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| Name: live555 | Distribution: SUSE Linux Enterprise 15 SP5 |
| Version: 2023.01.19 | Vendor: openSUSE |
| Release: bp155.1.6 | Build date: Mon May 22 11:54:06 2023 |
| Group: System/Libraries | Build host: lamb17 |
| Size: 531952 | Source RPM: live555-2023.01.19-bp155.1.6.src.rpm |
| Packager: https://bugs.opensuse.org | |
| Url: http://www.live555.com/liveMedia/ | |
| Summary: LIVE555 Streaming Media | |
This code forms a set of C++ libraries for multimedia streaming, using open standard protocols (RTP/RTCP, RTSP, SIP). These libraries can be used to build streaming applications
LGPL-2.1-only
* Sat Jan 21 2023 Dirk Müller <dmueller@suse.com>
- update to 2023.01.19:
- By default, we no longer compile "groupsock/NetAddress.cpp" for Windows to use
"gethostbyname()", because of a report that this breaks IPv6 name resolution.
* Mon Jan 16 2023 Dirk Müller <dmueller@suse.com>
- update to 2023.01.11:
* Updated the "BasicTaskScheduler"/"DelayQueue" implementation to make the 'token counter'
a field of the task scheduler object, rather than having it be a static variable.
This avoids potential problems if an application uses more than one thread (with each thread
having its own task scheduler).
* Fri Dec 02 2022 Dirk Müller <dmueller@suse.com>
- update to 2022.12.01:
- Yet another fix to the previous fix for RTSP-over-HTTP streaming.
- The previous version's fix to "RTSPClient" (for RTSP-over-HTTP streaming) was incomplete.
This should fix it for real.
- Fixed a bug in "RTSPClient" that would cause RTSP-over-HTTP streaming over TLS to
sometimes fail. (Thanks to Johannes Gajdosik for reporting this.)
- Fixed a bug that would cause a RTSP server to use an incorrect URL if it accepted connections
via TLS, but *without* streaming SRTP.
* Tue Nov 22 2022 Dirk Müller <dmueller@suse.com>
- update to 2022.11.19:
- Added a new global variable "ReceivingInterfaceAddr6" (analogous to the existing
variable "ReceivingInterfaceAddr" for IPv4) to allow applications to optionally
change the default receiving address for IPv6.
* Sun Oct 02 2022 Dirk Müller <dmueller@suse.com>
- update to 2022.10.01:
- Updated the previous revision so that the virtual function "specialHandlingOfAuthenticationFailure()"
is now called only if there is an actual authentication failure - not on the first time that
we send back a "401 Unauthorized" response.
- Added a new virtual function "specialHandlingOfAuthenticationFailure()" to "RTSPServer" to
allow a subclassed "RTSPServer" to take special action (e.g., statistics logging)
whenever an authentication failure occurs.
* Fri Jul 22 2022 Dirk Müller <dmueller@suse.com>
- update to 2022.07.14:
* use SHA-1 rather than MD5 to hash the latest tarball of our code
* Fri Jun 24 2022 Dominique Leuenberger <dimstar@opensuse.org>
- Update to version 2022.06.16:
+ Both our RTSP client and RTSP server implementations now
support (optional) RTSP-over-HTTPS streaming, when
RTSP-over-TLS would otherwise be available.
- Changes from version 2022.06.14:
+ Added optional support (via #ifdefs) to the
"testOnDemandRTSPServer" demo application for streaming via
RTSPS (RTSP-over-TLS) and optionally SRTP (encrypted RTP/RTCP).
To use this, you would need to define SERVER_USE_TLS, and
PATHNAME_TO_CERTIFICATE_FILE and PATHNAME_TO_PRIVATE_KEY_FILE.
* Wed May 04 2022 Christophe Giboudeaux <christophe@krop.fr>
- Update to 2022.04.26:
* Ensure that we don't call "delete[]" on an uninitialized
pointer.
- Changes from version 2022.04.15:
* Fixed a "fprintf()" argument-order-evaluation bug in the
"mikeyParse" demo application.
- Changes from version 2022.04.12:
* Updated the "openRTSP" application (RTSP command-line client)
to add an option '-L', meaning: receive only an "application"
(e.g., 'metadata') track, if present, outputting the data to
'stdout'.
* Tue Feb 15 2022 Dominique Leuenberger <dimstar@opensuse.org>
- Update to version 2022.02.07:
+ Updated the SRTP packet sending code in "MultiFramedRTPSink.cp"
to not allocate a variable-sized buffer on the stack, because
some compilers can't handle this.
+ Ensure that RTSP servers that serve SRTP do not also support
streaming over the TCP connection, because that would add extra
overhead for no benefit.
- Changes from version 2022.01.21:
+ Fixed a bug in the "groupsock" library that could cause
outgoing RTP packets to get duplicated when a RTSP "PLAY"
command is sent after a "PAUSE".
- Changes from version 2022.01.20:
+ More updates to the code for optional server SRTP streaming.
