Next version, in progress:

- Updated the "WindowsAudioInputDevice" code to make it work better with Unicode.  (Thanks to Jeff Shanab.)

​

2011.11.20:

- Fixed "RTCP.cpp" to not call a "BYE" handler (if one's set) until the very end of the incoming RTCP report handler member

  function.  This is to allow for the (common) possibility of the "BYE" handler causing "this" to get deleted.

  (Thanks to Jeff Shanab for noticing this.)

- Updated some counters used by "MPEG2TransportStreamFramer" from "unsigned long" to "u_int64_t", to better support streams

  that last for several days.  (Thanks to Tim Shackleton for this fix.)

- Modified the "RTSPServer" code to properly handle weird RTSP clients that ask for RAW UDP streaming, but also specify two

  port numbers in the "client_port=" part of the "Transport:" header.

  (Thanks to "homepuh(at)yandex.ru" for this suggestion.)

- Made a small change to the "VP8VideoRTPSource" implementation to reflect a minor change (the addition of the "K" header flag)

  in the most recent IETF Internet-Draft (draft-ietf-payload-vp8-02.txt)

- Added an #ifndef to "Boolean.hh" to deal with a Windows development environment that already defines (not "#define"s)

  "True" and "False".  (Thanks to Jeff Shanab for this suggestion.)

​

2011.11.08:

- Added "VorbisAudioRTPSink" and "VorbisAudioRTPSource" for sending/receiving Vorbos audio RTP streams (based on RFC 5215).

- Added "VP8VideoRTPSink" and "VP8VideoRTPSource" for sending/receiving VP8 video RTP streams.

- Added support for extracting and streaming Vorbis audio tracks from Matroska (including WEBM) files.

- Added support for extracting and streaming VP8 video tracks from Matroska (including WEBM) files.

- Updated the "testOnDemandRTSPServer" and "LIVE555MediaServer" (source-code version only) applications to support streaming

  from ".webm' files.

- Fixed frame durations for data extracted from Matroska tracks that don't have a 'default duration'.

- Fixed a memory leak in "RTSPClient::sendOptionsCmd()".  (Thanks to Gregg Mattinson for reporting this.)

  (Note, however, that "sendOptionsCmd()" - and other functions in the synchronous interface - are currently deprecated.

  Developers should now be using the asynchronous "RTSPClient" interface instead.)

​

2011.11.02:

- Updated the Matroska file parsing code to be more tolerant of extremely large frames (larger than the "StreamParser"

  "BANK_SIZE"), and bogus data - to avoid tripping into parser 'internal errors'.

- Added an #ifndef/#endif around the "NoReuse dummy;" statement in "RTSPServer.cpp".  This allows a developer - at compile time -

  to allow a "RTSPServer" to run even if there is already a server using the same port.  (The only time you should do this is

  if you anticipate restarting a server immediately after it exits, when the old server is still in a TIME_WAIT state, and

  you're sure that no other server will ever be running with the same port number.)

  (Suggestion by Chris Richardson.)

​

2011.10.27:

- Added support for seeking within streams that are demultiplexed from a Matroska file.

- Modified the signatures of the virtual functions "ServerMediaSubsession::seekStream()"

  and "OnDemandServerMediaSubsession::seekStreamSource()" to make the "seekNPT" parameter a "double&" rather than a "double".

  This makes it possible for subclasses to make slight changes to the specified seek position when they implement seeking.

  (This is necessary for seeking within Matroska-demuxed streams, because 'cue points' in Matroska files can be coarse-grain.)

- Made a slight modification to "openRTSP" so that it sets the play duration correctly if the RTSP seek position changes as

  a result of the RTSP "PLAY" command.

- Changed the implementation of "ReorderingPacketBuffer" in "MultiFramedRTPSource" to use a doubly-linked list rather than

  a singly-linked list.  This makes it more efficient to add new packets to the list after a packet loss has occurred

  (while we wait for the 'reordering threshold' to expire).  (As always, most of the "ReorderingPacketBuffer" code gets called

  only if a packet is lost (or reordered) in the input stream; not in the common case where all packets arrive in order.)

  (Thanks to Ralf Schroeder for this suggestion.)

- Made the implementation of the "our_random()" function more robust in case it's called concurrently by multiple threads.

  (Of course, as explained in the FAQ, you should calling LIVE555 code from multiple threads only if each thread uses its own

  "UsageEnvironment" and "TaskScheduler" object.)

  (Thanks to Michael Margold for raising this issue.)

- Fixed the way that you can specify - in the C/C++ preprocessor - that you want to use the system-supplied "random()" and

  "srandom()" functions, rather than our own versions.  You can do this by defining USE_SYSTEM_RANDOM (e.g., in a "config." file).

​

2011.10.18:

- Improved "RTSPServer" support for subdirectories in "rtsp://" URLs (handling this better for non-compliant clients

  that try to do a "SETUP" on agrregate URLs - when there is only a single subsession in the stream).

​

2011.10.09:

- Updated "RTSPServer" to better support multiple "/" characters (i.e., subdirectories) in URL - i.e., improving the modification

  that we made back in version 2011.08.13.

- Added an #include to "RTSPCommon.hh" to overcome a compilation problem that some

  people had when using Borland development tools on Windows.

​

2011.10.05:

- Added support for RTSP/RTP streaming, on demand, from a Matroska ('.mkv') file.  The video, audio, and subtitle tracks (if any)

  within the file are demultiplexed and streamed separately, via RTP.  We currently support H.264 video, AAC, AC3 and MP3 audio,

  and text subtitles.  Sometime in the future, we will also support WEBM ('.webm') Matroska files, with VP8 video and Vorbis audio.

  (We don't currently support seeking within the underlying  Matroska file (to implement RTSP seeking), but hope to do so

  in the future.)

  We also updated the "testOnDemandRTSPServer" demo application and the "LIVE555 Media Server" (currently, just the source-code

  version - not the prebuilt application binary versions) to support streaming from '.mkv' files.

- Added a new "RTPSink" subclass "T140TextRTPSink" to implement the 'text conversation' RTP payload format defined

  by RFC 4103.  (Note, however, that the optional support for 'redundancy' - as described in that RFC - is not yet supported.)

  We had already implemented the *reception* of this RTP payload format (using "SimpleRTPSource").

  We use this payload format to stream 'subtitle' tracks from Matroska files.

- Added a new configuration file "config.macosx-32bit" for generating 32-bit-only binaries for Mac OS X.

  (Thanks to Robert Wolff for contributing this.)

- Added a mechanism for (optionally) setting a callback function on a "RTPSink" - to be called if a send of data over

  the sink's socket fails.  (Contributed by Ralf Globisch.)

- Made a minor change to "MultiFramedRTPSink" to ensure that the "doSpecialFrameHandling()" virtual function gets called

  even when sending RTP packets that contain a 0-length payload.

- Fixed a minor memory leak in "H264VideoRTPSink" (the intermediate 'fragmenter' object wasn't getting deleted properly)

​

2011.09.19:

- We now set RTCP "RR" handlers for "PassiveServerMediaSubsession"s.  As a result, client sessions for multicast streams

  will no longer time out - as long as the clients send RTCP "RR" packets which the server can receive.

- Added a new public member function "setSPSandPPS()" to "H264VideoStreamFramer".  This lets you initialize a 

  "H264VideoStreamFramer" (or a "H264VideoStreamDiscreteFramer") with SPS and PPS NAL units, in case these do not appear

  early (or at all) in the input stream.

- Made some performance improvements to "H264VideoFileServerMediaSubsession" and "MPEG4VideoFileServerMediaSubsession".

- Fixed a bug in "H264VideoRTPSink" that could cause input source processing to not get stopped properly if the sink gets deleted.

​

2011.09.02:

- Updated "MultiFramedRTPSource" to more cleanly handle the situation where the sender's SSRC changes.  We don't yet support

  streams that contain multiple SSRCs, but we can handle streams where the SSRC changes occasionally - e.g., because a multicast

  source has been restarted.  (When the SSRC changes, we don't check the RTP sequence number of the first packet.)

  (Thanks to Dmitriy Vasil'ev for bringing this issue to our attention.)

​

2011.08.22:

- Updated the "RTSPServer" code to handle a rare condition whereby "handleRequestBytes()" gets called recursively.

  (This can happen only when we handle a "DESCRIBE" by re-calling the event loop (e.g., to wait for SDP parameters to

  become available).)  If this happens, we need to make sure that we don't delete the "RTSPClientSession" object until we leave the

  outermost call.  (Thanks to Vadim Kosarev for identifying the problem, and proposing the solution.)

​

2011.08.20:

- Modified the "MediaSession" and "MediaSubsession" classes to make it possible for developers to add support for receiving

  new RTP payload formats, without having to modify the "MediaSession" or "MediaSubsession" code itself.  To do this, developers

  define their own subclasses of "MediaSession" and "MediaSubsession".  For more details, see the comment near the top of

  "liveMedia/include/MediaSession.hh"

​

2011.08.13:

- Updated the "RTSPServer" implementation to allow multiple "/" characters in the "rtsp://" URL.  This allows the 

  "LIVE555 Media Server" to stream from subdirectories.  (Thanks to Mike Williams for this suggestion.)

- Updated the "RTSPClient" code to more cleanly handle our server's (empty) responses to RTSP "GET_PARAMETER" commands.

  (Thanks to Sebastien Escudier for noting this issue.)

- Updated the "StreamParser" code to allow parsers to test whether we've seen EOF on the input stream.  We also modified

  the "H264VideoStreamParser" code to use this, and fix a bug which was causing it to omit the last NAL unit in the input file.

- Updated "JPEGVideoRTPSource" to generate correct (or at least more correct) JPEG headers when the input stream contains just

  one quantization header.

- Added a 'cast' to the "RTSPServerSupportingHTTPStreaming" code to overcome a compilation problem that some people may

  have been seeing.

​

2011.07.21:

- Changed some member variables from "private:" to "protected:", to make subclassing easier.

  (Requested by Mojtaba Nouri and James Stafford.)

- Added a sanity check to the "RTP-over-TCP" handling code in "RTPInterface.cpp" to alleviate a strange condition (perhaps a bug)

  that Jeremy Noring saw.

​

2011.07.18:

- Added a new, optional extension to "RTSPServer" - called "RTSPServerSupportingHTTPStreaming" - that supports streaming

  to iPhones and iPads, using Apple's "HTTP Live Streaming" mechanism.  (For this to work, the files being streamed *must* be

  MPEG Transport Stream files with H.264 video, and must also be 'indexed' (using the same 'index file' mechanism that we use

  for 'trick play').  (The "live555MediaServer" source code has been updated to support this, as an option.)

- Fixed a bug in the various "*Streamer" demo applications that stream from the same input file, in a loop.  We weren't calling

  "stopPlaying()" on the "RTPSink" object at the end of each loop.  This bug directly affected only "testH264VideoStreamFramer",

  but was theoretically an issue for other demo applications as well.  (Thanks to Sergey Kosov for reporting this.)

​

2011.07.15a:

- Removed some experimental new code that had accidentally been left in the "mediaServer" directory.

​

2011.07.15:

- Added a new class "ByteStreamMemoryBufferSource" to encapsulate a (static) memory buffer that's used as a byte stream data source.

- Added a new class "TCPStreamSink" to encapsulate a writable TCP socket.  Being a "MediaSink", you can call "startPlaying()"

  on it, to stream from a "FramedSource" to a TCP socket.  This class could be used to implement a HTTP server, for example.

