VERSION_HH
-#define GROUPSOCK_LIBRARY_VERSION_STRING "2007.10.31"
-#define GROUPSOCK_LIBRARY_VERSION_INT 1193788800
+#define GROUPSOCK_LIBRARY_VERSION_STRING "2008.12.19"
+#define GROUPSOCK_LIBRARY_VERSION_INT 1229644800
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/groupsock/inet.c
^
|
| @@ -93,7 +93,8 @@
}
#endif
-#ifdef USE_SYSTEM_RANDOM
+#ifndef USE_OUR_RANDOM
+/* Use the system-supplied "random()" and "srandom()" functions */
#include <stdlib.h>
long our_random() {
#if defined(__WIN32__) || defined(_WIN32)
@@ -104,12 +105,15 @@
}
void our_srandom(unsigned int x) {
#if defined(__WIN32__) || defined(_WIN32)
- return srand(x);
+ srand(x);
#else
- return srandom(x);
+ srandom(x);
#endif
}
+
#else
+
+/* Use our own implementation of the "random()" and "srandom()" functions */
/*
* random.c:
*
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AC3AudioRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for AC3 audio
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AC3AudioRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// AC3 Audio RTP Sources
// Implementation
@@ -71,7 +71,7 @@
resultSpecialHeaderSize = 2;
return True;
-}
+}
char const* AC3AudioRTPSource::MIMEtype() const {
return "audio/AC3";
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AC3AudioStreamFramer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an AC3 audio elementary stream into frames
// Implementation
@@ -167,7 +167,7 @@
// We were able to acquire a frame from the input.
// It has already been copied to the reader's space.
fFrameSize = acquiredFrameSize;
-
+
// Also set the presentation time, and increment it for next time,
// based on the length of this frame:
fPresentationTime = fNextFramePresentationTime;
@@ -253,7 +253,7 @@
} else {
// Discard all of the data that was just read:
totNumValidBytes() = totNumValidBytes() - size;
-
+
return False;
}
}
@@ -270,7 +270,7 @@
try {
saveParserState();
-
+
// We expect an AC3 audio header (first 2 bytes == 0x0B77) at the start:
while (1) {
unsigned next4Bytes = test4Bytes();
@@ -280,10 +280,10 @@
}
fCurrentFrame.hdr0 = get4Bytes();
fCurrentFrame.hdr1 = test4Bytes();
-
+
fCurrentFrame.setParamsFromHeader();
fHaveParsedAFrame = True;
-
+
// Copy the frame to the requested destination:
unsigned frameSize = fCurrentFrame.frameSize;
if (frameSize > fMaxSize) {
@@ -299,7 +299,7 @@
fTo[3] = fCurrentFrame.hdr0;
getBytes(&fTo[4], frameSize-4);
skipBytes(numTruncatedBytes);
-
+
return frameSize;
} catch (int /*e*/) {
#ifdef DEBUG
@@ -316,7 +316,7 @@
fSavedFrameFlag = 0;
fUsingSource->getNextFrame(fSavedFrame, maxAC3FrameSize,
- afterGettingSavedFrame, this,
+ afterGettingSavedFrame, this,
onSavedFrameClosure, this);
fUsingSource->envir().taskScheduler().doEventLoop(&fSavedFrameFlag);
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/ADTSAudioFileServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from an AAC audio file in ADTS format
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/ADTSAudioFileSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A source object for AAC audio files in ADTS format
// Implementation
@@ -42,13 +42,13 @@
// to get the audio stream's parameters:
unsigned char fixedHeader[4]; // it's actually 3.5 bytes long
if (fread(fixedHeader, 1, sizeof fixedHeader, fid) < sizeof fixedHeader) break;
-
+
// Check the 'syncword':
if (!(fixedHeader[0] == 0xFF && (fixedHeader[1]&0xF0) == 0xF0)) {
env.setResultMsg("Bad 'syncword' at start of ADTS file");
break;
}
-
+
// Get and check the 'profile':
u_int8_t profile = (fixedHeader[2]&0xC0)>>6; // 2 bits
if (profile == 3) {
@@ -69,7 +69,11 @@
// If we get here, the frame header was OK.
// Reset the fid to the beginning of the file:
+#ifndef _WIN32_WCE
rewind(fid);
+#else
+ fseek(fid, SEEK_SET,0);
+#endif
#ifdef DEBUG
fprintf(stderr, "Read first frame: profile %d, "
"sampling_frequency_index %d => samplingFrequency %d, "
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AMRAudioFileServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from an AMR audio file.
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AMRAudioFileSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// AMR Audio File sinks
// Implementation
@@ -26,7 +26,7 @@
AMRAudioFileSink
::AMRAudioFileSink(UsageEnvironment& env, FILE* fid, unsigned bufferSize,
- char const* perFrameFileNamePrefix)
+ char const* perFrameFileNamePrefix)
: FileSink(env, fid, bufferSize, perFrameFileNamePrefix),
fHaveWrittenHeader(False) {
}
@@ -69,7 +69,7 @@
if (!fHaveWrittenHeader && fPerFrameFileNameBuffer == NULL) {
// Output the appropriate AMR header to the start of the file.
- // This header is defined in RFC 3267, section 5.
+ // This header is defined in RFC 3267, section 5.
// (However, we don't do this if we're creating one file per frame.)
char headerBuffer[100];
sprintf(headerBuffer, "#!AMR%s%s\n",
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AMRAudioFileSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A source object for AMR audio files (as defined in RFC 3267, section 5)
// Implementation
@@ -43,7 +43,7 @@
if (fread(buf, 1, 6, fid) < 6) break;
if (strncmp(buf, "#!AMR", 5) != 0) break; // bad magic #
unsigned bytesRead = 6;
-
+
// The next bytes must be "\n", "-WB\n", "_MC1.0\n", or "-WB_MC1.0\n"
if (buf[5] == '-') {
// The next bytes must be "WB\n" or "WB_MC1.0\n"
@@ -145,8 +145,8 @@
break;
}
}
- }
-
+ }
+
// Next, read the frame-block into the buffer provided:
fFrameSize *= fNumChannels; // because multiple channels make up a frame-block
if (fFrameSize > fMaxSize) {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AMRAudioRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for AMR audio (RFC 3267)
// Implementation
@@ -119,7 +119,7 @@
unsigned AMRAudioRTPSink::specialHeaderSize() const {
// For now, because we're packing only one frame per packet,
// there's just a 1-byte payload header, plus a 1-byte TOC #####
- return 2;
+ return 2;
}
char const* AMRAudioRTPSink::auxSDPLine() {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AMRAudioRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// AMR Audio RTP Sources (RFC 3267)
// Implementation
@@ -60,7 +60,7 @@
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
private:
Boolean fIsWideband, fIsOctetAligned, fIsInterleaved, fCRCsArePresent;
@@ -82,7 +82,7 @@
AMRDeinterleaver(UsageEnvironment& env,
Boolean isWideband, unsigned numChannels,
unsigned maxInterleaveGroupSize, RawAMRRTPSource* inputSource);
- // called only by "createNew()"
+ // called only by "createNew()"
virtual ~AMRDeinterleaver();
@@ -90,7 +90,7 @@
unsigned numTruncatedBytes,
struct timeval presentationTime,
unsigned durationInMicroseconds);
- void afterGettingFrame1(unsigned frameSize, struct timeval presentationTime);
+ void afterGettingFrame1(unsigned frameSize, struct timeval presentationTime);
private:
// Redefined virtual functions:
@@ -101,7 +101,7 @@
RawAMRRTPSource* fInputSource;
class AMRDeinterleavingBuffer* fDeinterleavingBuffer;
Boolean fNeedAFrame;
-
+
};
@@ -141,7 +141,7 @@
env << "AMRAudioRTPSource::createNew(): 'Bandwidth-efficient mode' was specified, along with interleaving, 'robust sorting order', and/or CRCs, so we assume 'octet-aligned mode' instead.\n";
isOctetAligned = True;
}
- }
+ }
Boolean isInterleaved;
unsigned maxInterleaveGroupSize; // in frames (not frame-blocks)
@@ -312,7 +312,7 @@
#endif
return True;
-}
+}
char const* RawAMRRTPSource::MIMEtype() const {
return fIsWideband ? "audio/AMR-WB" : "audio/AMR";
@@ -396,7 +396,7 @@
Boolean retrieveFrame(unsigned char* to, unsigned maxSize,
unsigned& resultFrameSize, unsigned& resultNumTruncatedBytes,
u_int8_t& resultFrameHeader,
- struct timeval& resultPresentationTime,
+ struct timeval& resultPresentationTime,
Boolean& resultIsSynchronized);
unsigned char* inputBuffer() { return fInputBuffer; }
@@ -460,7 +460,7 @@
// First, try getting a frame from the deinterleaving buffer:
if (fDeinterleavingBuffer->retrieveFrame(fTo, fMaxSize,
fFrameSize, fNumTruncatedBytes,
- fLastFrameHeader, fPresentationTime,
+ fLastFrameHeader, fPresentationTime,
fInputSource->isSynchronized())) {
// Success!
@@ -501,7 +501,7 @@
void AMRDeinterleaver
::afterGettingFrame1(unsigned frameSize, struct timeval presentationTime) {
RawAMRRTPSource* source = (RawAMRRTPSource*)fInputSource;
-
+
// First, put the frame into our deinterleaving buffer:
fDeinterleavingBuffer->deliverIncomingFrame(frameSize, source, presentationTime);
@@ -546,7 +546,7 @@
exit(1);
}
- --frameIndex; // because it was incremented by the source when this frame was read
+ --frameIndex; // because it was incremented by the source when this frame was read
u_int8_t frameHeader;
if (frameIndex >= source->TOCSize()) { // sanity check
frameHeader = FT_NO_DATA<<3;
@@ -578,7 +578,7 @@
fIncomingBankId ^= 1;
unsigned char tmp = fIncomingBinMax;
fIncomingBinMax = fOutgoingBinMax;
- fOutgoingBinMax = tmp;
+ fOutgoingBinMax = tmp;
fNextOutgoingBin = 0;
}
@@ -728,7 +728,7 @@
#endif
fromBV.skipBits(frameSizeBits);
toCount += frameSizeBytes;
- }
+ }
#ifdef DEBUG
if (fromBV.numBitsRemaining() > 7) {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AMRAudioSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A source object for AMR audio sources
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AVIFileSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A sink that generates an AVI file from a composite media session
// Implementation
@@ -42,7 +42,7 @@
unsigned char* dataEnd() { return &fData[fBytesInUse]; }
unsigned bytesInUse() const { return fBytesInUse; }
unsigned bytesAvailable() const { return fBufferSize - fBytesInUse; }
-
+
void setPresentationTime(struct timeval const& presentationTime) {
fPresentationTime = presentationTime;
}
@@ -98,15 +98,18 @@
AVIFileSink::AVIFileSink(UsageEnvironment& env,
MediaSession& inputSession,
- FILE* outFid,
+ char const* outputFileName,
unsigned bufferSize,
unsigned short movieWidth, unsigned short movieHeight,
unsigned movieFPS, Boolean packetLossCompensate)
- : Medium(env), fInputSession(inputSession), fOutFid(outFid),
+ : Medium(env), fInputSession(inputSession),
fBufferSize(bufferSize), fPacketLossCompensate(packetLossCompensate),
fAreCurrentlyBeingPlayed(False), fNumSubsessions(0), fNumBytesWritten(0),
fHaveCompletedOutputFile(False),
fMovieWidth(movieWidth), fMovieHeight(movieHeight), fMovieFPS(movieFPS) {
+ fOutFid = OpenOutputFile(env, outputFileName);
+ if (fOutFid == NULL) return;
+
// Set up I/O state for each input subsession:
MediaSubsessionIterator iter(fInputSession);
MediaSubsession* subsession;
@@ -151,11 +154,14 @@
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
AVISubsessionIOState* ioState
- = (AVISubsessionIOState*)(subsession->miscPtr);
+ = (AVISubsessionIOState*)(subsession->miscPtr);
if (ioState == NULL) continue;
delete ioState;
}
+
+ // Finally, close our output file:
+ CloseOutputFile(fOutFid);
}
AVIFileSink* AVIFileSink
@@ -164,16 +170,15 @@
unsigned bufferSize,
unsigned short movieWidth, unsigned short movieHeight,
unsigned movieFPS, Boolean packetLossCompensate) {
- do {
- FILE* fid = OpenOutputFile(env, outputFileName);
- if (fid == NULL) break;
-
- return new AVIFileSink(env, inputSession, fid, bufferSize,
- movieWidth, movieHeight, movieFPS,
- packetLossCompensate);
- } while (0);
+ AVIFileSink* newSink =
+ new AVIFileSink(env, inputSession, outputFileName, bufferSize,
+ movieWidth, movieHeight, movieFPS, packetLossCompensate);
+ if (newSink == NULL || newSink->fOutFid == NULL) {
+ Medium::close(newSink);
+ return NULL;
+ }
- return NULL;
+ return newSink;
}
Boolean AVIFileSink::startPlaying(afterPlayingFunc* afterFunc,
@@ -194,7 +199,7 @@
Boolean AVIFileSink::continuePlaying() {
// Run through each of our input session's 'subsessions',
// asking for a frame from each one:
- Boolean haveActiveSubsessions = False;
+ Boolean haveActiveSubsessions = False;
MediaSubsessionIterator iter(fInputSession);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
@@ -204,7 +209,7 @@
if (subsessionSource->isCurrentlyAwaitingData()) continue;
AVISubsessionIOState* ioState
- = (AVISubsessionIOState*)(subsession->miscPtr);
+ = (AVISubsessionIOState*)(subsession->miscPtr);
if (ioState == NULL) continue;
haveActiveSubsessions = True;
@@ -243,7 +248,7 @@
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
AVISubsessionIOState* ioState
- = (AVISubsessionIOState*)(subsession->miscPtr);
+ = (AVISubsessionIOState*)(subsession->miscPtr);
if (ioState == NULL) continue;
if (ioState->fOurSourceIsActive) return; // this source hasn't closed
@@ -290,7 +295,7 @@
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
AVISubsessionIOState* ioState
- = (AVISubsessionIOState*)(subsession->miscPtr);
+ = (AVISubsessionIOState*)(subsession->miscPtr);
if (ioState == NULL) continue;
maxBytesPerSecond += ioState->fMaxBytesPerSecond;
@@ -465,7 +470,7 @@
}
// Write the data into the file:
- fOurSink.fNumBytesWritten += fOurSink.addWord(fAVISubsessionTag);
+ fOurSink.fNumBytesWritten += fOurSink.addWord(fAVISubsessionTag);
fOurSink.fNumBytesWritten += fOurSink.addWord(frameSize);
fwrite(frameSource, 1, frameSize, fOurSink.fOutFid);
fOurSink.fNumBytesWritten += frameSize;
@@ -551,7 +556,7 @@
addFileHeader(RIFF,AVI);
size += addFileHeader_hdrl();
- size += addFileHeader_movi();
+ size += addFileHeader_movi();
fRIFFSizePosition = headerSizePosn;
fRIFFSizeValue = size-ignoredSize;
addFileHeaderEnd;
@@ -584,7 +589,7 @@
// Then add another JUNK entry
++fJunkNumber;
- size += addFileHeader_JUNK();
+ size += addFileHeader_JUNK();
addFileHeaderEnd;
#define AVIF_HASINDEX 0x00000010 // Index at end of file?
@@ -612,10 +617,10 @@
addFileHeaderEnd;
addFileHeader(LIST,strl);
- size += addFileHeader_strh();
- size += addFileHeader_strf();
+ size += addFileHeader_strh();
+ size += addFileHeader_strf();
fJunkNumber = 0;
- size += addFileHeader_JUNK();
+ size += addFileHeader_JUNK();
addFileHeaderEnd;
addFileHeader1(strh);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AudioInputDevice.cpp
^
|
| @@ -11,7 +11,7 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// Copyright (c) 2001-2003 Live Networks, Inc. All rights reserved.
// Generic audio input device (such as a microphone, or an input sound card)
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/AudioRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A generic RTP sink for audio codecs (abstract base class)
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/Base64.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Base64 encoding and decoding
// implementation
@@ -56,7 +56,7 @@
outTmp[i] = base64DecodeTable[(unsigned char)inTmp[i]];
if ((outTmp[i]&0x80) != 0) outTmp[i] = 0; // pretend the input was 'A'
}
-
+
out[k++] = (outTmp[0]<<2) | (outTmp[1]>>4);
out[k++] = (outTmp[1]<<4) | (outTmp[2]>>2);
out[k++] = (outTmp[2]<<6) | outTmp[3];
@@ -85,7 +85,7 @@
Boolean havePadding2 = origLength == numOrig24BitValues*3 + 2;
unsigned const numResultBytes = 4*(numOrig24BitValues + havePadding);
char* result = new char[numResultBytes+1]; // allow for trailing '\0'
-
+
// Map each full group of 3 input bytes into 4 output base-64 characters:
unsigned i;
for (i = 0; i < numOrig24BitValues; ++i) {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/BasicUDPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simple UDP sink (i.e., without RTP or other headers added); one frame per packet
// Implementation
@@ -30,7 +30,7 @@
unsigned maxPayloadSize)
: MediaSink(env),
fGS(gs), fMaxPayloadSize(maxPayloadSize) {
- fOutputBuffer = new unsigned char[fMaxPayloadSize];
+ fOutputBuffer = new unsigned char[fMaxPayloadSize];
}
BasicUDPSink::~BasicUDPSink() {
@@ -79,7 +79,7 @@
fNextSendTime.tv_usec += durationInMicroseconds;
fNextSendTime.tv_sec += fNextSendTime.tv_usec/1000000;
fNextSendTime.tv_usec %= 1000000;
-
+
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
int uSecondsToGo;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/BasicUDPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simple UDP source, where every UDP payload is a complete frame
// Implementation
@@ -45,7 +45,7 @@
void BasicUDPSource::doGetNextFrame() {
if (!fHaveStartedReading) {
// Await incoming packets:
- envir().taskScheduler().turnOnBackgroundReadHandling(fInputGS->socketNum(),
+ envir().taskScheduler().turnOnBackgroundReadHandling(fInputGS->socketNum(),
(TaskScheduler::BackgroundHandlerProc*)&incomingPacketHandler, this);
fHaveStartedReading = True;
}
@@ -67,7 +67,7 @@
// Read the packet into our desired destination:
struct sockaddr_in fromAddress;
if (!fInputGS->handleRead(fTo, fMaxSize, fFrameSize, fromAddress)) return;
-
+
// Tell our client that we have new data:
afterGetting(this); // we're preceded by a net read; no infinite recursion
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/BitVector.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Bit Vector data structure
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/BitVector.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Bit Vector data structure
// C++ header
@@ -41,7 +41,7 @@
unsigned curBitIndex() const { return fCurBitIndex; }
unsigned totNumBits() const { return fTotNumBits; }
- unsigned numBitsRemaining() const { return fTotNumBits - fCurBitIndex; }
+ unsigned numBitsRemaining() const { return fTotNumBits - fCurBitIndex; }
private:
unsigned char* fBaseBytePtr;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/ByteStreamFileSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A file source that is a plain byte stream (rather than frames)
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/ByteStreamMultiFileSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A source that consists of multiple byte-stream files, read sequentially
// Implementation
@@ -86,7 +86,7 @@
fPreferredFrameSize, fPlayTimePerFrame);
if (source == NULL) break;
fHaveStartedNewFile = True;
- }
+ }
// (Attempt to) read from the current source.
source->getNextFrame(fTo, fMaxSize,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/DarwinInjector.cpp
^
|
| @@ -11,14 +11,14 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// An object that redirects one or more RTP/RTCP streams - forming a single
// multimedia session - into a 'Darwin Streaming Server' (for subsequent
// reflection to potentially arbitrarily many remote RTSP clients).
-// Implementation.
+// Implementation
#include "DarwinInjector.hh"
#include <GroupsockHelper.hh>
@@ -27,7 +27,7 @@
class SubstreamDescriptor {
public:
- SubstreamDescriptor(RTPSink* rtpSink, RTCPInstance* rtcpInstance);
+ SubstreamDescriptor(RTPSink* rtpSink, RTCPInstance* rtcpInstance, unsigned trackId);
~SubstreamDescriptor();
SubstreamDescriptor*& next() { return fNext; }
@@ -72,10 +72,15 @@
: Medium(env),
fApplicationName(strDup(applicationName)), fVerbosityLevel(verbosityLevel),
fRTSPClient(NULL), fSubstreamSDPSizes(0),
- fHeadSubstream(NULL), fTailSubstream(NULL) {
-}
+ fHeadSubstream(NULL), fTailSubstream(NULL), fSession(NULL), fLastTrackId(0) {
+}
DarwinInjector::~DarwinInjector() {
+ if (fSession != NULL) { // close down and delete the session
+ fRTSPClient->teardownMediaSession(*fSession);
+ Medium::close(fSession);
+ }
+
delete fHeadSubstream;
delete[] (char*)fApplicationName;
Medium::close(fRTSPClient);
@@ -84,7 +89,7 @@
void DarwinInjector::addStream(RTPSink* rtpSink, RTCPInstance* rtcpInstance) {
if (rtpSink == NULL) return; // "rtpSink" should be non-NULL
- SubstreamDescriptor* newDescriptor = new SubstreamDescriptor(rtpSink, rtcpInstance);
+ SubstreamDescriptor* newDescriptor = new SubstreamDescriptor(rtpSink, rtcpInstance, ++fLastTrackId);
if (fHeadSubstream == NULL) {
fHeadSubstream = fTailSubstream = newDescriptor;
} else {
@@ -107,7 +112,6 @@
char const* sessionCopyright) {
char* sdp = NULL;
char* url = NULL;
- MediaSession* session = NULL;
Boolean success = False; // until we learn otherwise
do {
@@ -190,11 +194,11 @@
// Tell the remote server to start receiving the stream from us.
// (To do this, we first create a "MediaSession" object from the SDP description.)
- session = MediaSession::createNew(envir(), sdp);
- if (session == NULL) break;
+ fSession = MediaSession::createNew(envir(), sdp);
+ if (fSession == NULL) break;
ss = fHeadSubstream;
- MediaSubsessionIterator iter(*session);
+ MediaSubsessionIterator iter(*fSession);
MediaSubsession* subsession;
ss = fHeadSubstream;
unsigned streamChannelId = 0;
@@ -219,7 +223,7 @@
if (subsession != NULL) break; // an error occurred above
// Tell the RTSP server to start:
- if (!fRTSPClient->playMediaSession(*session)) break;
+ if (!fRTSPClient->playMediaSession(*fSession)) break;
// Finally, make sure that the output TCP buffer is a reasonable size:
increaseSendBufferTo(envir(), fRTSPClient->socketNum(), 100*1024);
@@ -228,8 +232,7 @@
} while (0);
delete[] sdp;
- delete[] url;
- Medium::close(session);
+ delete[] url;
return success;
}
@@ -240,10 +243,8 @@
////////// SubstreamDescriptor implementation //////////
-static unsigned lastTrackId = 0;
-
SubstreamDescriptor::SubstreamDescriptor(RTPSink* rtpSink,
- RTCPInstance* rtcpInstance)
+ RTCPInstance* rtcpInstance, unsigned trackId)
: fNext(NULL), fRTPSink(rtpSink), fRTCPInstance(rtcpInstance) {
// Create the SDP description for this substream
char const* mediaType = fRTPSink->sdpMediaType();
@@ -277,7 +278,7 @@
char const* auxSDPLine = fRTPSink->auxSDPLine();
if (auxSDPLine == NULL) auxSDPLine = "";
unsigned auxSDPLineSize = strlen(auxSDPLine);
-
+
char const* const sdpFmt =
"m=%s 0 RTP/AVP %u\r\n"
"%s" // "a=rtpmap:" line (if present)
@@ -294,9 +295,10 @@
rtpPayloadType, // m= <fmt list>
rtpmapLine, // a=rtpmap:... (if present)
auxSDPLine, // optional extra SDP line
- ++lastTrackId); // a=control:<track-id>
+ trackId); // a=control:<track-id>
fSDPLines = strDup(sdpLines);
delete[] sdpLines;
+ delete[] rtpmapLine;
}
SubstreamDescriptor::~SubstreamDescriptor() {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/DeviceSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A template for a MediaSource encapsulating an audio/video input device
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/DigestAuthentication.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A class used for digest authentication.
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/FileServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a file.
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/FileSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// File sinks
// Implementation
@@ -29,7 +29,7 @@
////////// FileSink //////////
FileSink::FileSink(UsageEnvironment& env, FILE* fid, unsigned bufferSize,
- char const* perFrameFileNamePrefix)
+ char const* perFrameFileNamePrefix)
: MediaSink(env), fOutFid(fid), fBufferSize(bufferSize) {
fBuffer = new unsigned char[bufferSize];
if (perFrameFileNamePrefix != NULL) {
@@ -43,7 +43,7 @@
FileSink::~FileSink() {
delete[] fPerFrameFileNameBuffer;
- delete[] fPerFrameFileNamePrefix;
+ delete[] fPerFrameFileNamePrefix;
delete[] fBuffer;
if (fOutFid != NULL) fclose(fOutFid);
}
@@ -86,7 +86,7 @@
unsigned /*durationInMicroseconds*/) {
FileSink* sink = (FileSink*)clientData;
sink->afterGettingFrame1(frameSize, presentationTime);
-}
+}
void FileSink::addData(unsigned char* data, unsigned dataSize,
struct timeval presentationTime) {
@@ -125,7 +125,7 @@
stopPlaying();
return;
}
-
+
if (fPerFrameFileNameBuffer != NULL) {
if (fOutFid != NULL) { fclose(fOutFid); fOutFid = NULL; }
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/FramedFileSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Framed File Sources
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/FramedFilter.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Framed Filters
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/FramedSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Framed Sources
// Implementation
@@ -77,33 +77,6 @@
doGetNextFrame();
}
-// ##### The following is for backwards-compatibility; remove it eventually:
-#ifdef BACKWARDS_COMPATIBLE_WITH_OLD_AFTER_GETTING_FUNC
-static void bwCompatHackAfterGetting(void* clientData, unsigned frameSize,
- unsigned /*numTruncatedBytes*/,
- struct timeval presentationTime,
- unsigned /*durationInMicroseconds*/) {
- FramedSource* source = (FramedSource*)clientData;
- FramedSource::bwCompatAfterGettingFunc* clientAfterGettingFunc
- = source->fSavedBWCompatAfterGettingFunc;
- void* afterGettingClientData = source->fSavedBWCompatAfterGettingClientData;
- if (clientAfterGettingFunc != NULL) {
- (*clientAfterGettingFunc)(afterGettingClientData, frameSize, presentationTime);
- }
-}
-void FramedSource::getNextFrame(unsigned char* to, unsigned maxSize,
- bwCompatAfterGettingFunc* afterGettingFunc,
- void* afterGettingClientData,
- onCloseFunc* onCloseFunc,
- void* onCloseClientData) {
- fSavedBWCompatAfterGettingFunc = afterGettingFunc;
- fSavedBWCompatAfterGettingClientData = afterGettingClientData;
- // Call the regular (new) "getNextFrame()":
- getNextFrame(to, maxSize, bwCompatHackAfterGetting, this,
- onCloseFunc, onCloseClientData);
-}
-#endif
-// ##### End of code for backwards-compatibility.
void FramedSource::afterGetting(FramedSource* source) {
source->fIsCurrentlyAwaitingData = False;
@@ -143,4 +116,4 @@
unsigned FramedSource::maxFrameSize() const {
// By default, this source has no maximum frame size.
return 0;
-}
+}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/GSMAudioRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for GSM audio
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H261VideoRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// H.261 Video RTP Sources
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H263plusVideoFileServerMediaSubsession.cpp
^
|
| @@ -11,131 +11,54 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a H263 video file.
// Implementation
// Author: Bernhard Feiten. // Based on MPEG4VideoFileServerMediaSubsession
+// Updated by Ross FInlayson (December 2007)
#include "H263plusVideoFileServerMediaSubsession.hh"
#include "H263plusVideoRTPSink.hh"
#include "ByteStreamFileSource.hh"
-#include "H263plusVideoStreamFramer.hh"
+#include "H263plusVideoStreamFramer.hh"
-////////////////////////////////////////////////////////////////////////////////
H263plusVideoFileServerMediaSubsession*
-H263plusVideoFileServerMediaSubsession::createNew(
- UsageEnvironment& env,
- char const* fileName,
- Boolean reuseFirstSource) {
+H263plusVideoFileServerMediaSubsession::createNew(UsageEnvironment& env,
+ char const* fileName,
+ Boolean reuseFirstSource) {
return new H263plusVideoFileServerMediaSubsession(env, fileName, reuseFirstSource);
}
-////////////////////////////////////////////////////////////////////////////////
-H263plusVideoFileServerMediaSubsession::H263plusVideoFileServerMediaSubsession(
- UsageEnvironment& env,
- char const* fileName,
- Boolean reuseFirstSource)
- : FileServerMediaSubsession(env, fileName, reuseFirstSource),
- fDoneFlag(0)
-{
+H263plusVideoFileServerMediaSubsession
+::H263plusVideoFileServerMediaSubsession(UsageEnvironment& env,
+ char const* fileName,
+ Boolean reuseFirstSource)
+ : FileServerMediaSubsession(env, fileName, reuseFirstSource) {
}
-////////////////////////////////////////////////////////////////////////////////
-H263plusVideoFileServerMediaSubsession::~H263plusVideoFileServerMediaSubsession()
-{
+H263plusVideoFileServerMediaSubsession::~H263plusVideoFileServerMediaSubsession() {
}
-////////////////////////////////////////////////////////////////////////////////
-#if 0
-static void afterPlayingDummy(void* clientData) {
- H263plusVideoFileServerMediaSubsession* subsess
- = (H263plusVideoFileServerMediaSubsession*)clientData;
- // Signal the event loop that we're done:
- subsess->setDoneFlag();
-}
-#endif
-
-static void checkForAuxSDPLine(void* clientData) {
- H263plusVideoFileServerMediaSubsession* subsess
- = (H263plusVideoFileServerMediaSubsession*)clientData;
- subsess->checkForAuxSDPLine1();
-}
-
-void H263plusVideoFileServerMediaSubsession::checkForAuxSDPLine1() {
- if (fDummyRTPSink->auxSDPLine() != NULL) {
- // Signal the event loop that we're done:
- setDoneFlag();
- } else {
- // try again after a brief delay:
- int uSecsToDelay = 100000; // 100 ms
- nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecsToDelay,
- (TaskFunc*)checkForAuxSDPLine, this);
- }
-}
-
-char const* H263plusVideoFileServerMediaSubsession::getAuxSDPLine(
- RTPSink* rtpSink,
- FramedSource* inputSource)
-{
- // Note: For MPEG-4 video files, the 'config' information isn't known
- // until we start reading the file. This means that "rtpSink"s
- // "auxSDPLine()" will be NULL initially, and we need to start reading
- // data from our file until this changes.
- // Is this true also for H263 ?? >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
- // asumption: no aux line needed for H263
-
-// fDummyRTPSink = rtpSink;
-
- // Start reading the file:
-// fDummyRTPSink->startPlaying(*inputSource, afterPlayingDummy, this);
-
- // Check whether the sink's 'auxSDPLine()' is ready:
-// checkForAuxSDPLine(this);
-
-// envir().taskScheduler().doEventLoop(&fDoneFlag);
-
-// char const* auxSDPLine = fDummyRTPSink->auxSDPLine();
-// return auxSDPLine;
- return NULL;
-}
-
-////////////////////////////////////////////////////////////////////////////////
-FramedSource* H263plusVideoFileServerMediaSubsession::createNewStreamSource(
- unsigned /*clientSessionId*/,
- unsigned& estBitrate)
-{
+FramedSource* H263plusVideoFileServerMediaSubsession
+::createNewStreamSource(unsigned /*clientSessionId*/, unsigned& estBitrate) {
estBitrate = 500; // kbps, estimate ??
// Create the video source:
- ByteStreamFileSource* fileSource
- = ByteStreamFileSource::createNew(envir(), fFileName);
- if (fileSource == NULL) return NULL;
- fFileSize = fileSource->fileSize();
-
- // Create a framer for the Video Elementary Stream:
- return H263plusVideoStreamFramer::createNew(envir(), fileSource);
+ ByteStreamFileSource* fileSource = ByteStreamFileSource::createNew(envir(), fFileName);
+ if (fileSource == NULL) return NULL;
+ fFileSize = fileSource->fileSize();
+
+ // Create a framer for the Video Elementary Stream:
+ return H263plusVideoStreamFramer::createNew(envir(), fileSource);
}
-///////////////////////////////////////////////////////////////////////////////
-RTPSink* H263plusVideoFileServerMediaSubsession::createNewRTPSink(
- Groupsock* rtpGroupsock,
- unsigned char rtpPayloadTypeIfDynamic,
- FramedSource* /*inputSource*/)
-{
- unsigned char payloadFormatCode;
-
- if (false) // some more logic is needed if h263 is dynamic PT
- payloadFormatCode = rtpPayloadTypeIfDynamic;
- else
- payloadFormatCode = 34;
-
- return H263plusVideoRTPSink::createNew(envir(), rtpGroupsock, payloadFormatCode);
+RTPSink* H263plusVideoFileServerMediaSubsession::createNewRTPSink(Groupsock* rtpGroupsock,
+ unsigned char rtpPayloadTypeIfDynamic,
+ FramedSource* /*inputSource*/) {
+ return H263plusVideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic);
}
-
-
-
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H263plusVideoRTPSink.cpp
^
|
| @@ -11,11 +11,11 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
-// RTP sink for H.263+ video (RFC 2429)
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
+// RTP sink for H.263+ video (RFC 4629)
// Implementation
#include "H263plusVideoRTPSink.hh"
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H263plusVideoRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// H.263+ Video RTP Sources
// Implementation
@@ -34,7 +34,7 @@
unsigned rtpTimestampFrequency)
: MultiFramedRTPSource(env, RTPgs,
rtpPayloadFormat, rtpTimestampFrequency),
- fNumSpecialHeaders(0), fSpecialHeaderBytesLength(0) {
+ fNumSpecialHeaders(0), fSpecialHeaderBytesLength(0) {
}
H263plusVideoRTPSource::~H263plusVideoRTPSource() {
@@ -62,7 +62,7 @@
++expectedHeaderSize;
if (packetSize < expectedHeaderSize) return False;
}
-
+
if (PLEN > 0) {
// There's an extra picture header at the end:
expectedHeaderSize += PLEN;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H263plusVideoStreamFramer.cpp
^
|
| @@ -11,13 +11,13 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Author Bernhard Feiten
// A filter that breaks up an H.263plus video stream into frames.
-//
+//
#include "H263plusVideoStreamFramer.hh"
#include "H263plusVideoStreamParser.hh"
@@ -46,7 +46,7 @@
FramedSource* inputSource,
Boolean createParser)
: FramedFilter(env, inputSource),
- fFrameRate(0.0), // until we learn otherwise
+ fFrameRate(0.0), // until we learn otherwise
fPictureEndMarker(False)
{
// Use the current wallclock time as the base 'presentation time':
@@ -94,12 +94,12 @@
///////////////////////////////////////////////////////////////////////////////
void H263plusVideoStreamFramer::continueReadProcessing()
{
- unsigned acquiredFrameSize;
-
+ unsigned acquiredFrameSize;
+
u_int64_t frameDuration; // in ms
- acquiredFrameSize = fParser->parse(frameDuration);
-// Calculate some average bitrate information (to be adapted)
+ acquiredFrameSize = fParser->parse(frameDuration);
+// Calculate some average bitrate information (to be adapted)
// avgBitrate = (totalBytes * 8 * H263_TIMESCALE) / totalDuration;
if (acquiredFrameSize > 0) {
@@ -110,10 +110,10 @@
fFrameRate = frameDuration == 0 ? 0.0 : 1000./(long)frameDuration;
- // Compute "fPresentationTime"
+ // Compute "fPresentationTime"
if (acquiredFrameSize == 5) // first frame
fPresentationTime = fPresentationTimeBase;
- else
+ else
fPresentationTime.tv_usec += (long) frameDuration*1000;
while (fPresentationTime.tv_usec >= 1000000) {
@@ -121,7 +121,7 @@
++fPresentationTime.tv_sec;
}
- // Compute "fDurationInMicroseconds"
+ // Compute "fDurationInMicroseconds"
fDurationInMicroseconds = (unsigned int) frameDuration*1000;;
// Call our own 'after getting' function. Because we're not a 'leaf'
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H263plusVideoStreamParser.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Author Bernhard Feiten
// A filter that breaks up an H.263plus video stream into frames.
// Based on MPEG4IP/mp4creator/h263.c
@@ -104,10 +104,10 @@
// Parse 1 frame
// For the first time, only the first frame's header is returned.
// The second time the full first frame is returned
- frameSize = parseH263Frame();
+ frameSize = parseH263Frame();
currentDuration = 0;
- if ((frameSize > 0)){
+ if ((frameSize > 0)){
// We were able to acquire a frame from the input.
// Parse the returned frame header (if any)
@@ -129,6 +129,8 @@
// if (h263Bitrates)
//GetMaxBitrate(&fMaxBitrateCtx, frameSize, prevTrDifference);
//prevTrDifference = trDifference;
+
+ setParseState(); // Needed for the parsing process in StreamParser
}
} catch (int /*e*/) {
#ifdef DEBUG
@@ -137,14 +139,12 @@
frameSize=0;
}
- setParseState(); // Needed for the parsing process in StreamParser
-
return frameSize;
}
///////////////////////////////////////////////////////////////////////////////
-// parseH263Frame derived from LoadNextH263Object of MPEG4IP
+// parseH263Frame derived from LoadNextH263Object of MPEG4IP
// - service routine that reads a single frame from the input file.
// It shall fill the input buffer with data up until - and including - the
// next start code and shall report back both the number of bytes read and a
@@ -169,11 +169,11 @@
u_int8_t * bufferIndex = fTo;
// The buffer end which will allow the loop to leave place for
// the additionalBytesNeeded
- u_int8_t * bufferEnd = fTo + fMaxSize - ADDITIONAL_BYTES_NEEDED - 1;
+ u_int8_t * bufferEnd = fTo + fMaxSize - ADDITIONAL_BYTES_NEEDED - 1;
memcpy(fTo, fNextHeader, H263_REQUIRE_HEADER_SIZE_BYTES);
bufferIndex += H263_REQUIRE_HEADER_SIZE_BYTES;
-
+
// The state table and the following loop implements a state machine enabling
// us to read bytes from the file until (and inclusing) the requested
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H263plusVideoStreamParser.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an H263 video stream into frames.
// derived from MPEG4IP h263.c
// Author Benhard Feiten
@@ -77,8 +77,8 @@
virtual ~H263plusVideoStreamParser();
void registerReadInterest(unsigned char* to, unsigned maxSize);
-
- unsigned parse(u_int64_t & currentDuration); // returns the size of the frame that was acquired, or 0 if none
+
+ unsigned parse(u_int64_t & currentDuration); // returns the size of the frame that was acquired, or 0 if none
unsigned numTruncatedBytes() const { return fNumTruncatedBytes; } // The number of truncated bytes (if any)
@@ -92,7 +92,7 @@
private:
- int parseH263Frame( );
+ int parseH263Frame( );
bool ParseShortHeader(u_int8_t *headerBuffer, H263INFO *outputInfoStruct);
void GetMaxBitrate( MaxBitrate_CTX *ctx, u_int32_t frameSize, u_int8_t frameTRDiff);
u_int64_t CalculateDuration(u_int8_t trDiff);
@@ -117,7 +117,7 @@
H263INFO fCurrentInfo; // Holds information about the current frame
MaxBitrate_CTX fMaxBitrateCtx; // Context for the GetMaxBitrate function
char fStates[3][256];
- u_int8_t fNextHeader[H263_REQUIRE_HEADER_SIZE_BYTES];
+ u_int8_t fNextHeader[H263_REQUIRE_HEADER_SIZE_BYTES];
u_int32_t fnextTR; // The next frame's presentation time in TR units
u_int64_t fcurrentPT; // The current frame's presentation time in milli-seconds
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H264VideoFileSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// H.264 Video File sinks
// Implementation
@@ -25,7 +25,7 @@
H264VideoFileSink
::H264VideoFileSink(UsageEnvironment& env, FILE* fid, unsigned bufferSize,
- char const* perFrameFileNamePrefix)
+ char const* perFrameFileNamePrefix)
: FileSink(env, fid, bufferSize, perFrameFileNamePrefix) {
}
@@ -64,7 +64,7 @@
struct timeval presentationTime) {
unsigned char start_code[4] = {0x00, 0x00, 0x00, 0x01};
addData(start_code, 4, presentationTime);
-
+
// Call the parent class to complete the normal file write with the input data:
FileSink::afterGettingFrame1(frameSize, presentationTime);
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H264VideoRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for H.264 video (RFC 3984)
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H264VideoRTPSource.cpp
^
|
| @@ -11,12 +11,12 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// H.264 Video RTP Sources
-// Implementation
+// Implementation
#include "H264VideoRTPSource.hh"
#include "Base64.hh"
@@ -67,11 +67,11 @@
unsigned& resultSpecialHeaderSize) {
unsigned char* headerStart = packet->data();
unsigned packetSize = packet->dataSize();
-
+
// The header has a minimum size of 0, since the NAL header is used
// as a payload header
unsigned expectedHeaderSize = 0;
-
+
// Check if the type field is 28 (FU-A) or 29 (FU-B)
fCurPacketNALUnitType = (headerStart[0]&0x1F);
switch (fCurPacketNALUnitType) {
@@ -92,7 +92,7 @@
expectedHeaderSize = 1;
if (packetSize < expectedHeaderSize) return False;
- headerStart[1] = (headerStart[0]&0xE0)+(headerStart[1]&0x1F);
+ headerStart[1] = (headerStart[0]&0xE0)+(headerStart[1]&0x1F);
fCurrentPacketBeginsFrame = True;
} else {
// If the startbit is not set, both the FU indicator and header
@@ -110,7 +110,7 @@
break;
}
}
-
+
resultSpecialHeaderSize = expectedHeaderSize;
return True;
}
@@ -138,7 +138,7 @@
*s = '\0';
}
}
-
+
// Allocate and fill in the result array:
SPropRecord* resultArray = new SPropRecord[numSPropRecords];
s = inStr;
@@ -190,7 +190,7 @@
default: {
// Common case: We use the entire packet data:
return dataSize;
- }
+ }
}
return (resultNALUSize <= dataSize) ? resultNALUSize : dataSize;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/H264VideoStreamFramer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Any source that feeds into a "H264VideoRTPSink" must be of this class.