- Changes from version 2022.01.17:
+ More updates to the code in preparation for optional server
SRTP streaming.
- Changes from version 2022.01.11:
+ Fixed a minor memory leak in "RTSPClient" when receiving a SRTP
stream.
+ Updates to "RTPSink" in preparation for optional server SRTP
streaming.
- Changes from version 2022.01.06:
+ Made "GenericMediaServer::addServerMediaSubsession()" a virtual
function, and redefine it in the subclass "RTSPServer" to call
the base function, then set the "ServerMediaSubsession"s
"streamingIsEncrypted" flag (if the RTSP server is streaming
SRTP).
* Mon Dec 20 2021 Dominique Leuenberger <dimstar@opensuse.org>
- Update to version 2021.12.18:
+ Fixed a bug in the way that "RTSPClient" handles its two
separate TCP connections when it does RTSP-over-HTTP.
+ Updated "RTPInterface::sendDataOverTCP()" so that if it's
necessary to do a blocking send(), we call
"makeSocketNonBlocking()" immediately after the call to
"send()".
+ Performed the annual update of the copyright years near the
start of each file.
* Sat Dec 18 2021 Dirk Müller <dmueller@suse.com>
- update to 2021.12.07:
- Added #ifndef NO_OPENSSL/#endif around "#include <openssl/err.h>" in
"liveMedia/TLSState.cpp", so that the code will compile if you're compiling
with no OpenSSL headers, and NO_OPENSSL
defined.
* Mon Dec 06 2021 Dirk Müller <dmueller@suse.com>
- update to 2021.11.23:
* Updated the "RTSPServer::setTLSState()" function to take an optional
parameter "weServeSRTP". For now, the default value of this parameter is
False, but it will get changed to True later, when we implement server-side
SRTP.
* Updated the RTSP server implementation to (optionally) support connections via TLS.
* Updated the "TLSState" interface and implementation to (1) reduce the
amount of stuff that the compiler gets to see if you're compiling with
NO_OPENSSL defined, and (2) add a new subclass "ServerTLSState" that
will eventually be used to implement optional TLS connections to our RTSP server.
* Split the "TLSState" class into two classes: "TLSState" (an abstract base
class), and "ClientTLSState" (a subclass). This is in preparation for later defining
second subclass "ServerTLSState" that will eventually be used to implement TLS connections
in our RTSP server.
* Updated the implementation of AES encryption/decryption (used by our client SRTP
implementation) to use the new OpenSSL EVP interface. This makes it possible to
use hardware acceleration (e.g., AES-NI), when it is available.
* Updated the "RTSPClient"s implementation of receiving RTP/RTCP-over-TCP so
that it will also work over a RTSP-over-TLS (including RTSPS) connection.
* Fixed a bug in "MatroskaFileParser" that could cause delivery of data to a downstream object
that wasn't expecting it (potentially causing an invalid memory access).
* The final (I hope!) update to eliminate a "depends on uninitialised value"
report from 'valgrind'.
* Mon Aug 23 2021 Fusion Future <qydwhotmail@gmail.com>
- Update to 2021.08.23:
* Updated the "readSocket()" code in "GroupsockHelper.cpp" yet
again to try to eliminate another (alleged) "depends on
uninitialised value" report from 'valgrind'.
(If, after this, you still see this, then your implementation
of "recvfrom()" is broken.)
- Changes in 2021.08.19:
* Updated the "readSocket()" code in "GroupsockHelper.cpp" to
eliminate another possible "depends on uninitialised value"
report from 'valgrind'.
- Changes in 2021.08.18:
* Updated the "readSocket()" code in "GroupsockHelper.cpp" to
eliminate a "depends on uninitialised value" report from
'valgrind'.
- Changes in 2021.08.17:
* Updated the 'groupsock' "setPortNum()" function to not rely
upon the "ss_family" family field, in case it's uninitialized.
- Changes in 2021.08.14:
* Fixed a minor bug in the previous release ("delete" should have
been "delete[]")
- Changes in 2021.08.13:
* Fixed a bug in "MPEG1or2Demux" that could cause a 'reading
twice at the same time" abort when streaming from a MPEG
Program Stream file. (boo#1189726, CVE-2021-39283)
* Fixed a potential memory leak in "AC3AudioStreamFramer".
(boo#1189725, CVE-2021-39282)
* Thu Aug 12 2021 Fusion Future <qydwhotmail@gmail.com>
- Update to 2021.08.09:
- Fixed a bug in the MPEG-1 or 2 file server demultiplexors that
could cause a RTSP server to crash if it received successive
RTSP "SETUP" commands for the same track. (Thanks to Ba
Jinsheng for reporting this.)(boo#1189352, CVE-2021-38381)
- Update to 2021.08.06:
- Fixed a bug in the Matroska and Ogg file server demultiplexors
that could cause a RTSP server to crash if it received
successive RTSP "SETUP" commands for the same track. (Thanks
to Ba Jinsheng for reporting this.)(boo#1189353, CVE-2021-38382)
- Update to 2021.08.04:
- In the "MP3FileSource" implementation, we no longer do a
recursive call to "doEventLoop()" when attempting to
synchronously read from a MP3 file. This avoids a possible
stack overflow in the RTSP server if multiple concurrent
requests are made. (Thanks to Ba Jinsheng for reporting this.)