- Changed the signature of "ServerMediaSubsession::seekStream()" to take an additional "numBytes" return parameter.

  This parameter is set to the number of bytes that will be streamed (or 0 if unlimited, or unknown).

  (This can be used to implement a HTTP server, with a "Content-Length:" response header.)

- Added a new virtual function "getStreamSource(streamToken)" to "ServerMediaSubsession".  This provides an easy way for

  "RTSPServer" subclasses to access the media source for each stream, if desired.

- Removed the classes "HTTPSink" and "MPEG1or2VideoHTTPSink".  They weren't being used anymore.

​

2011.07.08:

- Updated the RTSP server implementation to check the "Content-Length:" header - if present - in incoming requests.  Most commands

  don't include extra data after the <CR><LF><CR><LF>, and thus don't need a "Content-Length:" header.  However, two commands

  - GET_PARAMETER and SET_PARAMETER - include extra data, and thus have a "Content-Length:" header.  We now check this value,

  to make sure that we've received all of the required data for each request.

  (Thanks to Matt Schuckmannn for reporting this issue.)

​

2011.07.06:

- Modified the change to "ByteStreamFileSource" that we made in version 2011.06.12.  We now call "read()" only if we know that

  the underlying file is non-seekable.  Otherwise - if the file is seekable - we call "fread()", as we did prior to 2011.06.12.

  (Reading from the file using "read()" doesn't work properly if we're also seeking within the file.

  This is why Transport Stream 'trick play' got broken by 2011.06.12, but should be working again now.)

​

2011.07.05:

- Updated "MPEG2IndexFromTransportStream" to make "MPEG2TransportStreamIndexer" more robust when it sees PCR timestamps decreasing

  in time (something that shouldn't happen).  (Thanks to Mojtaba Nouri for providing an example Transport Stream file that

  illustrates this.)

- Fixed a bug in "MPEG2IndexFromTransportStream" that could cause the "MPEG2TransportStreamIndexer" utility to hang when it

  saw bogus data appear before a MPEG system code.  (Thanks to Julian Scheel for reporting this issue.)

- Modified the "BasicTaskScheduler" implementation to add "dumySocket" only to "fReadSet", but not to "fWriteSet" or "fExceptionSet".

  This should continue to work around the Windows bug (where "select()" returns an error if no sockets are present in "fReadSet"),

  but without causing "select()" to keep returning unnecessarily (chewing up CPU).  (Thanks to Matt Schuckmann for reporting this.)

- We now make all files in the source code distribution read-only, by default.  Developers can, of course, change the permissions

  of any of these files to make them writeable, but having them be read-only by default helps emphasize that modifying the

  supplied source code is discouraged.  Instead, developers should - if possible - extend the supplied source code using

  C++ subclassing.  For more information, see the FAQ: http://www.live555.com/liveMedia/faq.html#modifying-and-extending

​

2011.06.16:

- Updated "AC3AudioRTPSink" and "AC3AudioRTPSource" to conform to RFC 4184, which defines the RTP payload format for AC-3 audio.

  Previously, we were using a different, incompatible payload format (that had been defined by an initial draft document, but

  which differed from that defined in the final RFC.  Unfortunately, this means that older clients that use our code

  - including existing binary versions of VLC and MPlayer - will no longer be able to play AC-3/RTP streams from our server

  (until they are rebuilt using this newest version of our library).

- Updated "AC3AudioStreamFramer" - and added a new class "AC3AudioFileServerMediaSubsession" - to support streaming from an AC-3

  audio file.  We also added this support to the "testOnDemandRTSPServer" demo application, and to the "LIVE555 Media Server"

  (currently just the source code; not the prebuilt binary versions).

  (Thanks to Ashwani Kathuria for suggesting this.)

- Moved the definition of "class StreamState" from "OnDemandServerMediaSubsession.cpp" to "OnDemandServerMediaSubsession.hh".

  Even though "class StreamState" is used only internally, in the implementation of "OnDemandServerMediaSubsession", moving its

  definition to the header file makes it possible for subclasses of "OnDemandServerMediaSubsession" to access it.

​

2011.06.14a:

- Removed some junk (debugging) files that had accidentally been left in the "testProgs/" directory.

​

2011.06.14:

- Fixed a race condition in "H264VideoFileServerMediaSubsession" and "MPEG4VideoFileServerMediaSubsession" that could have been

  triggered when two separate clients tried to stream the same file concurrently.  (Thanks to Bruno Abreu for reporting this.)

- Fixed a minor bug in "Authenticator::operator=()": we were not freeing the LHS object's memory before doing the

  assignment.  (Thanks to Sawan Das for noting this.)

- Updated "config.iphoneos" to change LIBRARY_LINK from "ar cr " to "libtool -s -o ", just as we did for "config.macosx" in the

  previous release.

​

2011.06.12:

- Updated the RTCP implementation to (if we're a SSM source) reflect incoming RTCP packets back to the multicast group

  *only* if the packet originated from another host.  This means that we won't reflect packets that come from other

  processes on our own host.  The reason for this is that we can't reliably distinguish such packets from packets that were

  actually looped back from us.  (We must not 'reflect' such packets, otherwise we risk a packet storm.)

- Updated the "ByteStreamFileSource" code to make the underlying socket non-blocking.  We also read data using "read()"

  rather than "fread()", to eliminate blocking (if READ_FROM_FILES_SYNCHRONOUSLY is not defined).  (This change should

  affect only reading from pipes, not regular files.)  (Thanks to Tyson Wiser for noting this issue.) 

- Updated the "DelayQueue" destructor to make sure that any remaining delay queue entries are destroyed.

  (Thanks to "Mr. Xiao" and Luke Kucalaba for (independently) reporting this issue.)

- Updated "config.macosx" to change LIBRARY_LINK from "ar cr " to "libtool -s -o ", because someone reported that "ar"

  didn't work for them.

- Added support for receiving the "audio/EAC3" RTP payload format (which is handled the same way as "audio/AC3").

  (Thanks to Gregory McGarry for the patch.)

​

2011.05.25:

- Fixed a bug in "RTSPClient" which was causing clients to crash if servers returned a 401 error code, but without a

  "WWW-Authenticate:" header.  (Thanks to Sebastien Escudier for reporting this.)

​

2011.03.14:

- Updated the "MPEG2TransportFileServerMediaSubsession" to use the "streamDuration" parameter (if >0.0) to limit

  the number of Transport Packets that are streamed from the source file.  (This happens only if the file is indexed.)

  This allows the server to implement finite RTSP ranges for such streams.

- Fixed a minor bug in "OnDemandServerMediaSubsession" if "reuseFirstSource" is set.  (Thanks to Andreas Gaer for noting this.)

​

2011.03.06:

- Changed the implementation of "RTSPClient::teardownMediaSession()" and "RTSPClient::teardownMediaSubsession()"

  (i.e., in the old, now-deprecated synchronous "RTSPClient" interface) to not wait for, or handle, a response to the

  RTSP "TEARDOWN" command.  This avoids the client blocking indefinitely if the server (or its connection) happens to die before

  the RTSP response gets sent.

  (Note, however: All RTSP client applications should be updated to use the new, asynchronous interface.  The old synchronous

  interface will not be supported indefinitely.)

​

2011.03.05:

- Changed the signature of the "OnDemandServerMediaSubsession::seekStreamSource()"

  virtual function to add a "streamDuration" field.  This field (if >0.0) tells

  the implementation how much data to stream before ending with EOF.

  Our RTSPServer implementation now uses this to implement finite RTSP ranges

  (as specified in the "Range:" header).  As with seeking, only some codecs currently

  implement this: MP3 audio, WAV audio, and DV video.

- We now support streaming from IMA ADPCM ("DVI4") WAV files.

- When streaming MP3 audio files or indexed MPEG Transport Stream Files, we now compute a more accurate bitrate estimate

  (for use in computing RTCP packet frequencies).

​

2011.01.24:

- Fixed a bug that was accidentally introduced in version 2011.01.10, and which crashes VLC (when VLC is used to play a

  "rtsp://" URL).  To avoid the bug, VLC should therefore use this version or later.

​

2011.01.21:

- Fixed a bug in "H264VideoStreamFramer" that was introduced in the last release.

  (This affected the "testH264VideoToTransportStream" demo application.)  (Thanks to Dunling Li for reporting this.)

- Fixed a minor syntax bug in "H264VideoStreamFramer.cpp" which (fortunately) had not been an actual problem.

  (Thanks to Guillaume Le Neindre for the report.)

​

2011.01.20:

- Added a new demo application "testH264VideoToTransportStream", which takes a H.264 Video input file (named "in.264"),

  and converts it to a Transport Stream file (named "out.ts").

  (Note that for this conversion to work properly, the input H.264 file must contain sufficient timing information for us to be

  able to deduce the frame rate.)

  (Thanks to Dunling Li for this suggestion.)

- Changed our RTSP server implementation so that the "SET_PARAMETER" command succeeds (but does nothing) by default.

  (Subclasses can redefine its behavior, if necessary.)  We also now support the special "*" URL, which designates an

  operation on the entire server.  (The only commands for which this is allowed are "OPTIONS", "GET_PARAMETER" and "SET_PARAMETER".)

  (Thanks to Jeremy Noring for noting that some clients need this.)

​

2011.01.19:

- Fixed a bug in "OnDemandServerMediaSubsession" that was causing unicast RTSP/RTP

  servers to sometimes omit sending the very first packet of a stream.  (This seemed

  to occur only for servers running on Windows.)

​

2011.01.10:

- Updated "RTSPClient" so that the new asynchronous interface can handle RTSP URLs that contain a "<username>:<password>@"

  before the server host name/address.  This had been supported in the old synchronous interface, but when we implemented the

  new asynchronous interface, we had forgotten to implement it there as well.

- Made a change to the implementation (but not the default behavior) of "RTSPServer" to allow for the possibility of a subclass

  implementing HTTP streaming (using the same HTTP port that we use to support RTSP-over-HTTP tunneling).

​

2011.01.06:

- We added support for receiving the RTP payload formats "audio/L20", "audio/L24" and "audio/DAT12", as defined in RFC 3190.

  We also recognize the new "fmtp:"-line parameters "emphasis" and "channel-order" that this RFC also defines.

  (Thanks to Maciej Szlapka for noting that these payload formats had not previously been supported.)

​

2011.01.05:

- Made a small change to the sanity check introduced to "MultiFramedRTPSink" and "BasicUDPSink" in version 2010.12.05 to

  eliminate the possibility of small extra delays accumulating over time.  (Thanks to Warren Young for raising this issue.)

- Made a small change to the 'magic values' that are used to implement the "esds" atom for MPEG-4 video in "QuickTimeFileSink".

  (I have no idea what these values are supposed to do, but Stu Tomlinson reports that the new values work better for him.)

​

2010.12.31:

- We now support 'trick play' operations (seeking, fast-forward, reverse play) on MPEG Transport Stream files that contain

  H.264 video (rather than just MPEG-2 video, as previously).  To support this, the index file format has been extended in a

  backwards-compatible way, so that existing index files (for MPEG-2 video Transport Streams) will continue to work as before.

  New versions of the "MPEG2TransportStreamIndexer" and "testMPEG2TransportStreamTrickPlay" utilities - and the

  "live555MediaServer" - have also been released.

- Fixed a bug in the definition of the "profile_level_id" field in "H264VideoRTPSink".  (Thanks to Geoff Cleary for noting this.)