// This is a virtual base class; subclasses must implement the
// "currentNALUnitEndsAccessUnit()" virtual function.
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/HTTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// HTTP sinks
// Implementation
@@ -37,9 +37,9 @@
HTTPSink* newSink = new HTTPSink(env, ourSocket);
if (newSink == NULL) break;
-
+
appendPortNum(env, ourPort);
-
+
return newSink;
} while (0);
@@ -56,7 +56,7 @@
// Make sure we have a big send buffer:
if (!increaseSendBufferTo(env, ourSocket, 50*1024)) break;
-
+
if (listen(ourSocket, 1) < 0) { // we allow only one connection
env.setResultErrMsg("listen() failed: ");
break;
@@ -68,7 +68,7 @@
}
return ourSocket;
- } while (0);
+ } while (0);
if (ourSocket != -1) ::closeSocket(ourSocket);
return -1;
@@ -123,11 +123,11 @@
sprintf(okResponse, responseFmt, fSource->MIMEtype());
#else
snprintf(okResponse, sizeof okResponse, responseFmt, fSource->MIMEtype());
-#endif
+#endif
send(fClientSocket, okResponse, strlen(okResponse), 0);
}
}
-
+
fSource->getNextFrame(fBuffer, sizeof fBuffer,
afterGettingFrame, this,
ourOnSourceClosure, this);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/InputFile.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Common routines for opening/closing named input files
// Implementation
@@ -38,7 +38,7 @@
#if defined(__WIN32__) || defined(_WIN32)
_setmode(_fileno(stdin), _O_BINARY); // convert to binary mode
#endif
- } else {
+ } else {
fid = fopen(fileName, "rb");
if (fid == NULL) {
env.setResultMsg("unable to open file \"",fileName, "\"");
@@ -61,7 +61,7 @@
if (fileName == NULL) {
#endif
if (fid != NULL && SeekFile64(fid, 0, SEEK_END) >= 0) {
- fileSize = TellFile64(fid);
+ fileSize = (u_int64_t)TellFile64(fid);
if (fileSize == (u_int64_t)-1) fileSize = 0; // TellFile64() failed
SeekFile64(fid, 0, SEEK_SET);
}
@@ -78,7 +78,7 @@
return fileSize;
}
-u_int64_t SeekFile64(FILE *fid, int64_t offset, int whence) {
+int64_t SeekFile64(FILE *fid, int64_t offset, int whence) {
clearerr(fid);
fflush(fid);
#if (defined(__WIN32__) || defined(_WIN32)) && !defined(_WIN32_WCE)
@@ -92,7 +92,7 @@
#endif
}
-u_int64_t TellFile64(FILE *fid) {
+int64_t TellFile64(FILE *fid) {
clearerr(fid);
fflush(fid);
#if (defined(__WIN32__) || defined(_WIN32)) && !defined(_WIN32_WCE)
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/JPEGVideoRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for JPEG video (RFC 2435)
// Implementation
@@ -52,7 +52,7 @@
struct timeval frameTimestamp,
unsigned numRemainingBytes) {
// Our source is known to be a JPEGVideoSource
- JPEGVideoSource* source = (JPEGVideoSource*)fSource;
+ JPEGVideoSource* source = (JPEGVideoSource*)fSource;
if (source == NULL) return; // sanity check
u_int8_t mainJPEGHeader[8]; // the special header
@@ -73,7 +73,7 @@
u_int16_t length;
u_int8_t const* quantizationTables
= source->quantizationTables(precision, length);
-
+
unsigned const quantizationHeaderSize = 4 + length;
u_int8_t* quantizationHeader = new u_int8_t[quantizationHeaderSize];
@@ -105,7 +105,8 @@
unsigned JPEGVideoRTPSink::specialHeaderSize() const {
// Our source is known to be a JPEGVideoSource
- JPEGVideoSource* source = (JPEGVideoSource*)fSource;
+ JPEGVideoSource* source = (JPEGVideoSource*)fSource;
+ if (source == NULL) return 0; // sanity check
unsigned headerSize = 8; // by default
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/JPEGVideoRTPSource.cpp
^
|
| @@ -11,19 +11,12 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
-
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// JPEG Video (RFC 2435) RTP Sources
// Implementation
-// 09 26 2002 - Initial Implementation : Giom
-// Copyright (c) 1990-2002 Morgan Multimedia All rights reserved.
-
-// 02/2003: Cleaned up to add the synthesized JPEG header to the start
-// of each incoming frame.
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
#include "JPEGVideoRTPSource.hh"
@@ -55,18 +48,21 @@
JPEGVideoRTPSource*
JPEGVideoRTPSource::createNew(UsageEnvironment& env, Groupsock* RTPgs,
unsigned char rtpPayloadFormat,
- unsigned rtpTimestampFrequency) {
+ unsigned rtpTimestampFrequency,
+ unsigned defaultWidth, unsigned defaultHeight) {
return new JPEGVideoRTPSource(env, RTPgs, rtpPayloadFormat,
- rtpTimestampFrequency);
+ rtpTimestampFrequency, defaultWidth, defaultHeight);
}
JPEGVideoRTPSource::JPEGVideoRTPSource(UsageEnvironment& env,
Groupsock* RTPgs,
unsigned char rtpPayloadFormat,
- unsigned rtpTimestampFrequency)
+ unsigned rtpTimestampFrequency,
+ unsigned defaultWidth, unsigned defaultHeight)
: MultiFramedRTPSource(env, RTPgs,
rtpPayloadFormat, rtpTimestampFrequency,
- new JPEGBufferedPacketFactory) {
+ new JPEGBufferedPacketFactory),
+ fDefaultWidth(defaultWidth), fDefaultHeight(defaultHeight) {
}
JPEGVideoRTPSource::~JPEGVideoRTPSource() {
@@ -187,7 +183,7 @@
// MARKER_SOI:
*ptr++ = 0xFF; *ptr++ = MARKER_SOI;
-
+
// MARKER_APP_FIRST:
*ptr++ = 0xFF; *ptr++ = MARKER_APP_FIRST;
*ptr++ = 0x00; *ptr++ = 0x10; // size of chunk
@@ -197,14 +193,14 @@
*ptr++ = 0x00; *ptr++ = 0x01; // Horizontal pixel aspect ratio
*ptr++ = 0x00; *ptr++ = 0x01; // Vertical pixel aspect ratio
*ptr++ = 0x00; *ptr++ = 0x00; // no thumbnail
-
+
// MARKER_DRI:
if (dri > 0) {
*ptr++ = 0xFF; *ptr++ = MARKER_DRI;
*ptr++ = 0x00; *ptr++ = 0x04; // size of chunk
*ptr++ = (BYTE)(dri >> 8); *ptr++ = (BYTE)(dri); // restart interval
}
-
+
// MARKER_DQT (luma):
unsigned tableSize = numQtables == 1 ? qtlen : qtlen/2;
*ptr++ = 0xFF; *ptr++ = MARKER_DQT;
@@ -213,7 +209,7 @@
memcpy(ptr, qtables, tableSize);
qtables += tableSize;
ptr += tableSize;
-
+
if (numQtables > 1) {
unsigned tableSize = qtlen - qtlen/2;
// MARKER_DQT (chroma):
@@ -224,7 +220,7 @@
qtables += tableSize;
ptr += tableSize;
}
-
+
// MARKER_SOF0:
*ptr++ = 0xFF; *ptr++ = MARKER_SOF0;
*ptr++ = 0x00; *ptr++ = 0x11; // size of chunk
@@ -243,7 +239,7 @@
*ptr++ = 0x03; // id of component
*ptr++ = 0x11; // sampling ratio (h,v)
*ptr++ = 0x01; // quant table id
-
+
createHuffmanHeader(ptr, lum_dc_codelens, sizeof lum_dc_codelens,
lum_dc_symbols, sizeof lum_dc_symbols, 0, 0);
createHuffmanHeader(ptr, lum_ac_codelens, sizeof lum_ac_codelens,
@@ -340,8 +336,13 @@
unsigned type = Type & 1;
unsigned Q = (unsigned)headerStart[5];
unsigned width = (unsigned)headerStart[6] * 8;
- if (width == 0) width = 256*8; // special case
unsigned height = (unsigned)headerStart[7] * 8;
+ if ((width == 0 || height == 0) && fDefaultWidth != 0 && fDefaultHeight != 0) {
+ // Use the default width and height parameters instead:
+ width = fDefaultWidth;
+ height = fDefaultHeight;
+ }
+ if (width == 0) width = 256*8; // special case
if (height == 0) height = 256*8; // special case
if (Type > 63) {
@@ -386,11 +387,9 @@
if (packetSize < resultSpecialHeaderSize + Length) return False;
- if (qtables) delete [] qtables;
-
qtlen = Length;
qtables = &headerStart[resultSpecialHeaderSize];
-
+
resultSpecialHeaderSize += Length;
}
}
@@ -427,7 +426,7 @@
= fCurrentPacketCompletesFrame = packet->rtpMarkerBit();
return True;
-}
+}
char const* JPEGVideoRTPSource::MIMEtype() const {
return "video/JPEG";
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/JPEGVideoSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// JPEG video sources
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/Locale.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Support for temporarily setting the locale (e.g., to POSIX) for (e.g.) parsing or printing
// floating-point numbers in protocol headers, or calling toupper()/tolower() on human-input strings.
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3ADU.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// 'ADU' MP3 streams (for improved loss-tolerance)
// Implementation
@@ -89,7 +89,7 @@
struct timeval presentationTime,
unsigned durationInMicroseconds);
- Boolean sqAfterGettingCommon(Segment& seg, unsigned numBytesRead);
+ Boolean sqAfterGettingCommon(Segment& seg, unsigned numBytesRead);
Boolean isEmptyOrFull() {return headIndex() == nextFreeIndex();}
unsigned fHeadIndex, fNextFreeIndex, fTotalDataSize;
@@ -358,7 +358,7 @@
Segment* seg = &(fSegments->headSegment());
int const endOfHeadFrame = (int) seg->dataHere();
unsigned frameOffset = 0;
-
+
while (1) {
int endOfData = frameOffset - seg->backpointer + seg->aduSize;
if (endOfData >= endOfHeadFrame) {
@@ -366,14 +366,14 @@
needToEnqueue = False;
break;
}
-
+
frameOffset += seg->dataHere();
index = SegmentQueue::nextIndex(index);
if (index == fSegments->nextFreeIndex()) break;
seg = &(fSegments->s[index]);
}
}
-
+
return needToEnqueue;
}
@@ -385,11 +385,11 @@
// of the previous ADU, then we need to insert one or more
// empty, 'dummy' ADUs ahead of it. (This situation should occur
// only if an intermediate ADU was lost.)
-
+
unsigned tailIndex
= SegmentQueue::prevIndex(fSegments->nextFreeIndex());
Segment* tailSeg = &(fSegments->s[tailIndex]);
-
+
while (1) {
unsigned prevADUend; // relative to the start of the new ADU
if (fSegments->headIndex() != tailIndex) {
@@ -406,7 +406,7 @@
} else {
prevADUend = 0;
}
-
+
if (tailSeg->backpointer > prevADUend) {
// We need to insert a dummy ADU in front of the tail
#ifdef DEBUG
@@ -430,20 +430,20 @@
fprintf(stderr, "a->m:outputting frame for %d<-%d (fs %d, dh %d), (descriptorSize: %d)\n", seg->aduSize, seg->backpointer, seg->frameSize, seg->dataHere(), seg->descriptorSize);
#endif
unsigned char* toPtr = fTo;
-
+
// output header and side info:
fFrameSize = seg->frameSize;
fPresentationTime = seg->presentationTime;
fDurationInMicroseconds = seg->durationInMicroseconds;
memmove(toPtr, seg->dataStart(), seg->headerSize + seg->sideInfoSize);
toPtr += seg->headerSize + seg->sideInfoSize;
-
+
// zero out the rest of the frame, in case ADU data doesn't fill it all in
- unsigned bytesToZero = seg->dataHere();
+ unsigned bytesToZero = seg->dataHere();
for (unsigned i = 0; i < bytesToZero; ++i) {
toPtr[i] = '\0';
}
-
+
// Fill in the frame with appropriate ADU data from this and
// subsequent ADUs:
unsigned frameOffset = 0;
@@ -453,12 +453,12 @@
while (toOffset < endOfHeadFrame) {
int startOfData = frameOffset - seg->backpointer;
if (startOfData > (int)endOfHeadFrame) break; // no more ADUs needed
-
+
int endOfData = startOfData + seg->aduSize;
if (endOfData > (int)endOfHeadFrame) {
endOfData = endOfHeadFrame;
}
-
+
unsigned fromOffset;
if (startOfData <= (int)toOffset) {
fromOffset = toOffset - startOfData;
@@ -466,7 +466,7 @@
if (endOfData < startOfData) endOfData = startOfData;
} else {
fromOffset = 0;
-
+
// we may need some padding bytes beforehand
unsigned bytesToZero = startOfData - toOffset;
#ifdef DEBUG
@@ -474,7 +474,7 @@
#endif
toOffset += bytesToZero;
}
-
+
unsigned char* fromPtr
= &seg->dataStart()[seg->headerSize + seg->sideInfoSize + fromOffset];
unsigned bytesUsedHere = endOfData - startOfData;
@@ -483,13 +483,13 @@
#endif
memmove(toPtr + toOffset, fromPtr, bytesUsedHere);
toOffset += bytesUsedHere;
-
+
frameOffset += seg->dataHere();
index = SegmentQueue::nextIndex(index);
if (index == fSegments->nextFreeIndex()) break;
seg = &(fSegments->s[index]);
}
-
+
fSegments->dequeue();
return True;
@@ -505,7 +505,7 @@
}
return (unsigned)result;
-}
+}
////////// SegmentQueue //////////
@@ -516,9 +516,9 @@
FramedSource::handleClosure(usingSource);
return;
}
-
+
fUsingSource = usingSource;
-
+
Segment& seg = nextFreeSegment();
inputSource->getNextFrame(seg.buf, sizeof seg.buf,
sqAfterGettingSegment, this,
@@ -549,7 +549,7 @@
// Common code called after a new segment is enqueued
Boolean SegmentQueue::sqAfterGettingCommon(Segment& seg,
- unsigned numBytesRead) {
+ unsigned numBytesRead) {
unsigned char* fromPtr = seg.buf;
if (fIncludeADUdescriptors) {
@@ -570,7 +570,7 @@
seg.backpointer, seg.aduSize)) {
return False;
}
-
+
// If we've just read an ADU (rather than a regular MP3 frame), then use the
// entire "numBytesRead" data for the 'aduSize', so that we include any
// 'ancillary data' that may be present at the end of the ADU:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3ADURTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for 'ADUized' MP3 frames ("mpa-robust")
// Implementation
@@ -107,7 +107,7 @@
// Normally there's no special header.
// (The "ADU descriptor" is already present in the data.)
unsigned specialHeaderSize = 0;
-
+
// However, if we're about to output the second (or subsequent) fragment
// of a fragmented ADU, then we need to insert a new ADU descriptor at
// the front of the packet:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3ADURTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP source for 'ADUized' MP3 frames ("mpa-robust")
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3ADUTranscoder.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Transcoder for ADUized MP3 frames
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3ADUdescriptor.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Descriptor preceding frames of 'ADU' MP3 streams (for improved loss-tolerance)
// Implementation
@@ -59,7 +59,7 @@
return ((firstByte&0x3F)<<8) | secondByte;
} else {
// This is a 1-byte descriptor
- return (firstByte&0x3F);
+ return (firstByte&0x3F);
}
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3ADUdescriptor.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Descriptor preceding frames of 'ADU' MP3 streams (for improved loss-tolerance)
// C++ header
@@ -28,7 +28,7 @@
// Operations for generating a new descriptor
static unsigned computeSize(unsigned remainingFrameSize) {
return remainingFrameSize >= 64 ? 2 : 1;
- }
+ }
static unsigned generateDescriptor(unsigned char*& toPtr, unsigned remainingFrameSize);
// returns descriptor size; increments "toPtr" afterwards
static void generateTwoByteDescriptor(unsigned char*& toPtr, unsigned remainingFrameSize);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3ADUinterleaving.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Interleaving of MP3 ADUs
// Implementation
@@ -54,12 +54,12 @@
FramedSource* inputSource;
if (!FramedSource::lookupByName(env, inputSourceName, inputSource))
return NULL;
-
+
if (strcmp(inputSource->MIMEtype(), "audio/MPA-ROBUST") != 0) {
env.setResultMsg(inputSourceName, " is not an MP3 ADU source");
return NULL;
}
-
+
return inputSource;
}
@@ -188,7 +188,7 @@
public:
DeinterleavingFrames();
virtual ~DeinterleavingFrames();
-
+
Boolean haveReleaseableFrame();
void getIncomingFrameParams(unsigned char*& dataPtr,
unsigned& bytesAvailable);
@@ -222,6 +222,7 @@
}
MP3ADUdeinterleaver::~MP3ADUdeinterleaver() {
+ delete fFrames;
}
MP3ADUdeinterleaver* MP3ADUdeinterleaver::createNew(UsageEnvironment& env,
@@ -241,7 +242,7 @@
} else {
#ifdef TEST_LOSS
NOTE: This code no longer works, because it uses synchronous reads,
- which are no longer supported.
+ which are no longer supported.
static unsigned const framesPerPacket = 3;
static unsigned const frameCount = 0;
static Boolean packetIsLost;
@@ -286,7 +287,7 @@
icc, ii);
// Compare these to the values we saw last:
- if (icc != fICClastSeen || ii == fIIlastSeen) {
+ if (icc != fICClastSeen || ii == fIIlastSeen) {
// We've started a new interleave cycle
// (or interleaving was not used). Release all
// pending ADU frames to the ADU->MP3 conversion step:
@@ -465,7 +466,7 @@
desc.frameDataSize = frameSize;
desc.presentationTime = presentationTime;
desc.durationInMicroseconds = durationInMicroseconds;
-
+
// Advance over the ADU descriptor, to get to the MPEG 'syncword':
unsigned char* ptr = desc.frameData;
(void)ADUdescriptor::getRemainingFrameSize(ptr);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3AudioFileServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MP3 audio file.
// (Actually, any MPEG-1 or MPEG-2 audio file should work.)
@@ -48,7 +48,7 @@
}
void MP3AudioFileServerMediaSubsession
-::seekStreamSource(FramedSource* inputSource, float seekNPT) {
+::seekStreamSource(FramedSource* inputSource, double seekNPT) {
MP3FileSource* mp3Source;
if (fGenerateADUs) {
// "inputSource" is a filter; use its input source instead.
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3FileSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 File Sources
// Implementation
@@ -68,7 +68,7 @@
fStreamState->setPresentationTimeScale(scale);
}
-void MP3FileSource::seekWithinFile(float seekNPT) {
+void MP3FileSource::seekWithinFile(double seekNPT) {
fStreamState->seekWithinFile(seekNPT);
}
@@ -106,7 +106,7 @@
fPresentationTime = fFirstFramePresentationTime;
fHaveJustInitialized = False;
}
-
+
if (!fStreamState->readFrame(fTo, fMaxSize, fFrameSize, fDurationInMicroseconds)) {
char tmp[200];
sprintf(tmp,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3HTTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 HTTP Sources
// Implementation
@@ -47,7 +47,7 @@
// Make sure we have a big receive buffer:
if (!increaseReceiveBufferTo(env, ourSocket, 100*1024)) break;
-
+
// Try to make the new socket into a FILE*:
unsigned streamLength = 0; //#####
FILE* fid = NULL;
@@ -65,7 +65,7 @@
newSource = new MP3HTTPSource(env, fid);
if (newSource == NULL) break;
-
+
newSource->assignStream(fid, streamLength);
// Write the HTTP 'GET' command:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3Internals.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 internal implementation details
// Implementation
@@ -48,7 +48,7 @@
int shortDiff[13];
};
-static struct bandInfoStruct const bandInfo[7] = {
+static struct bandInfoStruct const bandInfo[7] = {
/* MPEG 1.0 */
{ {0,4,8,12,16,20,24,30,36,44,52,62,74, 90,110,134,162,196,238,288,342,418,576},
{4,4,4,4,4,4,6,6,8, 8,10,12,16,20,24,28,34,42,50,54, 76,158},
@@ -91,7 +91,7 @@
unsigned int n_slen2[512]; /* MPEG 2.0 slen for 'normal' mode */
unsigned int i_slen2[256]; /* MPEG 2.0 slen for intensity stereo */
-#define MPG_MD_MONO 3
+#define MPG_MD_MONO 3
////////// MP3FrameParams //////////
@@ -162,19 +162,19 @@
isMPEG2 = 1;
isMPEG2_5 = 1;
}
-
+
layer = 4-((hdr>>17)&3);
if (layer == 4) layer = 3; // layer==4 is not allowed
bitrateIndex = ((hdr>>12)&0xf);
-
+
if (isMPEG2_5) {
samplingFreqIndex = ((hdr>>10)&0x3) + 6;
} else {
samplingFreqIndex = ((hdr>>10)&0x3) + (isMPEG2*3);
}
-
+
hasCRC = ((hdr>>16)&0x1)^0x1;
-
+
padding = ((hdr>>9)&0x1);
extension = ((hdr>>8)&0x1);
mode = ((hdr>>6)&0x3);
@@ -182,9 +182,9 @@
copyright = ((hdr>>3)&0x1);
original = ((hdr>>2)&0x1);
emphasis = hdr & 0x3;
-
+
stereo = (mode == MPG_MD_MONO) ? 1 : 2;
-
+
if (((hdr>>10)&0x3) == 0x3) {
#ifdef DEBUG_ERRORS
fprintf(stderr,"Stream error - hdr: 0x%08x\n", hdr);
@@ -202,7 +202,7 @@
unsigned MP3FrameParams::computeSideInfoSize() {
unsigned size;
-
+
if (isMPEG2) {
size = isStereo ? 17 : 9;
} else {
@@ -277,7 +277,7 @@
fprintf(stderr, "updateSideInfoSizes (allowed: %d): %d->%d, %d->%d\n", allowedNumBits, p23L0+p23L0Trunc, p23L0, p23L1+p23L1Trunc, p23L1);
#endif
- // The truncations computed above are still estimates. We need to
+ // The truncations computed above are still estimates. We need to
// adjust them so that the new fields will continue to end on
// Huffman-encoded sample boundaries:
updateSideInfoForHuffman(sideInfo, isMPEG2, mainDataPtr,
@@ -358,7 +358,7 @@
si.main_data_begin = fr.getBits(9);
if (stereo == 1)
si.private_bits = fr.getBits(5);
- else
+ else
si.private_bits = fr.getBits(3);
for (ch=0; ch<stereo; ch++) {
@@ -402,11 +402,11 @@
fprintf(stderr,"Blocktype == 0 and window-switching == 1 not allowed.\n");
}
#endif
- /* region_count/start parameters are implicit in this case. */
+ /* region_count/start parameters are implicit in this case. */
gr_info.region1start = 36>>1;
gr_info.region2start = 576>>1;
}
- else
+ else
{
int i,r0c,r1c;
for (i=0; i<3; i++) {
@@ -441,7 +441,7 @@
si.main_data_begin = fr.getBits(8);
if (stereo == 1)
si.private_bits = fr.get1Bit();
- else
+ else
si.private_bits = fr.getBits(2);
for (ch=0; ch<stereo; ch++) {
@@ -481,7 +481,7 @@
fprintf(stderr,"Blocktype == 0 and window-switching == 1 not allowed.\n");
}
#endif
- /* region_count/start parameters are implicit in this case. */
+ /* region_count/start parameters are implicit in this case. */
/* check this again! */
if (gr_info.block_type == 2)
gr_info.region1start = 36>>1;
@@ -490,7 +490,7 @@
}
gr_info.region2start = 576>>1;
}
- else
+ else
{
int i,r0c,r1c;
for (i=0; i<3; i++) {
@@ -537,17 +537,17 @@
MP3SideInfo const& si, Boolean isStereo) {
int ch, gr, i;
int stereo = isStereo ? 2 : 1;
-
+
bv.putBits(si.main_data_begin,9);
if (stereo == 1)
bv.putBits(si.private_bits, 5);
else
bv.putBits(si.private_bits, 3);
-
+
for (ch=0; ch<stereo; ch++) {
bv.putBits(si.ch[ch].gr[1].scfsi, 4);
}
-
+
for (gr=0; gr<2; gr++) {
for (ch=0; ch<stereo; ch++) {
MP3SideInfo::gr_info_s_t const& gr_info = si.ch[ch].gr[gr];
@@ -570,8 +570,8 @@
bv.putBits(gr_info.table_select[i], 5);
bv.putBits(gr_info.region0_count, 4);
bv.putBits(gr_info.region1_count, 3);
- }
-
+ }
+
bv.put1Bit(gr_info.preflag);
bv.put1Bit(gr_info.scalefac_scale);
bv.put1Bit(gr_info.count1table_select);
@@ -583,16 +583,16 @@
MP3SideInfo const& si, Boolean isStereo) {
int ch, i;
int stereo = isStereo ? 2 : 1;
-
+
bv.putBits(si.main_data_begin,8);
if (stereo == 1)
bv.put1Bit(si.private_bits);
else
bv.putBits(si.private_bits, 2);
-
+
for (ch=0; ch<stereo; ch++) {
MP3SideInfo::gr_info_s_t const& gr_info = si.ch[ch].gr[0];
-
+
bv.putBits(gr_info.part2_3_length, 12);
bv.putBits(gr_info.big_values, 9);
bv.putBits(gr_info.global_gain, 8);
@@ -611,8 +611,8 @@
bv.putBits(gr_info.table_select[i], 5);
bv.putBits(gr_info.region0_count, 4);
bv.putBits(gr_info.region1_count, 3);
- }
-
+ }
+
bv.put1Bit(gr_info.scalefac_scale);
bv.put1Bit(gr_info.count1table_select);
}
@@ -670,7 +670,7 @@
// "bitrate" was larger than any possible, so return the largest possible:
return 14;
-}
+}
static void outputHeader(unsigned char* toPtr, unsigned hdr) {
toPtr[0] = (unsigned char)(hdr>>24);
@@ -688,7 +688,7 @@
unsigned backpointerSize = availableBytesForBackpointer;
if (backpointerSize > maxBackpointerSize) {
- backpointerSize = maxBackpointerSize;
+ backpointerSize = maxBackpointerSize;
}
// Store the new backpointer now:
@@ -780,22 +780,22 @@
// 'main data', using the new lengths
unsigned toBitOffset = 0;
unsigned fromBitOffset = 0;
-
+
/* rebuild portion 0a: */
memmove(toPtr, fromPtr, (part23Length0a+7)/8);
toBitOffset += part23Length0a;
fromBitOffset += part23Length0a + part23Length0aTruncation;
-
+
/* rebuild portion 0b: */
shiftBits(toPtr, toBitOffset, fromPtr, fromBitOffset, part23Length0b);
toBitOffset += part23Length0b;
fromBitOffset += part23Length0b + part23Length0bTruncation;
-
+
/* rebuild portion 1a: */
shiftBits(toPtr, toBitOffset, fromPtr, fromBitOffset, part23Length1a);
toBitOffset += part23Length1a;
fromBitOffset += part23Length1a + part23Length1aTruncation;
-
+
/* rebuild portion 1b: */
shiftBits(toPtr, toBitOffset, fromPtr, fromBitOffset, part23Length1b);
toBitOffset += part23Length1b;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3Internals.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 internal implementation details
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3InternalsHuffman.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 internal implementation details (Huffman encoding)
// Implementation
@@ -175,15 +175,15 @@
part23Length0bTruncation += adjustment;
/* Assign the bits we just shaved to granule 1 */
p23L1 += adjustment;
-
+
if (part23Length0aTruncation > 0) {
/* Change the granule's 'big_values' field to reflect the truncation */
gr->big_values = i;
}
- }
-
+ }
+
/* Process granule 1 (MPEG-1 only) */
-
+
if (isMPEG2) {
part23Length1a = part23Length1b = 0;
part23Length1aTruncation = part23Length1bTruncation = 0;
@@ -294,7 +294,7 @@
if (adjustment > part23Length1b) adjustment = part23Length1b; /*sanity*/
part23Length1b -= adjustment;
part23Length1bTruncation += adjustment;
-
+
if (part23Length1aTruncation > 0) {
/* Change the granule's 'big_values' field to reflect the truncation */
gr->big_values = i;
@@ -341,14 +341,14 @@
struct huffcodetab {
char tablename[3]; /*string, containing table_description */
- unsigned int xlen; /*max. x-index+ */
+ unsigned int xlen; /*max. x-index+ */
unsigned int ylen; /*max. y-index+ */
unsigned int linbits; /*number of linbits */
unsigned int linmax; /*max number to be stored in linbits */
int ref; /*a positive value indicates a reference*/
HUFFBITS *table; /*pointer to array[xlen][ylen] */
unsigned char *hlen; /*pointer to array[xlen][ylen] */
- unsigned char(*val)[2];/*decoder tree */
+ unsigned char(*val)[2];/*decoder tree */
unsigned int treelen; /*length of decoder tree */
};
@@ -357,7 +357,7 @@
/* 32,33 count1-tables */
/* read the huffman decoder table */
-static int read_decoder_table(unsigned char* fi) {
+static int read_decoder_table(unsigned char* fi) {
int n,i,nn,t;
unsigned int v0,v1;
char command[100],line[100];
@@ -365,11 +365,11 @@
rsf_ht[n].table = NULL;
rsf_ht[n].hlen = NULL;
- /* .table number treelen xlen ylen linbits */
+ /* .table number treelen xlen ylen linbits */
do {
rsf_getline(line,99,&fi);
} while ((line[0] == '#') || (line[0] < ' '));
-
+
sscanf(line,"%s %s %u %u %u %u",command,rsf_ht[n].tablename,
&rsf_ht[n].treelen, &rsf_ht[n].xlen, &rsf_ht[n].ylen, &rsf_ht[n].linbits);
if (strcmp(command,".end")==0)
@@ -381,14 +381,14 @@
return -1;
}
rsf_ht[n].linmax = (1<<rsf_ht[n].linbits)-1;
-
+
sscanf(rsf_ht[n].tablename,"%u",&nn);
if (nn != n) {
#ifdef DEBUG
fprintf(stderr,"wrong table number %u\n",n);
#endif
return(-2);
- }
+ }
do {
rsf_getline(line,99,&fi);
} while ((line[0] == '#') || (line[0] < ' '));
@@ -408,10 +408,10 @@
while ((line[0] == '#') || (line[0] < ' ') ) {
rsf_getline(line,99,&fi);
}
- }
+ }
else if (strcmp(command,".treedata")==0) {
rsf_ht[n].ref = -1;
- rsf_ht[n].val = (unsigned char (*)[2])
+ rsf_ht[n].val = (unsigned char (*)[2])
new unsigned char[2*(rsf_ht[n].treelen)];
if ((rsf_ht[n].val == NULL) && ( rsf_ht[n].treelen != 0 )){
#ifdef DEBUG
@@ -463,14 +463,14 @@
{12,12,12,0 } , {12, 9, 9,6 } , { 15,12,9,0} } ,
{ { 6, 9, 9,9 } , { 6, 9,12,6 } , { 15,18,0,0} ,
{ 6,15,12,0 } , { 6,12, 9,6 } , { 6,18,9,0} }
-};
+};
static unsigned rsf_get_scale_factors_1(MP3SideInfo::gr_info_s_t *gr_info) {
int numbits;
int num0 = slen[0][gr_info->scalefac_compress];
int num1 = slen[1][gr_info->scalefac_compress];
- if (gr_info->block_type == 2)
+ if (gr_info->block_type == 2)
{
numbits = (num0 + num1) * 18;
@@ -478,7 +478,7 @@
numbits -= num0; /* num0 * 17 + num1 * 18 */
}
}
- else
+ else
{
int scfsi = gr_info->scfsi;
@@ -535,7 +535,7 @@
gr_info->preflag = (slen>>15) & 0x1;
- n = 0;
+ n = 0;
if( gr_info->block_type == 2 ) {
n++;
if(gr_info->mixed_block_flag)
@@ -602,7 +602,7 @@
hei.reg2Start = bv.curBitIndex();
}
}
-
+
hei.allBitOffsets[i] = bv.curBitIndex();
rsf_huffman_decoder(bv, h, &x, &y, &v, &w);
if (hei.decodedValues != NULL) {
@@ -612,7 +612,7 @@
}
}
hei.bigvalStart = bv.curBitIndex();
-
+
/* Read count1 area. */
h = &rsf_ht[gr->count1table_select+32];
while (bv.curBitIndex() < bv.totNumBits() && i < SSLIMIT*SBLIMIT) {
@@ -658,21 +658,21 @@
error = 0;
break;
- }
+ }
if (bv.get1Bit()) {
- while (h->val[point][1] >= MXOFF) point += h->val[point][1];
+ while (h->val[point][1] >= MXOFF) point += h->val[point][1];
point += h->val[point][1];
}
else {
- while (h->val[point][0] >= MXOFF) point += h->val[point][0];
+ while (h->val[point][0] >= MXOFF) point += h->val[point][0];
point += h->val[point][0];
}
level >>= 1;
} while (level || (point < h->treelen) );
///// } while (level || (point < rsf_ht->treelen) );
-
+
/* Check for error. */
-
+
if (error) { /* set x and y to a medium value as a simple concealment */
printf("Illegal Huffman code in data.\n");
*x = ((h->xlen-1) << 1);
@@ -697,23 +697,23 @@
if (*y)
if (bv.get1Bit() == 1) *y = -*y;
}
-
+
/* Process sign and escape encodings for dual tables. */
-
+
else {
if (h->linbits)
- if ((h->xlen-1) == (unsigned)*x)
+ if ((h->xlen-1) == (unsigned)*x)
*x += bv.getBits(h->linbits);
if (*x)
if (bv.get1Bit() == 1) *x = -*x;
- if (h->linbits)
+ if (h->linbits)
if ((h->ylen-1) == (unsigned)*y)
*y += bv.getBits(h->linbits);
if (*y)
if (bv.get1Bit() == 1) *y = -*y;
}
-
- return error;
+
+ return error;
}
#ifdef DO_HUFFMAN_ENCODING
@@ -764,14 +764,14 @@
/* in region 2 */
h = &rsf_ht[gr->table_select[2]];
}
-
+
x = getNextSample(fromPtr);
y = getNextSample(fromPtr);
v = getNextSample(fromPtr);
w = getNextSample(fromPtr);
rsf_huffman_encoder(bv, h, x, y, v, w);
}
-
+
// Encode count1 area:
h = &rsf_ht[gr->count1table_select+32];
while (bv.curBitIndex() < bv.totNumBits() && i < SSLIMIT*SBLIMIT) {
@@ -803,16 +803,16 @@
return False;
}
}
-
+
if (numBitsTestedSoFar++ == bitsLength) {
// We don't yet have enough bits for this prefix
return False;
}
if (bits&mask) {
- while (h->val[point][1] >= MXOFF) point += h->val[point][1];
+ while (h->val[point][1] >= MXOFF) point += h->val[point][1];
point += h->val[point][1];
} else {
- while (h->val[point][0] >= MXOFF) point += h->val[point][0];
+ while (h->val[point][0] >= MXOFF) point += h->val[point][0];
point += h->val[point][0];
}
mask <<= 1;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3InternalsHuffman.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 internal implementation details (Huffman encoding)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3InternalsHuffmanTable.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 internal implementation details (Huffman encoding)
// Table
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3StreamState.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A class encapsulating the state of a MP3 stream
// Implementation
@@ -66,7 +66,7 @@
// result is numSamples/freq
unsigned const uSeconds
- = ((numSamples*2*MILLION)/freq + 1)/2; // rounds to nearest integer
+ = ((numSamples*2*MILLION)/freq + 1)/2; // rounds to nearest integer
struct timeval result;
result.tv_sec = uSeconds/MILLION;
@@ -86,7 +86,7 @@
return numFramesInFile*(pt.tv_sec + pt.tv_usec/(float)MILLION);
}
-void MP3StreamState::seekWithinFile(float seekNPT) {
+void MP3StreamState::seekWithinFile(double seekNPT) {
if (fFidIsReallyASocket) return; // it's not seekable
float fileDuration = filePlayTime();
@@ -95,7 +95,7 @@
} else if (seekNPT > fileDuration) {
seekNPT = fileDuration;
}
- float seekFraction = seekNPT/fileDuration;
+ float seekFraction = (float)seekNPT/fileDuration;
unsigned seekByteNumber;
if (fHasXingTOC) {
@@ -226,7 +226,7 @@
unsigned char hbuf[8];
unsigned l; int i;
int attempt = 0;
-
+
#ifdef DEBUGGING_INPUT
/* use this debugging code to generate a copy of the input stream */
FILE* fout;
@@ -238,15 +238,15 @@
fclose(fout);
exit(0);
#endif
-
+
read_again:
if (readFromStream(hbuf, 4) != 4) return False;
-
+
fr().hdr = ((unsigned long) hbuf[0] << 24)
| ((unsigned long) hbuf[1] << 16)
| ((unsigned long) hbuf[2] << 8)
| (unsigned long) hbuf[3];
-
+
#ifdef DEBUG_PARSE
fprintf(stderr, "fr().hdr: 0x%08x\n", fr().hdr);
#endif
@@ -314,7 +314,7 @@
fprintf(stderr,"Giving up searching valid MPEG header\n");
#endif
return False;
-
+
#ifdef DEBUG_ERRORS
fprintf(stderr,"Illegal Audio-MPEG-Header 0x%08lx at offset 0x%lx.\n",
fr().hdr,tell_stream(str)-4);
@@ -330,13 +330,13 @@
if (readFromStream(&hbuf[3],1) != 1) {
return False;
}
-
+
/* This is faster than combining fr().hdr from scratch */
fr().hdr = ((fr().hdr << 8) | hbuf[3]) & 0xffffffff;
-
+
if (!fr().oldHdr)
goto init_resync; /* "considered harmful", eh? */
-
+
} while ((fr().hdr & HDRCMPMASK) != (fr().oldHdr & HDRCMPMASK)
&& (fr().hdr & HDRCMPMASK) != (fr().firstHdr & HDRCMPMASK));
#ifdef DEBUG_ERRORS
@@ -346,19 +346,19 @@
if (!fr().firstHdr) {
fr().firstHdr = fr().hdr;
}
-
+
fr().setParamsFromHeader();
fr().setBytePointer(fr().frameBytes, fr().frameSize);
-
+
fr().oldHdr = fr().hdr;
-
+
if (fr().isFreeFormat) {
#ifdef DEBUG_ERRORS
fprintf(stderr,"Free format not supported.\n");
#endif
return False;
}
-
+
#ifdef MP3_ONLY
if (fr().layer != 3) {
#ifdef DEBUG_ERRORS
@@ -368,13 +368,13 @@
}
#endif
}
-
+
if ((l = readFromStream(fr().frameBytes, fr().frameSize))
!= fr().frameSize) {
if (l == 0) return False;
memset(fr().frameBytes+1, 0, fr().frameSize-1);
}
-
+
return True;
}
@@ -467,7 +467,7 @@
}
if (flags&XING_TOC_FLAG) {
- // Fill in the Xing 'table of contents':
+ // Fill in the Xing 'table of contents':
if (bytesAvailable < XING_TOC_LENGTH) return;
fHasXingTOC = True;
for (unsigned j = 0; j < XING_TOC_LENGTH; ++j) {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3StreamState.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A class encapsulating the state of a MP3 stream
// C++ header
@@ -58,7 +58,7 @@
float filePlayTime() const; // in seconds
void setPresentationTimeScale(unsigned scale) { fPresentationTimeScale = scale; }
- void seekWithinFile(float seekNPT);
+ void seekWithinFile(double seekNPT);
void checkForXingHeader(); // hack for Xing VBR files
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MP3Transcoder.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 Transcoder
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2AudioRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for MPEG audio (RFC 2250)
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2AudioRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG-1 or MPEG-2 Audio RTP Sources
// Implementation
@@ -54,7 +54,7 @@
resultSpecialHeaderSize = 4;
return True;
-}
+}
char const* MPEG1or2AudioRTPSource::MIMEtype() const {
return "audio/MPEG";
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2AudioStreamFramer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an MPEG (1,2) audio elementary stream into frames
// Implementation
@@ -136,7 +136,7 @@
// We were able to acquire a frame from the input.
// It has already been copied to the reader's space.
fFrameSize = acquiredFrameSize;
-
+
// Also set the presentation time, and increment it for next time,
// based on the length of this frame:
fPresentationTime = fNextFramePresentationTime;
@@ -179,15 +179,15 @@
unsigned MPEG1or2AudioStreamParser::parse(unsigned& numTruncatedBytes) {
try {
saveParserState();
-
+
// We expect a MPEG audio header (first 11 bits set to 1) at the start:
while (((fCurrentFrame.hdr = test4Bytes())&0xFFE00000) != 0xFFE00000) {
skipBytes(1);
saveParserState();
}
-
+
fCurrentFrame.setParamsFromHeader();
-
+
// Copy the frame to the requested destination:
unsigned frameSize = fCurrentFrame.frameSize + 4; // include header
if (frameSize > fMaxSize) {
@@ -199,7 +199,7 @@
getBytes(fTo, frameSize);
skipBytes(numTruncatedBytes);
-
+
return frameSize;
} catch (int /*e*/) {
#ifdef DEBUG
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2Demux.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Demultiplexer for a MPEG 1 or 2 Program Stream
// Implementation
@@ -30,7 +30,7 @@
PARSING_PACK_HEADER,
PARSING_SYSTEM_HEADER,
PARSING_PES_PACKET
-};
+};
class MPEGProgramStreamParser: public StreamParser {
public:
@@ -92,7 +92,7 @@
fOutput[i].isPotentiallyReadable = False;
fOutput[i].isCurrentlyActive = False;
fOutput[i].isCurrentlyAwaitingData = False;
- }
+ }
}
MPEG1or2Demux::~MPEG1or2Demux() {
@@ -157,14 +157,14 @@
FramedSource::onCloseFunc* onCloseFunc,
void* onCloseClientData) {
struct OutputDescriptor& out = fOutput[streamIdTag];
-
+
// Make sure this stream is not already being read:
if (out.isCurrentlyAwaitingData) {
envir() << "MPEG1or2Demux::registerReadInterest(): attempt to read stream id "
<< (void*)streamIdTag << " more than once!\n";
exit(1);
}
-
+
out.to = to; out.maxSize = maxSize;
out.fAfterGettingFunc = afterGettingFunc;
out.afterGettingClientData = afterGettingClientData;
@@ -229,7 +229,7 @@
if (acquiredStreamIdTag != 0) {
// We were able to acquire a frame from the input.