The server still does some synchronous reads, when
initializing, and when parsing MP3 frame headers. This should
be fixed sometime in the future. (boo#1189351, CVE-2021-38380)
- Update to 2021.07.20:
- If a "RTSPClient" receives a response to a RTSP "PLAY" that
changes the 'scale()' or 'speed()' of the whole session, then
those parameters also need to be changed in each subsession (as
that inheritance doesn't happen automatically). (Thanks to a
developer in China for reporting this.)
- Update to 2021.07.10:
- Updated "H264or5VideoStreamFramer.cpp" once again to set the
default value of "DeltaTfiDivisor" to 2.0 for H.265, and 1.0
for everything else. (This fixes the frame rate for another
stream supplied by Paul Westlund.)
- Update to 2021.06.29:
- In the proxy server implementation, if a client closes one
substream, but there are still other clients receiving other
substream(s), then we no send a single-track RTSP "PAUSE"
command downstream, because some back-end servers might handle
that by pausing all tracks of the stream. So now, in this
case, we don't send a RTSP "PAUSE" command at all. (Thanks to
Jose Maria Infanzon for noting this issue.)
- Update to 2021.06.25:
- Updated "H264or5VideoStreamFramer.cpp" to set the default value
of "DeltaTfiDivisor" to 1.0 (rather than 2.0), and to assume a
frame rate of 30 fps (rather than 25 fps) if there is no VPS or
SPS NAL unit that specifies a different frame rate. This seems
to work the best for most raw H.264 and H.265 video streams.
(Thanks to Paul Westlund for supplying an example file to
motivate this.)
- Change the so version of libliveMedia to 97
* Thu Jun 17 2021 tiwai@suse.de
- Update to 2021.05.22:
lots of fixes and updates, including the security fix for
CVE-2021-28899 boo#1185874
See the list in http://live555.com/liveMedia/public/changelog.txt
- Change the so version of libliveMedia to 94, libgroupsock to 30
* Sun Oct 18 2020 Dirk Mueller <dmueller@suse.com>
- update to 2020.10.16:
- Changed "TLSState::read()" to treat any "SSL_read()" result of <=0 as if the
TLS connection has closed (unless the error was SSL_ERROR_WANT_READ).
This fixes a problem that could cause 100% CPU usage in RTSP client applications.
(Thanks to Larry Wu for reporting this.)
- Updated "TLSState::setup()" to use "TLS_client_method()" instead of the (deprecated)
"SSLv23_client_method()".
* Sat Oct 03 2020 Dirk Mueller <dmueller@suse.com>
- update to 2020.08.19:
- Fixed a bug in "QuickTimeFileSink" that could cause malformed "esds" atoms to be
generated. (Thanks to Chris Paucar for reporting this issue.)
- In "MPEG2TransportStreamFromESSource.cpp", changed the name of the constant LOW_WATER_MARK
to TS_FROM_ES_LOW_WATER_MARK, and "#ifndef"d it, so that, if you wish, you can redefine it
at compile time.
- Fixed a bug in the handling of pausing, when streaming from (multi-track) Matroska files.
- Fixed another bug in the handling of seeking within Matroska files.
- Fixed a bug in the handling of seeking within Matroska files.
(Thanks to Jim Ham for reporting this problem.)
- Changed the parameter signature of the "RawVideoRTPSink" constructor and "createNew()"
functions so that the "width" parameter comes before the "height" parameter. This order
- "width", "height" - is more common, and is the order used when these parameters are
defined in RFC 4175.
IMPORTANT NOTE: Because the types of these two parameters are the same, existing application
code that uses "RawVideoRTPSink" will compile without error; however, it will not work
properly unless the order of the parameters in the call to "RawVideoRTPSink::createNew()"
is changed.
- More cleanup of the implementation of "RawVideoRTPSink".
- Cleaned up the implementation of "RawVideoRTPSink".
- Updated the "RawVideoRTPSource" implementation to not set "fCurrentPacketCompletesFrame"
until we are processing the last line in the packet.
(Thanks to Andrey Lisovoy for reporting this issue.)
- Fixed a potential buffer overflow bug in the server handling of a RTSP "PLAY" command,
when the command specifies seeking by absolute time.
(Thank to Xiaobo Xiang for reporting this.)
- Fixed a memory leak in the "sha1()" function (a "EVP_MD_CTX" object was not being deleted).
(Thanks to Amir Perlman for reporting this.)
- Moved all definitions of PREFIX from "Makefile.tail" files to "Makefile.head" (so that
it can be redefined by a "config.*" file, if desired.
Also changed the definition of EXE in "config.mingw" to be ".exe".
(Thanks to Eric Beuque for this suggestion.)