- Change the parsing of RTSP "Range:" headers to allow parameters of the form "clock=" or"smtpe=".  (However, we currently don't

  interpret parameters of this form; instead, we just ignore them.)

  (Thanks to Sebastien Escudier for this suggestion.)

- Fixed "openRTSP" to properly reset internal state before repeating the playing of a stream (if the '-c' (play continuously))

  option is used.  (Thanks to Anon Sricharoenchai for noting this.)

​

2010.12.14:

- Oops - there was a serious bug in the 'event trigger' implementation in the previous release.

  IMPORTANT: You should upgrade to this new version if you plan to use the 'event trigger' mechanism!

  (Thanks to "P.J." for noticing the bug in the previous version.)

​

2010.12.11:

- Added a new 'event trigger' mechanism to "TaskScheduler".  This makes it possible

  to define new events that can be handled within the event loop.  Unlike other

  library functions, events can be 'triggered' (i.e., fired) from a separate thread.

  This makes it easier to implement input devices than the old 'watchVariable'

  mechanism (which remains).

  Also, the "DeviceSource" class - which is a model for how to implement an input

  device class - has been significantly improved.  It now uses the new 'event trigger'

  mechanism.  Also, more of it has been implemented, making it clearer where new code

  needs to be written. 

​

2010.12.05:

- Significantly improved our support for streaming H.264 video.  In particular, "H264VideoStreamFramer"

  and "H264VideoStreamDiscreteFramer" (a new class) act like their corresponding MPEG4 versions: "H264VideoStreamFramer" reads

  a H.264 Video Elementary Stream byte stream (e.g., from a file), and "H264VideoStreamDiscreteFramer" reads discrete H.264 video

  NAL units (i.e., one-at-a-time), e.g., from a H.264 video encoder.  (Note that developers no longer need to subclass

  "H264VideoStreamFramer".)  We also added a new demo application - "testH264VideoStreamer" - for streaming from

  a H.264 Elementary Stream Video file via multicast.  "testOnDemandRTSPServer" and "live555MediaServer" were also updated to

  stream H.264 Video Elementary Stream files.

- Added a sanity check to "MultiFramedRTPSink" and "BasicUDPSink" to allow for the possibility of the system clock jumping ahead

  in time, and thereby messing up the calculation of how long to wait before sending the next packet.

  (Thanks to Anders Chen for noting this issue.)

- Fixed bugs in "AMRAudioRTPSource" and "QCELPAudioRTPSource" that might sometimes cause an event handler to try to reference

  objects that had already been deleted.  (Thanks to David Cailliere for detecting the problem with "AMRAudioRTPSource";

  it turns out that "QCELPAudioRTPSource" had the same problem.)

​

2010.11.17:

- Added new a member function "setAuthenticationDatabase()" to "RTSPServer".  This allows a server's manager to change

  (or disable) authentication at runtime.  (Thanks to Jeremy Norling for suggesting this functionality.)

​

2010.11.10:

- Fixed "openRTSP" to eliminate a recursive call to "shutdown()" if we receive a RTCP "BYE" while in the middle of doing

  a RTSP "TEARDOWN" command.  (Thanks to David Cailliere for noting this issue.)

​

2010.11.09:

- The previous release had accidentally included some experimental changes to "H264VideoStreamFramer" that were not (yet)

  intended to see the light of day.

​

2010.11.08:

- Fixed a minor problem with RTSP-over-HTTP support in "RTSPClient" that was causing some servers to complain due to the

  "CSeq:" header value not being incremented properly.  (Thanks to Kamil Dobkowski for the report.)

​

2010.11.04:

- Backed out and corrected the change to "RTSPClient" that we made in version 2010.10.22.  It turns out that existing RTSP-over-HTTP

  servers that return "401 Unauthorized" in response to the HTTP "GET" *do*, in fact, want clients to resend the "GET" command

  (with a filled-in "Authorization:" header).  However, at least one such server out there closes the TCP connection after

  sending back the "401 Unauthorized" response, so we need to send the second "GET" command using a new TCP connection.

  (Thanks to Kamil Dobkowski for providing access to a server that illustrated this problem.)

​

2010.10.28:

- Updated "JPEGVideoRTPSource" amd "JPEGVideoRTPSink" to support optional "Restart Marker Headers" in the outgoing RTP packet.

  (We already supported such headers in *incoming* JPEG/RTP packets ("JPEGVideoRTPSource").)

  If "type()" (defined by the "JPEGVideoSource" subclass) returns a value in [64,127], then the "JPEGVideoSource" subclass must

  also redefine "restartInterval()" to return a non-zero value.

  (Thanks to Cristiano Belloni for suggesting this addition.)

​

2010.10.23a:

- Fixed a bug in the way that "RTSPClient" generates ephemeral RTP/RTCP port number pairs for use when receiving unicast

  RTSP/RTP streams.  The RTP (even) port was guaranteed not to be in use elsewhere on the same host, but that guarantee was not

  necessarily true for the RTCP (next, i.e. odd) port.  This version fixes this.

​

2010.10.23:

- Our server implementation of "RTSP-over-HTTP tunneling" now allows for the possibility of receiving data from the 

  (Base-64-encoded) initial RTSP command at the same time that we receive the HTTP "POST" command.  (This is a possibility because

  the client does not wait for a response to the "POST" - because there isn't one.)

​

2010.10.22:

- Made a minor modification to the way that "RTSPClient" does "RTSP-over-HTTP tunneling".  If the initial "GET" request

  returns "401 Unauthorized", then we don't resend it (with an "Authorization:" header).

  Instead, we continue, as normal, sending the subsequent HTTP "POST" (with an "Authorization:" header).

  It's not clear (from the limited documentation on "RTSP-over-HTTP tunneling") which behavior is 'correct', but at least some

  servers seem to handle the new approach better.

​

2010.10.20:

- Made the implementation of 'RTSP-over-HTTP tunneling' in the RTSP server more tolerant of certain buggy clients.

  (Thanks to Cristiano Belloni for this suggestion.)

​

2010.10.15:

- Added server support for RTSP/RTP-over-HTTP tunneling.  We added a new "RTSPServer" member function

      Boolean setUpTunnelingOverHTTP(Port httpPort);

  to (attempt to) set up tunneling on the specified HTTP port number.

  We also updated the "testOnDemandRTSPServer" and "live555MediaServer" applications to try to set up HTTP tunneling

  on port 80, then port 8000, and then finally port 8080.

​

2010.10.06:

- Made a small change to "RTSPClient" to make it better handle RTSP servers that

  erroneously do not include "CSeq:" lines in their responses.

  (Thanks to Sebastien Escudier for this suggestion.)

- Made a minor update to "RTSPClient" to better handle any error in the HTTP "GET" command (if RTSP-over-HTTP tunneling

  is being used).

- Updated the "timestampString()" function in "groupsock/GroupsockHelper.cpp" to allow for the possibility of "ctime()" returning NULL.

- Removed the (never fully implemented) "RTSPOverHTTPServer" class.

  Instead, RTSP-over-HTTP tunneling will be implemented within the existing "RTSPServer"

  code, as that seems to be easier than using a separate class.

​

2010.09.25:

- Disabled the code for "Groupsock::multicastSendOnly()", because it was apparently causing SSM multicast packets to not be

  received by other applications on the same host (at least, for some systems).

​

2010.09.23a:

- Backed out the "RTSPServer" change that was in the previous release, because of a report that it doesn't work.

​

2010.09.23:

- Updated "H2564VideoFileSink" to take an optional "sprop parameter string" parameter as input.  If present, this string is

  decoded, and the resulting data (SPS/PPS NAL units) is prepended to the file.

  Also, updated "openRTSP" to call "MediaSubsession::fmtp_spropparametersets()", and pass this string when creating a

  "H2564VideoFileSink".

- Updated "RTSPServer" to ensure that we transmit RTP and RTCP packets over the same interface that is used by the requesting client

  (in case the server is multi-homed).  (Thanks to David Stegbauer for this suggestion.)

​

2010.09.22:

- Added a new virtual function "internalError()" to "UsageEnvironment" and "TaskScheduler".  This function is called

  a small number of times within the library code when an unexepected, 'should not occur'-type error condition occurs.

  The default implementation of this function calls "abort()", but subclasses can, of course, redefine this, if desired.

- Updated "RTSPClient::parseTransportParams()" to accept either "port=<num1>-<num2>" or "port=<num>".

  (The RTSP specification is ambiguous about whether or not both are allowed.)

- Made sure that calls to "ntohl()" and "htonl()" take (and return) "u_int32_t" rather than unsigned", for improved portability.

  (Thanks to Anon Sricharoenchai for this suggestion.)

​

2010.09.10:

- Added a sanity check to "MediaSubsession::getNormalPlayTime()" so that we ignore any old packet that might

  mess up the initial computation of the presentationTime-to-NPT offset following the arrival of new "RTP-Info" data.

  This should never happen with our servers (though John Shao Fa - who proposed this check - claims to be seeing it).

  We add the check, however, to protect against other servers that might be messing up.

​

2010.09.03:

- Made sure that various calls to "TaskScheduler::scheduleDelayedTask()" are called with a "int64_t" 'delay in microseconds'

  parameter, to avoid possible integer underflow if the system clock gets set back in time.

  (Thanks to Sebastien Escudier for reporting this problem.)

​

2010.08.31:

- Fixed some problems in the way that we implement RTP/RTCP-over-TCP streams that were showing up when we have more than one

  such stream sharing the same input source ("reuseFirstSource" == True).  (Thanks to John Tam for reporting this.)

​

2010.08.22:

- Updated the "DarwinInjector" class to use the new, asynchronous "RTSPClient" interface.

​

2010.07.29:

- Fixed a bug in RTP-over-TCP that was sometimes causing a socket to (attempt to be) read after it had already been closed.

  (Thanks to Wilton @ Invent Vision for reporting this bug.)

- Fixed some bugs in the way that RTP reception statistics were being generated and reported.  (Thanks to David Cailliere.)

- Added a new config file "config.linux-64bit" that should work better than "config.linux" for building for 64-bit Linux systems.

​

2010.07.13:

- Updated the "openRTSP" and "playSIP" applications to use the new asynchronous "RTSPClient" interface.

- Made the "readSocket()" routine more bullet-proof against Windows brain damage that was apparently causing problems with

  Windows RTSP clients that do RTP-over-TCP.

  (Thanks to Kamil Dobkowski for this suggestion.)

​

2010.07.07:

- Fixed a problem with RTP-over-TCP reception that was causing some RTP-over-TCP streams to not be received properly.

  All RTSP client applications that can request/receive RTP-over-TCP should upgrade to this new version.

  (Thanks to Kamil Dobkowski for providing a server that we could use to debug the problem.)

- Reinstated the functionality of the "forceMulticastOnUnspecified" flag in the "RTSPClient" "SETUP" command.

  (This functionality had temporarily been disabled, because I wasn't sure that it was still used by anyone.

  But apparently it is.)

​

2010.06.22:

- Fixed another bug in the new "RTSPClient" implementation (thanks to Sebastien Escudier); we were not recognizing

  "npt=now-" in "Range:" headers.

- Fixed a bug in "openRTSP" (reported by Kenneth Ljungh).  We were closing output files before issuing a RTSP "TEARDOWN" command.

  This problem was exposed only with the new "RTSPClient" implementation, because it reenters the event loop (thereby possibly

  handling incoming RTP packets) after the sending of the "TEARDOWN", before it gets handled and responded to.