- struct OutputDescriptor& newOut = fOutput[acquiredStreamIdTag];
+ struct OutputDescriptor& newOut = fOutput[acquiredStreamIdTag];
newOut.isCurrentlyAwaitingData = False;
// indicates that we can be read again
// (This needs to be set before the 'after getting' call below,
@@ -311,7 +311,7 @@
out.savedDataTotalSize = 0;
out.isPotentiallyReadable = out.isCurrentlyActive = out.isCurrentlyAwaitingData
= False;
- }
+ }
for (i = 0; i < numPending; ++i) {
(*savedPending[i].fOnCloseFunc)(savedPending[i].onCloseClientData);
}
@@ -411,9 +411,9 @@
skipBytes(1);
}
}
-
+
// The size of the pack header differs depending on whether it's
- // MPEG-1 or MPEG-2. The next byte tells us this:
+ // MPEG-1 or MPEG-2. The next byte tells us this:
unsigned char nextByte = get1Byte();
MPEG1or2Demux::SCR& scr = fUsingSource->fLastSeenSCR; // alias
if ((nextByte&0xF0) == 0x20) { // MPEG-1
@@ -460,10 +460,10 @@
<< (void*)nextByte
<< " following pack_start_code\n";
}
-
+
// Check for a System Header next:
setParseState(PARSING_SYSTEM_HEADER);
-}
+}
void MPEGProgramStreamParser::parseSystemHeader() {
#ifdef DEBUG
@@ -475,14 +475,14 @@
setParseState(PARSING_PES_PACKET);
return;
}
-
+
#ifdef DEBUG
fprintf(stderr, "saw system_header_start_code\n"); fflush(stderr);
#endif
skipBytes(4); // we've already seen the system_header_start_code
-
+
unsigned short remaining_header_length = get2Bytes();
-
+
// According to the MPEG-1 and MPEG-2 specs, "remaining_header_length" should be
// at least 6 bytes. Check this now:
if (remaining_header_length < 6) {
@@ -490,7 +490,7 @@
<< remaining_header_length << " < 6\n";
}
skipBytes(remaining_header_length);
-
+
// Check for a PES Packet next:
setParseState(PARSING_PES_PACKET);
}
@@ -517,7 +517,7 @@
}
}
-#define READER_NOT_READY 2
+#define READER_NOT_READY 2
unsigned char MPEGProgramStreamParser::parsePESPacket() {
#ifdef DEBUG
@@ -529,7 +529,7 @@
setParseState(PARSING_PACK_HEADER);
return 0;
}
-
+
#ifdef DEBUG
fprintf(stderr, "saw packet_start_code_prefix\n"); fflush(stderr);
#endif
@@ -561,7 +561,7 @@
static unsigned frameCount = 1;
fprintf(stderr, "%d, saw %s stream: 0x%02x\n", frameCount, streamTypeStr, streamNum); fflush(stderr);
#endif
-
+
unsigned short PES_packet_length = get2Bytes();
#ifdef DEBUG
fprintf(stderr, "PES_packet_length: %d\n", PES_packet_length); fflush(stderr);
@@ -572,7 +572,7 @@
if (fUsingSource->fOutput[RAW_PES].isPotentiallyReadable) {
// Hack: We've been asked to return raw PES packets, for every stream:
stream_id = RAW_PES;
- }
+ }
unsigned savedParserOffset = curOffset();
#ifdef DEBUG_TIMESTAMPS
unsigned char pts_highBit = 0;
@@ -691,12 +691,12 @@
} else {
PES_packet_length -= bytesSkipped;
#ifdef DEBUG
- unsigned next4Bytes = test4Bytes();
+ unsigned next4Bytes = test4Bytes();
#endif
// Check whether our using source is interested in this stream type.
// If so, deliver the frame to him:
- MPEG1or2Demux::OutputDescriptor_t& out = fUsingSource->fOutput[stream_id];
+ MPEG1or2Demux::OutputDescriptor_t& out = fUsingSource->fOutput[stream_id];
if (out.isCurrentlyAwaitingData) {
unsigned numBytesToCopy;
if (PES_packet_length > out.maxSize) {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2DemuxedElementaryStream.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A MPEG 1 or 2 Elementary Stream, demultiplexed from a Program Stream
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2DemuxedServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MPEG-1 or 2 demuxer.
// Implementation
@@ -94,7 +94,7 @@
}
void MPEG1or2DemuxedServerMediaSubsession
-::seekStreamSource(FramedSource* inputSource, float seekNPT) {
+::seekStreamSource(FramedSource* inputSource, double seekNPT) {
float const dur = duration();
unsigned const size = fOurDemux.fileSize();
unsigned absBytePosition = dur == 0.0 ? 0 : (unsigned)((seekNPT/dur)*size);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2FileServerDemux.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A server demultiplexer for a MPEG 1 or 2 Program Stream
// Implementation
@@ -142,7 +142,7 @@
if (!getMPEG1or2TimeCode(dataSource, *baseDemux, True, firstTimeCode)) break;
// Then, read the last time code from the file.
- // (Before doing this, flush the demux's input buffers,
+ // (Before doing this, flush the demux's input buffers,
// and seek towards the end of the file, for efficiency.)
baseDemux->flushInput();
unsigned const startByteFromEnd = 100000;
@@ -163,6 +163,8 @@
return duration;
}
+#define DUMMY_SINK_BUFFER_SIZE (6+65535) /* large enough for a PES packet */
+
class DummySink: public MediaSink {
public:
DummySink(MPEG1or2Demux& demux, Boolean returnFirstSeenCode);
@@ -184,7 +186,7 @@
private:
MPEG1or2Demux& fOurDemux;
Boolean fReturnFirstSeenCode;
- unsigned char fBuf[10000];
+ unsigned char fBuf[DUMMY_SINK_BUFFER_SIZE];
};
static void afterPlayingDummySink(DummySink* sink); // forward
@@ -201,7 +203,7 @@
sink.startPlaying(*dataSource,
(MediaSink::afterPlayingFunc*)afterPlayingDummySink, &sink);
env.taskScheduler().doEventLoop(&sink.watchVariable);
-
+
timeCode = computeSCRTimeCode(parentDemux.lastSeenSCR());
return parentDemux.lastSeenSCR().isValid;
}
@@ -246,7 +248,7 @@
}
static void afterPlayingDummySink(DummySink* sink) {
- // Return from the "doEventLoop()" call:
+ // Return from the "doEventLoop()" call:
sink->watchVariable = ~0;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2VideoFileServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MPEG-1 or 2 Elementary Stream video file.
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2VideoHTTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A HTTP Sink specifically for MPEG Video
// Implementation
@@ -31,9 +31,9 @@
MPEG1or2VideoHTTPSink* newSink = new MPEG1or2VideoHTTPSink(env, ourSocket);
if (newSink == NULL) break;
-
+
appendPortNum(env, ourPort);
-
+
return newSink;
} while (0);
@@ -65,7 +65,7 @@
= (framePtr[0]<<24)|(framePtr[1]<<16)|(framePtr[2]<<8)|framePtr[3];
if (first4Bytes == VIDEO_SEQUENCE_HEADER_START_CODE) {
- fHaveSeenFirstVSH = True;
+ fHaveSeenFirstVSH = True;
return True;
} else {
return False;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2VideoRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for MPEG video (RFC 2250)
// Implementation
@@ -51,7 +51,7 @@
// complete picture) can appear at other than the first position in a packet
// in all situations, EXCEPT when it follows the end of (i.e., the last slice
// of) a picture. I.e., the headers at the beginning of a picture must
- // appear at the start of a RTP packet.
+ // appear at the start of a RTP packet.
if (!fPreviousFrameWasSlice) return True;
// A slice is already packed into this packet. We allow this new 'frame'
@@ -72,7 +72,7 @@
unsigned numRemainingBytes) {
Boolean thisFrameIsASlice = False; // until we learn otherwise
if (isFirstFrameInPacket()) {
- fSequenceHeaderPresent = fPacketBeginsSlice = fPacketEndsSlice = False;
+ fSequenceHeaderPresent = fPacketBeginsSlice = fPacketEndsSlice = False;
}
if (fragmentationOffset == 0) {
@@ -94,7 +94,7 @@
unsigned char byte8 = numBytesInFrame == 8 ? 0 : frameStart[8];
fPictureState.temporal_reference = (next4Bytes&0xFFC00000)>>(32-10);
- fPictureState.picture_coding_type = (next4Bytes&0x00380000)>>(32-(10+3));
+ fPictureState.picture_coding_type = (next4Bytes&0x00380000)>>(32-(10+3));
unsigned char FBV, BFC, FFV, FFC;
FBV = BFC = FFV = FFC = 0;
@@ -141,7 +141,7 @@
// in the packet. That's OK, because this situation happens infrequently,
// and we want the video-specific header to reflect the most up-to-date
// information (in particular, from a Picture Header) anyway.
- unsigned videoSpecificHeader =
+ unsigned videoSpecificHeader =
// T == 0
(fPictureState.temporal_reference<<16) |
// AN == N == 0
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2VideoRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG-1 or MPEG-2 Video RTP Sources
// Implementation
@@ -56,7 +56,7 @@
resultSpecialHeaderSize = 4;
return True;
-}
+}
Boolean MPEG1or2VideoRTPSource
::packetIsUsableInJitterCalculation(unsigned char* packet,
@@ -74,7 +74,7 @@
} else { // a P, B, D, or other unknown frame type
return False;
}
-}
+}
char const* MPEG1or2VideoRTPSource::MIMEtype() const {
return "video/MPEG";
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2VideoStreamDiscreteFramer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simplified version of "MPEG1or2VideoStreamFramer" that takes only
// complete, discrete frames (rather than an arbitrary byte stream) as input.
// This avoids the parsing and data copying overhead of the full
@@ -32,7 +32,7 @@
return new MPEG1or2VideoStreamDiscreteFramer(env, inputSource,
iFramesOnly, vshPeriod);
}
-
+
MPEG1or2VideoStreamDiscreteFramer
::MPEG1or2VideoStreamDiscreteFramer(UsageEnvironment& env,
FramedSource* inputSource,
@@ -45,10 +45,10 @@
fLastNonBFramePresentationTime.tv_sec = 0;
fLastNonBFramePresentationTime.tv_usec = 0;
}
-
+
MPEG1or2VideoStreamDiscreteFramer::~MPEG1or2VideoStreamDiscreteFramer() {
}
-
+
void MPEG1or2VideoStreamDiscreteFramer::doGetNextFrame() {
// Arrange to read data (which should be a complete MPEG-1 or 2 video frame)
// from our data source, directly into the client's input buffer.
@@ -57,7 +57,7 @@
afterGettingFrame, this,
FramedSource::handleClosure, this);
}
-
+
void MPEG1or2VideoStreamDiscreteFramer
::afterGettingFrame(void* clientData, unsigned frameSize,
unsigned numTruncatedBytes,
@@ -145,7 +145,7 @@
}
}
- if (nextCode == 0x00 /*PICTURE_START_CODE*/ && i+2 < frameSize) {
+ if (nextCode == 0x00 /*PICTURE_START_CODE*/ && i+2 < frameSize) {
// Get the 'temporal_reference' and 'picture_coding_type' from the
// following 2 bytes:
++i;
@@ -191,7 +191,7 @@
// ##### Later:
// - do "iFramesOnly" if requested
-
+
// Complete delivery to the client:
fFrameSize = frameSize;
fNumTruncatedBytes = numTruncatedBytes;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG1or2VideoStreamFramer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an MPEG 1 or 2 video elementary stream into
// frames for: Video_Sequence_Header, GOP_Header, Picture_Header
// Implementation
@@ -33,7 +33,7 @@
PARSING_GOP_HEADER_SEEN_CODE,
PARSING_PICTURE_HEADER,
PARSING_SLICE
-};
+};
#define VSH_MAX_SIZE 1000
@@ -68,7 +68,7 @@
unsigned short fCurPicTemporalReference;
// used to compute slice timestamp
unsigned char fCurrentSliceNumber; // set when parsing a slice
-
+
// A saved copy of the most recently seen 'video_sequence_header',
// in case we need to insert it into the stream periodically:
unsigned char fSavedVSHBuffer[VSH_MAX_SIZE];
@@ -283,7 +283,7 @@
do {
saveToNextCode(next4Bytes);
} while (next4Bytes != GROUP_START_CODE && next4Bytes != PICTURE_START_CODE);
-
+
setParseState((next4Bytes == GROUP_START_CODE)
? PARSING_GOP_HEADER_SEEN_CODE : PARSING_PICTURE_HEADER);
@@ -355,7 +355,7 @@
// Compute this frame's timestamp:
usingSource()->computePresentationTime(0);
-
+
setParseState(PARSING_PICTURE_HEADER);
return curFrameSize();
@@ -397,7 +397,7 @@
saveToNextCode(next4Bytes);
} while (!isSliceStartCode(next4Bytes));
}
-
+
setParseState(PARSING_SLICE);
fCurrentSliceNumber = next4Bytes&0xFF;
@@ -410,7 +410,7 @@
if (fSkippingCurrentPicture) {
return parse(); // try again, until we get a non-skipped frame
- } else {
+ } else {
return curFrameSize();
}
}
@@ -429,7 +429,7 @@
// Copy all bytes that we see, up until we reach a code of some sort:
saveToNextCode(next4Bytes);
}
-
+
// The next thing to parse depends on the code that we just saw:
if (isSliceStartCode(next4Bytes)) { // common case
setParseState(PARSING_SLICE);
@@ -472,7 +472,7 @@
if (fSkippingCurrentPicture) {
return parse(); // try again, until we get a non-skipped frame
- } else {
+ } else {
return curFrameSize();
}
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG2IndexFromTransportStream.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that produces a sequence of I-frame indices from a MPEG-2 Transport Stream
// Implementation
@@ -36,23 +36,23 @@
IndexRecord(u_int8_t startOffset, u_int8_t size,
unsigned long transportPacketNumber, float pcr);
virtual ~IndexRecord();
-
+
RecordType& recordType() { return fRecordType; }
void setFirstFlag() { fRecordType = (RecordType)(((u_int8_t)fRecordType) | 0x80); }
u_int8_t startOffset() const { return fStartOffset; }
u_int8_t& size() { return fSize; }
float pcr() const { return fPCR; }
unsigned long transportPacketNumber() const { return fTransportPacketNumber; }
-
+
IndexRecord* next() const { return fNext; }
void addAfter(IndexRecord* prev);
void unlink();
-
+
private:
// Index records are maintained in a doubly-linked list:
IndexRecord* fNext;
IndexRecord* fPrev;
-
+
RecordType fRecordType;
u_int8_t fStartOffset; // within the Transport Stream packet
u_int8_t fSize; // in bytes, following "fStartOffset".
@@ -75,7 +75,7 @@
<< recordTypeStr[r.recordType()&0x7F] << ":"
<< (unsigned)r.transportPacketNumber() << ":" << r.startOffset()
<< "(" << r.size() << ")@" << r.pcr() << "]";
-}
+}
#endif
@@ -120,13 +120,13 @@
// Begin by trying to deliver an index record (for an already-parsed frame)
// to the client:
if (deliverIndexRecord()) return;
-
+
// No more index records are left to deliver, so try to parse a new frame:
if (parseFrame()) { // success - try again
doGetNextFrame();
return;
}
-
+
// We need to read some more Transport Stream packets. Check whether we have room:
if (fParseBufferSize - fParseBufferDataEnd < TRANSPORT_PACKET_SIZE) {
// There's no room left. Compact the buffer, and check again:
@@ -138,7 +138,7 @@
return;
}
}
-
+
// Arrange to read a new Transport Stream packet:
fInputSource->getNextFrame(fInputBuffer, sizeof fInputBuffer,
afterGettingFrame, this,
@@ -169,14 +169,14 @@
handleInputClosure1();
return;
}
-
+
++fInputTransportPacketCounter;
-
+
// Figure out how much of this Transport Packet contains PES data:
u_int8_t adaptation_field_control = (fInputBuffer[3]&0x30)>>4;
u_int8_t totalHeaderSize
= adaptation_field_control == 1 ? 4 : 5 + fInputBuffer[4];
-
+
// Check for a PCR:
if (totalHeaderSize > 5 && (fInputBuffer[5]&0x10) != 0) {
// There's a PCR:
@@ -187,14 +187,14 @@
if ((fInputBuffer[10]&0x80) != 0) pcr += 1/90000.0f; // add in low-bit (if set)
unsigned short pcrExt = ((fInputBuffer[10]&0x01)<<8) | fInputBuffer[11];
pcr += pcrExt/27000000.0f;
-
+
if (!fHaveSeenFirstPCR) {
fFirstPCR = pcr;
fHaveSeenFirstPCR = True;
}
fLastPCR = pcr;
}
-
+
// Get the PID from the packet, and check for special tables: the PAT and PMT:
u_int16_t PID = ((fInputBuffer[1]&0x1F)<<8) | fInputBuffer[2];
if (PID == PAT_PID) {
@@ -213,7 +213,7 @@
return;
}
fLastContinuityCounter = continuity_counter;
-
+
// Also, if this is the start of a PES packet, then skip over the PES header:
Boolean payload_unit_start_indicator = (fInputBuffer[1]&0x40) != 0;
//fprintf(stderr, "PUSI: %d\n", payload_unit_start_indicator);//#####
@@ -229,16 +229,16 @@
return;
}
}
-
+
// The remaining data is Video Elementary Stream data. Add it to our parse buffer:
unsigned vesSize = TRANSPORT_PACKET_SIZE - totalHeaderSize;
memmove(&fParseBuffer[fParseBufferDataEnd], &fInputBuffer[totalHeaderSize], vesSize);
fParseBufferDataEnd += vesSize;
-
+
// And add a new index record noting where it came from:
addToTail(new IndexRecord(totalHeaderSize, vesSize, fInputTransportPacketCounter,
fLastPCR - fFirstPCR));
-
+
// Try again:
doGetNextFrame();
}
@@ -266,7 +266,7 @@
fParseBuffer[fParseBufferDataEnd++] = 0;
fParseBuffer[fParseBufferDataEnd++] = 1;
fParseBuffer[fParseBufferDataEnd++] = PICTURE_START_CODE;
-
+
// Try again:
doGetNextFrame();
} else {
@@ -278,7 +278,7 @@
void MPEG2IFrameIndexFromTransportStream
::analyzePAT(unsigned char* pkt, unsigned size) {
// Get the PMT_PID (we assume that's there just 1 program):
- if (size < 16) return; // table too small
+ if (size < 16) return; // table too small
u_int16_t program_number = (pkt[9]<<8) | pkt[10];
if (program_number != 0) {
fPMT_PID = ((pkt[11]&0x1F)<<8) | pkt[12];
@@ -320,10 +320,10 @@
Boolean MPEG2IFrameIndexFromTransportStream::deliverIndexRecord() {
IndexRecord* head = fHeadIndexRecord;
if (head == NULL) return False;
-
+
// Check whether the head record has been parsed yet:
if (head->recordType() == RECORD_UNPARSED) return False;
-
+
// Remove the head record (the one whose data we'll be delivering):
IndexRecord* next = head->next();
head->unlink();
@@ -332,13 +332,13 @@
} else {
fHeadIndexRecord = next;
}
-
+
if (head->recordType() == RECORD_JUNK) {
// Don't actually deliver the data to the client:
delete head;
return True;
}
-
+
// Deliver data from the head record:
#ifdef DEBUG
envir() << "delivering: " << *head << "\n";
@@ -365,10 +365,10 @@
fTo[10] = (unsigned char)(tpn>>24);
fFrameSize = 11;
}
-
+
// Free the (former) head record (as we're now done with it):
delete head;
-
+
// Complete delivery to the client:
afterGetting(this);
return True;
@@ -380,7 +380,7 @@
// to "fParseBufferDataEnd". We now parse through this data, looking for
// a complete 'frame' (where a 'frame', in this case, means
// a Video Sequence Header, GOP Header, Picture Header, or Slice).
-
+
// Inspect the frame's initial 4-byte code, to make sure it starts with a system code:
if (fParseBufferDataEnd-fParseBufferFrameStart < 4) return False; // not enough data
unsigned numInitialBadBytes = 0;
@@ -393,14 +393,14 @@
}
unsigned char nextCode;
if (!parseToNextCode(nextCode)) return False;
-
+
numInitialBadBytes = fParseBufferParseEnd - fParseBufferFrameStart;
//fprintf(stderr, "#####numInitialBadBytes: 0x%x\n", numInitialBadBytes);
fParseBufferFrameStart = fParseBufferParseEnd;
fParseBufferParseEnd += 4; // skip over the code that we just saw
p = &fParseBuffer[fParseBufferFrameStart];
}
-
+
unsigned char curCode = p[3];
RecordType curRecordType;
unsigned char nextCode;
@@ -438,7 +438,7 @@
break;
}
}
-
+
if (curRecordType == RECORD_PIC_NON_IFRAME) {
if (curCode == VOP_START_CODE) { // MPEG-4
//fprintf(stderr, "#####parseFrame()1(4): 0x%x, 0x%x\n", curCode, fParseBuffer[fParseBufferFrameStart+4]&0xC0);
@@ -454,7 +454,7 @@
}
}
}
-
+
// There is now a parsed 'frame', from "fParseBufferFrameStart"
// to "fParseBufferParseEnd". Tag the corresponding index records to note this:
unsigned frameSize = fParseBufferParseEnd - fParseBufferFrameStart + numInitialBadBytes;
@@ -471,7 +471,7 @@
}
if (r == fHeadIndexRecord) r->setFirstFlag();
// indicates that this is the first record for this frame
-
+
if (r->size() > frameSize) {
// This record contains extra data that's not part of the frame.
// Shorten this record, and move the extra data to a new record
@@ -482,7 +482,7 @@
#ifdef DEBUG
envir() << "tagged record (modified): " << *r << "\n";
#endif
-
+
IndexRecord* newRecord
= new IndexRecord(newOffset, newSize, r->transportPacketNumber(), r->pcr());
newRecord->addAfter(r);
@@ -501,12 +501,12 @@
envir() << "!!!!!Internal consistency error!!!!!\n";
return False;
}
- }
-
+ }
+
// Finally, update our parse state (to skip over the now-parsed data):
fParseBufferFrameStart = fParseBufferParseEnd;
fParseBufferParseEnd += 4; // to skip over the next code (that we found)
-
+
return True;
}
@@ -524,7 +524,7 @@
return True;
} else p += 3;
}
-
+
fParseBufferParseEnd = p - &fParseBuffer[0]; // where we've gotten to
return False; // no luck this time
}
@@ -532,7 +532,7 @@
void MPEG2IFrameIndexFromTransportStream::compactParseBuffer() {
#ifdef DEBUG
envir() << "Compacting parse buffer: [" << fParseBufferFrameStart
- << "," << fParseBufferParseEnd << "," << fParseBufferDataEnd << "]";
+ << "," << fParseBufferParseEnd << "," << fParseBufferDataEnd << "]";
#endif
memmove(&fParseBuffer[0], &fParseBuffer[fParseBufferFrameStart],
fParseBufferDataEnd - fParseBufferFrameStart);
@@ -541,7 +541,7 @@
fParseBufferFrameStart = 0;
#ifdef DEBUG
envir() << "-> [" << fParseBufferFrameStart
- << "," << fParseBufferParseEnd << "," << fParseBufferDataEnd << "]\n";
+ << "," << fParseBufferParseEnd << "," << fParseBufferDataEnd << "]\n";
#endif
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG2TransportFileServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MPEG-2 Transport Stream file.
// Implementation
@@ -40,7 +40,7 @@
ClientTrickPlayState(MPEG2TransportStreamIndexFile* indexFile);
// Functions to bring "fNPT", "fTSRecordNum" and "fIxRecordNum" in sync:
- void updateStateFromNPT(float npt);
+ void updateStateFromNPT(double npt);
void updateStateOnScaleChange();
void updateStateOnPlayChange(Boolean reverseToPreviousVSH);
@@ -151,7 +151,7 @@
}
void MPEG2TransportFileServerMediaSubsession
-::seekStream(unsigned clientSessionId, void* streamToken, float seekNPT) {
+::seekStream(unsigned clientSessionId, void* streamToken, double seekNPT) {
if (fIndexFile != NULL) { // we support 'trick play'
ClientTrickPlayState* client = lookupClient(clientSessionId);
if (client != NULL) {
@@ -259,8 +259,8 @@
fTSRecordNum(0), fIxRecordNum(0) {
}
-void ClientTrickPlayState::updateStateFromNPT(float npt) {
- fNPT = npt;
+void ClientTrickPlayState::updateStateFromNPT(double npt) {
+ fNPT = (float)npt;
// Map "fNPT" to the corresponding Transport Stream and Index record numbers:
unsigned long tsRecordNum, ixRecordNum;
fIndexFile->lookupTSPacketNumFromNPT(fNPT, tsRecordNum, ixRecordNum);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG2TransportStreamFramer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that passes through (unchanged) chunks that contain an integral number
// of MPEG-2 Transport Stream packets, but returning (in "fDurationInMicroseconds")
// an updated estimate of the time gap between chunks.
@@ -32,6 +32,7 @@
// times that we expect receivers to play the incoming packets).
// (must be <= 1)
#define MAX_PLAYOUT_BUFFER_DURATION 0.1 // (seconds)
+#define PCR_PERIOD_VARIATION_RATIO 0.5
////////// PIDStatus //////////
@@ -58,7 +59,7 @@
MPEG2TransportStreamFramer
::MPEG2TransportStreamFramer(UsageEnvironment& env, FramedSource* inputSource)
: FramedFilter(env, inputSource),
- fTSPacketCount(0), fTSPacketDurationEstimate(0.0) {
+ fTSPacketCount(0), fTSPacketDurationEstimate(0.0), fTSPCRCount(0) {
fPIDStatusTable = HashTable::create(ONE_WORD_HASH_KEYS);
}
@@ -85,6 +86,7 @@
void MPEG2TransportStreamFramer::doStopGettingFrames() {
FramedFilter::doStopGettingFrames();
fTSPacketCount = 0;
+ fTSPCRCount = 0;
clearPIDStatusTable();
}
@@ -172,6 +174,7 @@
if (pcrFlag == 0) return; // no PCR
// There's a PCR. Get it, and the PID:
+ ++fTSPCRCount;
u_int32_t pcrBaseHigh = (pkt[6]<<24)|(pkt[7]<<16)|(pkt[8]<<8)|pkt[9];
double clock = pcrBaseHigh/45000.0;
if ((pkt[10]&0x80) != 0) clock += 1/90000.0; // add in low-bit (if set)
@@ -182,6 +185,7 @@
// Check whether we already have a record of a PCR for this PID:
PIDStatus* pidStatus = (PIDStatus*)(fPIDStatusTable->Lookup((char*)pid));
+
if (pidStatus == NULL) {
// We're seeing this PID's PCR for the first time:
pidStatus = new PIDStatus(clock, timeNow);
@@ -193,6 +197,15 @@
// We've seen this PID's PCR before; update our per-packet duration estimate:
double durationPerPacket
= (clock - pidStatus->lastClock)/(fTSPacketCount - pidStatus->lastPacketNum);
+
+ // Hack (suggested by "Romain"): Don't update our estimate if this PCR appeared unusually quickly.
+ // (This can produce more accurate estimates for wildly VBR streams.)
+ double meanPCRPeriod = 0.0;
+ if (fTSPCRCount > 0) {
+ meanPCRPeriod=(double)fTSPacketCount/fTSPCRCount;
+ if (fTSPacketCount - pidStatus->lastPacketNum < meanPCRPeriod*PCR_PERIOD_VARIATION_RATIO) return;
+ }
+
if (fTSPacketDurationEstimate == 0.0) { // we've just started
fTSPacketDurationEstimate = durationPerPacket;
} else if (discontinuity_indicator == 0 && durationPerPacket >= 0.0) {
@@ -216,7 +229,7 @@
pidStatus->firstRealTime = timeNow;
}
#ifdef DEBUG_PCR
- fprintf(stderr, "PID 0x%x, PCR 0x%08x+%d:%03x == %f @ %f (diffs %f @ %f), pkt #%lu, discon %d => this duration %f, new estimate %f\n", pid, pcrBaseHigh, pkt[10]>>7, pcrExt, clock, timeNow, clock - pidStatus->firstClock, timeNow - pidStatus->firstRealTime, fTSPacketCount, discontinuity_indicator != 0, durationPerPacket, fTSPacketDurationEstimate);
+ fprintf(stderr, "PID 0x%x, PCR 0x%08x+%d:%03x == %f @ %f (diffs %f @ %f), pkt #%lu, discon %d => this duration %f, new estimate %f, mean PCR period=%f\n", pid, pcrBaseHigh, pkt[10]>>7, pcrExt, clock, timeNow, clock - pidStatus->firstClock, timeNow - pidStatus->firstRealTime, fTSPacketCount, discontinuity_indicator != 0, durationPerPacket, fTSPacketDurationEstimate, meanPCRPeriod );
#endif
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG2TransportStreamFromESSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter for converting one or more MPEG Elementary Streams
// to a MPEG-2 Transport Stream
// Implementation
@@ -156,7 +156,7 @@
FramedSource* inputSource,
u_int8_t streamId, int mpegVersion,
InputESSourceRecord* next)
- : fNext(next), fParent(parent), fInputSource(inputSource),
+ : fNext(next), fParent(parent), fInputSource(inputSource),
fStreamId(streamId), fMPEGVersion(mpegVersion) {
fInputBuffer = new unsigned char[INPUT_BUFFER_SIZE];
reset();
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG2TransportStreamFromPESSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter for converting a stream of MPEG PES packets to a MPEG-2 Transport Stream
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG2TransportStreamIndexFile.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A class that encapsulates MPEG-2 Transport Stream 'index files'/
// These index files are used to implement 'trick play' operations
// (seek-by-time, fast forward, reverse play) on Transport Stream files.
@@ -68,14 +68,14 @@
tsPacketNumber = indexRecordNumber = 0;
return;
}
-
+
// If "npt" is the same as the one that we last looked up, return its cached result:
if (npt == fCachedPCR) {
tsPacketNumber = fCachedTSPacketNumber;
- indexRecordNumber = fCachedIndexRecordNumber;
+ indexRecordNumber = fCachedIndexRecordNumber;
return;
}
-
+
// Search for the pair of neighboring index records whose PCR values span "npt".
// Use the 'regula-falsi' method.
Boolean success = False;
@@ -87,7 +87,7 @@
pcrRight = pcrFromBuf();
if (npt > pcrRight) npt = pcrRight;
// handle "npt" too large by seeking to the last frame of the file
-
+
while (ixRight-ixLeft > 1 && pcrLeft < npt && npt <= pcrRight) {
unsigned long ixNew = ixLeft
+ (unsigned long)(((npt-pcrLeft)/(pcrRight-pcrLeft))*(ixRight-ixLeft));
@@ -106,12 +106,12 @@
}
}
if (ixRight-ixLeft > 1 || npt <= pcrLeft || npt > pcrRight) break; // bad PCR values in index file?
-
+
ixFound = ixRight;
// "Rewind' until we reach the start of a Video Sequence or GOP header:
success = rewindToVSH(ixFound);
} while (0);
-
+
if (success && readIndexRecord(ixFound)) {
// Return (and cache) information from record "ixFound":
npt = fCachedPCR = pcrFromBuf();
@@ -133,14 +133,14 @@
indexRecordNumber = 0;
return;
}
-
+
// If "tsPacketNumber" is the same as the one that we last looked up, return its cached result:
if (tsPacketNumber == fCachedTSPacketNumber) {
pcr = fCachedPCR;
- indexRecordNumber = fCachedIndexRecordNumber;
+ indexRecordNumber = fCachedIndexRecordNumber;
return;
}
-
+
// Search for the pair of neighboring index records whose TS packet #s span "tsPacketNumber".
// Use the 'regula-falsi' method.
Boolean success = False;
@@ -152,7 +152,7 @@
tsRight = tsPacketNumFromBuf();
if (tsPacketNumber > tsRight) tsPacketNumber = tsRight;
// handle "tsPacketNumber" too large by seeking to the last frame of the file
-
+
while (ixRight-ixLeft > 1 && tsLeft < tsPacketNumber && tsPacketNumber <= tsRight) {
unsigned long ixNew = ixLeft
+ (unsigned long)(((tsPacketNumber-tsLeft)/(tsRight-tsLeft))*(ixRight-ixLeft));
@@ -171,7 +171,7 @@
}
}
if (ixRight-ixLeft > 1 || tsPacketNumber <= tsLeft || tsPacketNumber > tsRight) break; // bad PCR values in index file?
-
+
ixFound = ixRight;
if (reverseToPreviousVSH) {
// "Rewind' until we reach the start of a Video Sequence or GOP header:
@@ -180,7 +180,7 @@
success = True;
}
} while (0);
-
+
if (success && readIndexRecord(ixFound)) {
// Return (and cache) information from record "ixFound":
pcr = fCachedPCR = pcrFromBuf();
@@ -200,7 +200,7 @@
unsigned long& transportPacketNum, u_int8_t& offset,
u_int8_t& size, float& pcr, u_int8_t& recordType) {
if (!readIndexRecord(indexRecordNum)) return False;
-
+
transportPacketNum = tsPacketNumFromBuf();
offset = offsetFromBuf();
size = sizeFromBuf();
@@ -211,7 +211,7 @@
float MPEG2TransportStreamIndexFile::getPlayingDuration() {
if (fNumIndexRecords == 0 || !readOneIndexRecord(fNumIndexRecords-1)) return 0.0f;
-
+
return pcrFromBuf();
}
@@ -221,15 +221,15 @@
fCurrentIndexRecordNum = 0;
}
}
-
+
return fFid != NULL;
}
Boolean MPEG2TransportStreamIndexFile::seekToIndexRecord(unsigned long indexRecordNumber) {
if (!openFid()) return False;
-
+
if (indexRecordNumber == fCurrentIndexRecordNum) return True; // we're already there
-
+
if (SeekFile64(fFid, (int64_t)(indexRecordNumber*INDEX_RECORD_SIZE), SEEK_SET) != 0) return False;
fCurrentIndexRecordNum = indexRecordNumber;
return True;
@@ -240,17 +240,17 @@
if (!seekToIndexRecord(indexRecordNum)) break;
if (fread(fBuf, INDEX_RECORD_SIZE, 1, fFid) != 1) break;
++fCurrentIndexRecordNum;
-
+
return True;
} while (0);
-
+
return False; // an error occurred
}
Boolean MPEG2TransportStreamIndexFile::readOneIndexRecord(unsigned long indexRecordNum) {
Boolean result = readIndexRecord(indexRecordNum);
closeFid();
-
+
return result;
}
@@ -273,17 +273,17 @@
Boolean MPEG2TransportStreamIndexFile::rewindToVSH(unsigned long&ixFound) {
Boolean success = False;
-
+
while (ixFound > 0) {
if (!readIndexRecord(ixFound)) break;
-
+
u_int8_t recordType = recordTypeFromBuf();
if ((recordType&0x80) != 0 && (recordType&0x7F) <= 2/*GOP*/) {
if ((recordType&0x7F) == 2) {
// This is a GOP. Hack: If the preceding record is for a Video Sequence Header,
// then use it instead:
unsigned long newIxFound = ixFound;
-
+
while (--newIxFound > 0) {
if (!readIndexRecord(newIxFound)) break;
recordType = recordTypeFromBuf();
@@ -301,6 +301,6 @@
--ixFound;
}
if (ixFound == 0) success = True; // use record 0 anyway
-
+
return success;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG2TransportStreamMultiplexor.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A class for generating MPEG-2 Transport Stream from one or more input
// Elementary Stream data sources
// Implementation
@@ -101,26 +101,30 @@
fInputBufferSize = 0; // then, ignore the buffer
} else {
fCurrentPID = stream_id;
-
+
// Set the stream's type:
u_int8_t& streamType = fPIDState[fCurrentPID].streamType; // alias
-
+
if (streamType == 0) {
// Instead, set the stream's type to default values, based on whether
// the stream is audio or video, and whether it's MPEG-1 or MPEG-2:
if ((stream_id&0xF0) == 0xE0) { // video
- streamType = mpegVersion == 1 ? 1 : 2;
- if (fPCR_PID == 0) fPCR_PID = fCurrentPID; // use this stream's SCR for PCR
+ streamType = mpegVersion == 1 ? 1 : mpegVersion == 2 ? 2 : 0x10;
} else if ((stream_id&0xE0) == 0xC0) { // audio
- streamType = mpegVersion == 1 ? 3 : 4;
- if (!fHaveVideoStreams && fPCR_PID == 0) fPCR_PID = fCurrentPID; // use this stream's SCR for PCR
+ streamType = mpegVersion == 1 ? 3 : mpegVersion == 2 ? 4 : 0xF;
} else if (stream_id == 0xBD) { // private_stream1 (usually AC-3)
- streamType = 0x06; // for DVB; for ATSC, use 0x81
+ streamType = 0x06; // for DVB; for ATSC, use 0x81
} else { // something else, e.g., AC-3 uses private_stream1 (0xBD)
- streamType = 0x81; // private
+ streamType = 0x81; // private
+ }
+ }
+
+ if (fPCR_PID == 0) { // set it to this stream, if it's appropriate:
+ if ((!fHaveVideoStreams && (streamType == 3 || streamType == 4 || streamType == 0xF))/* audio stream */ ||
+ (streamType == 1 || streamType == 2 || streamType == 0x10)/* video stream */) {
+ fPCR_PID = fCurrentPID; // use this stream's SCR for PCR
}
}
-
if (fCurrentPID == fPCR_PID) {
// Record the input's current SCR timestamp, for use as our PCR:
fPCR = scr;
@@ -175,7 +179,7 @@
}
// ASSERT: numHeaderBytes+numPCRBytes+numPaddingBytes+numDataBytes
// == TRANSPORT_PACKET_SIZE
-
+
// Fill in the header of the Transport Stream packet:
unsigned char* header = fTo;
*header++ = 0x47; // sync_byte
@@ -213,7 +217,7 @@
}
}
}
-
+
// Add any padding bytes:
for (unsigned i = 0; i < numPaddingBytes; ++i) *header++ = 0xFF;
@@ -247,18 +251,18 @@
*pat++ = OUR_PROGRAM_NUMBER>>8; *pat++ = OUR_PROGRAM_NUMBER; // program_number
*pat++ = 0xE0|(OUR_PROGRAM_MAP_PID>>8); // reserved; program_map_PID (high)
*pat++ = OUR_PROGRAM_MAP_PID; // program_map_PID (low)
-
+
// Compute the CRC from the bytes we currently have (not including "pointer_field"):
u_int32_t crc = calculateCRC(patBuffer+1, pat - (patBuffer+1));
*pat++ = crc>>24; *pat++ = crc>>16; *pat++ = crc>>8; *pat++ = crc;
-
+
// Fill in the rest of the packet with padding bytes:
while (pat < &patBuffer[patSize]) *pat++ = 0xFF;
// Deliver the packet:
unsigned startPosition = 0;
deliverDataToClient(PAT_PID, patBuffer, patSize, startPosition);
-
+
// Finally, remove the new buffer:
delete[] patBuffer;
}
@@ -286,7 +290,7 @@
*pmt++ = 0xF0; // reserved; program_info_length (high)
*pmt++ = 0; // program_info_length (low)
for (int pid = 0; pid < PID_TABLE_SIZE; ++pid) {
- if (fPIDState[pid].streamType != 0) {
+ if (fPIDState[pid].streamType != 0) {
// This PID gets recorded in the table
*pmt++ = fPIDState[pid].streamType;
*pmt++ = 0xE0; // reserved; elementary_pid (high)
@@ -297,18 +301,18 @@
}
unsigned section_length = pmt - (section_lengthPtr+1) + 4 /*for CRC*/;
*section_lengthPtr = section_length;
-
+
// Compute the CRC from the bytes we currently have (not including "pointer_field"):
u_int32_t crc = calculateCRC(pmtBuffer+1, pmt - (pmtBuffer+1));
*pmt++ = crc>>24; *pmt++ = crc>>16; *pmt++ = crc>>8; *pmt++ = crc;
-
+
// Fill in the rest of the packet with padding bytes:
while (pmt < &pmtBuffer[pmtSize]) *pmt++ = 0xFF;
// Deliver the packet:
unsigned startPosition = 0;
deliverDataToClient(OUR_PROGRAM_MAP_PID, pmtBuffer, pmtSize, startPosition);
-
+
// Finally, remove the new buffer:
delete[] pmtBuffer;
}
@@ -321,7 +325,7 @@
if ((u_int16_t)frameSize > 6+program_stream_map_length) {
frameSize = 6+program_stream_map_length;
}
-
+
u_int8_t versionByte = fInputBuffer[6];
if ((versionByte&0x80) == 0) return; // "current_next_indicator" is not set
fCurrentInputProgramMapVersion = versionByte&0x1F;
@@ -335,7 +339,7 @@
frameSize -= 4; // sizeof CRC_32
if (frameSize > offset + elementary_stream_map_length) {
frameSize = offset + elementary_stream_map_length;
- }
+ }
while (offset + 4 <= frameSize) {
u_int8_t stream_type = fInputBuffer[offset];
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG2TransportStreamTrickModeFilter.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that converts a MPEG Transport Stream file - with corresponding index file
// - to a corresponding Video Elementary Stream. It also uses a "scale" parameter
// to implement 'trick mode' (fast forward or reverse play, using I-frames) on
@@ -68,7 +68,7 @@
afterGetting(this);
return;
}
-
+
while (1) {
// Get the next record from our index file.