- Fixed a typo in the previous release that could cause a compilation problem for some
developers. (Thanks to Eric Beuque for reporting this.)
* Thu Jun 04 2020 Dominique Leuenberger <dimstar@opensuse.org>
- Update to version 2020.05.15:
+ Added a new filter class "ADTSAudioStreamDiscreteFramer" that
prepends ADTS headers to incoming AAC audio frames. This makes
the AAC audio playable (by media players).
+ Updated "openRTSP" to use a "ADTSAudioStreamDiscreteFramer"
when outputting a AAC audio stream.
+ Updated the "LIVE555 HLS Proxy" to support AAC audio tracks
(as well as H.264/5 video).
- Changes from version 2020.05.14:
+ Updated "H264or5VideoStreamDiscreteFramer" to add VPS,SPS,PPS
NAL units (if known) to the output stream, each time an
"access_unit_delimiter" NAL unit is added. This makes it more
likely that the Transport Stream segments produced by the
"LIVE555 HLS Proxy" will be understandable by a client browser.
+ Added support for H.265 video streams to the "LIVE555 HLS Proxy".
- Changes from version 2020.05.13:
+ Made the "MPEG2TransportStreamMultiplexor" segmentation mechanism
(used by "HLSSegmenter") more robust in case the Transport
Stream PTS is not monotonic non-decreasing.
* Tue Apr 28 2020 Dominique Leuenberger <dimstar@opensuse.org>
- Update to version 2020.04.24:
+ Fixed an apparent bug in "RTSPClient" that was causing it to
not always send an "Authorization:" header when sending a RTSP
"OPTIONS" command.
* Thu Apr 23 2020 munix9@googlemail.com
- Added pkgconfig(openssl) as a requirement for the devel package:
iveMedia/TLSState.hh #includes openssl/ssl.h.
* Tue Apr 14 2020 munix9@googlemail.com
- Update to version 2020.04.12:
+ Updated "config.linux-with-shared-libraries"
(and "liveMedia/Makefile.tail") to ensure that "libssl" and
"libcrypto" are linked when "libliveMedia" is built.
(Thanks to Felix Kaechele for reporting this.)
- Changes from version 2020.04.06:
+ Removed support for the classes "RTSPServerSupportingHTTPStreaming"
and "TCPStreamSink".
These were being used (in the "LIVE555 Media Server") for
streaming using "HTTP Live Streaming" (HLS). This was always
a hack; it is better to use a dedicated HTTP server to serve
HLS segments, rather than trying to implement a HTTP server
(serving 'virtual HLS segments) within our own (primarily RTSP)
server.
If you are looking for HLS support, note that we provide a
source-code demo application "testH264VideoToHLSSegments" that
converts a (static) H.264 Elementary Stream file to HLS
segments, and the "LIVE555 HLS Proxy", which proxies a live
RTSP/RTP stream to HLS segments.
* Thu Apr 02 2020 Dominique Leuenberger <dimstar@opensuse.org>
- Update to version 2020.03.06 (boo#1146283, CVE-2019-15232):
+ Fixed a problem in "H264or5VideoStreamFramer.cpp" that was
allegedly causing problems compiling for Windows.
- Changes from version 2020.02.25:
+ Added full support for the "LIVE555 HLS Proxy"
- Changes from version 2020.02.24:
+ Commented out a line of code that was preventing
"RTSPClientConnection" objects from being closed when a RTSP
server handles a "TEARDOWN" command (on a RTP-over-TCP stream).
- Changes from version 2020.02.23:
+ Updated our (unicast) RTSP server implementation to handle
"PAUSE" commands by calling "stopGettingFrames()" on the input
source.
+ Fixed a bug in "H264or5VideoStreamFramer" that was causing it
to not update its presentation times properly following a pause.
+ Updated "openRTSP" to improve the handling of the "-R <port-num>"
option.
- Changes from version 2020.02.11:
+ Added support for receiving SRTP (encrypted) RTSP streams.
- For changes between 2019.06.28 and 2020.02.10, please see the
http://www.live555.com/liveMedia/public/changelog.txt.
- Rebase live555-fpic.patch.
* Mon Jul 15 2019 Dave Plater <davejplater@gmail.com>
- Update to version 2019.06.28, fixes boo#1127341 VUL-1:
CVE-2019-9215: live555: malformed headers lead to invalid memory
access in the parseAuthorizationHeader function.
* Mon Jun 24 2019 Dominique Leuenberger <dimstar@opensuse.org>
- Convert to dynamic libraries (boo#1121995):
+ Use make ilinux-with-shared-libraries: build the dynamic libs
instead of the static one.
+ Use make install instead of a manual file copy script: this
also reveals that we missed quite a bit of code to be installed
before.
+ Split out shared library packages according the SLPP.
* Thu May 16 2019 Martin Liška <mliska@suse.cz>
- Use FAT LTO objects in order to provide proper static library.