  We have now fixed the "openRTSP" code to not close output files until after the RTSP "TEARDOWN" has been completed.

​

2010.06.18:

- Fixed a bug in the new "RTSPClient" implementation (reported by Sebastien Escudier) - that was causing problems

  when streaming from some RTSP-enabled video cameras.

​

2010.06.16a:

- Made "RTSPClient"s parsing of "Transport:" headers more bullet-proof against those sent by some servers

  (including, apparently, some recent versions of VLC's built-in RTSP server) that include a "client_port=" field,

  but no "server_port=" field.  (In this case, we assume that the server port is the same as the client port.)

​

2010.06.16:

- Fixed another bug in the new "RTSPClient" implementation of "GET_PARAMETER".

  (Thanks to Denis Charmet for reporting this.)

- Made "RTSPServer::RTSPClientSession::handleAlternativeRequestByte1()" more bullet-proof against an alleged bug noted by 

  Jeremy Noring.  (The actual bug hasn't been located, but this change apparently stops it from causing problems.)

​

2010.06.15:

- Removed the "blockUntilReadable()" function, because it's no longer needed, now that all socket reads are asynchronous.

  This means that "readSocket()" no longer calls "select()", which should improve the performance of applications.

- Improved the processing of RTP/RTCP/RTSP-over-TCP (possibly fixing some bugs in the process).

- Fixed bugs in the new "RTSPClient" implementation of "GET_PARAMETER" and "SET_PARAMETER".

  (Thanks to Denis Charmet for reporting this.)

​

2010.06.11:

- Fixed some potential memory access errors in the previous version of "RTSPClient".

- Removed code from "RTSPClient" that was supposed to deal with buggy (Microsoft?) servers, but which in reality was causing

  clients to not know the duration of time-bounded streams properly.

- Modified the new RTSP client implementation of "RTSP-over-HTTP" to make it more consistent with the old implementation.

  (Doug Porter reported a problem with the implementation in release 2010.05.29.)

- "AMRAudioSource" (and therefore its subclasses also) were not redefining the "MIMEtype()" virtual function.  Fixed this.

  (Thanks to Adam Mich for reporting this.)

- Fixed a minor problem that was causing some "UsageEnvironment"-allocated tables not to get deleted under some circumstances.

​

2010.05.29:

- Made a major change to the interface and implementation of "RTSPClient", to make it perform operations asynchronously

  - i.e., without blocking.  Each operation that performs a RTSP command now takes a 'response handler' function as parameter.

  The operation returns immediately, but the 'response handler' function will get called later, when a response arrives

  (or if an error occurs).

  For backwards compatibility, the old 'synchronous' interface still remains, and is implemented using the new 'asynchronous'

  interface.  (However, the old 'synchronous' interface should not be used for new applications, and it will likely be removed

  in a few months.)

  For now, the demo application "openRTSP" still uses the old interface, but will be rewritten over the next few weeks to use the

  new "RTSPClient" interface.

- Modified the "TaskScheduler" definition yet again, because it was too complicated.  There's now just one virtual function

  for setting/clearing socket handlers:

  "virtual void setBackgroundHandling(int socketNum, int conditionSet, BackgroundHandlerProc* handlerProc, void* clientData) = 0;"

  where "conditionSet" is a combination of SOCKET_READABLE, SOCKET_WRITABLE, and SOCKET_EXCEPTION.

  For backwards compatibility, the existing "turnOnBackgroundReadHandling()" and "turnOffBackgroundReadHandling()" functions

  are retained, but are now implemented using "setBackgroundHandling()".

  See "UsageEnvironment/include/UsageEnvironment.hh" for details.

​

2010.05.28:

- Updated the previous revision of "TaskScheduler" to add two more new virtual functions

  "turnOnBackgroundExceptionHandling()" and "turnOffBackgroundExceptionHandling()",

  because it turns out that Windows needs them for checking on a failed remote "connect()"

  (behaving differently from Posix, which requires only select()ing the socket for

  writing, not an error).

​

2010.05.27:

- Updated "TaskScheduler" (and our supplied example subclass "BasicTaskScheduler") to include new virtual functions

  "turnOnBackgroundWriteHandling()" and "turnOffBackgoundWriteHandling()".  These are the same as the "Read" versions,

  except that they are for checking when a socket becomes writable.  (They will be used to implement the new, upcoming

  'asynchronous' implementation of "RTSPClient".)

  We also modified the signature of "setResultErrMsg()" to take an optional "err" parameter.  (If  non-zero, this will be used

  instead of "getErrno()".) 

- Added a "config.iphoneos" (for cross-compiling on a Mac for the iPhone).  (Thanks to Mark Vasilkov.)

​

2010.04.09:

- Fixed a bug in the way that "AMRAudioRTPSource" reimplements "hasBeenSynchronizedUsingRTCP()".

  (Thanks to Ralf Globisch for reporting this bug.)

- Increased "SegmentQueueSize" in "MP3ADU.cpp" from 10 to 20, to allow for some unusually-VBR MP3 files that neeed this.

  (Thanks to Dirk Raffel for reporting this problem.)

- Changed the implementation of "parseRangeHeader()" in "RTSPCommon.cpp", in preparation for upcoming changes to "RTSPClient".

​

2010.04.01:

- Fixed the "openRTSP" code so that it no longer restricts the play time duration (using the "-d" option), because there is no

  no longer any practical limit on the duration that gets passed to "TaskScheduler::scheduleDelayedTask()"

  (because it is microseconds in 64 bits).

  (Thanks to Brad Thomas for this suggestion.)

- Removed some obsolete (and unused) code from the "RTSPClient" implementation, in preparation for a major reworking of this code.

​

2010.03.16:

- Made more improvements to "QuickTimeFileSink".  (Thanks to Wolfgang Breyha.)

​

2010.03.15:

- Fixed a bug in the Transport Stream file indexing mechanism that was preventing index files from properly being generated

  for some Transport Stream files.  (We sometimes weren't analyzing Program Association Tables correctly.)

- Removed the hack that we introduced in version 2009.11.27, because we have now fixed the bug that made that necessary.

- Removed the "RTPOverTCP_OK" hack from "RTPInterface.cpp", because it's no longer needed.

​

2010.03.14:

- Fixed the RTSP server implementation so that - when streaming via RTP-over-TCP - it can now handle RTSP commands after the

  first "PLAY".  (This fixes a long-standing bug.)

- Updated "QuickTimeFileSink" to support writing files larger than 2GB.  Also fixed a bug involving "esds" atoms for MPEG-4.

  (Thanks to Wolfgang Breyha for this patch.)

- Fixed a bug in the way that RTCP "SDES" items are constructed.  (Thanks to Jeremy Norling for bringing this to our attention.)

- Added a sanity check to the string version of "BasicUsageEnvironment::operator<<()", to check for a NULL string input.

  (Thanks to Paulo Rogerio Panhoto for this suggestion.)

​

2010.03.08:

- Fixed a rare bug that could sometimes cause RTSP clients to reuse the same (ephemeral) port number.

  (Thanks to Andro Debevere for reporting this.)

- Made changes to "config.armlinux", suggested by Rafael Carre.

- Removed a "|| defined(__CYGWIN32__)" test from "groupsock/GroupsockHelper.cpp", because it's no longer needed.

  (Cygwin has apparently now fixed the bug that made that necessary.)

  (Thanks to Tatterdemalian Anzulovic for reporting this.)

​

2010.02.10:

- Fixed a possible 1-byte overflow in "RTSPClient::setupMediaSubsession()".  (Thanks to Martin Bonnin for reporting this.)

- Make "parseCLine()" a bit more robust against certain technically illegal, but still parseable SDP "c=" lines.

  (Thanks to Aurelien Nephtali for this suggestion.)

​

2010.02.09:

- Fixed the way that "RTSPClient" operations handle the "timeout" parameter.

  We now make the socket blocking afterwards (having made it non-blocking beforehand).

  (Thanks to Jeremy Noring for this patch.)

​

2010.01.22:

- Changed the implementation of "Socket::changePort()" once again, beause it was reported that the change made in version

  2010.01.15 didn't work.  Note, however, that this change also introduces a new function

         virtual void moveSocketHandling(int oldSocketNum, int newSocketNum) = 0;

  to the "TaskScheduler" API.  We have implemented this for "BasicTaskScheduler".  If you implement your own subclass of

  "TaskScheduler", then you will need to implement this new function as well.

​

2010.01.16:

- Added ""#include <stdio.h>" to "groupsock/NetInterface.cpp".  This was tripping up some people.

  (Thanks to Ben Hutchings for noticing this.)

​

2010.01.15:

- Changed the implementation of "Socket::changePort()" to call "bind()" directly, rather than closing the socket and reopening a

  new one.  That old implementation sometimes caused the socket number to change, which messed up our event loop.

  (Thanks to Rob Krakora for noticing this.)

- Improved the implementation of DV video streaming (introduced in the previous release) - in particular,

  to implement "DVVideoRTPSource".

​

2010.01.13:

- Added support for streaming/receiving DV video via RTP (using the RTP payload format defined in RFC 3189.

  (Thanks to Ben Hutchings for his help.) 

​

2010.01.11:

- Increased the default "FileSink" buffer size in "openRTSP" from 20000 to 100000, because the previous limit was being bumped into

  too many times.

- Added #include "GroupsockHelper.hh" to "DVVideoStreamFramer.cpp", because it defines "gettimeofday()".

​

2010.01.09:

- In the various demo applications (in "testProgs/"), we changed the calls to "abort()" back to "exit(1)", because we want

  "abort()" to be used only for serious (unexpected) errors; not things like bad command-line arguments.

- Updated the RTSP server implementation to make session ids random (and 8 (hex) digits long), to conform to the RTSP specification.

  (Thanks to Guy Bonneau for this suggestion.)

​

2010.01.07:

- Fixed a bug in "parseGeneralConfigStr()" (it was mistakenly returning an extra byte).

- A few places in the code we called "exit()" if we encountered an unexpected error.  By request, these have now been changed to

  "abort()". 

- Made the "fSDPLines" member variable (in both "OnDemandServerMediaSubsession" and "PassiveServerMediaSubsession") "protected"

  rather than "private".  This allows subclasses to - if they wish - delete and NULL "fSDPLines" each time a client connects, in

  case the stream's SDP parameters have changed.

​

2009.11.27:

- Temporarily hacked the RTSP server code to not include "GET_PARAMETER" as an available option in the response to the "OPTIONS"

  command.  This will stop VLC from using "GET_PARAMETER" as a client 'liveness' indicator.  This is something that we don't

  need, because we already use clients' RTCP "RR" packets as a 'liveness' indicator, and because sending "GET_PARAMETER" over a

  RTP-over-TCP session causes problems with our current implementation.  (This hack will be removed when we fix the problem with

  RTP-over-TCP.)

​

2009.11.12:

- Made "BasicTaskScheduler::DelayQueue.cpp" more robust against an apparent compiler error on some platforms (e.g., Mac OS X)

  that was leading to "select() fails: Invalid argument" errors.

- Added a new compilation configuration file: "config.avr32-linux".  (Thanks to Jorge Munoz Castiner.)

- Made some minor changes to "groupsock/GroupsockHelper.cpp", "liveMedia/InputFile.cpp" and "liveMedia/OutputFile.cpp"

  to make these files compile (or compile better) for WinCE.  (Thanks to Pierre Ynard for these fixes.)