// This tells us the type of frame this data is, which Transport Stream packet
@@ -96,7 +96,7 @@
// fprintf(stderr, "#####read index record %ld: ts %ld: %c, PCR %f\n", fNextIndexRecordNum, fDesiredTSPacketNum, isIFrameStart(recordType) ? 'I' : isNonIFrameStart(recordType) ? 'j' : 'x', recordPCR);
fNextIndexRecordNum
+= (fState == DELIVERING_SAVED_FRAME) ? 1 : fDirection;
-
+
// Handle this index record, depending on the record type and our current state:
switch (fState) {
case SKIPPING_FRAME:
@@ -146,7 +146,7 @@
fState = SKIPPING_FRAME;
}
} else {
- // Not the start of a frame, but deliver it, if it's needed:
+ // Not the start of a frame, but deliver it, if it's needed:
if (fState == SAVING_AND_DELIVERING_FRAME) {
// fprintf(stderr, "\tdelivering\n");//#####
fDesiredDataPCR = recordPCR; // use this frame's PCR
@@ -196,7 +196,7 @@
fPresentationTime.tv_usec
= (unsigned long)((deliveryPCR - fPresentationTime.tv_sec)*1000000.0f);
// fprintf(stderr, "#####DGNF9\n");
-
+
afterGetting(this);
} else {
// Arrange to read the Transport Packet that we want:
@@ -206,11 +206,11 @@
void MPEG2TransportStreamTrickModeFilter::seekToTransportPacket(unsigned long tsPacketNum) {
if (tsPacketNum == fNextTSPacketNum) return; // we're already there
-
+
ByteStreamFileSource* tsFile = (ByteStreamFileSource*)fInputSource;
u_int64_t tsPacketNum64 = (u_int64_t)tsPacketNum;
tsFile->seekToByteAbsolute(tsPacketNum64*TRANSPORT_PACKET_SIZE);
-
+
fNextTSPacketNum = tsPacketNum;
}
@@ -236,10 +236,10 @@
onSourceClosure1();
return;
}
-
+
fCurrentTSPacketNum = fNextTSPacketNum; // i.e., the one that we just read
- ++fNextTSPacketNum;
-
+ ++fNextTSPacketNum;
+
// Attempt deliver again:
attemptDeliveryToClient();
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG4ESVideoRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for MPEG-4 Elementary Stream video (RFC 3016)
// Implementation
@@ -122,7 +122,7 @@
endPtr += 2;
}
sprintf(endPtr, "\r\n");
-
+
delete[] fAuxSDPLine;
fAuxSDPLine = strDup(fmtp);
delete[] fmtp;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG4ESVideoRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP4V-ES video RTP stream sources
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG4GenericRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG4-GENERIC ("audio", "video", or "application") RTP stream sinks
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG4GenericRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG4-GENERIC ("audio", "video", or "application") RTP stream sources
// Implementation
@@ -126,7 +126,7 @@
if (fSizeLength > 0) {
// The packet begins with a "AU Header Section". Parse it, to
- // determine the "AU-header"s for each frame present in this packet:
+ // determine the "AU-header"s for each frame present in this packet:
resultSpecialHeaderSize += 2;
if (packetSize < resultSpecialHeaderSize) return False;
@@ -223,7 +223,7 @@
result = samplingFrequencyFromIndex[samplingFrequencyIndex];
break;
}
-
+
// Index == 15 means that the actual frequency is next (24 bits):
if (configSize < 5) break;
result = ((config[1]&0x7F)<<17) | (config[2]<<9) | (config[3]<<1) | (config[4]>>7);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG4LATMAudioRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for MPEG-4 audio, using LATM multiplexing (RFC 3016)
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG4LATMAudioRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG-4 audio, using LATM multiplexing
// Implementation
@@ -153,7 +153,7 @@
// There's a second nibble, but it's malformed
return False;
}
-
+
resultByte = (firstNibble<<4)|secondNibble;
return True;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG4VideoFileServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MPEG-4 video file.
// Implementation
@@ -80,15 +80,15 @@
// "auxSDPLine()" will be NULL initially, and we need to start reading
// data from our file until this changes.
fDummyRTPSink = rtpSink;
-
+
// Start reading the file:
fDummyRTPSink->startPlaying(*inputSource, afterPlayingDummy, this);
-
+
// Check whether the sink's 'auxSDPLine()' is ready:
checkForAuxSDPLine(this);
-
+
envir().taskScheduler().doEventLoop(&fDoneFlag);
-
+
char const* auxSDPLine = fDummyRTPSink->auxSDPLine();
return auxSDPLine;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG4VideoStreamDiscreteFramer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simplified version of "MPEG4VideoStreamFramer" that takes only complete,
// discrete frames (rather than an arbitrary byte stream) as input.
// This avoids the parsing and data copying overhead of the full
@@ -143,7 +143,7 @@
mask >>= 1;
}
}
-
+
// If this is a "B" frame, then we have to tweak "presentationTime":
if (vop_coding_type == 2/*B*/
&& (fLastNonBFramePresentationTime.tv_usec > 0 ||
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEG4VideoStreamFramer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an MPEG-4 video elementary stream into
// frames for:
// - Visual Object Sequence (VS) Header + Visual Object (VO) Header
@@ -38,7 +38,7 @@
PARSING_GROUP_OF_VIDEO_OBJECT_PLANE,
PARSING_VIDEO_OBJECT_PLANE,
PARSING_VISUAL_OBJECT_SEQUENCE_END_CODE
-};
+};
class MPEG4VideoStreamParser: public MPEGVideoStreamParser {
public:
@@ -72,13 +72,13 @@
private:
MPEGParseState fCurrentParseState;
- unsigned fNumBitsSeenSoFar; // used by the getNextFrameBit*() routines
+ unsigned fNumBitsSeenSoFar; // used by the getNextFrameBit*() routines
u_int32_t vop_time_increment_resolution;
unsigned fNumVTIRBits;
// # of bits needed to count to "vop_time_increment_resolution"
u_int8_t fixed_vop_rate;
unsigned fixed_vop_time_increment; // used if 'fixed_vop_rate' is set
- unsigned fSecondsSinceLastTimeCode, fTotalTicksSinceLastTimeCode, fPrevNewTotalTicks;
+ unsigned fSecondsSinceLastTimeCode, fTotalTicksSinceLastTimeCode, fPrevNewTotalTicks;
unsigned fPrevPictureCountDelta;
Boolean fJustSawTimeCode;
};
@@ -129,7 +129,7 @@
// Copy the old, then the new, config bytes there:
memmove(configNew, fNewConfigBytes, fNumNewConfigBytes);
memmove(&configNew[fNumNewConfigBytes], newConfigBytes, numNewBytes);
-
+
delete[] fNewConfigBytes; fNewConfigBytes = configNew;
fNumNewConfigBytes += numNewBytes;
}
@@ -257,7 +257,7 @@
while (next4Bytes != VISUAL_OBJECT_START_CODE) {
saveToNextCode(next4Bytes);
}
-
+
setParseState(PARSING_VISUAL_OBJECT);
// Compute this frame's presentation time:
@@ -300,7 +300,7 @@
if (visual_object_type != 1) {
usingSource()->envir() << "MPEG4VideoStreamParser::parseVisualObject(): Warning: We don't handle visual_object_type " << visual_object_type << "\n";
}
-
+
// Now, copy all bytes that we see, up until we reach
// a video_object_start_code
u_int32_t next4Bytes = get4Bytes();
@@ -443,7 +443,7 @@
saveToNextCode(next4Bytes);
} while (next4Bytes != GROUP_VOP_START_CODE
&& next4Bytes != VOP_START_CODE);
-
+
analyzeVOLHeader();
setParseState((next4Bytes == GROUP_VOP_START_CODE)
@@ -470,7 +470,7 @@
// Next, extract the (18-bit) time code from the next 3 bytes:
u_int8_t next3Bytes[3];
getBytes(next3Bytes, 3);
- saveByte(next3Bytes[0]);saveByte(next3Bytes[1]);saveByte(next3Bytes[2]);
+ saveByte(next3Bytes[0]);saveByte(next3Bytes[1]);saveByte(next3Bytes[2]);
unsigned time_code
= (next3Bytes[0]<<10)|(next3Bytes[1]<<2)|(next3Bytes[2]>>6);
unsigned time_code_hours = (time_code&0x0003E000)>>13;
@@ -489,7 +489,7 @@
while (next4Bytes != VOP_START_CODE) {
saveToNextCode(next4Bytes);
}
-
+
// Compute this frame's presentation time:
usingSource()->computePresentationTime(fTotalTicksSinceLastTimeCode);
@@ -552,10 +552,10 @@
#ifdef DEBUG
fprintf(stderr, "vop_coding_type: %d(%c), modulo_time_base: %d, vop_time_increment: %d\n", vop_coding_type, "IPBS"[vop_coding_type], modulo_time_base, vop_time_increment);
#endif
-
+
// Now, copy all bytes that we see, up until we reach a code of some sort:
saveToNextCode(next4Bytes);
-
+
// Update our counters based on the frame timing information that we saw:
if (fixed_vop_time_increment > 0) {
// This is a 'fixed_vop_rate' stream. Use 'fixed_vop_time_increment':
@@ -606,9 +606,9 @@
}
}
fJustSawTimeCode = False; // for next time
-
+
// The next thing to parse depends on the code that we just saw,
- // but we are assumed to have ended the current picture:
+ // but we are assumed to have ended the current picture:
usingSource()->fPictureEndMarker = True; // HACK #####
switch (next4Bytes) {
case VISUAL_OBJECT_SEQUENCE_END_CODE: {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEGVideoStreamFramer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an MPEG video elementary stream into
// headers and frames
// Implementation
@@ -99,8 +99,8 @@
#ifdef DEBUG
if (firstPT.tv_sec == 0 && firstPT.tv_usec == 0) firstPT = fPresentationTime;
struct timeval diffPT;
- diffPT.tv_sec = fPresentationTime.tv_sec - firstPT.tv_sec;
- diffPT.tv_usec = fPresentationTime.tv_usec - firstPT.tv_usec;
+ diffPT.tv_sec = fPresentationTime.tv_sec - firstPT.tv_sec;
+ diffPT.tv_usec = fPresentationTime.tv_usec - firstPT.tv_usec;
if (fPresentationTime.tv_usec < firstPT.tv_usec) {
--diffPT.tv_sec;
diffPT.tv_usec += 1000000;
@@ -157,7 +157,7 @@
// It has already been copied to the reader's space.
fFrameSize = acquiredFrameSize;
fNumTruncatedBytes = fParser->numTruncatedBytes();
-
+
// "fPresentationTime" should have already been computed.
// Compute "fDurationInMicroseconds" now:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEGVideoStreamParser.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// An abstract parser for MPEG video streams
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MPEGVideoStreamParser.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// An abstract parser for MPEG video streams
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/Makefile.tail
^
|
| @@ -235,7 +235,7 @@
include/RTSPOverHTTPServer.hh: include/Media.hh
RTSPClient.$(CPP): include/RTSPClient.hh include/RTSPCommon.hh include/Base64.hh include/Locale.hh our_md5.h
include/RTSPClient.hh: include/MediaSession.hh include/DigestAuthentication.hh
-RTSPCommon.$(CPP): include/RTSPCommon.hh
+RTSPCommon.$(CPP): include/RTSPCommon.hh include/Locale.hh
SIPClient.$(CPP): include/SIPClient.hh
include/SIPClient.hh: include/MediaSession.hh include/DigestAuthentication.hh
MediaSession.$(CPP): include/liveMedia.hh include/Locale.hh
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/Media.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Media
// Implementation
@@ -33,7 +33,7 @@
void remove(char const* name);
void generateNewName(char* mediumName, unsigned maxLen);
-
+
protected:
MediaLookupTable(UsageEnvironment& env);
virtual ~MediaLookupTable();
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MediaSession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A data structure that represents a session that consists of
// potentially multiple (audio and/or video) sub-sessions
// Implementation
@@ -62,8 +62,9 @@
MediaSession::MediaSession(UsageEnvironment& env)
: Medium(env),
fSubsessionsHead(NULL), fSubsessionsTail(NULL),
- fConnectionEndpointName(NULL), fMaxPlayEndTime(0.0f),
- fScale(1.0f), fMediaSessionType(NULL), fSessionName(NULL), fSessionDescription(NULL) {
+ fConnectionEndpointName(NULL), fMaxPlayStartTime(0.0f), fMaxPlayEndTime(0.0f),
+ fScale(1.0f), fMediaSessionType(NULL), fSessionName(NULL), fSessionDescription(NULL),
+ fControlPath(NULL) {
#ifdef SUPPORT_REAL_RTSP
RealInitSDPAttributes(this);
#endif
@@ -89,6 +90,7 @@
delete[] fMediaSessionType;
delete[] fSessionName;
delete[] fSessionDescription;
+ delete[] fControlPath;
#ifdef SUPPORT_REAL_RTSP
RealReclaimSDPAttributes(this);
#endif
@@ -106,17 +108,18 @@
char const* nextSDPLine;
while (1) {
if (!parseSDPLine(sdpLine, nextSDPLine)) return False;
- //##### We should really check for:
+ //##### We should really check for:
// - "a=control:" attributes (to set the URL for aggregate control)
// - the correct SDP version (v=0)
if (sdpLine[0] == 'm') break;
sdpLine = nextSDPLine;
- if (sdpLine == NULL) break; // there are no m= lines at all
+ if (sdpLine == NULL) break; // there are no m= lines at all
// Check for various special SDP lines that we understand:
if (parseSDPLine_s(sdpLine)) continue;
if (parseSDPLine_i(sdpLine)) continue;
if (parseSDPLine_c(sdpLine)) continue;
+ if (parseSDPAttribute_control(sdpLine)) continue;
if (parseSDPAttribute_range(sdpLine)) continue;
if (parseSDPAttribute_type(sdpLine)) continue;
if (parseSDPAttribute_source_filter(sdpLine)) continue;
@@ -124,7 +127,7 @@
if (RealParseSDPAttributes(this, sdpLine)) continue;
#endif
}
-
+
while (sdpLine != NULL) {
// We have a "m=" line, representing a new sub-session:
MediaSubsession* subsession = new MediaSubsession(*this);
@@ -296,7 +299,7 @@
// Check for "s=<session name>" line
char* buffer = strDupSize(sdpLine);
Boolean parseSuccess = False;
-
+
if (sscanf(sdpLine, "s=%[^\r\n]", buffer) == 1) {
delete[] fSessionName; fSessionName = strDup(buffer);
parseSuccess = True;
@@ -310,7 +313,7 @@
// Check for "i=<session description>" line
char* buffer = strDupSize(sdpLine);
Boolean parseSuccess = False;
-
+
if (sscanf(sdpLine, "i=%[^\r\n]", buffer) == 1) {
delete[] fSessionDescription; fSessionDescription = strDup(buffer);
parseSuccess = True;
@@ -318,7 +321,7 @@
delete[] buffer;
return parseSuccess;
-}
+}
Boolean MediaSession::parseSDPLine_c(char const* sdpLine) {
// Check for "c=IN IP4 <connection-endpoint>"
@@ -349,8 +352,22 @@
return parseSuccess;
}
-static Boolean parseRangeAttribute(char const* sdpLine, float& endTime) {
- return sscanf(sdpLine, "a=range: npt = %*g - %g", &endTime) == 1;
+static Boolean parseRangeAttribute(char const* sdpLine, double& startTime, double& endTime) {
+ return sscanf(sdpLine, "a=range: npt = %lg - %lg", &startTime, &endTime) == 2;
+}
+
+Boolean MediaSession::parseSDPAttribute_control(char const* sdpLine) {
+ // Check for a "a=control:<control-path>" line:
+ Boolean parseSuccess = False;
+
+ char* controlPath = strDupSize(sdpLine); // ensures we have enough space
+ if (sscanf(sdpLine, "a=control: %s", controlPath) == 1) {
+ parseSuccess = True;
+ delete[] fControlPath; fControlPath = strDup(controlPath);
+ }
+ delete[] controlPath;
+
+ return parseSuccess;
}
Boolean MediaSession::parseSDPAttribute_range(char const* sdpLine) {
@@ -358,9 +375,13 @@
// (Later handle other kinds of "a=range" attributes also???#####)
Boolean parseSuccess = False;
- float playEndTime;
- if (parseRangeAttribute(sdpLine, playEndTime)) {
+ double playStartTime;
+ double playEndTime;
+ if (parseRangeAttribute(sdpLine, playStartTime, playEndTime)) {
parseSuccess = True;
+ if (playStartTime > fMaxPlayStartTime) {
+ fMaxPlayStartTime = playStartTime;
+ }
if (playEndTime > fMaxPlayEndTime) {
fMaxPlayEndTime = playEndTime;
}
@@ -375,7 +396,7 @@
// Note: At present, we don't check that <something> really matches
// one of our multicast addresses. We also don't support more than
// one <source> #####
- Boolean result = False; // until we succeed
+ Boolean result = False; // until we succeed
char* sourceName = strDupSize(sdpLine); // ensures we have enough space
do {
if (sscanf(sdpLine, "a=source-filter: incl IN IP4 %*s %s",
@@ -527,12 +548,13 @@
fSizelength(0), fStreamstateindication(0), fStreamtype(0),
fCpresent(False), fRandomaccessindication(False),
fConfig(NULL), fMode(NULL), fSpropParameterSets(NULL),
- fPlayEndTime(0.0),
- fVideoWidth(0), fVideoHeight(0), fVideoFPS(0), fNumChannels(1), fScale(1.0f),
+ fPlayStartTime(0.0), fPlayEndTime(0.0),
+ fVideoWidth(0), fVideoHeight(0), fVideoFPS(0), fNumChannels(1), fScale(1.0f), fNPT_PTS_Offset(0.0f),
fRTPSocket(NULL), fRTCPSocket(NULL),
fRTPSource(NULL), fRTCPInstance(NULL), fReadSource(NULL) {
+ rtpInfo.seqNum = 0; rtpInfo.timestamp = 0; rtpInfo.infoIsNew = False;
#ifdef SUPPORT_REAL_RTSP
- RealInitSDPAttributes(this);
+ RealInitSDPAttributes(this);
#endif
}
@@ -549,7 +571,13 @@
#endif
}
-float MediaSubsession::playEndTime() const {
+double MediaSubsession::playStartTime() const {
+ if (fPlayStartTime > 0) return fPlayStartTime;
+
+ return fParent.playStartTime();
+}
+
+double MediaSubsession::playEndTime() const {
if (fPlayEndTime > 0) return fPlayEndTime;
return fParent.playEndTime();
@@ -557,7 +585,7 @@
Boolean MediaSubsession::initiate(int useSpecialRTPoffset) {
if (fReadSource != NULL) return True; // has already been initiated
-
+
do {
if (fCodecName == NULL) {
env().setResultMsg("Codec is unspecified");
@@ -571,6 +599,7 @@
struct in_addr tempAddr;
tempAddr.s_addr = connectionEndpointAddress();
// This could get changed later, as a result of a RTSP "SETUP"
+ unsigned short firstRTPPortNum = fClientPortNum&~1;
while (1) {
unsigned short rtpPortNum = fClientPortNum&~1;
if (isSSM()) {
@@ -599,7 +628,8 @@
// Try again:
delete oldGroupsock;
oldGroupsock = fRTPSocket;
- fClientPortNum = 0;
+ if (firstRTPPortNum > 0) break; // Because we were told what port number to use, we can't choose another ourself
+ fClientPortNum += 2; // so that the OS doesn't choose the same port number again
}
delete oldGroupsock;
if (!success) break;
@@ -671,12 +701,12 @@
= MP3ADURTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,
fRTPTimestampFrequency);
if (fRTPSource == NULL) break;
-
+
// Add a filter that deinterleaves the ADUs after depacketizing them:
MP3ADUdeinterleaver* deinterleaver
= MP3ADUdeinterleaver::createNew(env(), fRTPSource);
if (deinterleaver == NULL) break;
-
+
// Add another filter that converts these ADUs to MP3 frames:
fReadSource = MP3FromADUSource::createNew(env(), deinterleaver);
} else if (strcmp(fCodecName, "X-MP3-DRAFT-00") == 0) {
@@ -687,7 +717,7 @@
fRTPTimestampFrequency,
"audio/MPA-ROBUST" /*hack*/);
if (fRTPSource == NULL) break;
-
+
// Add a filter that converts these ADUs to MP3 frames:
fReadSource = MP3FromADUSource::createNew(env(), fRTPSource,
False /*no ADU header*/);
@@ -745,7 +775,9 @@
fReadSource = fRTPSource
= JPEGVideoRTPSource::createNew(env(), fRTPSocket,
fRTPPayloadFormat,
- fRTPTimestampFrequency);
+ fRTPTimestampFrequency,
+ videoWidth(),
+ videoHeight());
} else if (strcmp(fCodecName, "X-QT") == 0
|| strcmp(fCodecName, "X-QUICKTIME") == 0) {
// Generic QuickTime streams, as defined in
@@ -792,7 +824,7 @@
env().setResultMsg("RTP payload format unknown or not supported");
break;
}
-
+
if (createSimpleRTPSource) {
char* mimeType
= new char[strlen(mediumName()) + strlen(codecName()) + 2] ;
@@ -810,7 +842,7 @@
env().setResultMsg("Failed to create read source");
break;
}
-
+
// Finally, create our RTCP instance. (It starts running automatically)
if (fRTPSource != NULL) {
unsigned totSessionBandwidth = 500; // HACK - later get from SDP#####
@@ -858,7 +890,7 @@
return True;
}
-unsigned MediaSubsession::connectionEndpointAddress() const {
+netAddressBits MediaSubsession::connectionEndpointAddress() const {
do {
// Get the endpoint name from with us, or our parent session:
char const* endpointString = connectionEndpointName();
@@ -871,17 +903,17 @@
NetAddressList addresses(endpointString);
if (addresses.numAddresses() == 0) break;
- return *(unsigned*)(addresses.firstAddress()->data());
+ return *(netAddressBits*)(addresses.firstAddress()->data());
} while (0);
// No address known:
return 0;
}
-void MediaSubsession::setDestinations(unsigned defaultDestAddress) {
+void MediaSubsession::setDestinations(netAddressBits defaultDestAddress) {
// Get the destination address from the connection endpoint name
// (This will be 0 if it's not known, in which case we use the default)
- unsigned destAddress = connectionEndpointAddress();
+ netAddressBits destAddress = connectionEndpointAddress();
if (destAddress == 0) destAddress = defaultDestAddress;
struct in_addr destAddr; destAddr.s_addr = destAddress;
@@ -900,6 +932,41 @@
}
}
+double MediaSubsession::getNormalPlayTime(struct timeval const& presentationTime) {
+ // First, check whether our "RTPSource" object has already been synchronized using RTCP.
+ // If it hasn't, then - as a special case - we need to use the RTP timestamp to compute the NPT.
+ if (rtpSource() == NULL || rtpSource()->timestampFrequency() == 0) return 0.0; // no RTP source, or bad freq!
+
+ if (!rtpSource()->hasBeenSynchronizedUsingRTCP()) {
+ if (!rtpInfo.infoIsNew) return 0.0; // the "rtpInfo" structure has not been filled in
+ u_int32_t timestampOffset = rtpSource()->curPacketRTPTimestamp() - rtpInfo.timestamp;
+ double nptOffset = (timestampOffset/(double)(rtpSource()->timestampFrequency()))*scale();
+ double npt = playStartTime() + nptOffset;
+
+ return npt;
+ } else {
+ // Common case: We have been synchronized using RTCP. This means that the "presentationTime" parameter
+ // will be accurate, and so we should use this to compute the NPT.
+ double ptsDouble = (double)(presentationTime.tv_sec + presentationTime.tv_usec/1000000.0);
+
+ if (rtpInfo.infoIsNew) {
+ // This is the first time we've been called with a synchronized presentation time since the "rtpInfo"
+ // structure was last filled in. Use this "presentationTime" to compute "fNPT_PTS_Offset":
+ u_int32_t timestampOffset = rtpSource()->curPacketRTPTimestamp() - rtpInfo.timestamp;
+ double nptOffset = (timestampOffset/(double)(rtpSource()->timestampFrequency()))*scale();
+ double npt = playStartTime() + nptOffset;
+ fNPT_PTS_Offset = npt - ptsDouble*scale();
+ rtpInfo.infoIsNew = False; // for next time
+
+ return npt;
+ } else {
+ // Use the precomputed "fNPT_PTS_Offset" to compute the NPT from the PTS:
+ if (fNPT_PTS_Offset == 0.0) return 0.0; // error: The "rtpInfo" structure was apparently never filled in
+ return (double)(ptsDouble*scale() + fNPT_PTS_Offset);
+ }
+ }
+}
+
Boolean MediaSubsession::parseSDPLine_c(char const* sdpLine) {
// Check for "c=IN IP4 <connection-endpoint>"
// or "c=IN IP4 <connection-endpoint>/<ttl+numAddresses>"
@@ -969,9 +1036,16 @@
// (Later handle other kinds of "a=range" attributes also???#####)
Boolean parseSuccess = False;
- float playEndTime;
- if (parseRangeAttribute(sdpLine, playEndTime)) {
+ double playStartTime;
+ double playEndTime;
+ if (parseRangeAttribute(sdpLine, playStartTime, playEndTime)) {
parseSuccess = True;
+ if (playStartTime > fPlayStartTime) {
+ fPlayStartTime = playStartTime;
+ if (playStartTime > fParent.playStartTime()) {
+ fParent.playStartTime() = playStartTime;
+ }
+ }
if (playEndTime > fPlayEndTime) {
fPlayEndTime = playEndTime;
if (playEndTime > fParent.playEndTime()) {
@@ -987,7 +1061,7 @@
// Check for a "a=fmtp:" line:
// TEMP: We check only for a handful of expected parameter names #####
// Later: (i) check that payload format number matches; #####
- // (ii) look for other parameters also (generalize?) #####
+ // (ii) look for other parameters also (generalize?) #####
do {
if (strncmp(sdpLine, "a=fmtp:", 7) != 0) break; sdpLine += 7;
while (isdigit(*sdpLine)) ++sdpLine;
@@ -1051,7 +1125,7 @@
delete[] fMode; fMode = strDup(valueStr);
} else if (sscanf(sdpLine, " sprop-parameter-sets = %[^; \t\r\n]", valueStr) == 1) {
// Note: We used "sdpLine" here, because the value is case-sensitive.
- delete[] fSpropParameterSets; fSpropParameterSets = strDup(valueStr);
+ delete[] fSpropParameterSets; fSpropParameterSets = strDup(valueStr);
} else {
// Some of the above parameters are Boolean. Check whether the parameter
// names appear alone, without a "= 1" at the end:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MediaSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Media Sinks
// Implementation
@@ -200,7 +200,7 @@
resetOverflowData();
}
-void OutPacketBuffer::adjustPacketStart(unsigned numBytes) {
+void OutPacketBuffer::adjustPacketStart(unsigned numBytes) {
fPacketStart += numBytes;
if (fOverflowDataOffset >= numBytes) {
fOverflowDataOffset -= numBytes;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MediaSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Media Sources
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MultiFramedRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for a common kind of payload format: Those which pack multiple,
// complete codec frames (as many as possible) into each RTP packet.
// Implementation
@@ -31,7 +31,7 @@
delete fOutBuf;
fOutBuf = new OutPacketBuffer(preferredPacketSize, maxPacketSize);
- fOurMaxPacketSize = maxPacketSize; // save value, in case subclasses need it
+ fOurMaxPacketSize = maxPacketSize; // save value, in case subclasses need it
}
MultiFramedRTPSink::MultiFramedRTPSink(UsageEnvironment& env,
@@ -322,8 +322,8 @@
// (iv) one frame per packet is allowed:
if (fOutBuf->isPreferredSize()
|| fOutBuf->wouldOverflow(numFrameBytesToUse)
- || (fPreviousFrameEndedFragmentation &&
- !allowOtherFramesAfterLastFragment())
+ || (fPreviousFrameEndedFragmentation &&
+ !allowOtherFramesAfterLastFragment())
|| !frameCanAppearAfterPacketStart(fOutBuf->curPtr() - frameSize,
frameSize) ) {
// The packet is ready to be sent now
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/MultiFramedRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP source for a common kind of payload format: Those that pack multiple,
// complete codec frames (as many as possible) into each RTP packet.
// Implementation
@@ -174,7 +174,7 @@
fPresentationTime, fCurPacketHasBeenSynchronizedUsingRTCP,
fCurPacketMarkerBit);
fFrameSize += frameSize;
-
+
if (!nextPacket->hasUsableData()) {
// We're completely done with this packet now
fReorderingBuffer->releaseUsedPacket(nextPacket);
@@ -229,22 +229,22 @@
// don't wait for 'lost' packets to arrive out-of-order later
if ((our_random()%10) == 0) break; // simulate 10% packet loss
#endif
-
+
// Check for the 12-byte RTP header:
if (bPacket->dataSize() < 12) break;
unsigned rtpHdr = ntohl(*(unsigned*)(bPacket->data())); ADVANCE(4);
Boolean rtpMarkerBit = (rtpHdr&0x00800000) >> 23;
unsigned rtpTimestamp = ntohl(*(unsigned*)(bPacket->data()));ADVANCE(4);
unsigned rtpSSRC = ntohl(*(unsigned*)(bPacket->data())); ADVANCE(4);
-
+
// Check the RTP version number (it should be 2):
if ((rtpHdr&0xC0000000) != 0x80000000) break;
-
+
// Skip over any CSRC identifiers in the header:
unsigned cc = (rtpHdr>>24)&0xF;
if (bPacket->dataSize() < cc) break;
ADVANCE(cc*4);
-
+
// Check for (& ignore) any RTP header extension
if (rtpHdr&0x10000000) {
if (bPacket->dataSize() < 4) break;
@@ -253,7 +253,7 @@
if (bPacket->dataSize() < remExtSize) break;
ADVANCE(remExtSize);
}
-
+
// Discard any padding bytes:
if (rtpHdr&0x20000000) {
if (bPacket->dataSize() == 0) break;
@@ -261,13 +261,13 @@
= (unsigned)(bPacket->data())[bPacket->dataSize()-1];
if (bPacket->dataSize() < numPaddingBytes) break;
bPacket->removePadding(numPaddingBytes);
- }
+ }
// Check the Payload Type.
if ((unsigned char)((rtpHdr&0x007F0000)>>16)
!= source->rtpPayloadFormat()) {
break;
}
-
+
// The rest of the packet is the usable data. Record and save it:
source->fLastReceivedSSRC = rtpSSRC;
unsigned short rtpSeqNo = (unsigned short)(rtpHdr&0xFFFF);
@@ -281,7 +281,7 @@
source->timestampFrequency(),
usableInJitterCalculation, presentationTime,
hasBeenSyncedUsingRTCP, bPacket->dataSize());
-
+
// Fill in the rest of the packet descriptor, and store it:
struct timeval timeNow;
gettimeofday(&timeNow, NULL);
@@ -317,6 +317,7 @@
void BufferedPacket::reset() {
fHead = fTail = 0;
fUseCount = 0;
+ fIsFirstPacket = False; // by default
}
// The following function has been deprecated:
@@ -336,11 +337,11 @@
// of more than one frame, on the assumption that the client doesn't
// care. (This is more efficient than delivering a frame at a time)
- // For backwards-compatibilty with existing uses of (the now deprecated)
+ // For backwards-compatibility with existing uses of (the now deprecated)
// "nextEnclosedFrameSize()", call that function to implement this one:
frameSize = nextEnclosedFrameSize(framePtr, dataSize);
- frameDurationInMicroseconds = 0; // by default. Subclasses should correct this.
+ frameDurationInMicroseconds = 0; // by default. Subclasses should correct this.
}
Boolean BufferedPacket::fillInData(RTPInterface& rtpInterface) {
@@ -381,7 +382,7 @@
void BufferedPacket::appendData(unsigned char* newData, unsigned numBytes) {
if (numBytes > fPacketSize-fTail) numBytes = fPacketSize - fTail;
- memmove(&fBuf[fTail], newData, numBytes);
+ memmove(&fBuf[fTail], newData, numBytes);
fTail += numBytes;
}
@@ -464,7 +465,7 @@
fSavedPacket = fPacketFactory->createNewPacket(ourSource);
fSavedPacketFree = True;
}
-
+
if (fSavedPacketFree == True) {
fSavedPacketFree = False;
return fSavedPacket;
@@ -478,19 +479,14 @@
if (!fHaveSeenFirstPacket) {
fNextExpectedSeqNo = rtpSeqNo; // initialization
+ bPacket->isFirstPacket() = True;
fHaveSeenFirstPacket = True;
}
// Ignore this packet if its sequence number is less than the one
// that we're looking for (in this case, it's been excessively delayed).
- // (But (sanity check) if the new packet's sequence number is a *lot*
- // less, then accept it anyway.)
- unsigned short const seqNoThreshold = 100;
- if (seqNumLT(rtpSeqNo, fNextExpectedSeqNo)
- && seqNumLT(fNextExpectedSeqNo, rtpSeqNo+seqNoThreshold)) {
- return False;
- }
-
+ if (seqNumLT(rtpSeqNo, fNextExpectedSeqNo)) return False;
+
// Figure out where the new packet will be stored in the queue:
BufferedPacket* beforePtr = NULL;
BufferedPacket* afterPtr = fHeadPacket;
@@ -500,11 +496,11 @@
// This is a duplicate packet - ignore it
return False;
}
-
+
beforePtr = afterPtr;
afterPtr = afterPtr->nextPacket();
}
-
+
// Link our new packet between "beforePtr" and "afterPtr":
bPacket->nextPacket() = afterPtr;
if (beforePtr == NULL) {
@@ -535,7 +531,8 @@
// of the queue:
// ASSERT: fHeadPacket->rtpSeqNo() >= fNextExpectedSeqNo
if (fHeadPacket->rtpSeqNo() == fNextExpectedSeqNo) {
- packetLossPreceded = False;
+ packetLossPreceded = fHeadPacket->isFirstPacket();
+ // (The very first packet is treated as if there was packet loss beforehand.)
return fHeadPacket;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/OnDemandServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand.
// Implementation
@@ -76,7 +76,7 @@
// We need to construct a set of SDP lines that describe this
// subsession (as a unicast stream). To do so, we first create
// dummy (unused) source and "RTPSink" objects,
- // whose parameters we use for the SDP lines:
+ // whose parameters we use for the SDP lines:
unsigned estBitrate; // unused
FramedSource* inputSource = createNewStreamSource(0, estBitrate);
if (inputSource == NULL) return NULL; // file not found
@@ -217,12 +217,12 @@
break; // success
}
-
+
unsigned char rtpPayloadType = 96 + trackNumber()-1; // if dynamic
rtpSink = createNewRTPSink(rtpGroupsock, rtpPayloadType, mediaSource);
udpSink = NULL;
}
-
+
// Turn off the destinations for each groupsock. They'll get set later
// (unless TCP is used instead):
if (rtpGroupsock != NULL) rtpGroupsock->removeAllDestinations();
@@ -234,7 +234,7 @@
streamBitrate, mediaSource,
rtpGroupsock, rtcpGroupsock);
}
-
+
// Record these destinations as being for this client session id:
Destinations* destinations;
if (tcpSocketNum < 0) { // UDP
@@ -251,7 +251,7 @@
void* rtcpRRHandlerClientData,
unsigned short& rtpSeqNum,
unsigned& rtpTimestamp) {
- StreamState* streamState = (StreamState*)streamToken;
+ StreamState* streamState = (StreamState*)streamToken;
Destinations* destinations
= (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));
if (streamState != NULL) {
@@ -270,17 +270,17 @@
// the same source:
if (fReuseFirstSource) return;
- StreamState* streamState = (StreamState*)streamToken;
+ StreamState* streamState = (StreamState*)streamToken;
if (streamState != NULL) streamState->pause();
}
void OnDemandServerMediaSubsession::seekStream(unsigned /*clientSessionId*/,
- void* streamToken, float seekNPT) {
+ void* streamToken, double seekNPT) {
// Seeking isn't allowed if multiple clients are receiving data from
// the same source:
if (fReuseFirstSource) return;
- StreamState* streamState = (StreamState*)streamToken;
+ StreamState* streamState = (StreamState*)streamToken;
if (streamState != NULL && streamState->mediaSource() != NULL) {
seekStreamSource(streamState->mediaSource(), seekNPT);
}
@@ -292,7 +292,7 @@
// from the same source:
if (fReuseFirstSource) return;
- StreamState* streamState = (StreamState*)streamToken;
+ StreamState* streamState = (StreamState*)streamToken;
if (streamState != NULL && streamState->mediaSource() != NULL) {
setStreamSourceScale(streamState->mediaSource(), scale);
}
@@ -300,7 +300,7 @@
void OnDemandServerMediaSubsession::deleteStream(unsigned clientSessionId,
void*& streamToken) {
- StreamState* streamState = (StreamState*)streamToken;
+ StreamState* streamState = (StreamState*)streamToken;
// Look up (and remove) the destinations for this client session:
Destinations* destinations
@@ -313,11 +313,11 @@
}
// Delete the "StreamState" structure if it's no longer being used:
- if (streamState != NULL && streamState->referenceCount() >= 0) {
+ if (streamState != NULL) {
if (streamState->referenceCount() > 0) --streamState->referenceCount();
if (streamState->referenceCount() == 0) {
delete streamState;
- if (fLastStreamToken == streamToken) fLastStreamToken = NULL;
+ if (fLastStreamToken == streamToken) fLastStreamToken = NULL;
streamToken = NULL;
}
}
@@ -333,7 +333,7 @@
}
void OnDemandServerMediaSubsession::seekStreamSource(FramedSource* /*inputSource*/,
- float /*seekNPT*/) {
+ double /*seekNPT*/) {
// Default implementation: Do nothing
}
@@ -358,7 +358,7 @@
char const* rangeLine = rangeSDPLine();
char const* auxSDPLine = getAuxSDPLine(rtpSink, inputSource);
if (auxSDPLine == NULL) auxSDPLine = "";
-
+
char const* const sdpFmt =
"m=%s %u RTP/AVP %d\r\n"
"c=IN IP4 %s\r\n"
@@ -384,7 +384,7 @@
auxSDPLine, // optional extra SDP line
trackId()); // a=control:<track-id>
delete[] (char*)rangeLine; delete[] rtpmapLine; delete[] ipAddressStr;
-
+
fSDPLines = strDup(sdpLines);
delete[] sdpLines;
}
@@ -416,7 +416,7 @@
fRTPSink(rtpSink), fUDPSink(udpSink), fStreamDuration(master.duration()),
fTotalBW(totalBW), fRTCPInstance(NULL) /* created later */,
fMediaSource(mediaSource), fRTPgs(rtpGS), fRTCPgs(rtcpGS) {
-}
+}
StreamState::~StreamState() {
reclaim();
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/OutputFile.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Common routines for opening/closing named output files
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/PassiveServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that represents an existing
// 'RTPSink', rather than one that creates new 'RTPSink's on demand.