* Mon Feb 04 2019 atoptsoglou@suse.com
- update to 2019.02.03:
* CVE-2019-7314:A use-after-free error causes the RTSP server to
crash (boo#1124159)
* Tue Jan 15 2019 astieger@suse.com
- update to 2018.12.14:
* Add support for sending (and handling) RTCP "BYE" packets that
contain a 'reason' string (breaking api change)
* Tue Jan 15 2019 astieger@suse.com
- update to 2018.11.26:
* CVE-2019-6256: DoS vulnerability in the server implementation
of RTSP-over-HTTP (boo#1121892)
* Tue Nov 06 2018 Mathias Homann <Mathias.Homann@opensuse.org>
- Update to 2018.10.17
* CVE-2018-4013: remote code execution vulnerability (boo#1114779)
* changes: see http://www.live555.com/liveMedia/public/changelog.txt
* removed xlocale.patch - xlocale.h is now conditionally included, only when
required. Upstream fix.
* Wed Nov 01 2017 Mathias.Homann@opensuse.org
- Update to 2017.10.28
2017.10.28:
- Fixed the handling of the LIVE555 Proxy Server's "-u <username> <password>" command-line option if
the "REGISTER" command is also implemented (i.e., with "-R"). Now, when we handle "REGISTER",
the <username> and <password> are used to access the REGISTER'ed back-end stream, if necessary.
- Changed the server handling of the RTSP "REGISTER" command to (if "reuseConnection" is True) delay
a short period of time (100ms) between replying to the "REGISTER" command, and actually
handling it. This is intended to reduce/avoid the possibility of a subsequent "DESCRIBE" command
ending up in the client ("REGISTER" sender)'s buffer, before the socket gets reused for handling
incoming RTSP commands. (Thanks to Durgesh Tanuku for noting this issue.)
- Made a change to "SIPClient" to better handle Asterisk SIP servers with authentication.
(Thanks to Rus.)
2017.09.12:
- Made some improvements/bug fixes to AVI indexes in "AVIFileSink".
(Thanks to Victor V. Vinokurov.)
- Updated the handling of the "writeTimeoutInMilliseconds" parameter in the "makeSocketBlocking()"
function to work correctly on Windows. (Thanks to Jeff Shanab for noting this issue.)
- Added support for adding Opus audio to MPEG Transport Streams. This is done by setting the
"mpegVersion" parameter to 3 in "MPEG2TransportStreamFromESSource::addNewAudioSource()" or
"MPEG2TransportStreamMultiplexor::handleNewBuffer()".
(Thanks to Praveen Mathad for suggesting this.)
2017.07.18:
- Updated "BitVector" to support a signed version of "get_expGolomb()", and fixed our H.264/265
parsing code to use the signed version where appropriate.
(Thanks to Toson Huang and Long Zhang for reporting this.)
* Mon Aug 07 2017 schwab@suse.de
- xlocale.patch: don't use obsolete <xlocale.h>
* Sat Jul 08 2017 jengelh@inai.de
- Replace silly -exec rm ; by -delete.
* Mon Jul 03 2017 ramaxlo@gmail.com
- Update to version 2017.06.04
2017.06.04:
* Fixed a bug in "RTPInterface::removeStreamSocket()" that could
cause not all 'TCP stream' records for a given socket number
to be removed if a TCP socket I/O error occurred (during
RTP/RTCP-over-TCP streaming). (Thanks to Gerald Hansink et al
for reporting this.)
2017.05.24:
* In "RTSPClient.cpp", moved the call to "clearServerRequestAlternativeByteHandler()"
from the "RTSPClient" destructor to the "resetTCPSockets()"
function (which is called more often). This should eliminate
a 'pointer to a deleted object' error. (Thanks to Gerald Hansink
et al for reporting this.)
2017.04.26:
* Added a new public member function "numClientSessions()"
to "GenericMediaServer" (and therefore to "RTSPServer", which
inherits from this). This allows a server to - at any time -
check how many clients are currently accessing the server.
* Updated the diagnostic output in "RTSPClient" to distinguish
between opening a new TCP socket and connect()ing on a TCP
socket. (The distinction is important for "REGISTER", which can
reuse an existing TCP socket.)
2017.04.10:
* Fixed a bug in "base64Decode()" that could be triggered
if (1) your RTSP server is streaming RTP/RTCP-over-HTTP, and (2)
the remote client sends bad Base64 data (containing an embedded '\0'
character). (Thanks to Arkady Bernov for reporting this.)
2017.01.26:
* Updated "ProxyServerMediaSession.cpp" to change all 'reset()'
operations so that they are now run as a 'scheduled task' from
the event loop - avoiding the possibility of bugs caused by
'reset()' being called while another operation is in progress.
(Thanks to Erik Montnemery for reporting this issue, and proposing a fix.)
2016.11.28:
* Our "RTSPClient" code now ignores "Connection: close" lines in
the responses to HTTP "GET" requests (that are used to set up
RTSP-over-HTTP tunneling). Because this tunneling requires that
the (separate) input and output TCP connections remain intact,
we assume that the server - if it includes such a line in the
response to a HTTP "GET" - doesn't really mean it.