​

2009.09.28:

- Added an optional "timeout" parameter to "DarwinInjector::setDestination()".  (Suggested by Jeremy Noring.)

- Added support for receiving the "text/T140" RTP payload format.  (Suggested by Pierre Ynard.)

​

2009.09.21:

- Added patches proposed by Ben Hutchings to (i) improve the top-level Makefile; (ii) build better static libraries (on

  Linux, FreeBSD, and Mac OS X); (iii) have RTSP servers set "b=AS:" lines in their SDP (and set an appropriate socket send buffer

  size), and (iv) have RTSP (and other SDP) clients interpret the "b=AS:" line (if present) to set the socket receive buffer.

- Changed two source files in the "groupsock" project to use the (current standard) <sstream> #include file.

- Increased RTSP_PARAM_STRING_MAX to 200 (request by Stuart Rawling)

​

2009.09.04:

- Fixed "BasicTaskScheduler"s workaround to a Windows bug (thanks to Vityusha Vinokurov).

- Fixed "DelayQueue::synchronize()" to allow for the possibility of the system clock being reset back in time.

  (Thanks to Sebastien Escudier for pointing out this issue.)

- Made "H264VideoRTPSink::auxSDPLine()" "protected:" rather than "private:" (following a request from Stuart Rawling)

​

2009.07.28:

- Updated "QuickTimeFileSink" to add a "stss" atom for video streams, following a suggestion by Gerardo Ares.

  (At present we just 'guess' which video 'samples' (frames) are 'key frames', so this might not work properly on some

  video streams.)

- Modified the "config.uClinux" configuration file, following a suggestion by Chetan Raj.

- Changed "RTSPClient"s implementation of the RTSP "TEARDOWN" command to always act as if the command succeeded, regardless of

  the actual response from the server (because, from the client's point of view, the session has ended).

  (This overcomes a potential memory leak, pointer out by Stuart Rawling.)

​

2009.07.09:

- Modified the RTSP server implementation to - for streams where there is a known duration - always include a range end time

  in the RTSP "PLAY" response, even if the client did not specify one in the "PLAY" request.  This allows VLC's client

  'trick play' to (mostly) work.

- Updated "MediaSession::initiate()" to eliminate a possible memory leak if we get an error in socket creation.

  (Thanks to Denis Charmet.)

- Made a minor change to "MultiFramedRTPSink" to make monitoring/debugging easier.  (Thanks to Guy Bonneau.)

- Begun adding support for DV video.  However, this implementation is still incomplete.  DO NOT USE IT!

​

2009.06.02:

- Updated the MPEG Transport Stream multiplexor implementation to allow for H.264 video.  (Thanks to Massimo Zito.)

- Updated "MultiFramedRTPSink" to allow for subclasses for RTP payload formats (such as DV, coming soon) that impose

  a granularity on RTP fragment sizes.

​

2009.04.20:

- Fixed "BasicUsageEnvironment::getErrno()" to always (under Windows) return 

  "WSAGetLastError()" (and to just ignore the "errno" variable>.

  Also fixed a few places in the code where we were still using "errno"

  instead of calling "getErrno()"

​

2009.04.07:

- Changed many "char*" variables to "char const*" to eliminate possible compiler warnings.

  (Thanks to Sebastien Escudier for pointing out this issue.)

​

2009.04.06:

- Modified our Windows-only version of "gettimeofday()" so that it now returns times based on the proper epoch.

  (Thanks to Patrick White for this suggestion.)

- Created a new config file for 64-bit Solaris, and renamed the old "config.solaris" file to make it

  clear that it's for 32-bit Solaris only.  (Thanks to Michael Skaastrup.)

- Modified "config.mingw" to add "-DLOCALE_NOT_USED" to the "COMPILE_OPTS =" line.  (The VLC folks seem to want

  this.)

- Made a minor change to some win32-specific code in "RTSPClient.cpp" that the VLC folks seem to like.

  (However, "RTSPClient" is about to undergo a major overhaul (for asynchronous I/O) anyway...)

- Made a small change to "mediaServer/DynamicRTSPServer.cpp" to eliminate compiler warnings on some platforms.

​

2009.03.22:

- Updated "MediaSession::initiate()" to make the selection of ephemeral client

  port numbers (for RTP and RTCP) more robust, and to eliminate the possibility of

  the same port numbers accidentally getting chosen more than once by different

  clients running on the same host.  (Thanks to Guido Marelli and others for helping

  track down this issue.)

​

2009.02.23:

- Changed the way in which 'trick mode' transport streams are generated, so that each

  video I-frame is included in the output Transport Stream no more than once.

  This means that now - for low scale values - the output frame rate, and bit rate,

  will be les than that of the original file.  (Beforehand, we maintained the same

  frame rate, which meant that - because each output frame is an I-frame - the output

  stream's bit rate was always significantly larger than the original.  This caused

  problems for some clients.)

- Changed the 'client liveness' checking code in the "RTSPServer" implementation so that it deletes the

  "RTSPClientSession" object even for multicast streams.  

- Made a minor fix to an error report in "RTSPClient".  (Thanks to Igor Milavec.)

​

2009.02.13:

- Changed calls to the "Locale()" constructor to use an explicit variable name (rather than an anonymous

  declaration), because some compilers weren't handling the anonymous declaration properly.

  (Thanks to Steven Kramer for noticing this.)

- Improved the "RTSPServer" definition and implementation to make it easier to subclass the

  "RTSPServer::RTSPClientSession" class (e.g., to add a custom implementation of the "SET_PARAMETER" command).

  Also added an iterator class for iterating over the "RTSPClientSession" objects within a "RTSPServer".

- Make "MediaSubsession::parseSDPAttribute_framerate()" a little more robust.

  (Thanks to Shawn Van Every for the suggestion.)

- Improved "BasicUsageEnvironment0::reportBackgroundError()" (thanks to Matt Schuckmann)

- Fixed a bug in the "AVIFileSink" code (thanks to Pramod Bhadwat for reporting this).

​

2009.01.26:

- Updated the header file for "rtcp_from_spec.c" to make the call to "exit()" a noop.

  (Thanks to Chris Silvers for raising this issue.)

- Changed the definition of "RTSPClientSession" (in "RTSPServer") to make it protected (and also to make its

  command handler functions protected and virtual), to allow developers to subclass it (to change the

  implementation of some commands).

  (Thanks to Yedidia Amit for suggesting this.)

- Fixed a minor bug in "scheduleNextQOSMeasurement()" (in the "openRTSP" implementation).

  (Thanks to Gabriele De Luca for noticing this.)

​

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.)

- Changed the default "reclamationTestSeconds" parameter (in "RTSPServer" and "DynamicRTSPServer") from 45 seconds

  to 65 seconds, because 60 seconds is the default timeout value expected by (broken) clients that send

  RTSP commands rather than RTCP "RR" reports to indicate liveness.

​

2008.07.25:

- Minor bug fix to "MPEG2TransportStreamFramer" (reset "fTSPCRCount" in "doStopGettingFrames()").

  (Thanks to Qiujian Shu for reporting this bug.)

- Modified the previous changes to "QuickTimeFileSink" and "AVIFileSink" so that the

  "createNew()" functions return NULL if the output file cannot be opened.

​

2008.07.24:

- Changed "MultiFramedRTPSource" to handle the very first packet that we receive as

  if there were packet loss beforehand.  This ensures that we correctly discard any

  initial packets that we receive that occur after the start of a (multi-packet) frame.

- Fixed "QuickTimeFileSink" and "AVIFileSink" to close the output file (if it's not stdout or stderr) in the destructor.

​

2008.07.22:

- Fixed "MPEG2TransportStreamMultiplexor" to use an appropriate stream for our SCR, even if the input contained

  its own Program Stream Map.

- More minor changes to the code to make Erik Hovland's code-checking tools happy.

​

2008.07.06:

- Corrected one of Erik Hovland's memory leak fixes made to the previous revision.

​

2008.07.05:

- Fixed some memory leaks - and several other mostly cosmetic fixes.

  (Thanks to Erik Hovland for noticing these.)

​

2008.06.26:

- Fixed a bug in "openRTSP" that was sometimes causing an incorrect "Range:" header

  to be sent in "PLAY" requests.

​

2008.06.25:

- Fixed a bug in the headers sent for the RTSP "GET_PARAMETER" and "SET_PARAMETER" commands.

  (Thanks to "beilyzhang (at) hotmail" for this report.)

​

2008.06.05:

- Renamed the "ourSourceAddressForMulticast()" function to the less confusing and more explanatory

  "ourIPAddress()".

- The "RTSPClient" change that was made in 2008.04.09 to accommodate multicast

  streams apparently broke (some?) unicast streams.  This has now been fixed.

- Updated "RTSPClient" to handle weird servers that include a "Content-Length:" header

  in the response to the RTSP "SETUP" command.  (If we see such a header, we skip over

  the data referred to in the header.)  (Thanks to Kristian Soerensen for this suggestion.)

- Changed the parsing of lines in "RTSPClient" so that it now accepts \r or \n alone

  at the end of each line.  (\r\n is preferred, but the standard also allows \r or \n)

  (Thanks to David Schueler for this suggestion.)

​

2008.05.12:

- Made a change to "RTCPInterface" to ensure that "turnOffBackgroundReadHandling()" is called on a socket

  as many times as "turnOnBackgroundReadHandling()".  Apparently some versions of Windows get upset if this

  doesn't happen.  (Thanks to Ken Seo for the suggested fix.)

- Fixed a bug in "H263plusVideoStreamParser".  (Thanks to Andrey Latin.)

- Added more sanity checking to "JPEGVideoRTPSink".

​

2008.04.09:

- Corrected a type-related bug in "RTPInterface::handleRead()".  (Thanks to Brain Lai for noting this.)

- Fixed a bug in "RTSPClient" that would prevent RTCP "RR" reports from being sent to

  the correct multicast address when we are receiving a multicast stream. (Thanks to Changjin Liu for noticing this.)

​

2008.04.03:

- Fixed a bug that was introduced in the "2007.12.27" release.  The "timeout" parameter to the call to

  "readSocket" in "SocketDescriptor::tcpReadHandler()" in "RTPInterface.cpp" needed to be initialized

  (to zero) first.  (Thanks to Lodewijk Loos for alerting us to this problem.)

- Added a new config file "config.bfin-linux-uclibc", and renamed "config.bfin_uclinux" as "config.bfin-uclinux".

  (Thanks to Mike Frysinger.)

​

2008.02.08:

- Added a hack (suggested by "Romain") to "MPEG2TransportStreamFramer" to (perhaps)

  produce more accurate per-transport-packet duration estimates for wildly VBR streams.

- Updated "MPEG2TransportStreamMultiplexor" to support the addition of MPEG-4 Audio or Video Elementary Streams.

- Updated "RTSPOverHTTPServer.cpp" (a work in progress) so that it complies properly for some versions of

  Visual Studio on Windows.  (Thanks to Eric Flickner for noting this.)

​

2008.01.19:

- Corrected the "getNormalPlayTime()" function - introduced in the previous release - to allow for 'trick play'

  scale factors other than 1.

​

2008.01.18:

- Added a new member function

    float MediaSubsession::getNormalPlayTime()

  which - given a stream's current presentation time - returns the "Normal Play Time".

  This function is useful for RTSP streams.

- Added support for a "a=control:" URL specified in the SDP description at the session level. 

​

2008.01.04:

- Changed the "RTSPClient" timeout - introduced in the previous release - from

  5s to 30s.