// Implementation
@@ -29,7 +29,7 @@
RTCPInstance* rtcpInstance) {
return new PassiveServerMediaSubsession(rtpSink, rtcpInstance);
}
-
+
PassiveServerMediaSubsession
::PassiveServerMediaSubsession(RTPSink& rtpSink, RTCPInstance* rtcpInstance)
: ServerMediaSubsession(rtpSink.envir()),
@@ -51,12 +51,12 @@
char const* rangeLine = rangeSDPLine();
char const* auxSDPLine = fRTPSink.auxSDPLine();
if (auxSDPLine == NULL) auxSDPLine = "";
-
+
char* const ipAddressStr = strDup(our_inet_ntoa(ipAddress));
-
+
char const* const sdpFmt =
"m=%s %d RTP/AVP %d\r\n"
- "c=IN IP4 %s/%d\r\n"
+ "c=IN IP4 %s/%d\r\n"
"%s"
"%s"
"%s"
@@ -69,7 +69,7 @@
+ strlen(auxSDPLine)
+ strlen(trackId());
char* sdpLines = new char[sdpFmtSize];
- sprintf(sdpLines, sdpFmt,
+ sprintf(sdpLines, sdpFmt,
mediaType, // m= <media>
portNum, // m= <port>
rtpPayloadType, // m= <fmt list>
@@ -80,7 +80,7 @@
auxSDPLine, // optional extra SDP line
trackId()); // a=control:<track-id>
delete[] ipAddressStr; delete[] (char*)rangeLine; delete[] rtpmapLine;
-
+
fSDPLines = strDup(sdpLines);
delete[] sdpLines;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/QCELPAudioRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Qualcomm "PureVoice" (aka. "QCELP") Audio RTP Sources
// Implementation
@@ -52,7 +52,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
virtual Boolean hasBeenSynchronizedUsingRTCP();
@@ -69,7 +69,7 @@
private:
QCELPDeinterleaver(UsageEnvironment& env,
RawQCELPRTPSource* inputSource);
- // called only by "createNew()"
+ // called only by "createNew()"
virtual ~QCELPDeinterleaver();
@@ -77,7 +77,7 @@
unsigned numTruncatedBytes,
struct timeval presentationTime,
unsigned durationInMicroseconds);
- void afterGettingFrame1(unsigned frameSize, struct timeval presentationTime);
+ void afterGettingFrame1(unsigned frameSize, struct timeval presentationTime);
private:
// Redefined virtual functions:
@@ -192,7 +192,7 @@
resultSpecialHeaderSize = 1;
return True;
-}
+}
char const* RawQCELPRTPSource::MIMEtype() const {
return "audio/QCELP";
@@ -361,13 +361,13 @@
void QCELPDeinterleaver
::afterGettingFrame1(unsigned frameSize, struct timeval presentationTime) {
RawQCELPRTPSource* source = (RawQCELPRTPSource*)fInputSource;
-
+
// First, put the frame into our deinterleaving buffer:
fDeinterleavingBuffer
->deliverIncomingFrame(frameSize, source->interleaveL(),
source->interleaveN(), source->frameIndex(),
source->curPacketRTPSeqNum(),
- presentationTime);
+ presentationTime);
// Then, try delivering a frame to the client (if he wants one):
if (fNeedAFrame) doGetNextFrame();
@@ -423,7 +423,7 @@
fIncomingBankId ^= 1;
unsigned char tmp = fIncomingBinMax;
fIncomingBinMax = fOutgoingBinMax;
- fOutgoingBinMax = tmp;
+ fOutgoingBinMax = tmp;
fNextOutgoingBin = 0;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/QuickTimeFileSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A sink that generates a QuickTime file from a composite media session
// Implementation
@@ -62,7 +62,7 @@
reset();
fData = new unsigned char[bufferSize];
}
- virtual ~SubsessionBuffer() { delete fData; }
+ virtual ~SubsessionBuffer() { delete[] fData; }
void reset() { fBytesInUse = 0; }
void addBytes(unsigned numBytes) { fBytesInUse += numBytes; }
@@ -70,7 +70,7 @@
unsigned char* dataEnd() { return &fData[fBytesInUse]; }
unsigned bytesInUse() const { return fBytesInUse; }
unsigned bytesAvailable() const { return fBufferSize - fBytesInUse; }
-
+
void setPresentationTime(struct timeval const& presentationTime) {
fPresentationTime = presentationTime;
}
@@ -205,7 +205,7 @@
QuickTimeFileSink::QuickTimeFileSink(UsageEnvironment& env,
MediaSession& inputSession,
- FILE* outFid,
+ char const* outputFileName,
unsigned bufferSize,
unsigned short movieWidth,
unsigned short movieHeight,
@@ -214,7 +214,7 @@
Boolean syncStreams,
Boolean generateHintTracks,
Boolean generateMP4Format)
- : Medium(env), fInputSession(inputSession), fOutFid(outFid),
+ : Medium(env), fInputSession(inputSession),
fBufferSize(bufferSize), fPacketLossCompensate(packetLossCompensate),
fSyncStreams(syncStreams), fGenerateMP4Format(generateMP4Format),
fAreCurrentlyBeingPlayed(False),
@@ -223,6 +223,9 @@
fHaveCompletedOutputFile(False),
fMovieWidth(movieWidth), fMovieHeight(movieHeight),
fMovieFPS(movieFPS), fMaxTrackDurationM(0) {
+ fOutFid = OpenOutputFile(env, outputFileName);
+ if (fOutFid == NULL) return;
+
fNewestSyncTime.tv_sec = fNewestSyncTime.tv_usec = 0;
fFirstDataTime.tv_sec = fFirstDataTime.tv_usec = (unsigned)(~0);
@@ -301,12 +304,15 @@
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
SubsessionIOState* ioState
- = (SubsessionIOState*)(subsession->miscPtr);
+ = (SubsessionIOState*)(subsession->miscPtr);
if (ioState == NULL) continue;
delete ioState->fHintTrackForUs; // if any
delete ioState;
}
+
+ // Finally, close our output file:
+ CloseOutputFile(fOutFid);
}
QuickTimeFileSink*
@@ -321,17 +327,15 @@
Boolean syncStreams,
Boolean generateHintTracks,
Boolean generateMP4Format) {
- do {
- FILE* fid = OpenOutputFile(env, outputFileName);
- if (fid == NULL) break;
-
- return new QuickTimeFileSink(env, inputSession, fid, bufferSize,
- movieWidth, movieHeight, movieFPS,
- packetLossCompensate, syncStreams,
- generateHintTracks, generateMP4Format);
- } while (0);
+ QuickTimeFileSink* newSink =
+ new QuickTimeFileSink(env, inputSession, outputFileName, bufferSize, movieWidth, movieHeight, movieFPS,
+ packetLossCompensate, syncStreams, generateHintTracks, generateMP4Format);
+ if (newSink == NULL || newSink->fOutFid == NULL) {
+ Medium::close(newSink);
+ return NULL;
+ }
- return NULL;
+ return newSink;
}
Boolean QuickTimeFileSink::startPlaying(afterPlayingFunc* afterFunc,
@@ -352,7 +356,7 @@
Boolean QuickTimeFileSink::continuePlaying() {
// Run through each of our input session's 'subsessions',
// asking for a frame from each one:
- Boolean haveActiveSubsessions = False;
+ Boolean haveActiveSubsessions = False;
MediaSubsessionIterator iter(fInputSession);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
@@ -362,7 +366,7 @@
if (subsessionSource->isCurrentlyAwaitingData()) continue;
SubsessionIOState* ioState
- = (SubsessionIOState*)(subsession->miscPtr);
+ = (SubsessionIOState*)(subsession->miscPtr);
if (ioState == NULL) continue;
haveActiveSubsessions = True;
@@ -406,7 +410,7 @@
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
SubsessionIOState* ioState
- = (SubsessionIOState*)(subsession->miscPtr);
+ = (SubsessionIOState*)(subsession->miscPtr);
if (ioState == NULL) continue;
if (ioState->fOurSourceIsActive) return; // this source hasn't closed
@@ -459,7 +463,7 @@
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
SubsessionIOState* ioState
- = (SubsessionIOState*)(subsession->miscPtr);
+ = (SubsessionIOState*)(subsession->miscPtr);
if (ioState == NULL) continue;
ChunkDescriptor* const headChunk = ioState->fHeadChunk;
@@ -473,7 +477,7 @@
iter.reset();
while ((subsession = iter.next()) != NULL) {
SubsessionIOState* ioState
- = (SubsessionIOState*)(subsession->miscPtr);
+ = (SubsessionIOState*)(subsession->miscPtr);
if (ioState == NULL) continue;
ioState->setFinalQTstate();
@@ -529,7 +533,6 @@
char const* noCodecWarning1 = "Warning: We don't implement a QuickTime ";
char const* noCodecWarning2 = " Media Data Type for the \"";
char const* noCodecWarning3 = "\" track, so we'll insert a dummy \"????\" Media Data Atom instead. A separate, codec-specific editing pass will be needed before this track can be played.\n";
- Boolean supportPartiallyOnly = False;
do {
fQTEnableTrack = True; // enable this track in the movie by default
@@ -622,13 +625,12 @@
break;
}
- if (supportPartiallyOnly) {
- envir() << "Warning: We don't have sufficient codec-specific information (e.g., sample sizes) to fully generate the \""
- << fOurSubsession.mediumName() << "/"
- << fOurSubsession.codecName()
- << "\" track, so we'll disable this track in the movie. A separate, codec-specific editing pass will be needed before this track can be played\n";
- fQTEnableTrack = False; // disable this track in the movie
- }
+#ifdef QT_SUPPORT_PARTIALLY_ONLY
+ envir() << "Warning: We don't have sufficient codec-specific information (e.g., sample sizes) to fully generate the \""
+ << fOurSubsession.mediumName() << "/" << fOurSubsession.codecName()
+ << "\" track, so we'll disable this track in the movie. A separate, codec-specific editing pass will be needed before this track can be played\n";
+ fQTEnableTrack = False; // disable this track in the movie
+#endif
return True;
} while (0);
@@ -636,7 +638,7 @@
envir() << ", so a track for the \"" << fOurSubsession.mediumName()
<< "/" << fOurSubsession.codecName()
<< "\" subsession will not be included in the output QuickTime file\n";
- return False;
+ return False;
}
void SubsessionIOState::setFinalQTstate() {
@@ -826,7 +828,7 @@
// If there has been a previous frame, then output a 'hint sample' for it.
// (We use the current frame's presentation time to compute the previous
// hint sample's duration.)
- RTPSource* const rs = fOurSubsession.rtpSource(); // abbrev
+ RTPSource* const rs = fOurSubsession.rtpSource(); // abbrev
struct timeval const& ppt = fPrevFrameState.presentationTime; //abbrev
if (ppt.tv_sec != 0 || ppt.tv_usec != 0) {
double duration = (presentationTime.tv_sec - ppt.tv_sec)
@@ -965,7 +967,7 @@
fQTTotNumSamples += useFrame1(hintSampleSize, ppt, hintSampleDuration,
hintSampleDestFileOffset);
}
-
+
// Remember this frame for next time:
fPrevFrameState.frameSize = frameSize;
fPrevFrameState.presentationTime = presentationTime;
@@ -1036,7 +1038,7 @@
return numSamples;
}
-
+
void SubsessionIOState::onSourceClosure() {
fOurSourceIsActive = False;
fOurSink.onSourceClosure1();
@@ -1063,7 +1065,7 @@
}
}
}
-
+
// Check again whether all subsessions have been synced:
if (s.fNumSyncedSubsessions < s.fNumSubsessions) return False;
@@ -1233,7 +1235,7 @@
MediaSubsessionIterator iter(fInputSession);
MediaSubsession* subsession;
while ((subsession = iter.next()) != NULL) {
- fCurrentIOState = (SubsessionIOState*)(subsession->miscPtr);
+ fCurrentIOState = (SubsessionIOState*)(subsession->miscPtr);
if (fCurrentIOState == NULL) continue;
if (strcmp(subsession->mediumName(), "audio") != 0) continue;
@@ -1247,7 +1249,7 @@
}
iter.reset();
while ((subsession = iter.next()) != NULL) {
- fCurrentIOState = (SubsessionIOState*)(subsession->miscPtr);
+ fCurrentIOState = (SubsessionIOState*)(subsession->miscPtr);
if (fCurrentIOState == NULL) continue;
if (strcmp(subsession->mediumName(), "audio") == 0) continue;
@@ -1372,7 +1374,7 @@
// Run through our chunks, looking at their presentation times.
// From these, figure out the edits that need to be made to keep
// the track media data in sync with the presentation times.
-
+
double const syncThreshold = 0.1; // 100 ms
// don't allow the track to get out of sync by more than this
@@ -1390,7 +1392,7 @@
+ (chunkStartTime.tv_usec - editStartTime.tv_usec)/1000000.0;
trackDurationOfEdit = (currentTrackPosition-editTrackPosition)
/ (double)(fCurrentIOState->fQTTimeScale);
-
+
double outOfSync = movieDurationOfEdit - trackDurationOfEdit;
if (outOfSync > syncThreshold) {
@@ -1710,6 +1712,7 @@
for (unsigned i = 0; i < configSize; ++i) {
size += addByte(config[i]);
}
+ delete[] config;
if (strcmp(subsession.mediumName(), "audio") == 0) {
// MPEG-4 audio
@@ -1773,10 +1776,12 @@
size += addHalfWord(0xffff); // Color table id
size += addAtom_avcC();
addAtomEnd;
-
+
addAtom(avcC);
// Begin by Base-64 decoding the "sprop" parameter sets strings:
char* psets = strDup(fCurrentIOState->fOurSubsession.fmtp_spropparametersets());
+ if (psets == NULL) return 0;
+
size_t comma_pos = strcspn(psets, ",");
psets[comma_pos] = '\0';
char* sps_b64 = psets;
@@ -2019,14 +2024,14 @@
addAtomEnd;
addAtom(hnti);
- size += addAtom_sdp();
+ size += addAtom_sdp();
addAtomEnd;
unsigned QuickTimeFileSink::addAtom_sdp() {
unsigned initFilePosn = ftell(fOutFid);
unsigned size = addAtomHeader("sdp ");
- // Add this subsession's SDP lines:
+ // Add this subsession's SDP lines:
char const* sdpLines = fCurrentIOState->fOurSubsession.savedSDPLines();
// We need to change any "a=control:trackID=" values to be this
// track's actual track id:
@@ -2039,12 +2044,12 @@
if (*p3 == '\0') {
// We found the end of the search string, at p2.
int beforeTrackNumPosn = p2-sdpLines;
- // Look for the subsequent track number, and skip over it:
+ // Look for the subsequent track number, and skip over it:
int trackNumLength;
if (sscanf(p2, " %*d%n", &trackNumLength) < 0) break;
int afterTrackNumPosn = beforeTrackNumPosn + trackNumLength;
-
- // Replace the old track number with the correct one:
+
+ // Replace the old track number with the correct one:
int i;
for (i = 0; i < beforeTrackNumPosn; ++i) newSDPLines[i] = sdpLines[i];
sprintf(&newSDPLines[i], "%d", fCurrentIOState->fTrackID);
@@ -2086,7 +2091,7 @@
size += addAtom_tmax();
size += addAtom_pmax();
size += addAtom_dmax();
- size += addAtom_payt();
+ size += addAtom_payt();
addAtomEnd;
addAtom(totl);
@@ -2148,10 +2153,10 @@
addAtomEnd;
addAtom(payt);
- MediaSubsession& ourSubsession = fCurrentIOState->fOurSubsession;
+ MediaSubsession& ourSubsession = fCurrentIOState->fOurSubsession;
RTPSource* rtpSource = ourSubsession.rtpSource();
size += addByte(rtpSource->rtpPayloadFormat());
-
+
// Also, add a 'rtpmap' string: <mime-subtype>/<rtp-frequency>
unsigned rtpmapStringLength = strlen(ourSubsession.codecName()) + 20;
char* rtpmapString = new char[rtpmapStringLength];
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/QuickTimeGenericRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP Sources containing generic QuickTime stream data, as defined in
// <http://developer.apple.com/quicktime/icefloe/dispatch026.html>
// Implementation
@@ -120,7 +120,7 @@
#endif
unsigned payloadDescriptionLength = (headerStart[2]<<8)|headerStart[3];
headerStart += 4;
-
+
#ifdef DEBUG
fprintf(stderr, "\tK: %d, F: %d, A: %d, Z: %d, payloadDescriptionLength: %d\n", K, F, A, Z, payloadDescriptionLength);
#endif
@@ -131,7 +131,7 @@
expectedHeaderSize += 3;
expectedHeaderSize -= expectedHeaderSize%4; // adds padding
if (packetSize < expectedHeaderSize) return False;
- unsigned char padding = expectedHeaderSize - nonPaddedSize;
+ unsigned char padding = expectedHeaderSize - nonPaddedSize;
#ifdef DEBUG
unsigned mediaType = (headerStart[0]<<24)|(headerStart[1]<<16)
@@ -193,7 +193,7 @@
unsigned ssInfoLength = (headerStart[2]<<8)|headerStart[3];
headerStart += 4;
-
+
#ifdef DEBUG
fprintf(stderr, "\tssInfoLength: %d\n", ssInfoLength);
#endif
@@ -204,7 +204,7 @@
expectedHeaderSize += 3;
expectedHeaderSize -= expectedHeaderSize%4; // adds padding
if (packetSize < expectedHeaderSize) return False;
- unsigned char padding = expectedHeaderSize - nonPaddedSize;
+ unsigned char padding = expectedHeaderSize - nonPaddedSize;
ssInfoLength -= 4;
while (ssInfoLength > 3) {
@@ -264,7 +264,7 @@
// later, extract and use the "timestamp" field #####
framePtr += 8;
dataSize -= 8;
-
+
return sampleLength < dataSize ? sampleLength : dataSize;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/RTCP.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTCP
// Implementation
@@ -142,7 +142,7 @@
}
if (isSSMSource) RTCPgs->multicastSendOnly(); // don't receive multicast
-
+
double timeNow = dTimeNow();
fPrevReportTime = fNextReportTime = timeNow;
@@ -161,7 +161,7 @@
TaskScheduler::BackgroundHandlerProc* handler
= (TaskScheduler::BackgroundHandlerProc*)&incomingReportHandler;
fRTCPInterface.startNetworkReading(handler);
-
+
// Send our first report.
fTypeOfEvent = EVENT_REPORT;
onExpire(this);
@@ -296,6 +296,9 @@
void RTCPInstance::addStreamSocket(int sockNum,
unsigned char streamChannelId) {
+ // First, turn off background read handling for the default (UDP) socket:
+ fRTCPInterface.stopNetworkReading();
+
// Add the RTCP-over-TCP interface:
fRTCPInterface.setStreamSocket(sockNum, streamChannelId);
@@ -349,7 +352,7 @@
// NOTE: Denial-of-service attacks are possible here.
// Users of this software may wish to add their own,
// application-specific mechanism for 'authenticating' the
- // validity of this packet before relecting it.
+ // validity of this packet before reflecting it.
fRTCPInterface.sendPacket(pkt, packetSize);
fHaveJustSentPacket = True;
fLastPacketSentSize = packetSize;
@@ -468,7 +471,7 @@
fromAddr = tcpReadStreamSocketNum;
fromPortNum = tcpReadStreamChannelId;
}
- Port fromPort(fromPortNum);
+ Port fromPort(fromPortNum);
RRHandlerRecord* rrHandler
= (RRHandlerRecord*)(fSpecificRRHandlerTable->Lookup(fromAddr, (~0), fromPort));
if (rrHandler != NULL) {
@@ -499,7 +502,7 @@
|| (fSink != NULL
&& fSink->transmissionStatsDB().lookup(reportSenderSSRC) != NULL))) {
fByeHandlerTask = NULL;
- // we call this only once by default
+ // we call this only once by default
(*byeHandler)(fByeHandlerClientData);
}
@@ -516,7 +519,7 @@
#endif
subPacketOK = True;
break;
- }
+ }
if (!subPacketOK) break;
// need to check for (& handle) SSRC collision! #####
@@ -524,7 +527,7 @@
#ifdef DEBUG
fprintf(stderr, "validated RTCP subpacket (type %d): %d, %d, %d, 0x%08x\n", typeOfPacket, rc, pt, length, reportSenderSSRC);
#endif
-
+
// Skip over any remaining bytes in this subpacket:
ADVANCE(length);
@@ -546,7 +549,7 @@
break;
}
}
-
+
if (!packetOK) {
#ifdef DEBUG
fprintf(stderr, "rejected bad RTCP subpacket: header 0x%08x\n", rtcpHdr);
@@ -557,7 +560,7 @@
fprintf(stderr, "validated entire RTCP packet\n");
#endif
}
-
+
onReceive(typeOfPacket, totPacketSize, reportSenderSSRC);
} while (0);
}
@@ -733,7 +736,7 @@
void RTCPInstance::enqueueCommonReportSuffix() {
// Output the report blocks for each source:
- if (fSource != NULL) {
+ if (fSource != NULL) {
RTPReceptionStatsDB& allReceptionStats
= fSource->receptionStatsDB();
@@ -768,7 +771,7 @@
unsigned numExpectedSinceLastReset
= highestExtSeqNumReceived - stats->lastResetExtSeqNumReceived();
int numLostSinceLastReset
- = numExpectedSinceLastReset - stats->numPacketsReceivedSinceLastReset();
+ = numExpectedSinceLastReset - stats->numPacketsReceivedSinceLastReset();
unsigned char lossFraction;
if (numExpectedSinceLastReset == 0 || numLostSinceLastReset < 0) {
lossFraction = 0;
@@ -776,7 +779,7 @@
lossFraction = (unsigned char)
((numLostSinceLastReset << 8) / numExpectedSinceLastReset);
}
-
+
fOutBuf->enqueueWord((lossFraction<<24) | totNumLost);
fOutBuf->enqueueWord(highestExtSeqNumReceived);
@@ -798,7 +801,7 @@
timeSinceLSR.tv_sec = timeNow.tv_sec - LSRtime.tv_sec;
timeSinceLSR.tv_usec = timeNow.tv_usec - LSRtime.tv_usec;
// The enqueued time is in units of 1/65536 seconds.
- // (Note that 65536/1000000 == 1024/15625)
+ // (Note that 65536/1000000 == 1024/15625)
unsigned DLSR;
if (LSR == 0) {
DLSR = 0;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/RTPInterface.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// An abstraction of a network interface used for RTP (or RTCP).
// (This allows the RTP-over-TCP hack (RFC 2326, section 10.12) to
// be implemented transparently.)
@@ -130,13 +130,24 @@
fTCPStreams = new tcpStreamRecord(sockNum, streamChannelId, fTCPStreams);
}
+static void deregisterSocket(UsageEnvironment& env, int sockNum, unsigned char streamChannelId) {
+ SocketDescriptor* socketDescriptor = lookupSocketDescriptor(env, sockNum);
+ if (socketDescriptor != NULL) {
+ socketDescriptor->deregisterRTPInterface(streamChannelId);
+ // Note: This may delete "socketDescriptor",
+ // if no more interfaces are using this socket
+ }
+}
+
void RTPInterface::removeStreamSocket(int sockNum,
unsigned char streamChannelId) {
for (tcpStreamRecord** streamsPtr = &fTCPStreams; *streamsPtr != NULL;
streamsPtr = &((*streamsPtr)->fNext)) {
if ((*streamsPtr)->fStreamSocketNum == sockNum
&& (*streamsPtr)->fStreamChannelId == streamChannelId) {
- // Remove the record pointed to by *streamsPtr :
+ deregisterSocket(envir(), sockNum, streamChannelId);
+
+ // Then remove the record pointed to by *streamsPtr :
tcpStreamRecord* next = (*streamsPtr)->fNext;
(*streamsPtr)->fNext = NULL;
delete (*streamsPtr);
@@ -195,9 +206,9 @@
// Read from the TCP connection:
bytesRead = 0;
unsigned totBytesToRead = fNextTCPReadSize;
- if (totBytesToRead > bufferMaxSize) totBytesToRead = bufferMaxSize;
+ if (totBytesToRead > bufferMaxSize) totBytesToRead = bufferMaxSize;
unsigned curBytesToRead = totBytesToRead;
- unsigned curBytesRead;
+ int curBytesRead;
while ((curBytesRead = readSocket(envir(), fNextTCPReadStreamSocketNum,
&buffer[bytesRead], curBytesToRead,
fromAddress)) > 0) {
@@ -229,13 +240,7 @@
// Also turn off read handling on each of our TCP connections:
for (tcpStreamRecord* streams = fTCPStreams; streams != NULL;
streams = streams->fNext) {
- SocketDescriptor* socketDescriptor
- = lookupSocketDescriptor(envir(), streams->fStreamSocketNum);
- if (socketDescriptor != NULL) {
- socketDescriptor->deregisterRTPInterface(streams->fStreamChannelId);
- // Note: This may delete "socketDescriptor",
- // if no more interfaces are using this socket
- }
+ deregisterSocket(envir(), streams->fStreamSocketNum, streams->fStreamChannelId);
}
}
@@ -330,9 +335,13 @@
// (Later, fix) #####
unsigned char c;
struct sockaddr_in fromAddress;
+ struct timeval timeout; timeout.tv_sec = 0; timeout.tv_usec = 0;
do {
- if (readSocket(env, socketNum, &c, 1, fromAddress) != 1) { // error reading TCP socket
- env.taskScheduler().turnOffBackgroundReadHandling(socketNum); // stops further calls to us
+ int result = readSocket(env, socketNum, &c, 1, fromAddress, &timeout);
+ if (result != 1) { // error reading TCP socket
+ if (result < 0) {
+ env.taskScheduler().turnOffBackgroundReadHandling(socketNum); // stops further calls to us
+ }
return;
}
} while (c != '$');
@@ -378,4 +387,4 @@
tcpStreamRecord::~tcpStreamRecord() {
delete fNext;
}
-
+
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/RTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP Sinks
// Implementation
@@ -215,7 +215,7 @@
RTPTransmissionStats*
RTPTransmissionStatsDB::Iterator::next() {
char const* key; // dummy
-
+
return (RTPTransmissionStats*)(fIter->next(key));
}
@@ -237,7 +237,7 @@
: fOurRTPSink(rtpSink), fSSRC(SSRC), fLastPacketNumReceived(0),
fPacketLossRatio(0), fTotNumPacketsLost(0), fJitter(0),
fLastSRTime(0), fDiffSR_RRTime(0), fFirstPacket(True),
- fTotalOctetCount_hi(0), fTotalOctetCount_lo(0),
+ fTotalOctetCount_hi(0), fTotalOctetCount_lo(0),
fTotalPacketCount_hi(0), fTotalPacketCount_lo(0) {
gettimeofday(&fTimeCreated, NULL);
@@ -272,7 +272,7 @@
#ifdef DEBUG_RR
fprintf(stderr, "RTCP RR data (received at %lu.%06ld): lossStats 0x%08x, lastPacketNumReceived 0x%08x, jitter 0x%08x, lastSRTime 0x%08x, diffSR_RRTime 0x%08x\n",
fTimeReceived.tv_sec, fTimeReceived.tv_usec, lossStats, lastPacketNumReceived, jitter, lastSRTime, diffSR_RRTime);
- unsigned rtd = roundTripDelay();
+ unsigned rtd = roundTripDelay();
fprintf(stderr, "=> round-trip delay: 0x%04x (== %f seconds)\n", rtd, rtd/65536.0);
#endif
@@ -300,7 +300,7 @@
unsigned RTPTransmissionStats::roundTripDelay() const {
// Compute the round-trip delay that was indicated by the most recently-received
// RTCP RR packet. Use the method noted in the RTP/RTCP specification (RFC 3350).
-
+
if (fLastSRTime == 0) {
// Either no RTCP RR packet has been received yet, or else the
// reporting receiver has not yet received any RTCP SR packets from us:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/RTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP Sources
// Implementation
@@ -57,7 +57,7 @@
fRTPPayloadFormat(rtpPayloadFormat),
fTimestampFrequency(rtpTimestampFrequency),
fSSRC(our_random32()) {
- fReceptionStatsDB = new RTPReceptionStatsDB(*this);
+ fReceptionStatsDB = new RTPReceptionStatsDB();
}
RTPSource::~RTPSource() {
@@ -71,9 +71,8 @@
////////// RTPReceptionStatsDB //////////
-RTPReceptionStatsDB::RTPReceptionStatsDB(RTPSource& rtpSource)
- : fOurRTPSource(rtpSource),
- fTable(HashTable::create(ONE_WORD_HASH_KEYS)), fTotNumPacketsReceived(0) {
+RTPReceptionStatsDB::RTPReceptionStatsDB()
+ : fTable(HashTable::create(ONE_WORD_HASH_KEYS)), fTotNumPacketsReceived(0) {
reset();
}
@@ -110,7 +109,7 @@
if (stats == NULL) {
// This is the first time we've heard from this SSRC.
// Create a new record for it:
- stats = new RTPReceptionStats(fOurRTPSource, SSRC, seqNum);
+ stats = new RTPReceptionStats(SSRC, seqNum);
if (stats == NULL) return;
add(SSRC, stats);
}
@@ -133,7 +132,7 @@
if (stats == NULL) {
// This is the first time we've heard of this SSRC.
// Create a new record for it:
- stats = new RTPReceptionStats(fOurRTPSource, SSRC);
+ stats = new RTPReceptionStats(SSRC);
if (stats == NULL) return;
add(SSRC, stats);
}
@@ -186,15 +185,12 @@
////////// RTPReceptionStats //////////
-RTPReceptionStats::RTPReceptionStats(RTPSource& rtpSource, u_int32_t SSRC,
- u_int16_t initialSeqNum)
- : fOurRTPSource(rtpSource) {
+RTPReceptionStats::RTPReceptionStats(u_int32_t SSRC, u_int16_t initialSeqNum) {
initSeqNum(initialSeqNum);
init(SSRC);
}
-RTPReceptionStats::RTPReceptionStats(RTPSource& rtpSource, u_int32_t SSRC)
- : fOurRTPSource(rtpSource) {
+RTPReceptionStats::RTPReceptionStats(u_int32_t SSRC) {
init(SSRC);
}
@@ -320,8 +316,7 @@
// (as long as "int" is 32 bits)
// Divide this by the timestamp frequency to get real time:
- double timeDiff
- = timestampDiff/(double)(fOurRTPSource.timestampFrequency());
+ double timeDiff = timestampDiff/(double)timestampFrequency;
// Add this to the 'sync time' to get our result:
unsigned const million = 1000000;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/RTSPClient.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A generic RTSP client
// Implementation
@@ -100,7 +100,7 @@
// Change the existing user agent header string:
char const* const formatStr = "User-Agent: %s\r\n";
- unsigned headerSize = strlen(formatStr) + strlen(userAgentStr);
+ unsigned const headerSize = strlen(formatStr) + strlen(userAgentStr) + 1;
delete[] fUserAgentHeaderStr;
fUserAgentHeaderStr = new char[headerSize];
sprintf(fUserAgentHeaderStr, formatStr, userAgentStr);
@@ -146,10 +146,15 @@
static char* getLine(char* startOfLine) {
// returns the start of the next line, or NULL if none
for (char* ptr = startOfLine; *ptr != '\0'; ++ptr) {
+ // Check for the end of line: \r\n (but also accept \r or \n by itself):
if (*ptr == '\r' || *ptr == '\n') {
// We found the end of the line
- *ptr++ = '\0';
- if (*ptr == '\n') ++ptr;
+ if (*ptr == '\r') {
+ *ptr++ = '\0';
+ if (*ptr == '\n') ++ptr;
+ } else {
+ *ptr++ = '\0';
+ }
return ptr;
}
}
@@ -158,20 +163,20 @@
}
char* RTSPClient::describeURL(char const* url, Authenticator* authenticator,
- Boolean allowKasennaProtocol) {
+ Boolean allowKasennaProtocol, int timeout) {
char* cmd = NULL;
fDescribeStatusCode = 0;
- do {
+ do {
// First, check whether "url" contains a username:password to be used:
char* username; char* password;
if (authenticator == NULL
&& parseRTSPURLUsernamePassword(url, username, password)) {
- char* result = describeWithPassword(url, username, password, allowKasennaProtocol);
+ char* result = describeWithPassword(url, username, password, allowKasennaProtocol, timeout);
delete[] username; delete[] password; // they were dynamically allocated
return result;
}
- if (!openConnectionFromURL(url, authenticator)) break;
+ if (!openConnectionFromURL(url, authenticator, timeout)) break;
// Send the DESCRIBE command:
@@ -260,7 +265,7 @@
} else if (strncmp(lineStart, "Content-Base:", 13) == 0) {
int cbIndex = 13;
- while (lineStart[cbIndex] == ' ' || lineStart[cbIndex] == '\t') ++cbIndex;
+ while (lineStart[cbIndex] == ' ' || lineStart[cbIndex] == '\t') ++cbIndex;
if (lineStart[cbIndex] != '\0'/*sanity check*/) {
delete[] fBaseURL; fBaseURL = strDup(&lineStart[cbIndex]);
}
@@ -270,7 +275,7 @@
#ifdef SUPPORT_REAL_RTSP
} else if (sscanf(lineStart, "ETag: %s", fRealETagStr) == 1) {
#endif
- } else if (wantRedirection) {
+ } else if (wantRedirection) {
if (sscanf(lineStart, "Location: %s", redirectionURL) == 1) {
// Try again with this URL
if (fVerbosityLevel >= 1) {
@@ -278,13 +283,14 @@
<< redirectionURL << "\"\n";
}
reset();
- char* result = describeURL(redirectionURL, authenticator, allowKasennaProtocol);
+ char* result = describeURL(redirectionURL, authenticator, allowKasennaProtocol, timeout);
delete[] redirectionURL;
delete[] serverType;
+ delete[] cmd;
return result;
}
}
- }
+ }
delete[] serverType;
// We're now at the end of the response header lines
@@ -369,10 +375,10 @@
delete[] fKasennaContentType;
fKasennaContentType = new char[fResponseBufferSize]; // ensures enough space
char* currentPos = bodyStart;
-
+
while (strcmp(currentWord, "</MediaDescription>") != 0) {
sscanf(currentPos, "%s", currentWord);
-
+
if (strcmp(currentWord, "VideoPid") == 0) {
currentPos += strlen(currentWord) + 1;
sscanf(currentPos, "%s", currentWord);
@@ -380,7 +386,7 @@
sscanf(currentPos, "%d", &videoPid);
currentPos += 3;
}
-
+
if (strcmp(currentWord, "AudioPid") == 0) {
currentPos += strlen(currentWord) + 1;
sscanf(currentPos, "%s", currentWord);
@@ -388,7 +394,7 @@
sscanf(currentPos, "%d", &audioPid);
currentPos += 3;
}
-
+
if (strcmp(currentWord, "Duration") == 0) {
currentPos += strlen(currentWord) + 1;
sscanf(currentPos, "%s", currentWord);
@@ -396,7 +402,7 @@
sscanf(currentPos, "%llu", &mh_duration);
currentPos += 3;
}
-
+
if (strcmp(currentWord, "TypeSpecificData") == 0) {
currentPos += strlen(currentWord) + 1;
sscanf(currentPos, "%s", currentWord);
@@ -405,24 +411,24 @@
currentPos += 3;
printf("Kasenna Content Type: %s\n", fKasennaContentType);
}
-
+
currentPos += strlen(currentWord) + 1;
}
-
+
if (fKasennaContentType != NULL
&& strcmp(fKasennaContentType, "PARTNER_41_MPEG-4") == 0) {
- char* describeSDP = describeURL(url, authenticator, True);
-
+ char* describeSDP = describeURL(url, authenticator, True, timeout);
+
delete[] currentWord;
delete[] cmd;
return describeSDP;
}
-
+
unsigned char byte1 = fServerAddress & 0x000000ff;
unsigned char byte2 = (fServerAddress & 0x0000ff00) >> 8;
unsigned char byte3 = (fServerAddress & 0x00ff0000) >> 16;
unsigned char byte4 = (fServerAddress & 0xff000000) >> 24;
-
+
char const* sdpFmt =
"v=0\r\n"
"o=NoSpacesAllowed 1 1 IN IP4 %u.%u.%u.%u\r\n"
@@ -445,18 +451,18 @@
byte1, byte2, byte3, byte4,
mh_duration/1000000,
videoPid);
-
+
char* result = strDup(sdpBuf);
delete[] sdpBuf; delete[] currentWord;
delete[] cmd;
return result;
}
////////// END Kasenna BS //////////
-
+
delete[] cmd;
return strDup(bodyStart);
} while (0);
-
+
delete[] cmd;
if (fDescribeStatusCode == 0) fDescribeStatusCode = 2;
return NULL;
@@ -465,10 +471,10 @@
char* RTSPClient
::describeWithPassword(char const* url,
char const* username, char const* password,
- Boolean allowKasennaProtocol) {
+ Boolean allowKasennaProtocol, int timeout) {
Authenticator authenticator;
authenticator.setUsernameAndPassword(username, password);
- char* describeResult = describeURL(url, &authenticator, allowKasennaProtocol);
+ char* describeResult = describeURL(url, &authenticator, allowKasennaProtocol, timeout);
if (describeResult != NULL) {
// We are already authorized
return describeResult;
@@ -481,7 +487,7 @@
}
// Try again:
- describeResult = describeURL(url, &authenticator, allowKasennaProtocol);
+ describeResult = describeURL(url, &authenticator, allowKasennaProtocol, timeout);
if (describeResult != NULL) {
// The authenticator worked, so use it in future requests:
fCurrentAuthenticator = authenticator;
@@ -492,7 +498,8 @@
char* RTSPClient::sendOptionsCmd(char const* url,
char* username, char* password,
- Authenticator* authenticator) {
+ Authenticator* authenticator,
+ int timeout) {
char* result = NULL;
char* cmd = NULL;
Boolean haveAllocatedAuthenticator = False;
@@ -502,9 +509,9 @@
// (and no username,password pair was supplied separately):
if (username == NULL && password == NULL
&& parseRTSPURLUsernamePassword(url, username, password)) {
- Authenticator authenticator;
- authenticator.setUsernameAndPassword(username, password);
- result = sendOptionsCmd(url, username, password, &authenticator);
+ Authenticator newAuthenticator;
+ newAuthenticator.setUsernameAndPassword(username, password);
+ result = sendOptionsCmd(url, username, password, &newAuthenticator, timeout);
delete[] username; delete[] password; // they were dynamically allocated
break;
} else if (username != NULL && password != NULL) {
@@ -513,7 +520,7 @@
haveAllocatedAuthenticator = True;
authenticator->setUsernameAndPassword(username, password);
- result = sendOptionsCmd(url, username, password, authenticator);
+ result = sendOptionsCmd(url, username, password, authenticator, timeout);
if (result != NULL) break; // We are already authorized
// The "realm" field should have been filled in:
@@ -525,7 +532,7 @@
}
}
- if (!openConnectionFromURL(url, authenticator)) break;
+ if (!openConnectionFromURL(url, authenticator, timeout)) break;
// Send the OPTIONS command:
@@ -602,22 +609,29 @@
return False;
}
+char const* RTSPClient::sessionURL(MediaSession const& session) const {
+ char const* url = session.controlPath();
+ if (url == NULL || strcmp(url, "*") == 0) url = fBaseURL;
+
+ return url;
+}
+
void RTSPClient::constructSubsessionURL(MediaSubsession const& subsession,
char const*& prefix,
char const*& separator,
char const*& suffix) {
// Figure out what the URL describing "subsession" will look like.
// The URL is returned in three parts: prefix; separator; suffix
- //##### NOTE: This code doesn't really do the right thing if "fBaseURL"
+ //##### NOTE: This code doesn't really do the right thing if "sessionURL()"
// doesn't end with a "/", and "subsession.controlPath()" is relative.
- // The right thing would have been to truncate "fBaseURL" back to the
+ // The right thing would have been to truncate "sessionURL()" back to the
// rightmost "/", and then add "subsession.controlPath()".
// In practice, though, each "DESCRIBE" response typically contains
- // a "Content-Base:" header that consists of "fBaseURL" followed by
+ // a "Content-Base:" header that consists of "sessionURL()" followed by
// a "/", in which case this code ends up giving the correct result.
// However, we should really fix this code to do the right thing, and
// also check for and use the "Content-Base:" header appropriately. #####
- prefix = fBaseURL;
+ prefix = sessionURL(subsession.parentSession());
if (prefix == NULL) prefix = "";
suffix = subsession.controlPath();
@@ -633,10 +647,11 @@
Boolean RTSPClient::announceSDPDescription(char const* url,
char const* sdpDescription,
- Authenticator* authenticator) {
+ Authenticator* authenticator,
+ int timeout) {
char* cmd = NULL;
do {
- if (!openConnectionFromURL(url, authenticator)) break;
+ if (!openConnectionFromURL(url, authenticator, timeout)) break;
// Send the ANNOUNCE command:
@@ -694,10 +709,10 @@
Boolean RTSPClient
::announceWithPassword(char const* url, char const* sdpDescription,
- char const* username, char const* password) {
+ char const* username, char const* password, int timeout) {
Authenticator authenticator;
authenticator.setUsernameAndPassword(username, password);
- if (announceSDPDescription(url, sdpDescription, &authenticator)) {
+ if (announceSDPDescription(url, sdpDescription, &authenticator, timeout)) {
// We are already authorized
return True;
}
@@ -710,7 +725,7 @@
// Try again:
Boolean secondTrySuccess
- = announceSDPDescription(url, sdpDescription, &authenticator);
+ = announceSDPDescription(url, sdpDescription, &authenticator, timeout);
if (secondTrySuccess) {
// The authenticator worked, so use it in future requests:
@@ -750,7 +765,7 @@
sessionStr = new char[20+strlen(fLastSessionId)];
sprintf(sessionStr, "Session: %s\r\n", fLastSessionId);
} else {
- sessionStr = "";
+ sessionStr = strDup("");
}
char* transportStr = NULL;
@@ -856,7 +871,7 @@
portTypeStr = ";interleaved";
rtpNumber = fTCPStreamIdCount++;
rtcpNumber = fTCPStreamIdCount++;
- } else { // normal RTP streaming
+ } else { // normal RTP streaming
unsigned connectionAddress = subsession.connectionEndpointAddress();
Boolean requestMulticastStreaming = IsMulticastAddress(connectionAddress)
|| (connectionAddress == 0 && forceMulticastOnUnspecified);
@@ -865,6 +880,7 @@
rtpNumber = subsession.clientPortNum();
if (rtpNumber == 0) {
envir().setResultMsg("Client port number unknown\n");
+ delete[] authenticatorStr; delete[] sessionStr; delete[] setupStr;
break;
}
rtcpNumber = rtpNumber + 1;
@@ -902,9 +918,7 @@
sessionStr,
authenticatorStr,
fUserAgentHeaderStr);
- delete[] authenticatorStr;
- if (sessionStr[0] != '\0') delete[] sessionStr;
- delete[] setupStr; delete[] transportStr;
+ delete[] authenticatorStr; delete[] sessionStr; delete[] setupStr; delete[] transportStr;
// And then send it:
if (!sendRequest(cmd, "SETUP")) break;
@@ -919,6 +933,7 @@
// For now, ignore other headers.
char* lineStart;
char* sessionId = new char[fResponseBufferSize]; // ensures we have enough space
+ unsigned cLength = 0;
while (1) {
lineStart = nextLineStart;
if (lineStart == NULL) break;
@@ -952,7 +967,11 @@
subsession.rtcpChannelId = rtcpChannelId;
continue;
}
- }
+
+ // Also check for a "Content-Length:" header. Some weird servers include this
+ // in the RTSP "SETUP" response.
+ if (sscanf(lineStart, "Content-Length: %d", &cLength) == 1) continue;
+ }
delete[] sessionId;
if (subsession.sessionId == NULL) {
@@ -960,6 +979,14 @@
break;
}
+ // If we saw a "Content-Length:" header in the response, then discard whatever
+ // included data it refers to:
+ if (cLength > 0) {
+ char* dummyBuf = new char[cLength];
+ getResponse1(dummyBuf, cLength);
+ delete[] dummyBuf;
+ }
+
if (streamUsingTCP) {
// Tell the subsession to receive RTP (and send/receive RTCP)
// over the RTSP stream:
@@ -972,8 +999,10 @@
} else {
// Normal case.