(Thanks to Nguyen Viet Hung for reporting a server that does this.)
2016.11.17:
* Fixed a bug in the handling of 'APP' RTCP subpackets. (Thanks to
Frederik de Ruyck for reporting this.)
* Fixed a bug in the "StreamReplicator" code. (Thanks to Bruno Abreu
for reporting this.)
2016.11.06:
* Increase the RTSP client's socket receive buffer when we'll be
receiving RTP/RTCP-over-TCP, and increase the RTSP server's
client connection socket send buffer when it's used to
"REGISTER" a stream.
2016.11.03:
* Fixed a bug (in the sending/ handling of the "REGISTER"/"DEREGISTER"
commands) that had been accidentally introduced in version
2016.09.19. (Thanks to Ralf Globisch for noting this.)
2016.10.29:
* Performed the annual update of the copyright years and license
near the start of each file
2016.10.21:
* Changed the "RTCPInstance error" message in "RTCP.cpp" to make
it clear that the problem is caused by the remote endpoint using
a buggy version of RTP/RTCP-over-TCP streaming.
* Updated "QuickTimeFileSink" to make the various creation/modification
times relative to January 1st 1904 in UTC (as Apple recommends),
rather than in US Pacific Time.
* Sun Oct 16 2016 aloisio@gmx.com
- Update to version 2016.10.11
2016.10.11:
* After building the source code, we now display a message
reminding the developer about our FAQ.
2016.09.22:
* Added a new "liveMedia" class
"MPEG2TransportStreamAccumulator" - a filter that can
be used to combine several (by default, 7) MPEG Transport
Stream 188-byte 'packets' into a larger chunk of data,
more appropriate for streaming via RTP (or raw UDP).
2016.09.19:
* Added support for an experimental RTSP "DEREGISTER"
command, which undoes the effect of a "REGISTER" command.
* Moved the REGISTER/DEREGISTER-specific functionality of
"RTSPServer.cpp" into a new file "RTSPServerRegister.cpp",
to make the base RTSP server code (in "RTSPServer.cpp")
easier to comprehend.
2016.09.12:
* Fixed "GenericMediaServer::createNewClientSessionWithId()"
to make sure that the new 'client session' object (returned
by a call to "createNewClientSession()") is not NULL
before it tries to add it to the 'fClientSessions' table.
(Thanks to Helmut Grohne for discovering this issue.)
2016.09.08:
* Updated "RTSPClient::reset()" to reset each of the 'request
queues' as well.
(Thanks to Erik Montnemery for noting a problem (with the
"LIVE555 Proxy Server") that this caused.)
* Updated "GenericMediaServer::ClientConnection::closeSockets()"
so that it doesn't try to call "closeSocket()" (=="close()")
on socket numbers <0.
2016.09.05:
* Fixed a problem whereby a 'delayed task' for a
"MPEG2TransportStreamMultiplexor" object might have gotten run
after such an object was deleted.
(Thanks to Bruno Basilio for providing debugging output to
help track this down.)
* Updated "Socket::reset()" (in "groupsock/NetInterface.cpp") so
that it doesn't try to call "closeSocket()" (=="close()") on
socket numbers <0.
* Added a comment to "UsageEnvironment/include/UsageEnvironment.hh"
to note that "triggerEvent()" should not be called with the
same 'event trigger id' from different threads.
(This was already noted in a comment in
"liveMedia/DeviceSource.cpp", but not in
"UsageEnvironment/include/UsageEnvironment.hh", which is
where "triggerEvent()" is defined.)
2016.08.27:
* Fixed a problem whereby a "Medium" object's "nextTask()"
(i.e., "fNextTask") could hold an invalid value after a
'scheduled task' has occurred (but before the next similar
task is scheduled) - which causes problems should the
"Medium" object be deleted during that window of time.
(Thanks to Helmut Grohne for noting this problem.)
* Added comments to "UsageEnvironment/include/UsageEnvironment.hh"
to make it clear that "unscheduleDelayedTask()" (or
"rescheduleDelayedTask()") must not be called on a
'scheduled task' after it has already occurred.
(Thanks to Helmut Grohne for motivating this.)
2016.08.07:
* Fixed a bug in the handling of the non-standard
"com.ses.streamID:" header (used by 'SAT>IP' servers)
that we had introduced in version 2016.01.12.
(Thanks to Yaobing Deng for noting this.)
2016.07.19:
* Fixed a bug in "RTSPServer" that could cause a crash if a
"RTSPServer" object is deleted after having been used
for RTSP-over-HTTP streaming. (Thanks to Pavel Aronov.)
* Updated "RTSPClient" to recognize a "Connection: Close"
header in a server's response. It handles this header by
closing the RTSP TCP connection (because the server is
assumed to not be using it again), so that we open a new
TCP connection for any subsequent commands.
(Thanks to Nathan (at MediaPortal) for this suggestion.)