- Added support for setting and returning the RTSP session start time (as well as the end time).

  (This eliminates the need for the "live-starttime.patch" file that VLC was using.)

- Updated some of the system-specific configuration files, to eliminate the need for many of the patches that

  the VLC developers added to their code.

​

2007.12.27:

- Added a 5s timeout to the "RTSPClient" code that checks for RTSP responses.

  This mirrors a change that was already being done to VLC's copy of the code.

  (This is still a short-term fix, until the "RTSPClient" code is rewritten to

  properly use asynchronous I/O, using the event loop.)

- Added a timeout to the "readSocket()" call in "SocketDescriptor::tcpReadHandler()" in "RTPInterface.cpp",

  to handle reading RTP-over-TCP data.  This allows for the possibility of non-cooperative RTSP clients.

  (Thanks to Peter Leese for this suggestion.)  This is probably not a complete solution;

  more thought is needed...

​

2007.12.07:

- Fixed "H2633plusVideoFileServerMediaSubsession" to properly use a dynamic RTP payload type, rather than the

  static type 34 (which is reserved for the now-obsolete old "video/H263" RTP payload format).

​

2007.12.06:

- Updated "JPEGVideoRTPSource" to take optional 'default width' and 'default height' parameters.

  These parameters can be set by fields in the SDP description, and can be used to specifiy unusually

  large frame widths and/or heights.  (Thanks to Andrey Filippov.)

​

2007.11.18:

- Fixed a couple of memory leaks in "DarwinInjector".  (Thanks to Eyal Beit-Halachmi for noting these.)

- Removed old 'backwards compatibility' stuff from "FramedSource".  (Noone should be relying upon this any more.)

- Fixed a bounds-checking error in "parseRTSPRequestString()" caused by an int vs. unsigned problem.

  (Thanks to Luigi Auriemma for noting this.)  

- In "RTSPClient.cpp", fixed a couple of "unsigned" vs. "int" nits.  (Thanks for Brain Lai for noting this.)

​

2007.11.01:

- Several of the options to "openRTSP" have now been changed, with two new options added:

      -s <initial-seek-time>

      -z <scale>

  See the "openRTSP" documentation <http://www.live555.com/openRTSP/> for details.

- Fixed a bug in the way that "MPEG4VideoStreamDiscreteFramer" parses 'config' information (for inclusion in

  the stream's SDP description).  (Thanks to Nicola Bova for helping to identify this bug.)

- Eliminated a potential memory (and socket) leak when allocating server RTP,RTCP socket pairs.

  (Thanks to David Pan for reporting this.)

- Updated "ByteStreamFileSource" to treat a 0-byte file read the same as EOF.

- Ensure that we have reasonable OS buffering for writes on non-blocking sockets.

​

2007.08.03a:

- Removed a debugging printf() that had been left in by mistake.  (Thanks to Massimo Zito for noticing this.)

​

2007.08.03:

- Updated the "RTSPServer" implementation to work better on systems with more than one IP address.

  The server will now include - in its RTSP responses - the IP address on which the corresponding incoming request

  was received.

​

2007.07.25:

- Added some sanity checks to various "MediaSink" subclasses, in case "fSource" is NULL.

  (Thanks to Andrey Kaminsky for noting one of these.)

​

2007.07.10:

- Made sure that "MPEG2TransportStreamFramer"s estimate of the average duration of each Transport Packet

  gets updated correctly after each 'seek' operation.  (Thanks to Massimo Zito for suggesting this.)

- Fixed a bug in "MPEG2TransportStreamFromESSource" that was causing presentation

  timestamps to not be generated correctly.  (Thanks to Massimo Zito for noticing this.)

​

2007.07.01:

- Fixed a bug in "RTSPServer" and "RTSPClient" that would cause problems when streaming

  files whose names have spaces in them.

- Fixed a bug in "OnDemandServerMediaSubsession::deleteStream".  (Thanks to Igor Bukanov.)

- Make the sockets for "RTPSource", "RTCPInstance" and "BasicUDPSource" non-blocking, even though they will be read

  from only asynchronously, when packets arrive.  The reason for this is that, in some OSs, reads on a blocking

  socket can (allegedly) sometimes block, even if the socket was previously reported (e.g., by "select()") as

  having data available.  (This can supposedly happen if the UDP checksum fails, for example.)

  (Thanks to Marc Neuberger for pointing this out.)

- Fixed a bug in "MPEG2TransportFileServerMediaSubsession.cpp" that was causing 2x

  fast-forward to fail.  (Some debugging code had been left in by mistake.)

​

2007.05.24:

- Modified "setupDatagramSocket()" to better handle setting "ReceivingInterfaceAddr" when creating a socket

  to be used to send/receive multicast.

- Made a couple of minor changes to overcome some compilation errors that some people were apparently seeing.

​

2007.05.23:

- The RTSP server now includes the "source=" parameter in the response to a RTSP "SETUP" command.

  This works around a bug in QuickTime Player that would cause it to not send RTCP "RR" packets

  (when playing a unicast stream, and running on Mac OS X).  (Thanks to Dave Singer et al at Apple for tracking

  down this problem.)

- The RTSP server implementation no longer terminates the session if it returns 400 or 405 errors.

​

2007.04.24a:

- Fixed a problem in the new "RTSPOverHTTPServer" code that was causing some people compilation problems.

​

2007.04.24:

- Fixed a typo in "testOnDemandRTSPServer.cpp".  (Thanks to Nils Grundback for noticing this.)

- Modified the signature of "RTSPServer::specialClientAccessCheck()" (defined in the previous revision) to add

  a new "clientAddr" parameter (a "struct sockaddr_in"), to allow special access checking based on clients' IP

  address.

- Make sure that the locale is set to "POSIX" when calling "toupper()" or "tolower()" on human-supplied strings.

  (Thanks to Ismail Doenmez for noting this.)

​

2007.04.20:

- Modified "MPEG2TransportStreamMultiplexor" to set the "discontinuity_indicator" flag for the first

  "adaptation_field" in the output Transport Stream.

- Changed "AMRAudioRTPSource" to more accurately report whether/when an interleaved frame's timestamp has

  been synchronized using RTCP.

  (Thanks to David Bertrand for this patch.)

- Made to a small fix to the previous revision's support for multicast streaming of raw UDP (nonstandard)

- Added a virtual function to "OnDemandServerMediaSubsession" for closing the stream source.

  Subclasses can redefine this, if they wish, to do something smarter.  (Thanks to Igor Bukanov for this patch.)

- Added support to "RTSPServer" for optionally performing special per-client access control, beyond the

  standard Digest Authentication method.  (Thanks to Igor Bukanov for this patch.)

- Made a minor optimization to "Groupsock.cpp".  (Thanks to Maxim Petrov for this suggestion.)

- Added an initial implementation of RTSP-over-HTTP at the server level (it was already supported at the client level).

  (Note: This code has not yet been completed, and doesn't yet work, so don't try to use it.)

​

2007.02.20:

- Updated "RTSPServer" to support multicast streaming of raw UDP streams (nonstandard).

  (Thanks to Aesmund Grammeltvedt for this modification.)

- Made "RTSPClient" a little more robust in case the TCP connection fails.

- Made "RTPSink::rtpmapLine()" virtual (as requested by Andrew Voznytsa).

- Removed an archaic (no longer used) declaration in "groupsock/include/Groupsock.hh".

- Added an optional "allowKasennaProtocol" parameter to "RTSPClient::describeWithPassword()"

  (similar to "RTSPClient::describeURL()").  (Thanks to Igor Bukanov for this suggestion.)s

​

2007.01.17:

- Fixed a bug that would cause the "MPEG2TransportStreamIndexer" application to fail when run on a big-endian

  architecture.

- Added a new class "H264VideoFileSink", that prepends each incoming H.264 NAL unit with the prefix 0x000001,

  before writing it to the file.  Also updated the "openRTSP" code to use this, when receiving H.264/RTP streams.

  (Thanks to Chris Kuiper for contributing this.)

- Added a temporary #ifdef to "GroupsockHelper.cpp" to work around a compliation problem when building

  for Cygwin.

​

2007.01.11:

- Improved the "MPEG2IndexFromTransportStream" class to recogize Transport Stream

  PAT and PMT (tables).  This in turn makes the "MPEG2TransportStreamIndexer"

  utility more robust.

- Fixed a minor bug in "MPEG2TransportStreamFromESSource".

​

2007.01.09:

- Added RTSP server support for 'trick play' operations on MPEG-2 Transport Stream

  files.  This requires the presence of a ".tsx" index file for each ".ts" file.

  (This functionality will shortly be documented on the LIVE555 web site,

  and announced on the "live-devel" mailing list.) 

- Improved the performance of asynchronous file reading in "ByteStreamFileSource".

  (Thanks to Aesmund Grammeltvedt for this suggestion.)

- Added "-D_FILE_OFFSET_BITS=64" to the "COMPILE_OPTS" line for each of the Linux config files, in order to

  alleviate possible problems with I/O on large files.

​

2006.12.31:

- We now implement "ByteStreamFileSource" on Windows using synchronous file reads,

  by default.  I had thought that Windows XP had fixed the problem - present in

  earlier versions of Windows - whereby open files are not treated as select()able

  sockets.  But apparently the problem is still there.

- Added code that will - probably in the next release, very soon - support server

  'trick mode' operations on MPEG-2 Transport Stream files.  At present, we have two

  new applications - in the "testProgs" directory - that help support this:

  "MPEG2TransportStreamIndexer" (to create a special index file for a Transport Stream

  file), and "testMPEG2TransportStreamTrickPlay" (to generate a new Transport Stream

  file that simulates a 'trick play' operation performed on the original Transport

  Stream file).  See my forthcoming post to the "live-devel" mailing list for more

  details.

- Fixed a bug in the Base64 encoding routine.  (Thanks to Sebastian Gracias for reporting this.)

​

2006.12.08:

- Made sure that each TCP socket used by a "RTSPserver" is non-blocking, so that a slow or hanging client

  cannot hang a server.  (Thanks to "jers (at) inwind.it" for this suggestion.)

​

2006.11.16a:

- Oops - fixed a problem with the packaging of the new "mediaServer" code.

​

2006.11.16:

- Included the code for the new "LIVE555 Media Server" application

  (in the "mediaServer" subdirectory).  This is a complete RTSP/RTP server, that

  will serve any appropriately-formatted file that resides in the current directory.

  (More documentation will follow.)

- Updated "QuickTimeFileSink" to support H.264 video in ".mov" or ".mp4" output files.

  (Thanks to David Arnold for this.)

- Fixed a potential reference-count related bug in "OnDemandServerMediaSubsession".  (Thanks to Jers.)

- Changed the "RTSPClient" constructor from "private:" to "protected:".

​

2006.10.27:

- "RTSPClient" now recognizes the "Content-Base:" header in RTSP "DESCRIBE" responses.

  (Thanks to Derk-Jan Hartman for this suggestion.)

- Updated "MultiFramedRTPSource" to avoid excessive recursion (and thus potential

  stack overflow) in some cases when receiving streams that have packet loss.

  (Thanks to Erwin Beckers for raising this as an issue.)

- Updated "RTPSink" and "MultiFramedRTPSink" to ensure that a proper RTP timestamp

  is set in the RTSP "RTP-Info" header that follows a "PAUSE", and that this timestamp

  gets used in the next outgoing RTP packet.  (Thanks to David Bertrand for this suggestion.)