// Set the RTP and RTCP sockets' destination address and port
- // from the information in the SETUP response:
- subsession.setDestinations(fServerAddress);
+ // from the information in the SETUP response (if present):
+ netAddressBits destAddress = subsession.connectionEndpointAddress();
+ if (destAddress == 0) destAddress = fServerAddress;
+ subsession.setDestinations(destAddress);
}
delete[] cmd;
@@ -990,35 +1019,35 @@
// This is the default value; we don't need a "Scale:" header:
buf[0] = '\0';
} else {
- Locale("POSIX", LC_NUMERIC);
+ Locale("C", LC_NUMERIC);
sprintf(buf, "Scale: %f\r\n", scale);
}
return strDup(buf);
}
-
-static char* createRangeString(float start, float end) {
+
+static char* createRangeString(double start, double end) {
char buf[100];
if (start < 0) {
// We're resuming from a PAUSE; there's no "Range:" header at all
buf[0] = '\0';
} else if (end < 0) {
// There's no end time:
- Locale("POSIX", LC_NUMERIC);
+ Locale("C", LC_NUMERIC);
sprintf(buf, "Range: npt=%.3f-\r\n", start);
} else {
// There's both a start and an end time; include them both in the "Range:" hdr
- Locale("POSIX", LC_NUMERIC);
+ Locale("C", LC_NUMERIC);
sprintf(buf, "Range: npt=%.3f-%.3f\r\n", start, end);
}
return strDup(buf);
}
-
+
static char const* NoSessionErr = "No RTSP session is currently in progress\n";
Boolean RTSPClient::playMediaSession(MediaSession& session,
- float start, float end, float scale) {
+ double start, double end, float scale) {
#ifdef SUPPORT_REAL_RTSP
if (session.isRealNetworksRDT) {
// This is a RealNetworks stream; set the "Subscribe" parameter before proceeding:
@@ -1055,8 +1084,9 @@
"%s"
"\r\n";
+ char const* sessURL = sessionURL(session);
unsigned cmdSize = strlen(cmdFmt)
- + strlen(fBaseURL)
+ + strlen(sessURL)
+ 20 /* max int len */
+ strlen(fLastSessionId)
+ strlen(scaleStr)
@@ -1065,7 +1095,7 @@
+ fUserAgentHeaderStrSize;
cmd = new char[cmdSize];
sprintf(cmd, cmdFmt,
- fBaseURL,
+ sessURL,
++fCSeq,
fLastSessionId,
scaleStr,
@@ -1091,7 +1121,23 @@
nextLineStart = getLine(lineStart);
- if (parseScaleHeader(lineStart, session.scale())) break;
+ if (parseScaleHeader(lineStart, session.scale())) continue;
+ if (parseRangeHeader(lineStart, session.playStartTime(), session.playEndTime())) continue;
+
+ u_int16_t seqNum; u_int32_t timestamp;
+ if (parseRTPInfoHeader(lineStart, seqNum, timestamp)) {
+ // This is data for our first subsession. Fill it in, and do the same for our other subsessions:
+ MediaSubsessionIterator iter(session);
+ MediaSubsession* subsession;
+ while ((subsession = iter.next()) != NULL) {
+ subsession->rtpInfo.seqNum = seqNum;
+ subsession->rtpInfo.timestamp = timestamp;
+ subsession->rtpInfo.infoIsNew = True;
+
+ if (!parseRTPInfoHeader(lineStart, seqNum, timestamp)) break;
+ }
+ continue;
+ }
}
if (fTCPStreamIdCount == 0) { // we're not receiving RTP-over-TCP
@@ -1109,7 +1155,7 @@
}
Boolean RTSPClient::playMediaSubsession(MediaSubsession& subsession,
- float start, float end, float scale,
+ double start, double end, float scale,
Boolean hackForDSS) {
char* cmd = NULL;
do {
@@ -1185,13 +1231,16 @@
nextLineStart = getLine(lineStart);
- if (parseRTPInfoHeader(lineStart,
- subsession.rtpInfo.trackId,
- subsession.rtpInfo.seqNum,
- subsession.rtpInfo.timestamp)) {
+ if (parseScaleHeader(lineStart, subsession.scale())) continue;
+ if (parseRangeHeader(lineStart, subsession._playStartTime(), subsession._playEndTime())) continue;
+
+ u_int16_t seqNum; u_int32_t timestamp;
+ if (parseRTPInfoHeader(lineStart, seqNum, timestamp)) {
+ subsession.rtpInfo.seqNum = seqNum;
+ subsession.rtpInfo.timestamp = timestamp;
+ subsession.rtpInfo.infoIsNew = True;
continue;
}
- if (parseScaleHeader(lineStart, subsession.scale())) continue;
}
delete[] cmd;
@@ -1217,6 +1266,7 @@
char* authenticatorStr
= createAuthenticatorString(&fCurrentAuthenticator, "PAUSE", fBaseURL);
+ char const* sessURL = sessionURL(session);
char* const cmdFmt =
"PAUSE %s RTSP/1.0\r\n"
"CSeq: %d\r\n"
@@ -1226,14 +1276,14 @@
"\r\n";
unsigned cmdSize = strlen(cmdFmt)
- + strlen(fBaseURL)
+ + strlen(sessURL)
+ 20 /* max int len */
+ strlen(fLastSessionId)
+ strlen(authenticatorStr)
+ fUserAgentHeaderStrSize;
cmd = new char[cmdSize];
sprintf(cmd, cmdFmt,
- fBaseURL,
+ sessURL,
++fCSeq,
fLastSessionId,
authenticatorStr,
@@ -1265,13 +1315,13 @@
envir().setResultMsg(NoSessionErr);
break;
}
-
+
// Send the PAUSE command:
-
+
// First, construct an authenticator string:
char* authenticatorStr
= createAuthenticatorString(&fCurrentAuthenticator, "PAUSE", fBaseURL);
-
+
char* const cmdFmt =
"PAUSE %s%s%s RTSP/1.0\r\n"
"CSeq: %d\r\n"
@@ -1279,7 +1329,7 @@
"%s"
"%s"
"\r\n";
-
+
char const *prefix, *separator, *suffix;
constructSubsessionURL(subsession, prefix, separator, suffix);
if (fServerIsKasenna) separator = suffix = "";
@@ -1298,7 +1348,7 @@
authenticatorStr,
fUserAgentHeaderStr);
delete[] authenticatorStr;
-
+
if (!sendRequest(cmd, "PAUSE")) break;
if (fTCPStreamIdCount == 0) { // When TCP streaming, don't look for a response
@@ -1307,11 +1357,11 @@
char* firstLine; char* nextLineStart;
if (!getResponse("PAUSE", bytesRead, responseCode, firstLine, nextLineStart)) break;
}
-
+
delete[] cmd;
return True;
} while (0);
-
+
delete[] cmd;
return False;
}
@@ -1398,16 +1448,18 @@
"Session: %s\r\n"
"%s"
"%s"
- "%s: %s\r\n"
- "\r\n";
+ "Content-length: %d\r\n\r\n"
+ "%s: %s\r\n";
+ unsigned parameterNameLen = strlen(parameterName);
+ unsigned parameterValueLen = strlen(parameterValue);
unsigned cmdSize = strlen(cmdFmt)
+ strlen(fBaseURL)
+ 20 /* max int len */
+ strlen(fLastSessionId)
+ strlen(authenticatorStr)
+ fUserAgentHeaderStrSize
- + strlen(parameterName) + strlen(parameterValue);
+ + parameterNameLen + parameterValueLen;
cmd = new char[cmdSize];
sprintf(cmd, cmdFmt,
fBaseURL,
@@ -1415,6 +1467,7 @@
fLastSessionId,
authenticatorStr,
fUserAgentHeaderStr,
+ parameterNameLen + parameterValueLen + 2, // the "+ 2" is for the \r\n after the parameter "name: value"
parameterName, parameterValue);
delete[] authenticatorStr;
@@ -1461,16 +1514,16 @@
"%s"
"Content-type: text/parameters\r\n"
"Content-length: %d\r\n\r\n"
- "%s\r\n"
- "\r\n";
-
+ "%s\r\n";
+
+ unsigned parameterNameLen = strlen(parameterName);
unsigned cmdSize = strlen(cmdFmt)
+ strlen(fBaseURL)
+ 20 /* max int len */
+ strlen(fLastSessionId)
+ strlen(authenticatorStr)
+ fUserAgentHeaderStrSize
- + strlen(parameterName);
+ + parameterNameLen;
cmd = new char[cmdSize];
sprintf(cmd, cmdFmt,
fBaseURL,
@@ -1478,7 +1531,7 @@
fLastSessionId,
authenticatorStr,
fUserAgentHeaderStr,
- strlen(parameterName)+2,
+ parameterNameLen + 2, // the "+ 2" is for the \r\n after the parameter name
parameterName);
} else {
char* const cmdFmt =
@@ -1488,7 +1541,7 @@
"%s"
"%s"
"\r\n";
-
+
unsigned cmdSize = strlen(cmdFmt)
+ strlen(fBaseURL)
+ 20 /* max int len */
@@ -1511,7 +1564,7 @@
// This section was copied/modified from the RTSPClient::describeURL func
unsigned bytesRead; unsigned responseCode;
char* firstLine; char* nextLineStart;
- if (!getResponse("GET_PARAMETER", bytesRead, responseCode, firstLine,
+ if (!getResponse("GET_PARAMETER", bytesRead, responseCode, firstLine,
nextLineStart, False /*don't check for response code 200*/)) break;
// Inspect the first line to check whether it's a result code that
@@ -1541,12 +1594,12 @@
break;
}
}
- }
+ }
delete[] serverType;
// We're now at the end of the response header lines
if (lineStart == NULL) {
- envir().setResultMsg("no content following header lines: ",
+ envir().setResultMsg("no content following header lines: ",
fResponseBuffer);
break;
}
@@ -1603,7 +1656,7 @@
delete[] cmd;
return True;
} while (0);
-
+
delete[] cmd;
return False;
}
@@ -1624,6 +1677,7 @@
= createAuthenticatorString(&fCurrentAuthenticator,
"TEARDOWN", fBaseURL);
+ char const* sessURL = sessionURL(session);
char* const cmdFmt =
"TEARDOWN %s RTSP/1.0\r\n"
"CSeq: %d\r\n"
@@ -1633,14 +1687,14 @@
"\r\n";
unsigned cmdSize = strlen(cmdFmt)
- + strlen(fBaseURL)
+ + strlen(sessURL)
+ 20 /* max int len */
+ strlen(fLastSessionId)
+ strlen(authenticatorStr)
+ fUserAgentHeaderStrSize;
cmd = new char[cmdSize];
sprintf(cmd, cmdFmt,
- fBaseURL,
+ sessURL,
++fCSeq,
fLastSessionId,
authenticatorStr,
@@ -1654,7 +1708,7 @@
unsigned bytesRead; unsigned responseCode;
char* firstLine; char* nextLineStart;
if (!getResponse("TEARDOWN", bytesRead, responseCode, firstLine, nextLineStart)) break;
-
+
// Run through each subsession, deleting its "sessionId":
MediaSubsessionIterator iter(session);
MediaSubsession* subsession;
@@ -1666,7 +1720,7 @@
delete[] fLastSessionId; fLastSessionId = NULL;
// we're done with this session
}
-
+
delete[] cmd;
return True;
} while (0);
@@ -1725,7 +1779,7 @@
char* firstLine; char* nextLineStart;
if (!getResponse("TEARDOWN", bytesRead, responseCode, firstLine, nextLineStart)) break;
}
-
+
delete[] (char*)subsession.sessionId;
subsession.sessionId = NULL;
// we're done with this session
@@ -1739,7 +1793,7 @@
}
Boolean RTSPClient
-::openConnectionFromURL(char const* url, Authenticator* authenticator) {
+::openConnectionFromURL(char const* url, Authenticator* authenticator, int timeout) {
do {
// Set this as our base URL:
delete[] fBaseURL; fBaseURL = strDup(url); if (fBaseURL == NULL) break;
@@ -1752,26 +1806,60 @@
if (!parseRTSPURL(envir(), url, destAddress, urlPortNum, &urlSuffix)) break;
portNumBits destPortNum
= fTunnelOverHTTPPortNum == 0 ? urlPortNum : fTunnelOverHTTPPortNum;
-
+
if (fInputSocketNum < 0) {
// We don't yet have a TCP socket. Set one up (blocking) now:
fInputSocketNum = fOutputSocketNum
= setupStreamSocket(envir(), 0, False /* =>blocking */);
if (fInputSocketNum < 0) break;
-
+
// Connect to the remote endpoint:
fServerAddress = *(unsigned*)(destAddress.data());
MAKE_SOCKADDR_IN(remoteName, fServerAddress, htons(destPortNum));
+ //Start change for timeout on connect
+
+/*
if (connect(fInputSocketNum, (struct sockaddr*)&remoteName, sizeof remoteName)
!= 0) {
envir().setResultErrMsg("connect() failed: ");
break;
+*/
+ fd_set set;
+ FD_ZERO(&set);
+ timeval tvout = {0,0};
+ if (timeout > 0) {
+ FD_SET((unsigned)fInputSocketNum, &set);
+ tvout.tv_sec = timeout;
+ tvout.tv_usec = 0;
+ makeSocketNonBlocking(fInputSocketNum);
+ }
+ if (connect(fInputSocketNum, (struct sockaddr*) &remoteName, sizeof remoteName) != 0) {
+#if defined(__WIN32__) || defined(_WIN32)
+ if (errno != WSAEINPROGRESS) {
+#else
+ if (errno != EINPROGRESS) {
+#endif
+ envir().setResultErrMsg("connect() failed: ");
+ break;
+ }
+ if (timeout > 0 && (select(fInputSocketNum + 1, NULL, &set, NULL, &tvout) <= 0)) {
+ envir().setResultErrMsg("select/connect() failed: ");
+ break;
+ }
+/*
+errno = 0;
+if (getsockopt(fd, SOL_SOCKET, SO_ERROR, &err, &len) < 0 || errno != 0 )
+{
+break;
+}
+*/
+ //End change for timeout on connect
}
-
+
if (fTunnelOverHTTPPortNum != 0 && !setupHTTPTunneling(urlSuffix, authenticator)) break;
}
- return True;
+ return True;
} while (0);
fDescribeStatusCode = 1;
@@ -1927,7 +2015,7 @@
sprintf(usernamePassword, "%s:%s", authenticator->username(), authenticator->password());
char* response = base64Encode(usernamePassword, usernamePasswordLength);
- unsigned authBufSize = strlen(authFmt) + strlen(response);
+ unsigned const authBufSize = strlen(authFmt) + strlen(response) + 1;
authenticatorStr = new char[authBufSize];
sprintf(authenticatorStr, authFmt, response);
delete[] response; delete[] usernamePassword;
@@ -1969,7 +2057,7 @@
}
delete[] realm; delete[] nonce;
if (foundAuthenticateHeader) break;
- }
+ }
}
}
@@ -2022,7 +2110,7 @@
nextLineStart = getLine(firstLine);
if (!parseResponseCode(firstLine, responseCode)) break;
-
+
if (responseCode != 200 && checkFor200Response) {
envir().setResultMsg(tag, ": cannot handle response: ", firstLine);
break;
@@ -2038,7 +2126,7 @@
unsigned RTSPClient::getResponse1(char*& responseBuffer,
unsigned responseBufferSize) {
struct sockaddr_in fromAddress;
-
+
if (responseBufferSize == 0) return 0; // just in case...
responseBuffer[0] = '\0'; // ditto
@@ -2048,7 +2136,9 @@
Boolean success = False;
while (1) {
unsigned char firstByte;
- if (readSocket(envir(), fInputSocketNum, &firstByte, 1, fromAddress)
+ struct timeval timeout;
+ timeout.tv_sec = 30; timeout.tv_usec = 0;
+ if (readSocket(envir(), fInputSocketNum, &firstByte, 1, fromAddress, &timeout)
!= 1) break;
if (firstByte != '$') {
// Normal case: This is the start of a regular response; use it:
@@ -2075,7 +2165,7 @@
if (tmpBuffer == NULL) break;
unsigned bytesRead = 0;
unsigned bytesToRead = size;
- unsigned curBytesRead;
+ int curBytesRead;
while ((curBytesRead = readSocket(envir(), fInputSocketNum,
&tmpBuffer[bytesRead], bytesToRead,
fromAddress)) > 0) {
@@ -2090,7 +2180,7 @@
}
}
if (!success) return 0;
-
+
// Keep reading data from the socket until we see "\r\n\r\n" (except
// at the start), or until we fill up our buffer.
// Don't read any more than this.
@@ -2098,23 +2188,24 @@
Boolean haveSeenNonCRLF = False;
int bytesRead = 1; // because we've already read the first byte
while (bytesRead < (int)responseBufferSize) {
- unsigned bytesReadNow
+ int bytesReadNow
= readSocket(envir(), fInputSocketNum,
(unsigned char*)(responseBuffer+bytesRead),
1, fromAddress);
- if (bytesReadNow == 0) {
+ if (bytesReadNow <= 0) {
envir().setResultMsg("RTSP response was truncated");
break;
}
bytesRead += bytesReadNow;
-
- // Check whether we have "\r\n\r\n":
+
+ // Check whether we have "\r\n\r\n" (or "\r\r" or "\n\n"):
char* lastToCheck = responseBuffer+bytesRead-4;
if (lastToCheck < responseBuffer) continue;
for (; p <= lastToCheck; ++p) {
if (haveSeenNonCRLF) {
- if (*p == '\r' && *(p+1) == '\n' &&
- *(p+2) == '\r' && *(p+3) == '\n') {
+ if ((*p == '\r' && *(p+1) == '\n' && *(p+2) == '\r' && *(p+3) == '\n')
+ || (*(p+2) == '\r' && *(p+3) == '\r')
+ || (*(p+2) == '\n' && *(p+3) == '\n')) {
responseBuffer[bytesRead] = '\0';
// Before returning, trim any \r or \n from the start:
@@ -2131,12 +2222,12 @@
}
}
}
-
+
envir().setResultMsg("We received a response not ending with <CR><LF><CR><LF>");
return 0;
}
-Boolean RTSPClient::parseResponseCode(char const* line,
+Boolean RTSPClient::parseResponseCode(char const* line,
unsigned& responseCode) {
if (sscanf(line, "%*s%u", &responseCode) != 1) {
envir().setResultMsg("no response code in line: \"", line, "\"");
@@ -2146,7 +2237,7 @@
return True;
}
-Boolean RTSPClient::parseTransportResponse(char const* line,
+Boolean RTSPClient::parseTransportResponse(char const* line,
char*& serverAddressStr,
portNumBits& serverPortNum,
unsigned char& rtpChannelId,
@@ -2218,25 +2309,29 @@
return False;
}
-Boolean RTSPClient::parseRTPInfoHeader(char const* line,
- unsigned& trackId,
- u_int16_t& seqNum,
- u_int32_t& timestamp) {
- if (_strncasecmp(line, "RTP-Info: ", 10) != 0) return False;
- line += 10;
- char const* fields = line;
- char* field = strDupSize(fields);
-
- while (sscanf(fields, "%[^;]", field) == 1) {
- if (sscanf(field, "url=trackID=%u", &trackId) == 1 ||
- sscanf(field, "url=trackid=%u", &trackId) == 1 ||
- sscanf(field, "seq=%hu", &seqNum) == 1 ||
+Boolean RTSPClient::parseRTPInfoHeader(char*& line, u_int16_t& seqNum, u_int32_t& timestamp) {
+ // At this point in the parsing, "line" should begin with either "RTP-Info: " (for the start of the header),
+ // or ",", indicating the RTP-Info parameter list for the 2nd-through-nth subsessions:
+ if (_strncasecmp(line, "RTP-Info: ", 10) == 0) {
+ line += 10;
+ } else if (line[0] == ',') {
+ ++line;
+ } else {
+ return False;
+ }
+
+ // "line" now consists of a ';'-separated list of parameters, ending with ',' or '\0'.
+ char* field = strDupSize(line);
+
+ while (sscanf(line, "%[^;,]", field) == 1) {
+ if (sscanf(field, "seq=%hu", &seqNum) == 1 ||
sscanf(field, "rtptime=%u", ×tamp) == 1) {
}
-
- fields += strlen(field);
- if (fields[0] == '\0') break;
- ++fields; // skip over the ';'
+
+ line += strlen(field);
+ if (line[0] == '\0' || line[0] == ',') break;
+ // ASSERT: line[0] == ';'
+ ++line; // skip over the ';'
}
delete[] field;
@@ -2247,14 +2342,14 @@
if (_strncasecmp(line, "Scale: ", 7) != 0) return False;
line += 7;
- Locale("POSIX", LC_NUMERIC);
+ Locale("C", LC_NUMERIC);
return sscanf(line, "%f", &scale) == 1;
}
-Boolean RTSPClient::parseGetParameterHeader(char const* line,
+Boolean RTSPClient::parseGetParameterHeader(char const* line,
const char* param,
char*& value) {
- if ((param != NULL && param[0] != '\0') &&
+ if ((param != NULL && param[0] != '\0') &&
(line != NULL && line[0] != '\0')) {
int param_len = strlen(param);
int line_len = strlen(line);
@@ -2265,7 +2360,7 @@
}
return False;
}
-
+
// Strip \r\n from the end if it's there.
if (line[line_len-2] == '\r' && line[line_len-1] == '\n') {
line_len -= 2;
@@ -2281,7 +2376,7 @@
param_len += 2;
}
}
-
+
// Get the string we want out of the line:
value = strDup(line+param_len);
return True;
@@ -2313,7 +2408,7 @@
our_MD5Data((unsigned char*)(&seedData), sizeof seedData, sessionCookie);
// DSS seems to require that the 'session cookie' string be 22 bytes long:
sessionCookie[23] = '\0';
-
+
// Construct an authenticator string:
char* authenticatorStr
= createAuthenticatorString(authenticator, "GET", urlSuffix);
@@ -2341,7 +2436,7 @@
sessionCookie);
delete[] authenticatorStr;
if (!sendRequest(cmd, "HTTP GET", False/*don't base64-encode*/)) break;
-
+
// Get the response from the server:
unsigned bytesRead; unsigned responseCode;
char* firstLine; char* nextLineStart;
@@ -2358,7 +2453,7 @@
// this socket.
fOutputSocketNum = setupStreamSocket(envir(), 0, False /* =>blocking */);
if (fOutputSocketNum < 0) break;
-
+
// Connect to the remote endpoint:
MAKE_SOCKADDR_IN(remoteName, fServerAddress, htons(fTunnelOverHTTPPortNum));
if (connect(fOutputSocketNum,
@@ -2393,7 +2488,7 @@
sessionCookie);
delete[] authenticatorStr;
if (!sendRequest(cmd, "HTTP POST", False/*don't base64-encode*/)) break;
-
+
// Note that there's no response to the "POST".
delete[] cmd;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/RTSPCommon.cpp
^
|
| @@ -11,14 +11,17 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Common routines used by both RTSP clients and servers
// Implementation
#include "RTSPCommon.hh"
+#include "Locale.hh"
+#include <string.h>
+#include <stdio.h>
Boolean parseRTSPRequestString(char const* reqStr,
unsigned reqStrSize,
@@ -46,11 +49,11 @@
}
resultCmdName[i] = '\0';
if (!parseSucceeded) return False;
-
+
// Skip over the prefix of any "rtsp://" or "rtsp:/" URL that follows:
unsigned j = i+1;
while (j < reqStrSize && (reqStr[j] == ' ' || reqStr[j] == '\t')) ++j; // skip over any additional white space
- for (j = i+1; j < reqStrSize-8; ++j) {
+ for (; (int)j < (int)(reqStrSize-8); ++j) {
if ((reqStr[j] == 'r' || reqStr[j] == 'R')
&& (reqStr[j+1] == 't' || reqStr[j+1] == 'T')
&& (reqStr[j+2] == 's' || reqStr[j+2] == 'S')
@@ -72,7 +75,7 @@
// Look for the URL suffix (before the following "RTSP/"):
parseSucceeded = False;
- for (unsigned k = i+1; k < reqStrSize-5; ++k) {
+ for (unsigned k = i+1; (int)k < (int)(reqStrSize-5); ++k) {
if (reqStr[k] == 'R' && reqStr[k+1] == 'T' &&
reqStr[k+2] == 'S' && reqStr[k+3] == 'P' && reqStr[k+4] == '/') {
while (--k >= i && reqStr[k] == ' ') {} // go back over all spaces before "RTSP/"
@@ -87,7 +90,7 @@
resultURLSuffix[n] = '\0';
// Also look for the URL 'pre-suffix' before this:
- unsigned k3 = --k1;
+ unsigned k3 = (k1 == 0) ? 0 : --k1;
while (k3 > i && reqStr[k3] != '/') --k3;
// the URL pre-suffix comes from [k3+1,k1]
@@ -107,7 +110,7 @@
// Look for "CSeq:", skip whitespace,
// then read everything up to the next \r or \n as 'CSeq':
parseSucceeded = False;
- for (j = i; j < reqStrSize-5; ++j) {
+ for (j = i; (int)j < (int)(reqStrSize-5); ++j) {
if (reqStr[j] == 'C' && reqStr[j+1] == 'S' && reqStr[j+2] == 'e' &&
reqStr[j+3] == 'q' && reqStr[j+4] == ':') {
j += 5;
@@ -130,3 +133,29 @@
return True;
}
+
+Boolean parseRangeHeader(char const* buf, double& rangeStart, double& rangeEnd) {
+ // First, find "Range:"
+ while (1) {
+ if (*buf == '\0') return False; // not found
+ if (_strncasecmp(buf, "Range: ", 7) == 0) break;
+ ++buf;
+ }
+
+ // Then, run through each of the fields, looking for ones we handle:
+ char const* fields = buf + 7;
+ while (*fields == ' ') ++fields;
+ double start, end;
+ Locale("C", LC_NUMERIC);
+ if (sscanf(fields, "npt = %lf - %lf", &start, &end) == 2) {
+ rangeStart = start;
+ rangeEnd = end;
+ } else if (sscanf(fields, "npt = %lf -", &start) == 1) {
+ rangeStart = start;
+ rangeEnd = 0.0;
+ } else {
+ return False; // The header is malformed
+ }
+
+ return True;
+}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/RTSPOverHTTPServer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simple HTTP server that acts solely to implement RTSP-over-HTTP tunneling
// (to a separate RTSP server), as described in
// http://developer.apple.com/documentation/QuickTime/QTSS/Concepts/chapter_2_section_14.html
@@ -24,7 +24,9 @@
#include "RTSPCommon.hh"
#include <GroupsockHelper.hh>
+#include <string.h>
#if defined(__WIN32__) || defined(_WIN32) || defined(_QNX4)
+#define snprintf _snprintf
#else
#include <signal.h>
#define USE_SIGNALS 1
@@ -42,7 +44,7 @@
int ourSocket = -1;
do {
- int ourSocket = setUpOurSocket(env, ourHTTPPort);
+ ourSocket = setUpOurSocket(env, ourHTTPPort);
if (ourSocket == -1) break;
return new RTSPOverHTTPServer(env, ourSocket, rtspServerPort, rtspServerHostName);
@@ -163,7 +165,7 @@
struct sockaddr_in dummy; // 'from' address, meaningless in this case
Boolean endOfMsg = False;
unsigned char* ptr = &fRequestBuffer[fRequestBytesAlreadySeen];
-
+
int bytesRead = readSocket(envir(), fClientSocket,
ptr, fRequestBufferBytesLeft, dummy);
if (bytesRead <= 0 || (unsigned)bytesRead >= fRequestBufferBytesLeft) {
@@ -195,7 +197,7 @@
}
++tmpPtr;
}
-
+
fRequestBufferBytesLeft -= bytesRead;
fRequestBytesAlreadySeen += bytesRead;
@@ -238,7 +240,7 @@
}
#endif
}
-
+
#ifdef DEBUG
fprintf(stderr, "sending response: %s", fResponseBuffer);
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/RTSPServer.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A RTSP server
// Implementation
@@ -40,7 +40,7 @@
int ourSocket = -1;
do {
- int ourSocket = setUpOurSocket(env, ourPort);
+ ourSocket = setUpOurSocket(env, ourPort);
if (ourSocket == -1) break;
return new RTSPServer(env, ourSocket, ourPort, authDatabase,
@@ -105,7 +105,7 @@
// Use our default IP address in the URL:
ourAddress.sin_addr.s_addr = ReceivingInterfaceAddr != 0
? ReceivingInterfaceAddr
- : ourSourceAddressForMulticast(envir()); // hack
+ : ourIPAddress(envir()); // hack
} else {
SOCKLEN_T namelen = sizeof ourAddress;
getsockname(clientSocket, (struct sockaddr*)&ourAddress, &namelen);
@@ -160,7 +160,7 @@
}
return ourSocket;
- } while (0);
+ } while (0);
if (ourSocket != -1) ::closeSocket(ourSocket);
return -1;
@@ -179,7 +179,7 @@
: Medium(env),
fServerSocket(ourSocket), fServerPort(ourPort),
fAuthDB(authDatabase), fReclamationTestSeconds(reclamationTestSeconds),
- fServerMediaSessions(HashTable::create(STRING_HASH_KEYS)),
+ fServerMediaSessions(HashTable::create(STRING_HASH_KEYS)),
fSessionIdCounter(0) {
#ifdef USE_SIGNALS
// Ignore the SIGPIPE signal, so that clients on the same host that are killed
@@ -312,7 +312,7 @@
struct sockaddr_in dummy; // 'from' address, meaningless in this case
Boolean endOfMsg = False;
unsigned char* ptr = &fRequestBuffer[fRequestBytesAlreadySeen];
-
+
int bytesRead = readSocket(envir(), fClientSocket,
ptr, fRequestBufferBytesLeft, dummy);
if (bytesRead <= 0 || (unsigned)bytesRead >= fRequestBufferBytesLeft) {
@@ -343,7 +343,7 @@
}
++tmpPtr;
}
-
+
fRequestBufferBytesLeft -= bytesRead;
fRequestBytesAlreadySeen += bytesRead;
@@ -385,7 +385,7 @@
handleCmd_notSupported(cseq);
}
}
-
+
#ifdef DEBUG
fprintf(stderr, "sending response: %s", fResponseBuffer);
#endif
@@ -420,7 +420,7 @@
WCHAR timeFormat[] = L"HH:mm:ss GMT\r\n";
WCHAR inBuf[200];
DWORD locale = LOCALE_NEUTRAL;
-
+
int ret = GetDateFormat(locale, 0, &SystemTime,
(LPTSTR)dateFormat, (LPTSTR)inBuf, sizeof inBuf);
inBuf[ret - 1] = ' ';
@@ -530,7 +530,7 @@
RTP_UDP,
RTP_TCP,
RAW_UDP
-};
+} StreamingMode;
static void parseTransportHeader(char const* buf,
StreamingMode& streamingMode,
@@ -548,7 +548,7 @@
destinationAddressStr = NULL;
destinationTTL = 255;
clientRTPPortNum = 0;
- clientRTCPPortNum = 1;
+ clientRTCPPortNum = 1;
rtpChannelId = rtcpChannelId = 0xFF;
portNumBits p1, p2;
@@ -594,33 +594,6 @@
delete[] field;
}
-static Boolean parseRangeHeader(char const* buf, float& rangeStart, float& rangeEnd) {
- // Initialize the result parameters to default values:
- rangeStart = rangeEnd = 0.0;
-
- // First, find "Range:"
- while (1) {
- if (*buf == '\0') return False; // not found
- if (_strncasecmp(buf, "Range: ", 7) == 0) break;
- ++buf;
- }
-
- // Then, run through each of the fields, looking for ones we handle:
- char const* fields = buf + 7;
- while (*fields == ' ') ++fields;
- float start, end;
- if (sscanf(fields, "npt = %f - %f", &start, &end) == 2) {
- rangeStart = start;
- rangeEnd = end;
- } else if (sscanf(fields, "npt = %f -", &start) == 1) {
- rangeStart = start;
- } else {
- return False; // The header is malformed
- }
-
- return True;
-}
-
static Boolean parsePlayNowHeader(char const* buf) {
// Find "x-playNow:" header, if present
while (1) {
@@ -708,7 +681,7 @@
// Look for a "Transport:" header in the request string,
// to extract client parameters:
StreamingMode streamingMode;
- char* streamingModeString; // set when RAW_UDP streaming is specified
+ char* streamingModeString = NULL; // set when RAW_UDP streaming is specified
char* clientsDestinationAddressStr;
u_int8_t clientsDestinationTTL;
portNumBits clientRTPPortNum, clientRTCPPortNum;
@@ -730,7 +703,7 @@
// Next, check whether a "Range:" header is present in the request.
// This isn't legal, but some clients do this to combine "SETUP" and "PLAY":
- float rangeStart, rangeEnd;
+ double rangeStart = 0.0, rangeEnd = 0.0;
fStreamAfterSETUP = parseRangeHeader(fullRequestStr, rangeStart, rangeEnd) ||
parsePlayNowHeader(fullRequestStr);
@@ -792,7 +765,6 @@
dateHeader(),
streamingModeString, destAddrStr, sourceAddrStr, ntohs(serverRTPPort.num()), destinationTTL,
fOurSessionId);
- delete[] streamingModeString;
break;
}
} else {
@@ -834,12 +806,11 @@
dateHeader(),
streamingModeString, destAddrStr, sourceAddrStr, ntohs(clientRTPPort.num()), ntohs(serverRTPPort.num()),
fOurSessionId);
- delete[] streamingModeString;
break;
}
}
- delete[] destAddrStr; delete[] sourceAddrStr;
}
+ delete[] destAddrStr; delete[] sourceAddrStr; delete[] streamingModeString;
}
void RTSPServer::RTSPClientSession
@@ -928,7 +899,7 @@
char* rtspURL = fOurServer.rtspURL(fOurServerMediaSession, fClientSocket);
unsigned rtspURLSize = strlen(rtspURL);
- //// Parse the client's "Scale:" header, if any:
+ //// Parse the client's "Scale:" header, if any:
float scale;
Boolean sawScaleHeader = parseScaleHeader(fullRequestStr, scale);
@@ -948,8 +919,8 @@
}
scaleHeader = strDup(buf);
- //// Parse the client's "Range:" header, if any:
- float rangeStart, rangeEnd;
+ //// Parse the client's "Range:" header, if any:
+ double rangeStart = 0.0, rangeEnd = 0.0;
Boolean sawRangeHeader = parseRangeHeader(fullRequestStr, rangeStart, rangeEnd);
// Use this information, plus the stream's duration (if known), to create
@@ -982,7 +953,7 @@
// Create a "RTP-Info:" line. It will get filled in from each subsession's state:
char const* rtpInfoFmt =
"%s" // "RTP-Info:", plus any preceding rtpInfo items
- "%s" // comma separator, if needed
+ "%s" // comma separator, if needed
"url=%s/%s"
";seq=%d"
#ifdef RTPINFO_INCLUDE_RTPTIME
@@ -1032,7 +1003,7 @@
#ifdef RTPINFO_INCLUDE_RTPTIME
+ 10 /*max unsigned (32-bit) len*/
#endif
- + 2 /*allows for trailing \r\n at final end of string*/;
+ + 2 /*allows for trailing \r\n at final end of string*/;
rtpInfo = new char[rtpInfoSize];
sprintf(rtpInfo, rtpInfoFmt,
prevRTPInfo,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/SIPClient.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A generic SIP client
// Implementation
@@ -59,7 +59,7 @@
fApplicationNameSize = strlen(fApplicationName);
struct in_addr ourAddress;
- ourAddress.s_addr = ourSourceAddressForMulticast(env); // hack
+ ourAddress.s_addr = ourIPAddress(env); // hack
fOurAddressStr = strDup(our_inet_ntoa(ourAddress));
fOurAddressStrSize = strlen(fOurAddressStr);
@@ -89,7 +89,7 @@
<< env.getResultMsg() << "\n";
}
}
-
+
// Set various headers to be used in each request:
char const* formatStr;
unsigned headerSize;
@@ -202,7 +202,7 @@
char* rtpmapLine;
unsigned rtpmapLineSize;
if (fMIMESubtypeSize > 0) {
- char* const rtpmapFmt =
+ char* const rtpmapFmt =
"a=rtpmap:%u %s/8000\r\n";
unsigned rtpmapFmtSize = strlen(rtpmapFmt)
+ 3 /* max char len */ + fMIMESubtypeSize;
@@ -215,7 +215,7 @@
rtpmapLine = strDup("");
rtpmapLineSize = 0;
}
- char* const inviteSDPFmt =
+ char* const inviteSDPFmt =
"v=0\r\n"
"o=- %u %u IN IP4 %s\r\n"
"s=%s session\r\n"
@@ -371,10 +371,10 @@
fInviteClientState = Calling;
if (!sendINVITE()) doInviteStateTerminated(0);
} else {
- // Turn off timers A & B before moving to a new state:
+ // Turn off timers A & B before moving to a new state:
sched.unscheduleDelayedTask(fTimerA);
sched.unscheduleDelayedTask(fTimerB);
-
+
if (responseCode == timerBFires) {
envir().setResultMsg("No response from server");
doInviteStateTerminated(0);
@@ -458,7 +458,7 @@
char* firstLine = NULL;
char* nextLineStart = NULL;
unsigned bytesRead = getResponse(readBuf, readBufSize);
- if (bytesRead < 0) break;
+ if (bytesRead == 0) break;
if (fVerbosityLevel >= 1) {
envir() << "Received INVITE response: " << readBuf << "\n";
}
@@ -502,7 +502,7 @@
}
delete[] realm; delete[] nonce;
if (foundAuthenticateHeader) break;
- }
+ }
}
envir().setResultMsg("cannot handle INVITE response: ", firstLine);
break;
@@ -528,7 +528,7 @@
fToTagStrSize = strlen(fToTagStr);
}
delete[] toTagStr;
-
+
if (sscanf(lineStart, "Content-Length: %d", &contentLength) == 1
|| sscanf(lineStart, "Content-length: %d", &contentLength) == 1) {
if (contentLength < 0) {
@@ -549,7 +549,7 @@
// the "Content-length:" header (if any) that we saw. We may need to
// read more data, or we may have extraneous data in the buffer.
char* bodyStart = nextLineStart;
- if (contentLength >= 0) {
+ if (bodyStart != NULL && contentLength >= 0) {
// We saw a "Content-length:" header
unsigned numBodyBytes = &readBuf[bytesRead] - bodyStart;
if (contentLength > (int)numBodyBytes) {
@@ -658,7 +658,7 @@
fURL, fToTagStr,
fCallId, fOurAddressStr,
fCSeq /* note: it's the same as before; not incremented */);
-
+
if (!sendRequest(cmd, strlen(cmd))) {
envir().setResultErrMsg("ACK send() failed: ");
break;
@@ -699,7 +699,7 @@
fURL, fToTagStr,
fCallId, fOurAddressStr,
++fCSeq);
-
+
if (!sendRequest(cmd, strlen(cmd))) {
envir().setResultErrMsg("BYE send() failed: ");
break;
@@ -721,14 +721,14 @@
NetAddress destAddress;
if (!parseSIPURL(envir(), url, destAddress, fServerPortNum)) break;
fServerAddress.s_addr = *(unsigned*)(destAddress.data());
-
+
if (fOurSocket != NULL) {
fOurSocket->changeDestinationParameters(fServerAddress,
fServerPortNum, 255);
}
}
- return True;
+ return True;
} while (0);
fInviteStatusCode = 1;
@@ -908,7 +908,7 @@
break;
}
bytesRead += bytesReadNow;
-
+
// Check whether we have "\r\n\r\n":
char* lastToCheck = responseBuffer+bytesRead-4;
if (lastToCheck < responseBuffer) continue;
@@ -932,11 +932,11 @@
}
}
}
-
+
return 0;
}
-Boolean SIPClient::parseResponseCode(char const* line,
+Boolean SIPClient::parseResponseCode(char const* line,
unsigned& responseCode) {
if (sscanf(line, "%*s%u", &responseCode) != 1) {
envir().setResultMsg("no response code in line: \"", line, "\"");
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/ServerMediaSession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A data structure that represents a session that consists of
// potentially multiple (audio and/or video) sub-sessions
// (This data structure is used for media *streamers* - i.e., servers.
@@ -183,11 +183,10 @@
}
char* ServerMediaSession::generateSDPDescription() {
- struct in_addr ourIPAddress;
- ourIPAddress.s_addr = ourSourceAddressForMulticast(envir());
- char* const ourIPAddressStr
- = strDup(our_inet_ntoa(ourIPAddress));
- unsigned ourIPAddressStrSize = strlen(ourIPAddressStr);
+ struct in_addr ipAddress;
+ ipAddress.s_addr = ourIPAddress(envir());
+ char* const ipAddressStr = strDup(our_inet_ntoa(ipAddress));
+ unsigned ipAddressStrSize = strlen(ipAddressStr);
// For a SSM sessions, we need a "a=source-filter: incl ..." line also:
char* sourceFilterLine;
@@ -195,12 +194,10 @@
char const* const sourceFilterFmt =
"a=source-filter: incl IN IP4 * %s\r\n"
"a=rtcp-unicast: reflection\r\n";
- unsigned sourceFilterFmtSize = strlen(sourceFilterFmt)
- + ourIPAddressStrSize;
+ unsigned const sourceFilterFmtSize = strlen(sourceFilterFmt) + ipAddressStrSize + 1;
sourceFilterLine = new char[sourceFilterFmtSize];
- sprintf(sourceFilterLine, sourceFilterFmt,
- ourIPAddressStr);
+ sprintf(sourceFilterLine, sourceFilterFmt, ipAddressStr);
} else {
sourceFilterLine = strDup("");
}
@@ -233,7 +230,7 @@
} else { // subsessions have differing durations, so "a=range:" lines go there
rangeLine = strDup("");
}
-
+
char const* const sdpPrefixFmt =
"v=0\r\n"
"o=- %ld%06ld %d IN IP4 %s\r\n"
@@ -249,7 +246,7 @@
"a=x-qt-text-inf:%s\r\n"
"%s";
sdpLength += strlen(sdpPrefixFmt)
- + 20 + 6 + 20 + ourIPAddressStrSize
+ + 20 + 6 + 20 + ipAddressStrSize
+ strlen(fDescriptionSDPString)
+ strlen(fInfoSDPString)
+ strlen(libNameStr) + strlen(libVersionStr)
@@ -260,12 +257,12 @@
+ strlen(fMiscSDPLines);
sdp = new char[sdpLength];
if (sdp == NULL) break;
-
+
// Generate the SDP prefix (session-level lines):
sprintf(sdp, sdpPrefixFmt,
fCreationTime.tv_sec, fCreationTime.tv_usec, // o= <session id>
1, // o= <version> // (needs to change if params are modified)
- ourIPAddressStr, // o= <address>
+ ipAddressStr, // o= <address>
fDescriptionSDPString, // s= <description>
fInfoSDPString, // i= <info>
libNameStr, libVersionStr, // a=tool:
@@ -284,7 +281,7 @@
}
} while (0);
- delete[] rangeLine; delete[] sourceFilterLine; delete[] ourIPAddressStr;
+ delete[] rangeLine; delete[] sourceFilterLine; delete[] ipAddressStr;
return sdp;
}
@@ -342,7 +339,7 @@
// default implementation: do nothing
}
void ServerMediaSubsession::seekStream(unsigned /*clientSessionId*/,
- void* /*streamToken*/, float /*seekNPT*/) {
+ void* /*streamToken*/, double /*seekNPT*/) {
// default implementation: do nothing
}
void ServerMediaSubsession::setStreamScale(unsigned /*clientSessionId*/,
@@ -386,5 +383,5 @@
char buf[100];
sprintf(buf, "a=range:npt=0-%.3f\r\n", ourDuration);
return strDup(buf);
- }
+ }
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/SimpleRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simple RTP sink that packs frames into each outgoing
// packet, without any fragmentation or special headers.