* Made a small optimization to "RTSPServer"s handling of the
first "SETUP" command from each client. (Thanks to Maxim
Dementiev for the suggestion.)
2016.06.26:
* Added a new (public) function "canDeliverNewFrameImmediately()"
to "MPEG2TransportStreamMultiplexor".
This function may be used by a downstream reader to test
whether the next call to "doGetNextFrame()" will deliver
data immediately. It can be useful if you want to decide
whether or not to keep accumulating multiple Transport Stream
'packets' into an outgoing RTP packet.
(Thanks to Gilles Chanteperdrix for suggesting this.)
* Made a minor syntactic change to "MediaTranscodingTable.hh"
to eliminate compiler warnings.
2016.06.23:
* Changed the constant "MAX_INPUT_ES_FRAME_SIZE" to a static
member variable
"MPEG2TransportStreamFromESSource::maxInputESFrameSize"
that can, if desired, be increased at run time (before a
"MPEG2TransportStreamFromESSource" object is created).
(Thanks to Gilles Chanteperdrix for motivating this.)
2016.06.22:
* Changed "~ProxyServerMediaSession()" so that it no longer
deletes the "MediaTranscodingTable" object that it had
been passed in its constructor. (The reason for this is
that the same "MediaTranscodingTable" can be used by more
than one "ProxyServerMediaSession".)
* Made the "parseTransportHeaderForREGISTER()" function
(that's used in the "RTSPServer" implementation) non-static,
so that it can be used in other, non-RTSP server
implementations that want to handle the "REGISTER" command.
* Made the "RTPSink::SSRC()" function "public:" rather than
"protected:".
(Thanks to Jean-Luc Bonnet for this suggestion.)
2016.05.20:
* Added a new virtual function "noteLiveness()" to the
"ServerMediaSession" class. This function is called
(by a "GenericMediaServer") whenever there's 'liveness'
on a "ClientSession". The default implementation of this
function is a 'noop', but subclasses can redefine it - e.g.,
if you want to remove long-unused "ServerMediaSession"s
from the server.
* Fixed a bug in the options handling for the command
"live555ProxyServer" that could erroneously produce a
"usage" error if the '-R' option is used, but no
back-end "rtsp://" URL is given.
2016.05.18:
* Backed out the change to "MultiFramedRTPSink" that was
made in 2016.05.17; the 2016.05.16 version turned out
to be correct.
* Rearranged "#include"s to avoid an 'excessive #include
nesting' error with some old compilers.
2016.05.17:
* Made a (mostly inconsequential) fix to the previous bugfix
for "MultiFramedRTPSink".
2016.05.16:
* Fixed a bug in "MultiFramedRTPSink" that affected
subclasses that redefine "frameSpecificHeaderSize()"
(for frame-specific headers that precede multiple frames
in a RTP packet). (Currently, the only subclass that
this affected was "VorbisAudioRTPSink".)
(Thanks to Gilles Chanteperdrix for reporting this bug.)
* Made a minor update to the "ProxyServerMediaSession" code
to better support optional media transcoding.
2016.04.21:
* Made it easier to set the MTU for all outgoing RTP
packets, instead of having to call "setPacketSizes()"
after each "MultiFramedRTPSink" is created. If you wish,
you can define the compile-time constants (macros)
RTP_PAYLOAD_MAX_SIZE and (optionally) RTP_PAYLOAD_PREFERRED_SIZE
when compiling "MultiFramedRTPSink.cpp". (These constants
have default values of 1456 and 1000 respectively, just as
before.)
* Updated "GroupsockHelper.{hh,cpp}" to (supposedly) support
'MinGW' better
2016.04.01:
* Fixed a bug the "ProxyServerMediaSubsession" code that
could cause an infinite loop if the 'back-end' server was
slow to respond to "SETUP" requests.
(Thanks to Erik Montnemery for helping to debug this.)
* Added support for parsing/streaming Matroska files that
contain PCM audio tracks.
(Thanks to Michel Promonet.)
2016.03.16:
* Added some more debugging fprintf()s to the
"ProxyServerMediaSubsession" code to try to track down a
bug.
* Simplified the "genMakefiles" script (moving duplicate
code into a 'for' loop).
2016.03.14:
* Updated the proxy server implementation to better handle
'front-end' clients that have asked to stream only some
of the substreams of a multi-stream session. Now, if a
substream is closed (because all 'front-end' clients have
stopped requesting it), but other front-end clients are
still streaming other substreams, then we will send - to
the 'back-end' server - only a substream-specific "PAUSE"
command; not a "PAUSE" command for the entire stream.
(Thanks to Lakshmi Narayanan for noting this issue.)
* Added an optional "-p <RTSP-port-number>" option to the
"LIVE555 Proxy Server", to allow the user to specify a
RTSP server port number other than the standard port
numbers: 554 and 8554. (These standard port numbers are
still tried if the specified port number can't be used.)
(Thanks to Denis Genestier for this suggestion.)
2016.02.22:
* Updated the "ProxyServerMediaSession" to add a Boolean
virtual function "allowProxyingForSubsession()".