- Changed the special implementation of "gettimeofday()" for Windows to return

  more accurate results.  (Thanks to David Arnold for this change.)

​

2006.10.18a:

- A minor update to the "MultiFramedRTPSink" change made in the previous release.

​

2006.10.18:

- Added support for adding optional RTP padding bytes to "MultiFramedRTPSink".

  (Thanks to Victor Kozhuhov for this suggestion.)

​

2006.10.12a:

- Backed out the change that we made in the 2006.09.18 release: We now don't attempt

  to setsockopt() SO_REUSEPORT and IP_MULTICAST_LOOP on Windows. (Dave Arnold reports

  that doing so leads to UDP packets with TTL=1.)

​

2006.10.12:

- Fixed a rare memory leak in "MultiFramedRTPSource".  (Thanks to David Bertrand for this fix.)

- Made "RTSPServer::lookupServerMediaSession()" virtual, to allow for future

  subclasses of "RTSPServer" that - for example - create and add "ServerMediaSession"s

  dynamically (depending on whether or not the named file exists).

- Added a "rtspURLPrefix()" member function to "RTSPServer".  This returns the

  "rtsp://" URL prefix that's common to all of the server's "rtsp://" URLs.

​

2006.10.07:

- The demo applications (in the "testProgs" directory) that stream MPEG-4 

  Elementary Stream video files now use a file name "test.m4e" rather than "test.m4v".

  This is to avoid confusion, because Apple has coopted the ".m4v" suffix for

  MPEG-4 (audio+video) files, which are different.

- Made "MPEG4VideoStreamFramer" more bullet proof against some non-standard MPEG-4 video data.

  (Thanks to Maxim Petrov for this suggestion.)

​

2006.10.05:

- Fixed a minor bug in "MPEG2TransportStreamFromESSource" (thanks to "jeremy at electrosilk.net")

- Changed some class member function and variables from "private" to "protected", in response to a request

  from David Bertrand.

- Made "DarwinInjector::createNew()" more bullet proof, by giving the "applicationName" parameter

  a default value.

​

2006.09.20:

- Corrected the bug fix to "H264VideoRTPSink" that was made two revisions ago.

​

2006.09.18:

- Fixed a bug in the way that "RTPSource" records the maximum sequence number seen so far.

  (The bug could be triggered by packets arriving out of order.)  Thanks to Mallikharjuna Reddy

  for pointing this out. 

- When creating new datagram sockets, we now attempt to setsockopt() SO_REUSEPORT and IP_MULTICAST_LOOP,

  even on Windows.  (A version of Windows prior to XP had problems with this, but it's apparently no longer

  a problem.)

​

2006.09.12:

- Fixed a minor bug in "H264VideoRTPSink".  (Thanks to Emiliano Parasassi for noting this.)

- If the "totSessionBW" parameter to "RTCPInstance::RTCPInstance()" is zero, print out an error message,

  and use a value of 1 (kbps) instead.

​

2006.09.09:

- Reenabled asynchronous file reading in "ByteStreamFileSource", after fixing a bug

  that could lead to a 'race condition' in some circumstances.  (The fix was to call

  "FramedSource::afterGetting()" directly - rather than via the event loop - after

  delivering data to the downstream object.)

- Updated "MPEG2TransportStreamFromESSource" again.  Now, we set the

  "PES_packet_length" field to zero if the input data was larger than 2^16 bytes

  long.  (This is in accordance with the MPEG specifications; thanks to "jeremy at electrosilk.net"

  for noticing this..)

​

2006.09.08:

- Temporarily disabled asynchronous file reading in "ByteStreamFileSource" until we

  fix an apparent bug that is causing MPEG streaming to fail.

- Updated "MPEG2TransportStreamFromESSource" to print a warning message if the input source delivers

  data in chunks that are too big to make up a PES packet.

​

2006.09.07:

- Fixed a bug that was preventing the RTCP "RR" packet handler from being called

  when we were streaming RTP/RTCP-over-TCP.  (Thanks to "zhangzx at rcs-9000.com"

  for noticing this.)

​

2006.08.26:

- Changed the implementation of "ByteStreamFileSource" so that reads from the

  open file are now done asynchronously, when data becomes available.  This matches

  the code's event-driven execution model better than the previous implementation,

  which read from files synchronously.  It should also perform better in systems

  that read from more than one file concurrently.

- Added Victor Kozhuhov's modifications to the WAV file support code, to allow for WAV files that contain

  u-law or a-law content (instead of the usual raw PCM).

​

2006.08.24:

- Fixed the way in which "RTSPServer" handles requests that span more than one

  TCP socket read.  These reads are now all handled asynchronously, in accordance

  with the library's event-driven execution model.  (Thanks to Marc Neuberger for noting this.)

- Fixed a memory leak in "MultiFramedRTPSource" that would occur if we received

  packets with duplicate sequence numbers, or if we received excessively delayed

  packets.  (Thanks to David Bertrand for noticing this.)

- Changed the order of statements in "~RTSPClient()" to fix a bug (noted by Brain Lai).

- Changed "base64Encode()" to take a 'length' parameter, so that it will work with input data that

  contains '\0' characters.  (Thanks to Derk-Jan Hartman for this suggestion.)

- Added support - in "RTSPServer" - for the (non-standard) "x-playNow:" header in RTSP requests.

  (Thanks to Rob Casey for this patch.)

- Updated "MediaSession" to parse the (standard) SDP "framerate" attribute (in addition to "x-framerate", which

  is non-standard).

- Changed the parent class of "H264VideoStreamFramer" to be "FramedFilter" (to be consistent with other

  'Framer' classes) instead of "FramedSource".  (Thanks to Jinfeng Zhang for noticing this.)

- Made the "FramedFilter" implementation more robust, in case the supplied input source parameter is NULL.

​

2006.08.07:

- Changed the way that "OnDemandServerMediaSubsession" creates server ports.

  It now chooses server port numbers starting with a specific port number

  (which is now an optional parameter to the "OnDemandServerMediaSubsession"

  constructor).  The default value of this initial port number parameter is 6970.

  This matches the port number range used by other common RTSP server implementations,

  including Darwin Streaming Server and Helix.

- Changed the criterion for whether to send out a RTCP "SR".  (Thanks to David Bertrand.)

- Fixed an obscure bug in "OnDemandServerMediaSubsession::deleteStream()"

  (reported by Scott Hayes)

- Removed some archaic code that was developed for a customer several years ago, but

  which is no longer used.

​

2006.07.04:

- Another update to "H264VideoRTPSink" (requested by David Bertrand)

​

2006.06.28:

- Another update to "H264VideoRTPSink".  (Thanks to David Bertrand.)

​

2006.06.27:

- Updated "H364VideoRTPSink"

- Corrected the definition of MAX_PES_PACKET_SIZE in MPEG2TransportStreamFromPESSource.cpp.

  (Thanks to Jiri Pinkava for this fix.)

​

2006.06.23:

- Fixed some minor issues with the previous addition of "H264VideoRTPSink"

​

2006.06.22:

- Added "H264VideoRTPSink", for transmitting H.264/RTP streams.  (Thanks to

  Benoit Quirynen and David Bertrand for funding this work.)

- Changed the default definitions of IP_ADD_SOURCE_MEMBERSHIP and

  IP_DROP_SOURCE_MEMBERSHIP (for use if they're not already defined in header files),

  to make it more likely that SSM joins will properly fail if SSM is not actually

  implemented on the host OS.

- Changed some "delete"s into "delete[]".  (Thanks to Wojciech Matyjewicz

  and Mathur for reporting these bugs.)

- Made "sapWatch a bit more robust.  (Thanks to Jiri Pinkava for this fix.)

​

2006.05.17:

- Fixed "OnDemandRTSPServer" so that it sets a RTCP "RR" handler,

  even if "reuseFirstStream" is True.

- Changed "RTSPServer" (and the "testOnDemandRTSPServer" demo application)

  to use a 45-second client liveness timeout, by default.  (Previously,

  no client liveness checking was being done by default.)

  (Note that client liveness timeouts are done only for unicast streams, not multicast.)

- Fixed a bug in the implementation of "BasicHashTable" that arose when

  we were using hash tables with multi-word-sized keys.

​

2006.05.15:

- No change from the previous release, but the installed ".tar.gz" file had

  accidentally been deleted.

​

2006.05.11:

- Cleaned up some code in "BasicTaskScheduler" that was allegedly causing

  problems on a Zaurus.

- Changed "TaskScheduler::rescheduleDelayedTask()" so that its "microseconds"

  parameter is a "int64_t" (to match "scheduleDelayedTask").

  (Thanks to David Bertrand for suggesting this.)

- Updated "ServerMediaSession:::generateSDPDescription()" to change

  "a=rtcp: unicast-reflection" to "a=rtcp-unicast: reflection", to

  reflect the latest version of the IETF's "draft-ietf-avt-rtcpssm"

  Internet-Draft.  (This is for Source-Specific Multicast sessions only.)

- Cleaned up "H263plusVideoRTPSink" slightly, to remove an unnecessary

  member variable.

- Removed support for the (experimental and non-standard) "X-MCT-TEXT"

  payload type.

​

2006.03.17:

- Fixed a bug in "RTSPClient" (that had been introduced in the 2006.02.15

  release) that was stopping RTP-over-TCP reception from working.

​

2006.03.15:

- Updated "MediaSession" to parse the "s=" and "i=" lines of input SDP

  descriptions, and added member functions for returning these string values.

  (Thanks to Scott Hays or this.)

- Updated "MPEG2TransportStreamMultiplexor" to use the audio track for PCR

  iff there is no video track present.

- Made a change to "liveMedia/include/H263plusVideoStreamFramer.hh" to overcome

  an error reported by some nit-picking compilers.

- Fixed a minor bug in "MPEG2TransportStreamFromESSource" (in the way that it

  handles EOF on input).

​

2006.03.03:

- Updated "MPEG4LATMAudioRTPSource" to allow the LATM data length field to

  be omitted from the start of each returned audio frame.  (The "faad" decoder

  (used by VLC, for example) needed this.)

- Renamed "SECOND" in the "BasicUsageEnvironment" library to "DELAY_SECOND"

  (and similarly for "ZERO", "MINUTE", "HOUR", "DAY"), to avoid an alleged

  name conflict with some other code used by VLC.  (Thanks to

  Derk-Jan Hartman for this suggestion.)

- Removed some unnecessary code from "testProgs/playCommon.cpp".

​

2006.02.25:

- Updated the previous change to "MultiFramedRTPSource::doStopGettingFrames()",

  to also reset the object's fields.  (Thanks to David Bertrand.)

​

2006.02.15:

- Changed "GetFileSize()" (defined in "liveMedia/include/InputFile.hh") and

  "ByteStreamFileSource::fileSize()" to return a "u_int64_t", to allow for

  (potentially) large files.

- Changed the "microseconds" parameter to "TaskScheduler::scheduleDelayedTask()"

  from "int" to "int64_t".  This allows for durations greater than the previous

  ~2000 second limit.

- Updated "RTSPClient" to support handling RTSP requests sent by the server.

  (Some servers send periodic requests back to the client as a keep-alive test.)

  At present, we just respond to such requests with a "not allowed" error.

  (Thanks to Khanh Mai for this change.)

- Updated "MultiFramedRTPSource::doStopGettingFrames()" to reset the

  "ReorderingPacketBuffer".  (Thanks to David Bertrand for this fix.)