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/SimpleRTPSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A RTP source for a simple RTP payload format that
// - doesn't have any special headers following the RTP header
// - doesn't have any special framing apart from the packet data itself
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/StreamParser.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Abstract class for parsing a byte stream
// Implementation
@@ -61,7 +61,7 @@
// Swap banks, but save any still-needed bytes from the old bank:
unsigned numBytesToSave = fTotNumValidBytes - fSavedParserIndex;
unsigned char const* from = &curBank()[fSavedParserIndex];
-
+
fCurBankNum = (fCurBankNum + 1)%2;
fCurBank = fBank[fCurBankNum];
memmove(curBank(), from, numBytesToSave);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/StreamParser.hh
^
|
| @@ -11,17 +11,17 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Abstract class for parsing a byte stream
// C++ header
#ifndef _STREAM_PARSER_HH
#define _STREAM_PARSER_HH
-#ifndef _FRAMED_SOURCE_HH
+#ifndef _FRAMED_SOURCE_HH
#include "FramedSource.hh"
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/VideoRTPSink.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A generic RTP sink for video codecs (abstract base class)
// Implementation
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/WAVAudioFileServerMediaSubsession.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from an WAV audio file.
// Implementation
@@ -43,7 +43,7 @@
}
void WAVAudioFileServerMediaSubsession
-::seekStreamSource(FramedSource* inputSource, float seekNPT) {
+::seekStreamSource(FramedSource* inputSource, double seekNPT) {
WAVAudioFileSource* wavSource;
if (fBitsPerSample == 16) {
// "inputSource" is a filter; its input source is the original WAV file source:
@@ -55,7 +55,7 @@
unsigned seekSampleNumber = (unsigned)(seekNPT*fSamplingFrequency);
unsigned seekByteNumber = (seekSampleNumber*fNumChannels*fBitsPerSample)/8;
-
+
wavSource->seekToPCMByte(seekByteNumber);
}
@@ -109,7 +109,7 @@
= uLawFromPCMAudioSource::createNew(envir(), wavSource, 1/*little-endian*/);
bitsPerSecond /= 2;
} else {
- // Add a filter that converts from little-endian to network (big-endian) order:
+ // Add a filter that converts from little-endian to network (big-endian) order:
resultSource = EndianSwap16::createNew(envir(), wavSource);
}
} else { // fBitsPerSample == 8
@@ -133,7 +133,7 @@
do {
char* mimeType;
unsigned char payloadFormatCode;
- if (fAudioFormat == WA_PCM) {
+ if (fAudioFormat == WA_PCM) {
if (fBitsPerSample == 16) {
if (fConvertToULaw) {
mimeType = "PCMU";
@@ -170,10 +170,10 @@
} else {
payloadFormatCode = rtpPayloadTypeIfDynamic;
}
- } else { //unknown format
- break;
+ } else { //unknown format
+ break;
}
-
+
return SimpleRTPSink::createNew(envir(), rtpGroupsock,
payloadFormatCode, fSamplingFrequency,
"audio", mimeType, fNumChannels);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/WAVAudioFileSource.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A WAV audio file source
// Implementation
@@ -75,26 +75,25 @@
#define nextc fgetc(fid)
-#define ucEOF ((unsigned char)EOF)
static Boolean get4Bytes(FILE* fid, unsigned& result) { // little-endian
- unsigned char c0, c1, c2, c3;
- if ((c0 = nextc) == ucEOF || (c1 = nextc) == ucEOF ||
- (c2 = nextc) == ucEOF || (c3 = nextc) == ucEOF) return False;
+ int c0, c1, c2, c3;
+ if ((c0 = nextc) == EOF || (c1 = nextc) == EOF ||
+ (c2 = nextc) == EOF || (c3 = nextc) == EOF) return False;
result = (c3<<24)|(c2<<16)|(c1<<8)|c0;
return True;
}
static Boolean get2Bytes(FILE* fid, unsigned short& result) {//little-endian
- unsigned char c0, c1;
- if ((c0 = nextc) == ucEOF || (c1 = nextc) == ucEOF) return False;
+ int c0, c1;
+ if ((c0 = nextc) == EOF || (c1 = nextc) == EOF) return False;
result = (c1<<8)|c0;
return True;
}
static Boolean skipBytes(FILE* fid, int num) {
while (num-- > 0) {
- if (nextc == ucEOF) return False;
+ if (nextc == EOF) return False;
}
return True;
}
@@ -127,7 +126,7 @@
if (!get2Bytes(fid, audioFormat)) break;
fAudioFormat = (unsigned char)audioFormat;
- if (fAudioFormat != WA_PCM && fAudioFormat != WA_PCMA && fAudioFormat != WA_PCMU) {
+ if (fAudioFormat != WA_PCM && fAudioFormat != WA_PCMA && fAudioFormat != WA_PCMU) {
// not PCM/PCMU/PCMA - we can't handle this
env.setResultMsg("Audio format is not PCM/PCMU/PCMA");
break;
@@ -157,7 +156,7 @@
if (!skipBytes(fid, formatLength - 16)) break;
// FACT chunk (optional):
- unsigned char c = nextc;
+ int c = nextc;
if (c == 'f') {
if (nextc != 'a' || nextc != 'c' || nextc != 't') break;
unsigned factLength;
@@ -174,7 +173,7 @@
fWAVHeaderSize = ftell(fid);
success = True;
} while (0);
-
+
if (!success) {
env.setResultMsg("Bad WAV file format");
// Set "fBitsPerSample" to zero, to indicate failure:
@@ -219,7 +218,7 @@
fFrameSize = fread(fTo, 1, bytesToRead, fFid);
} else {
// We read every 'fScaleFactor'th sample:
- fFrameSize = 0;
+ fFrameSize = 0;
while (bytesToRead > 0) {
size_t bytesRead = fread(fTo, 1, bytesPerSample, fFid);
if (bytesRead <= 0) break;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AC3AudioRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for AC3 audio
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AC3AudioRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// AC3 Audio RTP Sources
// C++ header
@@ -45,7 +45,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AC3AudioStreamFramer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an AC3 audio elementary stream into frames
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/ADTSAudioFileServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from an AAC audio file in ADTS format
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/ADTSAudioFileSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A source object for AAC audio files in ADTS format
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AMRAudioFileServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from an AMR audio file.
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AMRAudioFileSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// AMR Audio File Sinks
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AMRAudioFileSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A source object for AMR audio files (as defined in RFC 3267, section 5)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AMRAudioRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for AMR audio (RFC 3267)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AMRAudioRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// AMR Audio RTP Sources (RFC 3267)
// C++ header
@@ -39,7 +39,7 @@
Boolean isOctetAligned = True,
unsigned interleaving = 0,
// relevant only if "isOctetAligned"
- // The maximum # of frame-blocks in a group
+ // The maximum # of frame-blocks in a group
// 0 means: no interleaving
Boolean robustSortingOrder = False,
// relevant only if "isOctetAligned"
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AMRAudioSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A source object for AMR audio sources
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AVIFileSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A sink that generates an AVI file from a composite media session
// C++ header
@@ -44,7 +44,7 @@
private:
AVIFileSink(UsageEnvironment& env, MediaSession& inputSession,
- FILE* outFid, unsigned bufferSize,
+ char const* outputFileName, unsigned bufferSize,
unsigned short movieWidth, unsigned short movieHeight,
unsigned movieFPS, Boolean packetLossCompensate);
// called only by createNew()
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AudioInputDevice.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Generic audio input device (such as a microphone, or an input sound card)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/AudioRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A generic RTP sink for audio codecs (abstract base class)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/Base64.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Base64 encoding and decoding
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/BasicUDPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simple UDP sink (i.e., without RTP or other headers added); one frame per packet
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/BasicUDPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simple UDP source, where every UDP payload is a complete frame
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/ByteStreamFileSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A file source that is a plain byte stream (rather than frames)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/ByteStreamMultiFileSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A source that consists of multiple byte-stream files, read sequentially
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/DarwinInjector.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// An object that redirects one or more RTP/RTCP streams - forming a single
// multimedia session - into a 'Darwin Streaming Server' (for subsequent
// reflection to potentially arbitrarily many remote RTSP clients).
@@ -48,8 +48,8 @@
</Limit>
require any-user
Use the "remoteUserName" and "remotePassword" parameters to
- "setDestination()", as appropriate.
- 4/ Call "startPlaying" on each RTP sink (from the corresponding 'source').
+ "setDestination()", as appropriate.
+ 4/ Call "startPlaying" on each RTP sink (from the corresponding 'source').
*/
class SubstreamDescriptor; // forward
@@ -65,7 +65,7 @@
void addStream(RTPSink* rtpSink, RTCPInstance* rtcpInstance);
- Boolean setDestination(char const* remoteRTSPServerNameOrAddress,
+ Boolean setDestination(char const* remoteRTSPServerNameOrAddress,
char const* remoteFileName,
char const* sessionName = "",
char const* sessionInfo = "",
@@ -92,6 +92,8 @@
unsigned fSubstreamSDPSizes;
SubstreamDescriptor* fHeadSubstream;
SubstreamDescriptor* fTailSubstream;
+ MediaSession* fSession;
+ unsigned fLastTrackId;
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/DeviceSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A template for a MediaSource encapsulating an audio/video input device
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/DigestAuthentication.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A class used for digest authentication.
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/FileServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a file.
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/FileSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// File Sinks
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/FramedFileSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Framed File Sources
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/FramedFilter.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Framed Filters
// C++ header
@@ -29,6 +29,9 @@
public:
FramedSource* inputSource() const { return fInputSource; }
+ // Call before destruction if you want to prevent the destructor from closing the input source
+ void detachInputSource() { fInputSource = NULL; }
+
protected:
FramedFilter(UsageEnvironment& env, FramedSource* inputSource);
// abstract base class
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/FramedSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Framed Sources
// C++ header
@@ -43,20 +43,7 @@
void* afterGettingClientData,
onCloseFunc* onCloseFunc,
void* onCloseClientData);
- // ##### The following is for backwards-compatibility; remove it eventually:
-#ifdef BACKWARDS_COMPATIBLE_WITH_OLD_AFTER_GETTING_FUNC
- typedef void (bwCompatAfterGettingFunc)(void* clientData, unsigned frameSize,
- struct timeval presentationTime);
- void getNextFrame(unsigned char* to, unsigned maxSize,
- bwCompatAfterGettingFunc* afterGettingFunc,
- void* afterGettingClientData,
- onCloseFunc* onCloseFunc,
- void* onCloseClientData);
- bwCompatAfterGettingFunc* fSavedBWCompatAfterGettingFunc;
- void* fSavedBWCompatAfterGettingClientData;
-#endif
- // ##### End of code for backwards-compatibility.
-
+
static void handleClosure(void* clientData);
// This should be called (on ourself) if the source is discovered
// to be closed (i.e., no longer readable)
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/GSMAudioRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for GSM audio
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/H261VideoRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// H.261 Video RTP Sources
// C++ header
@@ -47,7 +47,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
private:
u_int32_t fLastSpecialHeader;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/H263plusVideoFileServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MPEG-4 video file.
// C++ header
@@ -31,27 +31,18 @@
static H263plusVideoFileServerMediaSubsession*
createNew(UsageEnvironment& env, char const* fileName, Boolean reuseFirstSource);
- void setDoneFlag() { fDoneFlag = ~0; }
- void checkForAuxSDPLine1();
-
private:
H263plusVideoFileServerMediaSubsession(UsageEnvironment& env,
- char const* fileName, Boolean reuseFirstSource);
+ char const* fileName, Boolean reuseFirstSource);
// called only by createNew();
virtual ~H263plusVideoFileServerMediaSubsession();
private: // redefined virtual functions
- virtual char const* getAuxSDPLine(RTPSink* rtpSink,
- FramedSource* inputSource);
virtual FramedSource* createNewStreamSource(unsigned clientSessionId,
unsigned& estBitrate);
virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock,
unsigned char rtpPayloadTypeIfDynamic,
FramedSource* inputSource);
-
-private:
- char fDoneFlag; // used when setting up "fSDPLines"
- RTPSink* fDummyRTPSink; // ditto
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/H263plusVideoRTPSink.hh
^
|
| @@ -11,11 +11,11 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
-// RTP sink for H.263+ video (RFC 2429)
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
+// RTP sink for H.263+ video (RFC 4629)
// C++ header
#ifndef _H263_PLUS_VIDEO_RTP_SINK_HH
@@ -30,7 +30,7 @@
static H263plusVideoRTPSink* createNew(UsageEnvironment& env, Groupsock* RTPgs,
unsigned char rtpPayloadFormat,
u_int32_t rtpTimestampFrequency = 90000);
-
+
protected:
H263plusVideoRTPSink(UsageEnvironment& env, Groupsock* RTPgs,
unsigned char rtpPayloadFormat,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/H263plusVideoRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// H.263+ Video RTP Sources
// C++ header
@@ -54,7 +54,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/H263plusVideoStreamFramer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an H263 video elementary stream into frames.
// Author Benhard Feiten
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/H264VideoFileSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// H.264 Video File Sinks
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/H264VideoRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for H.264 video (RFC 3984)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/H264VideoRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// H.264 Video RTP Sources
// C++ header
@@ -46,7 +46,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
private:
friend class H264BufferedPacket;
@@ -67,6 +67,6 @@
// Returns the binary value of each 'parameter set' specified in a
// "sprop-parameter-sets" string (in the SDP description for a H.264/RTP stream).
// The value is returned as an array (length "numSPropRecords") of "SPropRecord"s.
- // This array is dynamically allocated by this routine, and must be delete[]d by the caller.
+ // This array is dynamically allocated by this routine, and must be delete[]d by the caller.
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/H264VideoStreamFramer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Any source that feeds into a "H264VideoRTPSink" must be of this class.
// This is a virtual base class; subclasses must implement the
// "currentNALUnitEndsAccessUnit()" virtual function.
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/HTTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// HTTP Sinks
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/InputFile.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Common routines for opening/closing named input files
// C++ header
@@ -31,10 +31,10 @@
u_int64_t GetFileSize(char const* fileName, FILE* fid);
// 0 means zero-length, unbounded, or unknown
-u_int64_t SeekFile64(FILE *fid, int64_t offset, int whence);
+int64_t SeekFile64(FILE *fid, int64_t offset, int whence);
// A platform-independent routine for seeking within (possibly) large files
-u_int64_t TellFile64(FILE *fid);
+int64_t TellFile64(FILE *fid);
// A platform-independent routine for reporting the position within
// (possibly) large files
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/JPEGVideoRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for JPEG video (RFC 2435)
// C++ header
@@ -28,7 +28,7 @@
class JPEGVideoRTPSink: public VideoRTPSink {
public:
static JPEGVideoRTPSink* createNew(UsageEnvironment& env, Groupsock* RTPgs);
-
+
protected:
JPEGVideoRTPSink(UsageEnvironment& env, Groupsock* RTPgs);
// called only by createNew()
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/JPEGVideoRTPSource.hh
^
|
| @@ -11,11 +11,11 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
-// JPEG Video (RFC 2435) RTP Sources
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
+// JPEG Video (RFC 2435) RTP Sources
// C++ header
#ifndef _JPEG_VIDEO_RTP_SOURCE_HH
@@ -32,7 +32,8 @@
static JPEGVideoRTPSource*
createNew(UsageEnvironment& env, Groupsock* RTPgs,
unsigned char rtpPayloadFormat = 26,
- unsigned rtpPayloadFrequency = 90000);
+ unsigned rtpPayloadFrequency = 90000,
+ unsigned defaultWidth = 0, unsigned defaultHeight = 0);
protected:
virtual ~JPEGVideoRTPSource();
@@ -40,15 +41,19 @@
private:
JPEGVideoRTPSource(UsageEnvironment& env, Groupsock* RTPgs,
unsigned char rtpPayloadFormat,
- unsigned rtpTimestampFrequency);
+ unsigned rtpTimestampFrequency,
+ unsigned defaultWidth, unsigned defaultHeight);
// called only by createNew()
+ // Image dimensions from the SDP description, if any
+ unsigned fDefaultWidth, fDefaultHeight;
+
private:
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
-
- virtual char const* MIMEtype() const;
+
+ virtual char const* MIMEtype() const;
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/JPEGVideoSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// JPEG video sources
// C++ header
@@ -36,7 +36,7 @@
u_int16_t& length);
// If "qFactor()" returns a value >= 128, then this function is called
// to tell us the quantization tables that are being used.
- // (The default implementation of this function just returns NULL.)
+ // (The default implementation of this function just returns NULL.)
// "precision" and "length" are as defined in RFC 2435, section 3.1.8.
protected:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/Locale.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Support for temporarily setting the locale (e.g., to POSIX) for (e.g.) parsing or printing
// floating-point numbers in protocol headers, or calling toupper()/tolower() on human-input strings.
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MP3ADU.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// 'ADU' MP3 streams (for improved loss-tolerance)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MP3ADURTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for 'ADUized' MP3 frames ("mpa-robust")
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MP3ADURTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP source for 'ADUized' MP3 frames ("mpa-robust")
// C++ header
@@ -43,7 +43,7 @@
private:
// redefined virtual functions:
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MP3ADUTranscoder.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Transcoder for ADUized MP3 frames
// C++ header
@@ -31,7 +31,7 @@
unsigned outBitrate /* in kbps */,
FramedSource* inputSource);
- unsigned outBitrate() const { return fOutBitrate; }
+ unsigned outBitrate() const { return fOutBitrate; }
protected:
MP3ADUTranscoder(UsageEnvironment& env,
unsigned outBitrate /* in kbps */,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MP3ADUinterleaving.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Interleaving of MP3 ADUs
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MP3AudioFileServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from an MP3 audio file.
// (Actually, any MPEG-1 or MPEG-2 audio file should work.)
@@ -47,7 +47,7 @@
virtual ~MP3AudioFileServerMediaSubsession();
private: // redefined virtual functions
- virtual void seekStreamSource(FramedSource* inputSource, float seekNPT);
+ virtual void seekStreamSource(FramedSource* inputSource, double seekNPT);
virtual void setStreamSourceScale(FramedSource* inputSource, float scale);
virtual FramedSource* createNewStreamSource(unsigned clientSessionId,
unsigned& estBitrate);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MP3FileSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 File Sources
// C++ header
@@ -33,7 +33,7 @@
float filePlayTime() const;
void setPresentationTimeScale(unsigned scale);
- void seekWithinFile(float seekNPT);
+ void seekWithinFile(double seekNPT);
protected:
MP3FileSource(UsageEnvironment& env, FILE* fid);
@@ -50,7 +50,7 @@
private:
// redefined virtual functions:
virtual void doGetNextFrame();
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
virtual void getAttributes() const;
private:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MP3HTTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 HTTP Sources
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MP3Transcoder.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP3 Transcoder
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2AudioRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for MPEG audio (RFC 2250)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2AudioRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG-1 or MPEG-2 Audio RTP Sources
// C++ header
@@ -45,7 +45,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2AudioStreamFramer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an MPEG (1,2) audio elementary stream into frames
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2Demux.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Demultiplexer for a MPEG 1 or 2 Program Stream
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2DemuxedElementaryStream.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A MPEG 1 or 2 Elementary Stream, demultiplexed from a Program Stream
// C++ header
@@ -43,7 +43,7 @@
// redefined virtual functions:
virtual void doGetNextFrame();
virtual void doStopGettingFrames();
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
virtual unsigned maxFrameSize() const;
private:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2DemuxedServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MPEG-1 or 2 demuxer.
// C++ header
@@ -45,7 +45,7 @@
virtual ~MPEG1or2DemuxedServerMediaSubsession();
private: // redefined virtual functions
- virtual void seekStreamSource(FramedSource* inputSource, float seekNPT);
+ virtual void seekStreamSource(FramedSource* inputSource, double seekNPT);
virtual FramedSource* createNewStreamSource(unsigned clientSessionId,
unsigned& estBitrate);
virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2FileServerDemux.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A server demultiplexer for a MPEG 1 or 2 Program Stream
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2VideoFileServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MPEG-1 or 2 Elementary Stream video file.
// C++ header
@@ -40,7 +40,7 @@
char const* fileName,
Boolean reuseFirstSource,
Boolean iFramesOnly,
- double vshPeriod);
+ double vshPeriod);
// called only by createNew();
virtual ~MPEG1or2VideoFileServerMediaSubsession();
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2VideoHTTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A HTTP Sink specifically for MPEG Video
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2VideoRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for MPEG video (RFC 2250)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2VideoRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG-1 or MPEG-2 Video RTP Sources
// C++ header
@@ -47,7 +47,7 @@
unsigned& resultSpecialHeaderSize);
virtual Boolean packetIsUsableInJitterCalculation(unsigned char* packet,
unsigned packetSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2VideoStreamDiscreteFramer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simplified version of "MPEG1or2VideoStreamFramer" that takes only
// complete, discrete frames (rather than an arbitrary byte stream) as input.
// This avoids the parsing and data copying overhead of the full
@@ -36,18 +36,18 @@
createNew(UsageEnvironment& env, FramedSource* inputSource,
Boolean iFramesOnly = False,
double vshPeriod = 5.0); // see MPEG1or2VideoStreamFramer.hh
-
+
private:
MPEG1or2VideoStreamDiscreteFramer(UsageEnvironment& env,
FramedSource* inputSource,
Boolean iFramesOnly, double vshPeriod);
// called only by createNew()
virtual ~MPEG1or2VideoStreamDiscreteFramer();
-
+
private:
// redefined virtual functions:
virtual void doGetNextFrame();
-
+
private:
static void afterGettingFrame(void* clientData, unsigned frameSize,
unsigned numTruncatedBytes,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG1or2VideoStreamFramer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an MPEG 1 or 2 video elementary stream into
// frames for: Video_Sequence_Header, GOP_Header, Picture_Header
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG2IndexFromTransportStream.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that produces a sequence of I-frame indices from a MPEG-2 Transport Stream
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG2TransportFileServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MPEG-2 Transport Stream file.
// C++ header
@@ -57,7 +57,7 @@
unsigned short& rtpSeqNum,
unsigned& rtpTimestamp);
virtual void pauseStream(unsigned clientSessionId, void* streamToken);
- virtual void seekStream(unsigned clientSessionId, void* streamToken, float seekNPT);
+ virtual void seekStream(unsigned clientSessionId, void* streamToken, double seekNPT);
virtual void setStreamScale(unsigned clientSessionId, void* streamToken, float scale);
virtual void deleteStream(unsigned clientSessionId, void*& streamToken);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG2TransportStreamFramer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that passes through (unchanged) chunks that contain an integral number
// of MPEG-2 Transport Stream packets, but returning (in "fDurationInMicroseconds")
// an updated estimate of the time gap between chunks.
@@ -66,6 +66,7 @@
unsigned long fTSPacketCount;
double fTSPacketDurationEstimate;
HashTable* fPIDStatusTable;
+ unsigned long fTSPCRCount;
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG2TransportStreamFromESSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter for converting one or more MPEG Elementary Streams
// to a MPEG-2 Transport Stream
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG2TransportStreamFromPESSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter for converting a stream of MPEG PES packets to a MPEG-2 Transport Stream
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG2TransportStreamIndexFile.hh
^
|
| @@ -11,14 +11,14 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A class that encapsulates MPEG-2 Transport Stream 'index files'/
// These index files are used to implement 'trick play' operations
// (seek-by-time, fast forward, reverse play) on Transport Stream files.
-//
+//
// C++ header
#ifndef _MPEG2_TRANSPORT_STREAM_INDEX_FILE_HH
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG2TransportStreamMultiplexor.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A class for generating MPEG-2 Transport Stream from one or more input
// Elementary Stream data sources
// C++ header
@@ -69,7 +69,7 @@
u_int8_t streamType; // for use in Program Maps
} fPIDState[PID_TABLE_SIZE];
u_int8_t fPCR_PID, fCurrentPID;
- // Note: We map 8-bit stream_ids directly to PIDs
+ // Note: We map 8-bit stream_ids directly to PIDs
MPEG1or2Demux::SCR fPCR;
unsigned char* fInputBuffer;
unsigned fInputBufferSize, fInputBufferBytesUsed;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG2TransportStreamTrickModeFilter.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.// A filter that converts a MPEG Transport Stream file - with corresponding index file
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.// A filter that converts a MPEG Transport Stream file - with corresponding index file
// - to a corresponding Video Elementary Stream. It also uses a "scale" parameter
// to implement 'trick mode' (fast forward or reverse play, using I-frames) on
// the video stream.
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG4ESVideoRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for MPEG-4 Elementary Stream video (RFC 3016)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG4ESVideoRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MP4V-ES video RTP stream sources
// C++ header
@@ -45,7 +45,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG4GenericRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG4-GENERIC ("audio", "video", or "application") RTP stream sinks
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG4GenericRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG4-GENERIC ("audio", "video", or "application") RTP stream sources
// C++ header
@@ -59,7 +59,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
private:
char* fMIMEType;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG4LATMAudioRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for MPEG-4 audio, using LATM multiplexing (RFC 3016)
// (Note that the initial 'size' field is assumed to be present at the start of
// each frame.)
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG4LATMAudioRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// MPEG-4 audio, using LATM multiplexing
// C++ header
@@ -51,7 +51,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
private:
Boolean fIncludeLATMDataLengthField;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG4VideoFileServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from a MPEG-4 video file.
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG4VideoStreamDiscreteFramer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simplified version of "MPEG4VideoStreamFramer" that takes only complete,
// discrete frames (rather than an arbitrary byte stream) as input.
// This avoids the parsing and data copying overhead of the full
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEG4VideoStreamFramer.hh
^
|
| @@ -11,16 +11,16 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an MPEG-4 video elementary stream into
// frames for:
// - Visual Object Sequence (VS) Header + Visual Object (VO) Header
// + Video Object Layer (VOL) Header
// - Group of VOP (GOV) Header
-// - VOP frame
+// - VOP frame
// C++ header
#ifndef _MPEG4_VIDEO_STREAM_FRAMER_HH
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MPEGVideoStreamFramer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A filter that breaks up an MPEG video elementary stream into
// headers and frames
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/Media.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Medium
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MediaSession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A data structure that represents a session that consists of
// potentially multiple (audio and/or video) sub-sessions
// (This data structure is used for media *receivers* - i.e., clients.
@@ -39,7 +39,8 @@
MediaSession*& resultSession);
Boolean hasSubsessions() const { return fSubsessionsHead != NULL; }
- float& playEndTime() { return fMaxPlayEndTime; }
+ double& playStartTime() { return fMaxPlayStartTime; }
+ double& playEndTime() { return fMaxPlayEndTime; }
char* connectionEndpointName() const { return fConnectionEndpointName; }
char const* CNAME() const { return fCNAME; }
struct in_addr const& sourceFilterAddr() const { return fSourceFilterAddr; }
@@ -47,6 +48,7 @@
char* mediaSessionType() const { return fMediaSessionType; }
char* sessionName() const { return fSessionName; }
char* sessionDescription() const { return fSessionDescription; }
+ char const* controlPath() const { return fControlPath; }
Boolean initiateByMediaType(char const* mimeType,
MediaSubsession*& resultSubsession,
@@ -78,6 +80,7 @@
Boolean parseSDPLine_i(char const* sdpLine);
Boolean parseSDPLine_c(char const* sdpLine);
Boolean parseSDPAttribute_type(char const* sdpLine);
+ Boolean parseSDPAttribute_control(char const* sdpLine);
Boolean parseSDPAttribute_range(char const* sdpLine);
Boolean parseSDPAttribute_source_filter(char const* sdpLine);
@@ -97,12 +100,14 @@
// Fields set from a SDP description:
char* fConnectionEndpointName;
- float fMaxPlayEndTime;
+ double fMaxPlayStartTime;
+ double fMaxPlayEndTime;
struct in_addr fSourceFilterAddr; // used for SSM
float fScale; // set from a RTSP "Scale:" header
char* fMediaSessionType; // holds a=type value
char* fSessionName; // holds s=<session name> value
char* fSessionDescription; // holds i=<session description> value
+ char* fControlPath; // holds optional a=control: string
};
@@ -110,10 +115,10 @@
public:
MediaSubsessionIterator(MediaSession& session);
virtual ~MediaSubsessionIterator();
-
+
MediaSubsession* next(); // NULL if none
void reset();
-
+
private:
MediaSession& fOurSession;
MediaSubsession* fNextPtr;
@@ -147,7 +152,11 @@
// This is the source that client sinks read from. It is usually
// (but not necessarily) the same as "rtpSource()"
- float playEndTime() const;
+ double playStartTime() const;
+ double playEndTime() const;
+ // Used only to set the local fields:
+ double& _playStartTime() { return fPlayStartTime; }
+ double& _playEndTime() { return fPlayEndTime; }
Boolean initiate(int useSpecialRTPoffset = -1);
// Creates a "RTPSource" for this subsession. (Has no effect if it's
@@ -190,9 +199,9 @@
char const* fmtp_mode() const { return fMode; }
char const* fmtp_spropparametersets() const { return fSpropParameterSets; }
- unsigned connectionEndpointAddress() const;
+ netAddressBits connectionEndpointAddress() const;
// Converts "fConnectionEndpointName" to an address (or 0 if unknown)
- void setDestinations(unsigned defaultDestAddress);
+ void setDestinations(netAddressBits defaultDestAddress);
// Uses "fConnectionEndpointName" and "serverPortNum" to set
// the destination address and port of the RTP and RTCP objects.
// This is typically called by RTSP clients after doing "SETUP".
@@ -207,11 +216,19 @@
// Parameters set from a RTSP "RTP-Info:" header:
struct {
- unsigned trackId;
u_int16_t seqNum;
u_int32_t timestamp;
+ Boolean infoIsNew; // not part of the RTSP header; instead, set whenever this struct is filled in
} rtpInfo;
+ double getNormalPlayTime(struct timeval const& presentationTime);
+ // Computes the stream's "Normal Play Time" (NPT) from the given "presentationTime".
+ // (For the definition of "Normal Play Time", see RFC 2326, section 3.6.)
+ // This function is useful only if the "rtpInfo" structure was previously filled in
+ // (e.g., by a "RTP-Info:" header in a RTSP response).
+ // Also, for this function to work properly, the RTP stream's presentation times must (eventually) be
+ // synchronized via RTCP.
+
#ifdef SUPPORT_REAL_RTSP
// Attributes specific to RealNetworks streams:
unsigned fRealMaxBitRate, fRealAvgBitRate, fRealMaxPacketSize, fRealAvgPacketSize, fRealPreroll;
@@ -255,7 +272,7 @@
char* fCodecName;
char* fProtocolName;
unsigned fRTPTimestampFrequency;
- char* fControlPath;
+ char* fControlPath; // holds optional a=control: string
struct in_addr fSourceFilterAddr; // used for SSM
// Parameters set by "a=fmtp:" SDP lines:
@@ -268,7 +285,8 @@
Boolean fCpresent, fRandomaccessindication;
char *fConfig, *fMode, *fSpropParameterSets;
- float fPlayEndTime;
+ double fPlayStartTime;
+ double fPlayEndTime;
unsigned short fVideoWidth, fVideoHeight;
// screen dimensions (set by an optional a=x-dimensions: <w>,<h> line)
unsigned fVideoFPS;
@@ -276,6 +294,7 @@
unsigned fNumChannels;
// optionally set by "a=rtpmap:" lines for audio sessions. Default: 1
float fScale; // set from a RTSP "Scale:" header
+ double fNPT_PTS_Offset; // set by "getNormalPlayTime()"; add this to a PTS to get NPT
// Fields set by initiate():
Groupsock* fRTPSocket; Groupsock* fRTCPSocket; // works even for unicast
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MediaSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Media Sinks
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MediaSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Media Sources
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MultiFramedRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP sink for a common kind of payload format: Those which pack multiple,
// complete codec frames (as many as possible) into each RTP packet.
// C++ header
@@ -80,9 +80,11 @@
unsigned ourMaxPacketSize() const { return fOurMaxPacketSize; }
+public: // redefined virtual functions:
+ virtual void stopPlaying();
+
protected: // redefined virtual functions:
virtual Boolean continuePlaying();
- virtual void stopPlaying();
private:
void buildAndSendPacket(Boolean isFirstPacket);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/MultiFramedRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP source for a common kind of payload format: Those which pack multiple,
// complete codec frames (as many as possible) into each RTP packet.
// C++ header
@@ -93,7 +93,7 @@
Boolean fillInData(RTPInterface& rtpInterface);
void assignMiscParams(unsigned short rtpSeqNo, unsigned rtpTimestamp,
struct timeval presentationTime,
- Boolean hasBeenSyncedUsingRTCP,
+ Boolean hasBeenSyncedUsingRTCP,
Boolean rtpMarkerBit, struct timeval timeReceived);
void skip(unsigned numBytes); // used to skip over an initial header
void removePadding(unsigned numBytes); // used to remove trailing bytes
@@ -112,6 +112,7 @@
unsigned char* data() const { return &fBuf[fHead]; }
unsigned dataSize() const { return fTail-fHead; }
Boolean rtpMarkerBit() const { return fRTPMarkerBit; }
+ Boolean& isFirstPacket() { return fIsFirstPacket; }
protected:
virtual void reset();
@@ -137,6 +138,7 @@
struct timeval fPresentationTime; // corresponding to "fRTPTimestamp"
Boolean fHasBeenSyncedUsingRTCP;
Boolean fRTPMarkerBit;
+ Boolean fIsFirstPacket;
struct timeval fTimeReceived;
};
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/OnDemandServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand.
// C++ header
@@ -56,14 +56,14 @@
unsigned short& rtpSeqNum,
unsigned& rtpTimestamp);
virtual void pauseStream(unsigned clientSessionId, void* streamToken);
- virtual void seekStream(unsigned clientSessionId, void* streamToken, float seekNPT);
+ virtual void seekStream(unsigned clientSessionId, void* streamToken, double seekNPT);
virtual void setStreamScale(unsigned clientSessionId, void* streamToken, float scale);
virtual void deleteStream(unsigned clientSessionId, void*& streamToken);
protected: // new virtual functions, possibly redefined by subclasses
virtual char const* getAuxSDPLine(RTPSink* rtpSink,
FramedSource* inputSource);
- virtual void seekStreamSource(FramedSource* inputSource, float seekNPT);
+ virtual void seekStreamSource(FramedSource* inputSource, double seekNPT);
virtual void setStreamSourceScale(FramedSource* inputSource, float scale);
virtual void closeStreamSource(FramedSource *inputSource);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/OutputFile.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Common routines for opening/closing named output files
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/PassiveServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that represents an existing
// 'RTPSink', rather than one that creates new 'RTPSink's on demand.
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/QCELPAudioRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Qualcomm "PureVoice" (aka. "QCELP") Audio RTP Sources
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/QuickTimeFileSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A sink that generates a QuickTime file from a composite media session
// C++ header
@@ -47,7 +47,7 @@
private:
QuickTimeFileSink(UsageEnvironment& env, MediaSession& inputSession,
- FILE* outFid, unsigned bufferSize,
+ char const* outputFileName, unsigned bufferSize,
unsigned short movieWidth, unsigned short movieHeight,
unsigned movieFPS, Boolean packetLossCompensate,
Boolean syncStreams, Boolean generateHintTracks,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/QuickTimeGenericRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP Sources containing generic QuickTime stream data, as defined in
// <http://developer.apple.com/quicktime/icefloe/dispatch026.html>
// C++ header
@@ -59,7 +59,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
private:
char const* fMIMEtypeString;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/RTCP.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTCP
// C++ header
@@ -66,7 +66,7 @@
// This prevents (for example) the handler for a multicast receiver being
// called if some other multicast receiver happens to exit.
// If "handleActiveParticipantsOnly" is False, then the handler is called
- // for any incoming RTCP "BYE".
+ // for any incoming RTCP "BYE".
void setSRHandler(TaskFunc* handlerTask, void* clientData);
void setRRHandler(TaskFunc* handlerTask, void* clientData);
// Assigns a handler routine to be called if a "SR" or "RR"
@@ -84,7 +84,7 @@
void setStreamSocket(int sockNum, unsigned char streamChannelId);
void addStreamSocket(int sockNum, unsigned char streamChannelId);
void removeStreamSocket(int sockNum, unsigned char streamChannelId) {
- fRTCPInterface.removeStreamSocket(sockNum, streamChannelId);
+ fRTCPInterface.removeStreamSocket(sockNum, streamChannelId);
}
// hacks to allow sending RTP over TCP (RFC 2236, section 10.12)
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/RTPInterface.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// An abstraction of a network interface used for RTP (or RTCP).
// (This allows the RTP-over-TCP hack (RFC 2326, section 10.12) to
// be implemented transparently.)
@@ -89,7 +89,7 @@
int fNextTCPReadStreamSocketNum;
unsigned char fNextTCPReadStreamChannelId;
TaskScheduler::BackgroundHandlerProc* fReadHandlerProc; // if any
-
+
AuxHandlerFunc* fAuxReadHandlerFunc;
void* fAuxReadHandlerClientData;
};
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/RTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP Sinks
// C++ header
@@ -75,10 +75,10 @@
fRTPInterface.setStreamSocket(sockNum, streamChannelId);
}
void addStreamSocket(int sockNum, unsigned char streamChannelId) {
- fRTPInterface.addStreamSocket(sockNum, streamChannelId);
+ fRTPInterface.addStreamSocket(sockNum, streamChannelId);
}
void removeStreamSocket(int sockNum, unsigned char streamChannelId) {
- fRTPInterface.removeStreamSocket(sockNum, streamChannelId);
+ fRTPInterface.removeStreamSocket(sockNum, streamChannelId);
}
// hacks to allow sending RTP over TCP (RFC 2236, section 10.12)
@@ -137,7 +137,7 @@
HashTable::Iterator* fIter;
};
- // The following is called whenever a RTCP RR packet is received:
+ // The following is called whenever a RTCP RR packet is received:
void noteIncomingRR(u_int32_t SSRC, struct sockaddr_in const& lastFromAddress,
unsigned lossStats, unsigned lastPacketNumReceived,
unsigned jitter, unsigned lastSRTime, unsigned diffSR_RRTime);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/RTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// RTP Sources
// C++ header
@@ -35,8 +35,6 @@
static Boolean lookupByName(UsageEnvironment& env, char const* sourceName,
RTPSource*& resultSource);
- u_int16_t curPacketRTPSeqNum() const { return fCurPacketRTPSeqNum; }
- u_int32_t curPacketRTPTimestamp() const { return fCurPacketRTPTimestamp; }
Boolean curPacketMarkerBit() const { return fCurPacketMarkerBit; }
unsigned char rtpPayloadFormat() const { return fRTPPayloadFormat; }
@@ -72,6 +70,12 @@
handlerClientData);
}
+ // Note that RTP receivers will usually not need to call either of the following two functions, because
+ // RTP sequence numbers and timestamps are usually not useful to receivers.
+ // (Our implementation of RTP reception already does all needed handling of RTP sequence numbers and timestamps.)
+ u_int16_t curPacketRTPSeqNum() const { return fCurPacketRTPSeqNum; }
+ u_int32_t curPacketRTPTimestamp() const { return fCurPacketRTPTimestamp; }
+
protected:
RTPSource(UsageEnvironment& env, Groupsock* RTPgs,
unsigned char rtpPayloadFormat, u_int32_t rtpTimestampFrequency);
@@ -125,7 +129,7 @@
HashTable::Iterator* fIter;
};
- // The following is called whenever a RTP packet is received:
+ // The following is called whenever a RTP packet is received:
void noteIncomingPacket(u_int32_t SSRC, u_int16_t seqNum,
u_int32_t rtpTimestamp,
unsigned timestampFrequency,
@@ -134,7 +138,7 @@
Boolean& resultHasBeenSyncedUsingRTCP,
unsigned packetSize /* payload only */);
- // The following is called whenever a RTCP SR packet is received:
+ // The following is called whenever a RTCP SR packet is received:
void noteIncomingSR(u_int32_t SSRC,
u_int32_t ntpTimestampMSW, u_int32_t ntpTimestampLSW,
u_int32_t rtpTimestamp);
@@ -146,7 +150,7 @@
protected: // constructor and destructor, called only by RTPSource:
friend class RTPSource;
- RTPReceptionStatsDB(RTPSource& rtpSource);
+ RTPReceptionStatsDB();
virtual ~RTPReceptionStatsDB();
protected:
@@ -154,7 +158,6 @@
protected:
friend class Iterator;
- RTPSource& fOurRTPSource;
unsigned fNumActiveSourcesSinceLastReset;
private:
@@ -200,9 +203,8 @@
protected:
// called only by RTPReceptionStatsDB:
friend class RTPReceptionStatsDB;
- RTPReceptionStats(RTPSource& rtpSource, u_int32_t SSRC,
- u_int16_t initialSeqNum);
- RTPReceptionStats(RTPSource& rtpSource, u_int32_t SSRC);
+ RTPReceptionStats(u_int32_t SSRC, u_int16_t initialSeqNum);
+ RTPReceptionStats(u_int32_t SSRC);
virtual ~RTPReceptionStats();
private:
@@ -221,7 +223,6 @@
// generate a reception report)
protected:
- RTPSource& fOurRTPSource;
u_int32_t fSSRC;
unsigned fNumPacketsReceivedSinceLastReset;
unsigned fTotNumPacketsReceived;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/RTSPClient.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A generic RTSP client
// C++ header
@@ -48,32 +48,35 @@
RTSPClient*& resultClient);
char* describeURL(char const* url, Authenticator* authenticator = NULL,
- Boolean allowKasennaProtocol = False);
+ Boolean allowKasennaProtocol = False, int timeout = -1);
// Issues a RTSP "DESCRIBE" command
// Returns the SDP description of a session, or NULL if none
// (This is dynamically allocated, and must later be freed
// by the caller - using "delete[]")
char* describeWithPassword(char const* url,
char const* username, char const* password,
- Boolean allowKasennaProtocol = False);
+ Boolean allowKasennaProtocol = False,
+ int timeout = -1);
// Uses "describeURL()" to do a "DESCRIBE" - first
// without using "password", then (if we get an Unauthorized
// response) with an authentication response computed from "password"
Boolean announceSDPDescription(char const* url,
char const* sdpDescription,
- Authenticator* authenticator = NULL);
+ Authenticator* authenticator = NULL,
+ int timeout = -1);
// Issues a RTSP "ANNOUNCE" command
// Returns True iff this command succeeds
Boolean announceWithPassword(char const* url, char const* sdpDescription,
- char const* username, char const* password);
+ char const* username, char const* password, int timeout = -1);
// Uses "announceSDPDescription()" to do an "ANNOUNCE" - first
// without using "password", then (if we get an Unauthorized
// response) with an authentication response computed from "password"
char* sendOptionsCmd(char const* url,
char* username = NULL, char* password = NULL,
- Authenticator* authenticator = NULL);
+ Authenticator* authenticator = NULL,
+ int timeout = -1);
// Issues a RTSP "OPTIONS" command
// Returns a string containing the list of options, or NULL
@@ -86,16 +89,16 @@
// If "forceMulticastOnUnspecified" is True (and "streamUsingTCP" is False),
// then the client will request a multicast stream if the media address
// in the original SDP response was unspecified (i.e., 0.0.0.0).