- Corrected a small bug in the previous "RTSPClient" modification

  (noticed by Glen Gray).

- Fixed a bug in "ADTSAudioFileSource" when parsing the optional "crc_check"

  field.  (Thanks to Paul Li for noticing this.)

​

2006.01.27:

- Updated "RTSPClient" to:

  - allow the "parameterName" parameter to "getMediaSessionParameter()" to be NULL

  - add a "setUserAgentString()" member function, to allow clients to

    specify an alternative string for "User-Agent:" headers

  - parse the optional "timeout" parameter in response "Session:" headers,

    and added a "sessionTimeoutParameter()" member function to retrieve it.

  (Thanks to Glen Gray for proposing these changes.)

- Fixed "H263plusVideoRTPSource" to properly initialize two member variables

  in its constructor.  (Thanks to David Bertrand for this fix.)

- Added Bernhard Feiten's support for H.263plus parsing/framing

​

2006.01.24:

- Corrected the maximum PES packet size (in "MPEG1or2DemuxedElementaryStream.cpp")

  to allow for a 6-byte header.  (Thanks to Jiri Pinkava for this fix.)

- Fixed "MPEG1or2AudioStreamFramer" to avoid a potential divide-by-zero when

  it's fed certain malformed MPEG audio data.  (Thanks to Eric Peters for this.)

- Updated the top-level "Makefile.tail" to make it clear that "testProgs" can't

  be built until after the 'library' directories are built. This ensures that

  parallelizing "make"s don't do the wrong thing.

​

2006.01.05:

- Updated "MPEG4VideoStreamFramer" to make the parsing of MPEG-4 video a bit

  more robust.  (Thanks to Khanh Mai for this suggestion.) 

​

2006.01.04:

- Updated "OnDemandServerMediaSubsession" to tear down the stream (thus sending

  a RTCP "BYE") when the input source ends, iff the stream does not have a known

  duration.  (If the stream has a known duration, then we keep the stream alive,

  in case a client wants to seek backwards in it.) 

​

2005.12.30:

- Extended the previous release's fix to also work with the RTSP

  "OPTIONS" command.

​

2005.12.29:

- Include (if necessary) authentication headers in the initial HTTP

  "GET" and "POST" commands, if we're setting up RTSP-over-HTTP tunneling.

  This allows "RTSPClient" to work properly with some Axis cameras, which

  require authentication on these HTTP commands.

​

2005.12.23:

- Fixed the RTCP port number in the RTSP "Transport:" header "port=" part

  (for multicast streams).

- Updated the change that we made in version 2005.11.02 so that it also

  sets the port number correctly.

​

2005.12.15:

- Made calls to "select()" more robust, by checking for (and ignoring)

  EAGAIN or EINTR errors.

- Fixed a bug in "MPEG4VideoFileServerMediaSubsession" that could occur if a

  bad MPEG-4 video file (not containing any initial 'config' data) is read.

  (Thanks to Trevor Pering for noticing this.)

- Removed an undocumented, experimental and rarely-used option from "openRTSP".

​

2005.12.09:

- Updated the RTCP implementation so that reception stats for a SSRC are now kept

  around even after a RTCP BYE is received, in case we want to use these stats

  for QOS reports.  (However, if a SSRC is reaped due to inactivity, then the

  reception stats are deleted.)

​

2005.12.05:

- On some systems (notably FreeBSD), "struct sockaddr_in" includes a "sin_len"

  field, which should be set before the struct is used.  We now replace declarations

  of "struct sockaddr_in" with a macro MAKE_SOCKADDR_IN() (defined in

  "groupsock/include/GroupsockHelper.hh").  This macro sets the "sin_len" field

  iff "HAVE_SOCKADDR_LEN" is defined on the command line.

  (Thanks to Boris Nikolaus for noting this issue.)

- Fixed a bug in the timestamp generation for certain rare MPEG-4 video streams.

  (Thanks to Bernhard Feiten for reporting this.)

​

2005.11.30a:

- Improved the calculation of the per-Transport Packet duration in

  "MPEG2TransportStreamFramer", so that the overall transmit duration

  tries to match the client playout duration (as determined by PCRs).

- Minor patch to avoid errors when compiling with GCC 4.1.

  (Patch contributed by Hanno Boeck.)

- Patch to the special 'Kasenna'-specific support in "RTSPClient".

  (Patch contributed by Glen Gray.)

​

2005.11.11:

- Modified "MPEG2TransportStreamFramer" to take the PCR "discontinuity_indicator"

  flag into account when updating the estimated duration of each TS packet.

​

2005.11.03:

- Minor bug fix to the previous version's update to "RTSPClient".

​

2005.11.02a:

- Updated the previous release to add a new (optional)

  "forceMulticastOnUnspecified" parameter to "RTSPClient::setupMediaSubsession()".

  This forces the client to request a multicast stream if the original SDP

  response address was unspecified (0.0.0.0).  (Note that not all servers

  will be able to handle this.)  We also handle the "port=" field in RTSP

  "SETUP" response "Transport:" headers.

- Fixed "RTSPServer" to include a proper "port=" field in the "SETUP"

  "Transport:" response when streaming multicast.

​

2005.11.02:

- Updated "RTSPClient" to use the "destination=" address in the "SETUP"

  response's "Transport:" header - for multicast streams.  We do this because

  some weird servers (e.g., Axis cameras, reportedly) do not specify the

  multicast address earlier, in the "DESCRIBE" response's SDP.

​

2005.10.28b:

- Added a "rtpTimestampFrequency" parameter (with default value 90000)

  to "H263plusVideoRTPSink", "MPEG4ESVideoRTPSink".

​

2005.10.28a:

- Removed some unnecessary "delete"s, and changed some "delete"s to

  "Medium::close()"s.  (Minor bug noted by Zhixue Zhang.)

​

2005.10.28:

- Fixed a potential 'divide by zero' problem in "MP3Internals.cpp".

  (Thanks to Eric Peters for noting this.)

​

2005.10.27:

- Added a "numEntries()" pure virtual function to "HashTable".

  (Thanks to David Bertrand for this suggestion.)

​

2005.10.23:

- When creating a Transport Stream from AC-3 audio input, we now use a stream_id

  of 0x06 rather than 0x81.

- Add support for receiving the various G.726 RTP payload formats, described

  in RFC 3551.

- Added an implementation of the "doStopGettingFrames()" virtual function to

  "BasicUDPSource".  (Thanks to Adrian Hornsby for this suggestion.)

- Changed the reimplemented virtual functions in "BasicUsageEnvironment0" and

  "BasicTaskScheduler0" from "protected" to "public".  (Thanks to David Bertrand

  for this suggestion.)

- Added support to "RTSPClient" for the "GET_PARAMETER" RTSP command.

  (Thanks to Glen Gray for this.)

- Added a Windows version of the "genWindowsMakefiles" script.  This should

  make it easier for Windows users to unpack and build the code.

  (Thanks to Erwin Herzog for this script.)

​

2005.10.05:

- Fixed a couple of bugs noticed by David Bertrand.

​

2005.09.23:

- This software is now called "LIVE555 Streaming Media", and our preferred domain name

  is now "live555.com".

​

2005.09.20:

- Updated "MPEG2TransportStreamMultiplexor" to increase the frequency of PMTs

  and PATs in the output Transport Stream. 

- More improvements to "MPEG2TransportStreamFromESSource".

- Updated "RTSPServer" to allow for empty stream names from certain non-standard

  RTSP clients (such as Amino STBs).

- Fixed "RTSPClient::tearDownMediaSession()" to delete each subsession's

  "sessionId" field.

- Fixed a minor memory leak in "RTSPClient" (when implementing HTTP tunneling)

​

2005.09.09:

- Made the destructors for "BasicUsageEnvironment" and "BasicUsageEnvironment0"

  protected rather than public, to match the base class "UsageEnvironment".

- More improvements to "MPEG2TransportStreamFromPESSource" and

  "MPEG2TransportStreamFromESSource".

​

2005.09.06:

- Modified "MPEG2TransportStreamMultiplexor", and added

  "MPEG2TransportStreamFromESSource".

​

2005.08.26:

- Updated "MediaSession" to parse the SDP "a=type:" attribute.  (Patch by

  Derk-Jan Hartman.)

- Added support to "RTSPClient" for handling Microsoft servers' non-standard use

  of the RTSP/RTP protocol.  Also, fixed the parsing and generation of floating

  point numbers in the RTSP "Range:" and "Scale:" headers to work correctly in

  non-POSIX locales.  (Thanks to Derk-Jan Hartman for this suggestion.)

- Fixed a memory leak in "MPEG2TransportStreamFromPESSource".

- Changed "MP3StreamState::readFromStream()" from "private" to "protected", to

  satisfy a request by Pierre l'Hussiez.

​

2005.08.09:

- Improved the Transport Stream generation software to (in the future) allow

  for generating Transport Streams from Elementary Stream data as well as

  from PES packet data.

​

2005.07.23:

- Improved "H264VideoRTPSource" to properly handle 'aggregate' packets

  that contain more than one NAL unit.

​

2005.07.21:

- We no longer set a RTCP "RR" handler when we're streaming, unicast on-demand,

  from a shared input source.  (The reason for this is that, in this case,

  we also share a RTCP instance for all client sessions (streams).

  A better fix will eventually be needed.)

​

2005.07.19:

- Updated H.264/RTP support.

​

2005.07.15:

- Oops, DEBUG was not supposed to be defined in "RTSPServer.cpp".

- Added "-D_LARGEFILE_SOURCE=1" to the "COMPILE_OPTS =" line of the various

  "config.*linux*" files, to ensure that code that uses "fseeko()" will

  compile properly for all Linux systems.

​

2005.07.13:

- When handling a 'passive' session, we no longer set a RTCP "RR" handler,

  because (unlike unicast, on-demand sessions) the RTCP instance usually

  outlives RTSP client sessions.  (Thanks to Jon Sheller for reporting

  this bug.)

​

2005.07.12:

- Minor updates to the Base64 code for H.264/RTP receiving.

​

2005.07.10:

- Removed a debugging version of "testOnDemandRTSPServer" that had mistakenly been

  left in the proevious revision.  If you downloaded the previous revision, then

  please download this version instead!

- Fixed "MPEG1or2VideoStreamFramer" to better handle MPEG video streams where the

  first GOP time code has a non-zero 'pictures' count.

  (Thanks to Eric Peters for reporting this problem.)

- Some enhancements to the Base64 code, for eventual use by "H264VideoRTPSource".

​

2005.07.08:

- Added - to "RTCPInstance" - the ability to assign 'handler' functions

  to be called on the arrival of RTCP "SR" or "RR" packets (as well as "BYE").

  We now use this mechanism (specifically, a RTCP "RR" handler) to improve the

  optional 'liveness' test for "RTSPServer".  (Liveness is now indicated by

  either a RTSP command over the TCP connection, or the arrival of a RTCP

  "RR" packet.)

- Moved routines for Base64 encoding and decoding into the "liveMedia"

  library.  (Base64 encoding was already being used by "RTSPClient";

  Base64 decoding will soon be used by "H264VideoRTPSource".)

​

2005.07.01:

- Updated "testOnDemandRTSPServer" to demonstrate AAC audio streaming (from an

  ADTS-format file)

- Made some modifications to (supposedly) make the code compile better for

  WinCE. 

​

2005.06.30:

- Added support for reading and streaming from AAC audio files in ADTS format.

  (Thanks to Manthan Systems for funding this work.)