- // Note, however, that not all servers will support this.
+ // Note, however, that not all servers will support this.
Boolean playMediaSession(MediaSession& session,
- float start = 0.0f, float end = -1.0f,
+ double start = 0.0f, double end = -1.0f,
float scale = 1.0f);
// Issues an aggregate RTSP "PLAY" command on "session".
// Returns True iff this command succeeds
// (Note: start=-1 means 'resume'; end=-1 means 'play to end')
Boolean playMediaSubsession(MediaSubsession& subsession,
- float start = 0.0f, float end = -1.0f,
+ double start = 0.0f, double end = -1.0f,
float scale = 1.0f,
Boolean hackForDSS = False);
// Issues a RTSP "PLAY" command on "subsession".
@@ -164,7 +167,8 @@
void reset();
void resetTCPSockets();
- Boolean openConnectionFromURL(char const* url, Authenticator* authenticator);
+ Boolean openConnectionFromURL(char const* url, Authenticator* authenticator,
+ int timeout = -1);
char* createAuthenticatorString(Authenticator const* authenticator,
char const* cmd, char const* url);
static void checkForAuthenticationFailure(unsigned responseCode,
@@ -183,12 +187,12 @@
portNumBits& serverPortNum,
unsigned char& rtpChannelId,
unsigned char& rtcpChannelId);
- Boolean parseRTPInfoHeader(char const* line, unsigned& trackId,
- u_int16_t& seqNum, u_int32_t& timestamp);
+ Boolean parseRTPInfoHeader(char*& line, u_int16_t& seqNum, u_int32_t& timestamp);
Boolean parseScaleHeader(char const* line, float& scale);
- Boolean parseGetParameterHeader(char const* line,
+ Boolean parseGetParameterHeader(char const* line,
const char* param,
char*& value);
+ char const* sessionURL(MediaSession const& session) const;
void constructSubsessionURL(MediaSubsession const& subsession,
char const*& prefix,
char const*& separator,
@@ -215,7 +219,7 @@
Authenticator fCurrentAuthenticator;
unsigned char fTCPStreamIdCount; // used for (optional) RTP/TCP
char* fLastSessionId;
- unsigned fSessionTimeoutParameter; // optionally set in response "Session:" headers
+ unsigned fSessionTimeoutParameter; // optionally set in response "Session:" headers
#ifdef SUPPORT_REAL_RTSP
char* fRealChallengeStr;
char* fRealETagStr;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/RTSPCommon.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Common routines used by both RTSP clients and servers
// C++ header
@@ -36,11 +36,14 @@
Boolean parseRTSPRequestString(char const *reqStr, unsigned reqStrSize,
char *resultCmdName,
- unsigned resultCmdNameMaxSize,
+ unsigned resultCmdNameMaxSize,
char* resultURLPreSuffix,
- unsigned resultURLPreSuffixMaxSize,
+ unsigned resultURLPreSuffixMaxSize,
char* resultURLSuffix,
- unsigned resultURLSuffixMaxSize,
+ unsigned resultURLSuffixMaxSize,
char* resultCSeq,
- unsigned resultCSeqMaxSize);
+ unsigned resultCSeqMaxSize);
+
+Boolean parseRangeHeader(char const* buf, double& rangeStart, double& rangeEnd);
+
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/RTSPOverHTTPServer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simple HTTP server that acts solely to implement RTSP-over-HTTP tunneling
// (to a separate RTSP server), as described in
// http://developer.apple.com/documentation/QuickTime/QTSS/Concepts/chapter_2_section_14.html
@@ -57,7 +57,7 @@
UsageEnvironment& envir() { return fOurServer.envir(); }
void resetRequestBuffer();
Boolean parseHTTPRequestString(char* resultCmdName,
- unsigned resultCmdNameMaxSize,
+ unsigned resultCmdNameMaxSize,
char* sessionCookie,
unsigned sessionCookieMaxSize,
char* acceptStr,
@@ -113,7 +113,7 @@
#if 0 //#####@@@@@
Authenticator fCurrentAuthenticator; // used if access control is needed
unsigned char fTCPStreamIdCount; // used for (optional) RTP/TCP
- unsigned fNumStreamStates;
+ unsigned fNumStreamStates;
struct streamState {
ServerMediaSubsession* subsession;
void* streamToken;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/RTSPServer.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A RTSP server
// C++ header
@@ -63,7 +63,7 @@
public:
static RTSPServer* createNew(UsageEnvironment& env, Port ourPort = 554,
UserAuthenticationDatabase* authDatabase = NULL,
- unsigned reclamationTestSeconds = 45);
+ unsigned reclamationTestSeconds = 65);
// If ourPort.num() == 0, we'll choose the port number
// Note: The caller is responsible for reclaiming "authDatabase"
// If "reclamationTestSeconds" > 0, then the "RTSPClientSession" state for
@@ -168,7 +168,7 @@
Boolean fIsMulticast, fSessionIsActive, fStreamAfterSETUP;
Authenticator fCurrentAuthenticator; // used if access control is needed
unsigned char fTCPStreamIdCount; // used for (optional) RTP/TCP
- unsigned fNumStreamStates;
+ unsigned fNumStreamStates;
struct streamState {
ServerMediaSubsession* subsession;
void* streamToken;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/SIPClient.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A generic SIP client
// C++ header
@@ -88,7 +88,7 @@
Boolean processURL(char const* url);
Boolean sendINVITE();
static void inviteResponseHandler(void* clientData, int mask);
- void doInviteStateMachine(unsigned responseCode);
+ void doInviteStateMachine(unsigned responseCode);
void doInviteStateTerminated(unsigned responseCode);
TaskToken fTimerA, fTimerB, fTimerD;
static void timerAHandler(void* clientData);
@@ -126,7 +126,7 @@
char const* fURL;
unsigned fURLSize;
struct in_addr fServerAddress;
- portNumBits fServerPortNum; // in host order
+ portNumBits fServerPortNum; // in host order
portNumBits fClientStartPortNum; // in host order
unsigned fCallId, fFromTag; // set by us
char const* fToTagStr; // set by the responder
@@ -142,7 +142,7 @@
inviteClientState fInviteClientState;
char fEventLoopStopFlag;
unsigned fInviteStatusCode;
- // 0: OK; 1: connection failed; 2: stream unavailable
+ // 0: OK; 1: connection failed; 2: stream unavailable
};
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/ServerMediaSession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A data structure that represents a session that consists of
// potentially multiple (audio and/or video) sub-sessions
// (This data structure is used for media *streamers* - i.e., servers.
@@ -41,7 +41,7 @@
char const* description = NULL,
Boolean isSSM = False,
char const* miscSDPLines = NULL);
-
+
virtual ~ServerMediaSession();
static Boolean lookupByName(UsageEnvironment& env,
@@ -98,10 +98,10 @@
public:
ServerMediaSubsessionIterator(ServerMediaSession& session);
virtual ~ServerMediaSubsessionIterator();
-
+
ServerMediaSubsession* next(); // NULL if none
void reset();
-
+
private:
ServerMediaSession& fOurSession;
ServerMediaSubsession* fNextPtr;
@@ -135,7 +135,7 @@
unsigned short& rtpSeqNum,
unsigned& rtpTimestamp) = 0;
virtual void pauseStream(unsigned clientSessionId, void* streamToken);
- virtual void seekStream(unsigned clientSessionId, void* streamToken, float seekNPT);
+ virtual void seekStream(unsigned clientSessionId, void* streamToken, double seekNPT);
virtual void setStreamScale(unsigned clientSessionId, void* streamToken, float scale);
virtual void deleteStream(unsigned clientSessionId, void*& streamToken);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/SimpleRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A simple RTP sink that packs frames into each outgoing
// packet, without any fragmentation or special headers.
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/SimpleRTPSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A RTP source for a simple RTP payload format that
// - doesn't have any special headers following the RTP header
// (if necessary, the "offset" parameter can be used to specify a
@@ -56,7 +56,7 @@
// redefined virtual functions:
virtual Boolean processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize);
- virtual char const* MIMEtype() const;
+ virtual char const* MIMEtype() const;
private:
char const* fMIMEtypeString;
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/VideoRTPSink.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A generic RTP sink for video codecs (abstract base class)
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/WAVAudioFileServerMediaSubsession.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A 'ServerMediaSubsession' object that creates new, unicast, "RTPSink"s
// on demand, from an WAV audio file.
// C++ header
@@ -41,7 +41,7 @@
virtual ~WAVAudioFileServerMediaSubsession();
protected: // redefined virtual functions
- virtual void seekStreamSource(FramedSource* inputSource, float seekNPT);
+ virtual void seekStreamSource(FramedSource* inputSource, double seekNPT);
virtual void setStreamSourceScale(FramedSource* inputSource, float scale);
virtual FramedSource* createNewStreamSource(unsigned clientSessionId,
unsigned& estBitrate);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/WAVAudioFileSource.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// A WAV audio file source
// NOTE: Samples are returned in little-endian order (the same order in which
// they were stored in the file).
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/liveMedia.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Inclusion of header files representing the interface
// for the entire library
//
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/liveMedia_version.hh
^
|
| @@ -1,10 +1,10 @@
// Version information for the "liveMedia" library
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
#ifndef _LIVEMEDIA_VERSION_HH
#define _LIVEMEDIA_VERSION_HH
-#define LIVEMEDIA_LIBRARY_VERSION_STRING "2007.10.31"
-#define LIVEMEDIA_LIBRARY_VERSION_INT 1193788800
+#define LIVEMEDIA_LIBRARY_VERSION_STRING "2008.12.19"
+#define LIVEMEDIA_LIBRARY_VERSION_INT 1229644800
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/include/uLawAudioFilter.hh
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Filters for converting between raw PCM audio and uLaw
// C++ header
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/liveMedia/uLawAudioFilter.cpp
^
|
| @@ -11,10 +11,10 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
-// Copyright (c) 1996-2007 Live Networks, Inc. All rights reserved.
+// Copyright (c) 1996-2008 Live Networks, Inc. All rights reserved.
// Filters for converting between raw PCM audio and uLaw
// Implementation
@@ -28,7 +28,7 @@
if (byteOrdering < 0 || byteOrdering > 2) {
env.setResultMsg("uLawFromPCMAudioSource::createNew(): bad \"byteOrdering\" parameter");
return NULL;
- }
+ }
return new uLawFromPCMAudioSource(env, inputSource, byteOrdering);
}
@@ -37,11 +37,11 @@
int byteOrdering)
: FramedFilter(env, inputSource),
fByteOrdering(byteOrdering), fInputBuffer(NULL), fInputBufferSize(0) {
-}
+}
uLawFromPCMAudioSource::~uLawFromPCMAudioSource() {
delete[] fInputBuffer;
-}
+}
void uLawFromPCMAudioSource::doGetNextFrame() {
// Figure out how many bytes of input data to ask for, and increase
@@ -154,11 +154,11 @@
FramedSource* inputSource)
: FramedFilter(env, inputSource),
fInputBuffer(NULL), fInputBufferSize(0) {
-}
+}
PCMFromuLawAudioSource::~PCMFromuLawAudioSource() {
delete[] fInputBuffer;
-}
+}
void PCMFromuLawAudioSource::doGetNextFrame() {
// Figure out how many bytes of input data to ask for, and increase
@@ -230,10 +230,10 @@
::NetworkFromHostOrder16(UsageEnvironment& env,
FramedSource* inputSource)
: FramedFilter(env, inputSource) {
-}
+}
NetworkFromHostOrder16::~NetworkFromHostOrder16() {
-}
+}
void NetworkFromHostOrder16::doGetNextFrame() {
// Arrange to read data directly into the client's buffer:
@@ -284,10 +284,10 @@
::HostFromNetworkOrder16(UsageEnvironment& env,
FramedSource* inputSource)
: FramedFilter(env, inputSource) {
-}
+}
HostFromNetworkOrder16::~HostFromNetworkOrder16() {
-}
+}
void HostFromNetworkOrder16::doGetNextFrame() {
// Arrange to read data directly into the client's buffer:
@@ -337,10 +337,10 @@
EndianSwap16::EndianSwap16(UsageEnvironment& env,
FramedSource* inputSource)
: FramedFilter(env, inputSource) {
-}
+}
EndianSwap16::~EndianSwap16() {
-}
+}
void EndianSwap16::doGetNextFrame() {
// Arrange to read data directly into the client's buffer:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/mediaServer/DynamicRTSPServer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A subclass of "RTSPServer" that creates "ServerMediaSession"s on demand,
// based on whether or not the specified stream name exists as a file
// Implementation
@@ -24,14 +24,15 @@
DynamicRTSPServer*
DynamicRTSPServer::createNew(UsageEnvironment& env, Port ourPort,
- UserAuthenticationDatabase* authDatabase) {
+ UserAuthenticationDatabase* authDatabase,
+ unsigned reclamationTestSeconds) {
int ourSocket = -1;
do {
int ourSocket = setUpOurSocket(env, ourPort);
if (ourSocket == -1) break;
- return new DynamicRTSPServer(env, ourSocket, ourPort, authDatabase);
+ return new DynamicRTSPServer(env, ourSocket, ourPort, authDatabase, reclamationTestSeconds);
} while (0);
if (ourSocket != -1) ::closeSocket(ourSocket);
@@ -40,8 +41,8 @@
DynamicRTSPServer::DynamicRTSPServer(UsageEnvironment& env, int ourSocket,
Port ourPort,
- UserAuthenticationDatabase* authDatabase)
- : RTSPServer(env, ourSocket, ourPort, authDatabase, 45/*reclamationTestSeconds*/) {
+ UserAuthenticationDatabase* authDatabase, unsigned reclamationTestSeconds)
+ : RTSPServer(env, ourSocket, ourPort, authDatabase, reclamationTestSeconds) {
}
DynamicRTSPServer::~DynamicRTSPServer() {
@@ -55,11 +56,11 @@
// First, check whether the specified "streamName" exists as a local file:
FILE* fid = fopen(streamName, "rb");
Boolean fileExists = fid != NULL;
-
+
// Next, check whether we already have a "ServerMediaSession" for this file:
ServerMediaSession* sms = RTSPServer::lookupServerMediaSession(streamName);
Boolean smsExists = sms != NULL;
-
+
// Handle the four possibilities for "fileExists" and "smsExists":
if (!fileExists) {
if (smsExists) {
@@ -148,7 +149,7 @@
// change the following to True:
Boolean convertToULaw = False;
sms->addSubsession(WAVAudioFileServerMediaSubsession::createNew(env, fileName, reuseSource, convertToULaw));
- }
+ }
return sms;
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/mediaServer/DynamicRTSPServer.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A subclass of "RTSPServer" that creates "ServerMediaSession"s on demand,
// based on whether or not the specified stream name exists as a file
// Header file
@@ -28,11 +28,12 @@
class DynamicRTSPServer: public RTSPServer {
public:
static DynamicRTSPServer* createNew(UsageEnvironment& env, Port ourPort,
- UserAuthenticationDatabase* authDatabase);
+ UserAuthenticationDatabase* authDatabase,
+ unsigned reclamationTestSeconds = 65);
private:
DynamicRTSPServer(UsageEnvironment& env, int ourSocket, Port ourPort,
- UserAuthenticationDatabase* authDatabase);
+ UserAuthenticationDatabase* authDatabase, unsigned reclamationTestSeconds);
// called only by createNew();
virtual ~DynamicRTSPServer();
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/mediaServer/live555MediaServer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// LIVE555 Media Server
// main program
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/mediaServer/version.hh
^
|
| @@ -1,10 +1,10 @@
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// Version information for the LIVE555 Media Server application
// Header file
#ifndef _MEDIA_SERVER_VERSION_HH
#define _MEDIA_SERVER_VERSION_HH
-#define MEDIA_SERVER_VERSION_STRING "0.18"
+#define MEDIA_SERVER_VERSION_STRING "0.20"
#endif
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/MPEG2TransportStreamIndexer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A program that reads an existing MPEG-2 Transport Stream file,
// and generates a separate index file that can be used - by our RTSP server
// implementation - to support 'trick play' operations when streaming the
@@ -74,7 +74,7 @@
*env << "Failed to open output file \"" << outputFileName << "\"\n";
exit(1);
}
-
+
// Start playing, to generate the output index file:
*env << "Writing index file \"" << outputFileName << "\"...";
output->startPlaying(*indexer, afterPlaying, NULL);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/openRTSP.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A RTSP client test program that opens a RTSP URL argument,
// and extracts the data from each incoming RTP stream.
@@ -63,15 +63,16 @@
extern double initialSeekTime, duration, scale;
double endTime = initialSeekTime;
if (scale > 0) {
- endTime = initialSeekTime + duration;
- } else if (scale < 0) {
+ if (duration <= 0) endTime = -1.0f;
+ else endTime = initialSeekTime + duration;
+ } else {
endTime = initialSeekTime - duration;
if (endTime < 0) endTime = 0.0f;
}
if (client == NULL || session == NULL) return False;
RTSPClient* rtspClient = (RTSPClient*)client;
- return rtspClient->playMediaSession(*session, (float)initialSeekTime, (float)endTime, (float)scale);
+ return rtspClient->playMediaSession(*session, initialSeekTime, endTime, (float)scale);
}
Boolean clientTearDownSession(Medium* client,
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/playCommon.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A common framework, used for the "openRTSP" and "playSIP" applications
// Implementation
@@ -173,7 +173,7 @@
break;
}
- case 'I': { // specify input interface...
+ case 'I': { // specify input interface...
NetAddressList addresses(argv[2]);
if (addresses.numAddresses() == 0) {
*env << "Failed to find network address for \"" << argv[2] << "\"";
@@ -410,8 +410,8 @@
}
case 's': { // specify initial seek time (trick play)
- float arg;
- if (sscanf(argv[2], "%g", &arg) != 1 || arg < 0) {
+ double arg;
+ if (sscanf(argv[2], "%lg", &arg) != 1 || arg < 0) {
usage();
}
initialSeekTime = arg;
@@ -579,7 +579,7 @@
// Because we're saving the incoming data, rather than playing
// it in real time, allow an especially large time threshold
// (1 second) for reordering misordered incoming packets:
- unsigned const thresh = 1000000; // 1 second
+ unsigned const thresh = 1000000; // 1 second
subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);
if (socketInputBufferSize > 0) {
@@ -605,7 +605,7 @@
<< subsession->mediumName()
<< "/" << subsession->codecName()
<< "\" subsession. (Try adding the \"-p <portNum>\" option.)\n";
- } else {
+ } else {
madeProgress = True;
}
}
@@ -677,11 +677,10 @@
while ((subsession = iter.next()) != NULL) {
if (subsession->readSource() == NULL) continue; // was not initiated
- fileSink->startPlaying(*(subsession->readSource()),
- subsessionAfterPlaying,
- subsession);
- madeProgress = True;
- break; // play from one subsession only
+ fileSink->startPlaying(*(subsession->readSource()),
+ subsessionAfterPlaying, subsession);
+ madeProgress = True;
+ break; // play from one subsession only
}
if (!madeProgress) shutdown();
#endif
@@ -691,7 +690,7 @@
iter.reset();
while ((subsession = iter.next()) != NULL) {
if (subsession->readSource() == NULL) continue; // was not initiated
-
+
// Create an output file for each desired stream:
char outFileName[1000];
if (singleMedium == NULL) {
@@ -752,7 +751,7 @@
subsession->sink->startPlaying(*(subsession->readSource()),
subsessionAfterPlaying,
subsession);
-
+
// Also set a handler to be called if a RTCP "BYE" arrives
// for this subsession:
if (subsession->rtcpInstance() != NULL) {
@@ -766,7 +765,7 @@
if (!madeProgress) shutdown();
}
}
-
+
// Finally, start playing each subsession, to start the data flow:
startPlayingStreams();
@@ -959,7 +958,7 @@
}
}
virtual ~qosMeasurementRecord() { delete fNext; }
-
+
void periodicQOSMeasurement(struct timeval const& timeNow);
public:
@@ -1108,16 +1107,16 @@
*env << "subsession\t" << subsession->mediumName()
<< "/" << subsession->codecName() << "\n";
-
+
unsigned numPacketsReceived = 0, numPacketsExpected = 0;
-
+
if (curQOSRecord != NULL) {
numPacketsReceived = curQOSRecord->totNumPacketsReceived;
numPacketsExpected = curQOSRecord->totNumPacketsExpected;
}
*env << "num_packets_received\t" << numPacketsReceived << "\n";
*env << "num_packets_lost\t" << numPacketsExpected - numPacketsReceived << "\n";
-
+
if (curQOSRecord != NULL) {
unsigned secsDiff = curQOSRecord->measurementEndTime.tv_sec
- curQOSRecord->measurementStartTime.tv_sec;
@@ -1125,11 +1124,11 @@
- curQOSRecord->measurementStartTime.tv_usec;
double measurementTime = secsDiff + usecsDiff/1000000.0;
*env << "elapsed_measurement_time\t" << measurementTime << "\n";
-
+
*env << "kBytes_received_total\t" << curQOSRecord->kBytesTotal << "\n";
-
+
*env << "measurement_sampling_interval_ms\t" << qosMeasurementIntervalMS << "\n";
-
+
if (curQOSRecord->kbits_per_second_max == 0) {
// special case: we didn't receive any data:
*env <<
@@ -1142,7 +1141,7 @@
<< (measurementTime == 0.0 ? 0.0 : 8*curQOSRecord->kBytesTotal/measurementTime) << "\n";
*env << "kbits_per_second_max\t" << curQOSRecord->kbits_per_second_max << "\n";
}
-
+
*env << "packet_loss_percentage_min\t" << 100*curQOSRecord->packet_loss_fraction_min << "\n";
double packetLossFraction = numPacketsExpected == 0 ? 1.0
: 1.0 - numPacketsReceived/(double)numPacketsExpected;
@@ -1150,7 +1149,7 @@
*env << "packet_loss_percentage_ave\t" << 100*packetLossFraction << "\n";
*env << "packet_loss_percentage_max\t"
<< (packetLossFraction == 1.0 ? 100.0 : 100*curQOSRecord->packet_loss_fraction_max) << "\n";
-
+
#ifdef SUPPORT_REAL_RTSP
if (session->isRealNetworksRDT) {
RealRDTSource* rdt = (RealRDTSource*)src;
@@ -1178,11 +1177,11 @@
#ifdef SUPPORT_REAL_RTSP
}
#endif
-
+
curQOSRecord = curQOSRecord->fNext;
}
}
- }
+ }
*env << "end_QOS_statistics\n";
delete qosRecordHead;
@@ -1220,7 +1219,7 @@
}
void checkForPacketArrival(void* /*clientData*/) {
- if (!notifyOnPacketArrival) return; // we're not checking
+ if (!notifyOnPacketArrival) return; // we're not checking
// Check each subsession, to see whether it has received data packets:
unsigned numSubsessionsChecked = 0;
@@ -1278,7 +1277,7 @@
}
void checkInterPacketGaps(void* /*clientData*/) {
- if (interPacketGapMaxTime == 0) return; // we're not checking
+ if (interPacketGapMaxTime == 0) return; // we're not checking
// Check each subsession, counting up how many packets have been received:
unsigned newTotNumPacketsReceived = 0;
@@ -1299,7 +1298,7 @@
sessionAfterPlaying();
} else {
totNumPacketsReceived = newTotNumPacketsReceived;
- // Check again, after the specified delay:
+ // Check again, after the specified delay:
interPacketGapCheckTimerTask
= env->taskScheduler().scheduleDelayedTask(interPacketGapMaxTime*1000000,
(TaskFunc*)checkInterPacketGaps, NULL);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/playCommon.hh
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2003, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A common framework, used for the "openRTSP" and "playSIP" applications
// Interfaces
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/playSIP.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A SIP client test program that opens a SIP URL argument,
// and extracts the data from each incoming RTP stream.
@@ -64,7 +64,7 @@
unsigned proxyServerAddress // later, allow for IPv6 #####
= *(unsigned*)(address.data());
if (proxyServerPortNum == 0) proxyServerPortNum = 5060; // default
-
+
sipClient->setProxyServer(proxyServerAddress, proxyServerPortNum);
}
}
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/sapWatch.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A program that receives and prints SDP/SAP announcements
// (on the default SDP/SAP directory: 224.2.127.254/9875)
@@ -38,9 +38,9 @@
const Port port(9875);
const unsigned char ttl = 0; // we're only reading from this mcast group
-
+
Groupsock inputGroupsock(*env, sessionAddress, port, ttl);
-
+
// Start reading and printing incoming packets
// (Because this is the only thing we do, we can just do this
// synchronously, in a loop, so we don't need to set up an asynchronous
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testAMRAudioStreamer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that reads an AMR audio file (as defined in RFC 3267)
// and streams it using RTP
// main program
@@ -114,7 +114,7 @@
<< env->getResultMsg() << "\n";
exit(1);
}
-
+
// Finally, start playing:
*env << "Beginning to read from file...\n";
audioSink->startPlaying(*audioSource, afterPlaying, audioSink);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testGSMStreamer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that streams GSM audio via RTP/RTCP
// main program
@@ -78,12 +78,12 @@
const unsigned short rtpPortNum = 6666;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 1; // low, in case routers don't admin scope
-
+
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
-
+
sessionState.rtpGroupsock
= new Groupsock(*env, destinationAddress, rtpPort, ttl);
sessionState.rtcpGroupsock
@@ -92,11 +92,11 @@
sessionState.rtpGroupsock->multicastSendOnly();
sessionState.rtcpGroupsock->multicastSendOnly();
#endif
-
+
// Create a 'GSM RTP' sink from the RTP 'groupsock':
sessionState.sink
= GSMAudioRTPSink::createNew(*env, sessionState.rtpGroupsock);
-
+
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 160; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
@@ -141,7 +141,7 @@
*env << "Failed to create GSM source\n";
exit(1);
}
-
+
// Finally, start the streaming:
*env << "Beginning streaming...\n";
sessionState.sink->startPlaying(*sessionState.source, afterPlaying, NULL);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMP3Receiver.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that receives a RTP/RTCP multicast MP3 stream,
// and outputs the resulting MP3 file stream to 'stdout'
// main program
@@ -67,12 +67,12 @@
#ifndef USE_SSM
const unsigned char ttl = 1; // low, in case routers don't admin scope
#endif
-
+
struct in_addr sessionAddress;
sessionAddress.s_addr = our_inet_addr(sessionAddressStr);
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
-
+
#ifdef USE_SSM
char* sourceAddressStr = "aaa.bbb.ccc.ddd";
// replace this with the real source address
@@ -87,7 +87,7 @@
Groupsock rtpGroupsock(*env, sessionAddress, rtpPort, ttl);
Groupsock rtcpGroupsock(*env, sessionAddress, rtcpPort, ttl);
#endif
-
+
RTPSource* rtpSource;
#ifndef STREAM_USING_ADUS
// Create the data source: a "MPEG Audio RTP source"
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMP3Streamer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that streams a MP3 file via RTP/RTCP
// main program
@@ -80,12 +80,12 @@
const unsigned short rtpPortNum = 6666;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 1; // low, in case routers don't admin scope
-
+
struct in_addr destinationAddress;
destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
-
+
sessionState.rtpGroupsock
= new Groupsock(*env, destinationAddress, rtpPort, ttl);
sessionState.rtcpGroupsock
@@ -94,7 +94,7 @@
sessionState.rtpGroupsock->multicastSendOnly();
sessionState.rtcpGroupsock->multicastSendOnly();
#endif
-
+
// Create a 'MP3 RTP' sink from the RTP 'groupsock':
#ifdef STREAM_USING_ADUS
unsigned char rtpPayloadFormat = 96; // A dynamic payload format code
@@ -105,7 +105,7 @@
sessionState.sink
= MPEG1or2AudioRTPSink::createNew(*env, sessionState.rtpGroupsock);
#endif
-
+
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = 160; // in kbps; for RTCP b/w share
const unsigned maxCNAMElen = 100;
@@ -155,7 +155,7 @@
<< "\" as a MP3 file source\n";
exit(1);
}
-
+
#ifdef STREAM_USING_ADUS
// Add a filter that converts the source MP3s to ADUs:
sessionState.source
@@ -170,7 +170,7 @@
unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own order...
unsigned const interleaveCycleSize
= (sizeof interleaveCycle)/(sizeof (unsigned char));
- Interleaving interleaving(interleaveCycleSize, interleaveCycle);
+ Interleaving interleaving(interleaveCycleSize, interleaveCycle);
sessionState.source
= MP3ADUinterleaver::createNew(*env, interleaving, sessionState.source);
if (sessionState.source == NULL) {
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG1or2AudioVideoStreamer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that reads a MPEG-1 or 2 Program Stream file,
// splits it into Audio and Video Elementary Streams,
// and streams both using RTP
@@ -156,7 +156,7 @@
// wait until its sink ends playing also:
if (audioSource->isCurrentlyAwaitingData()
|| videoSource->isCurrentlyAwaitingData()) return;
-
+
// Now that both sinks have ended, close both input sources,
// and start playing again:
*env << "...done reading from file\n";
@@ -182,7 +182,7 @@
<< "\" as a byte-stream file source\n";
exit(1);
}
-
+
// We must demultiplex Audio and Video Elementary Streams
// from the input source:
mpegDemux = MPEG1or2Demux::createNew(*env, fileSource);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG1or2AudioVideoToDarwin.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that reads a MPEG-1 or 2 Program Stream file,
// splits it into Audio and Video Elementary Streams,
// and streams both using RTP, through a remote Darwin Streaming Server.
@@ -143,7 +143,7 @@
// wait until its sink ends playing also:
if (audioSource->isCurrentlyAwaitingData()
|| videoSource->isCurrentlyAwaitingData()) return;
-
+
// Now that both sinks have ended, close both input sources,
// and start playing again:
*env << "...done reading from file\n";
@@ -169,7 +169,7 @@
<< "\" as a byte-stream file source\n";
exit(1);
}
-
+
// We must demultiplex Audio and Video Elementary Streams
// from the input source:
mpegDemux = MPEG1or2Demux::createNew(*env, fileSource);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG1or2ProgramToTransportStream.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A program that converts a MPEG-1 or 2 Program Stream file into
// a Transport Stream file.
// main program
@@ -40,7 +40,7 @@
<< "\" as a byte-stream file source\n";
exit(1);
}
-
+
// Create a MPEG demultiplexor that reads from that source.
MPEG1or2Demux* baseDemultiplexor = MPEG1or2Demux::createNew(*env, inputSource);
@@ -57,7 +57,7 @@
*env << "Unable to open file \"" << outputFileName << "\" as a file sink\n";
exit(1);
}
-
+
// Finally, start playing:
*env << "Beginning to read...\n";
outputSink->startPlaying(*tsFrames, afterPlaying, NULL);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG1or2Splitter.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that splits a MPEG-1 or 2 Program Stream file into
// video and audio output files.
// main program
@@ -53,7 +53,7 @@
<< "\" as a byte-stream file source\n";
exit(1);
}
-
+
// Create a MPEG demultiplexor that reads from that source.
sessionState.baseDemultiplexor = MPEG1or2Demux::createNew(*env, inputSource);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG1or2VideoReceiver.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2000, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that receives a RTP/RTCP multicast MPEG video stream,
// and outputs the resulting MPEG file stream to 'stdout'
// main program
@@ -62,12 +62,12 @@
#ifndef USE_SSM
const unsigned char ttl = 1; // low, in case routers don't admin scope
#endif
-
+
struct in_addr sessionAddress;
sessionAddress.s_addr = our_inet_addr(sessionAddressStr);
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
-
+
#ifdef USE_SSM
char* sourceAddressStr = "aaa.bbb.ccc.ddd";
// replace this with the real source address
@@ -82,7 +82,7 @@
Groupsock rtpGroupsock(*env, sessionAddress, rtpPort, ttl);
Groupsock rtcpGroupsock(*env, sessionAddress, rtcpPort, ttl);
#endif
-
+
// Create the data source: a "MPEG Video RTP source"
sessionState.source = MPEG1or2VideoRTPSource::createNew(*env, &rtpGroupsock);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG1or2VideoStreamer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that reads a MPEG-1 or 2 Video Elementary Stream file,
// and streams it using RTP
// main program
@@ -152,7 +152,7 @@
<< "\" as a byte-stream file source\n";
exit(1);
}
-
+
FramedSource* videoES;
#ifdef SOURCE_IS_PROGRAM_STREAM
// We must demultiplex a Video Elementary Stream from the input source:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG2TransportStreamTrickPlay.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A program that tests 'trick mode' operations on a MPEG-2 Transport Stream file,
// by generating a new Transport Stream file that represents the result of the
// 'trick mode' operation (seeking and/or fast forward/reverse play).
@@ -114,7 +114,7 @@
*env << "Failed to open output file \"" << outputFileName << "\"\n";
exit(1);
}
-
+
// Start playing, to generate the output file:
*env << "Writing output file \"" << outputFileName
<< "\" (start time " << startTime
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG2TransportStreamer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that reads a MPEG-2 Transport Stream file,
// and streams it using RTP
// main program
@@ -147,7 +147,7 @@
<< "\" as a byte-stream file source\n";
exit(1);
}
-
+
// Create a 'framer' for the input source (to give us proper inter-packet gaps):
videoSource = MPEG2TransportStreamFramer::createNew(*env, fileSource);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG4VideoStreamer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that reads a MPEG-4 Video Elementary Stream file,
// and streams it using RTP
// main program
@@ -113,12 +113,12 @@
<< "\" as a byte-stream file source\n";
exit(1);
}
-
+
FramedSource* videoES = fileSource;
// Create a framer for the Video Elementary Stream:
videoSource = MPEG4VideoStreamFramer::createNew(*env, videoES);
-
+
// Finally, start playing:
*env << "Beginning to read from file...\n";
videoSink->startPlaying(*videoSource, afterPlaying, videoSink);
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testMPEG4VideoToDarwin.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that reads a MPEG-4 Video Elementary Stream file,
// and streams both using RTP, through a remote Darwin Streaming Server.
// main program
@@ -137,7 +137,7 @@
<< "\" as a byte-stream file source\n";
exit(1);
}
-
+
FramedSource* videoES = fileSource;
// Create a framer for the Video Elementary Stream:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testOnDemandRTSPServer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that demonstrates how to stream - via unicast RTP
// - various kinds of file on demand, using a built-in RTSP server.
// main program
@@ -51,7 +51,7 @@
#endif
// Create the RTSP server:
- RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB, 45);
+ RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, authDB);
if (rtspServer == NULL) {
*env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
exit(1);
@@ -126,13 +126,13 @@
descriptionString);
Boolean useADUs = False;
Interleaving* interleaving = NULL;
-#ifdef STREAM_USING_ADUS
+#ifdef STREAM_USING_ADUS
useADUs = True;
#ifdef INTERLEAVE_ADUS
unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...
unsigned const interleaveCycleSize
= (sizeof interleaveCycle)/(sizeof (unsigned char));
- interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);
+ interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);
#endif
#endif
sms->addSubsession(MP3AudioFileServerMediaSubsession
@@ -228,7 +228,7 @@
static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,
char const* streamName, char const* inputFileName) {
char* url = rtspServer->rtspURL(sms);
- UsageEnvironment& env = rtspServer->envir();
+ UsageEnvironment& env = rtspServer->envir();
env << "\n\"" << streamName << "\" stream, from the file \""
<< inputFileName << "\"\n";
env << "Play this stream using the URL \"" << url << "\"\n";
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testRelay.cpp
^
|
| @@ -11,11 +11,11 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that receives a UDP multicast stream
-// and retransmits it to another (multicast or unicast) address & port
+// and retransmits it to another (multicast or unicast) address & port
// main program
#include <liveMedia.hh>
@@ -44,7 +44,7 @@
Port const inputPort(8888);
unsigned char const inputTTL = 0; // we're only reading from this mcast group
-
+
#ifdef USE_SSM
char* sourceAddressStr = "aaa.bbb.ccc.ddd";
// replace this with the real source address
@@ -55,7 +55,7 @@
#else
Groupsock inputGroupsock(*env, inputAddress, inputPort, inputTTL);
#endif
-
+
// Then create a liveMedia 'source' object, encapsulating this groupsock:
FramedSource* source = BasicUDPSource::createNew(*env, &inputGroupsock);
@@ -67,10 +67,10 @@
// address as "inputAddressStr".
struct in_addr outputAddress;
outputAddress.s_addr = our_inet_addr(outputAddressStr);
-
+
Port const outputPort(4444);
unsigned char const outputTTL = 255;
-
+
Groupsock outputGroupsock(*env, outputAddress, outputPort, outputTTL);
// Then create a liveMedia 'sink' object, encapsulating this groupsock:
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/testWAVAudioStreamer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that streams a WAV audio file via RTP/RTCP
// main program
@@ -83,7 +83,7 @@
*env << bitsPerSample << " bits-per-sample, ";
*env << numChannels << " channels => ";
*env << bitsPerSecond << " bits-per-second\n";
-
+
// Add in any filter necessary to transform the data prior to streaming.
// (This is where any audio compression would get added.)
char* mimeType;
@@ -143,24 +143,24 @@
const unsigned short rtpPortNum = 2222;
const unsigned short rtcpPortNum = rtpPortNum+1;
const unsigned char ttl = 255;
-
+
const Port rtpPort(rtpPortNum);
const Port rtcpPort(rtcpPortNum);
-
+
sessionState.rtpGroupsock
= new Groupsock(*env, destinationAddress, rtpPort, ttl);
sessionState.rtpGroupsock->multicastSendOnly(); // we're a SSM source
sessionState.rtcpGroupsock
= new Groupsock(*env, destinationAddress, rtcpPort, ttl);
sessionState.rtcpGroupsock->multicastSendOnly(); // we're a SSM source
-
+
// Create an appropriate audio RTP sink (using "SimpleRTPSink")
// from the RTP 'groupsock':
sessionState.sink
= SimpleRTPSink::createNew(*env, sessionState.rtpGroupsock,
payloadFormatCode, samplingFrequency,
"audio", mimeType, numChannels);
-
+
// Create (and start) a 'RTCP instance' for this RTP sink:
const unsigned estimatedSessionBandwidth = bitsPerSecond/1000;
// in kbps; for RTCP b/w share
|
|
[-]
[+]
|
Changed |
live.2008.12.20.tar.bz2/testProgs/vobStreamer.cpp
^
|
| @@ -11,9 +11,9 @@
You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
-59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
-// Copyright (c) 1996-2007, Live Networks, Inc. All rights reserved
+// Copyright (c) 1996-2008, Live Networks, Inc. All rights reserved
// A test program that reads a VOB file
// splits it into Audio (AC3) and Video (MPEG) Elementary Streams,
// and streams both using RTP.
@@ -122,7 +122,7 @@
if (iFramesOnly && (mediaToStream&VOB_VIDEO) == 0) {
*env << "Warning: Because we're not streaming video, the -i flag has no effect.\n";
}
-
+
inputFileNames = &argv[1];
curInputFileName = inputFileNames;
@@ -155,7 +155,7 @@
audioSink
= AC3AudioRTPSink::createNew(*env, rtpGroupsockAudio, 96, 0);
// set the RTP timestamp frequency 'for real' later
-
+
// Create (and start) a 'RTCP instance' for this RTP sink:
rtcpGroupsockAudio
= new Groupsock(*env, destinationAddress, rtcpPortAudio, ttl);
@@ -173,10 +173,10 @@
rtpGroupsockVideo
= new Groupsock(*env, destinationAddress, rtpPortVideo, ttl);
rtpGroupsockVideo->multicastSendOnly(); // because we're a SSM source
-
+
// Create a 'MPEG Video RTP' sink from the RTP 'groupsock':
videoSink = MPEG1or2VideoRTPSink::createNew(*env, rtpGroupsockVideo);
-
+
// Create (and start) a 'RTCP instance' for this RTP sink:
rtcpGroupsockVideo
= new Groupsock(*env, destinationAddress, rtcpPortVideo, ttl);
@@ -229,7 +229,7 @@
|| videoSource != NULL && videoSource->isCurrentlyAwaitingData()) {
return;
}
-
+
// Now that both sinks have ended, close both input sources,
// and start playing again:
*env << "...done reading from file\n";
@@ -257,7 +257,7 @@
haveReadOneFile = False;
curInputFileName = inputFileNames;
}
-
+
// Open the current input file as a 'byte-stream file source':
ByteStreamFileSource* fileSource
= ByteStreamFileSource::createNew(*env, *curInputFileName);
@@ -270,7 +270,7 @@
return;
}
haveReadOneFile = True;
-
+
// We must demultiplex Audio and Video Elementary Streams
// from the input source:
mpegDemux = MPEG1or2Demux::createNew(*env, fileSource);
|