By default, this always returns True. However,
subclasses can redefine this if they wish to restrict
which subsessions of a stream get proxied - e.g., if
you want to proxy only video tracks.
* Improved the "WAVAudioFileSource" code (for parsing
WAV-format audio files) to make it more tolerant of
unusual formats.
* Made it possible to build a version of the "liveMedia"
library that doesn't contain any RTSP server code;
e.g., if you are developing only a RTSP client, and
want to save space. To do this, omit any files that
contain "Server" or "RTPSink" in their name, and define
OMIT_REGISTER_HANDLING when compiling "RTSPClient.cpp".
(Thanks to Jeff Shanab for this suggestion.)
2016.02.09:
* Added an option "-E <absolute-seek-end-time>" to
"openRTSP".
(Thanks to Hans Maes for suggesting this.)
2016.02.08:
* Fixed a bug that was causing "playSIP" to crash.
(Thanks to Vilaysak Thipavong for reporting this.)
2016.01.29:
* Updated "QuickTimeFileSink" to make it usable with non-RTP
input sources. It still needs to have a "MediaSession"
that describes the input source; however, this input source
no longer needs to be RTP; it can, instead, be a UDP or
other type of source. (Of course, audio/video synchronization
and hint tracks can't be done in this case.)
* Changed the name of a variable in the "Makefile.tail" file
for the "BasicUsageEnvironment" project, in response to
a complaint that the old name clashed with something in
some Windows development environment
2016.01.24:
* Updated "ProxyServerMediaSession.cpp" to add some
'internal error' debugging fprintf()s to try to catch a
possible bug that was reported recently.
2016.01.20:
* When a server calls "startStream()" to start a RTSP stream
for a client, we now no longer make a slight adjustment to
the RTP timestamp sequence (using the "presetNextTimestamp()"
call) if there is already another ongoing stream using the
same "RTPSink". The effect of this is only minor, but it
ensures that the addition of an addition 'destination' to
an ongoing RTSP/RTP stream does not cause any change to the
contents of the RTP/RTCP packets.
(Thanks to Erik Montnemery for noting this issue.)
2016.01.16:
* This release has no source-code changes from the previous
release. However, a test file was mistakenly left in the
previous version; this produced an excessively-large tar file.
This has now been removed.
2016.01.12:
* Added a hack to "RTSPClient" to handle the non-standard
"com.ses.streamID:" header - used by 'SAT>IP' servers -
by using its value in the 'base URL' for subsequent requests.
(Thanks to Julian Scheel for proposing this.)
2015.12.22:
* Updated "QuickTimeFileSink" to add a sanity check to try
to prevent an occasional problem with H.264 video tracks
that contain 'sync frames'.
* Updated the "config.linux-with-shared-libraries"
configuration file to use the $(CC) and $(CXX)
macros, to allow for cross-compiling. (Thanks to Michel
Promonet.)
* Updated the years in the copyright notice on each file.
2015.11.09:
* Changed the "ProxyServerMediaSession" code once again. We
backed out the changes in the previous two releases,
and now respond to failures of the back-end "SETUP"
or "PLAY" commands by doing a full reset - which involves
deleting the "ProxyServerMediaSubsession" object, and
doing another "DESCRIBE" to create a new one. However, we
can't do this immediately - because the "SETUP" and "PLAY"
commands can be sent from within
"ProxyServerMediaSubsession::createNewStreamSource()".
Instead, we wait until the next 'liveness' command, which
will get sent immediately when we return to the event loop.
* Our proxy server code no longer converts the "mode" string
to lower case before passing it to
"MPEG4GenericRTPSink::createNew()". (This turned out to
be unnecessary, and was breaking some clients that weren't
treating this string as case-insensitive when they saw it
in the stream's SDP descriptor.) (Thanks to Craig Matsuura
for noting this issue.)
2015.10.29:
* Updated the fix in the previous revision to apply to the
back-end "PLAY" command as well as the back-end "SETUP"
command, because both of these back-end commands can get
sent from within
"ProxyServerMediaSubsession::createNewStreamSource()", so
we can't allow the "ProxyServerMediaSubsession" object
to get deleted in either case.
* Fri Oct 16 2015 aloisio@gmx.com
- Update to version 2015.10.12:
* The change that we made to the "ProxyServerMediaSession" code
in version 2015.07.31 (to reset the proxy server's state if
a back-end "SETUP" command fails) was too aggressive; it was
deleting the "ProxyServerMediaSubsession" object. This was
a problem, because "SETUP" commands can be called from within
"ProxyServerMediaSubsession::createNewStreamSource()".
Instead, we now deal with a failed back-end "SETUP" command
simply by resetting the 'back-end' connection. (Thanks to
Hardik Sangani for reporting this issue.)
- 2015.09.24:
* Fixed a bug in "RTSPClient" that could cause a crash if the
TCP connection was lost while resending a RTSP command.
(Thanks to ChaSeop Im for reporting this.)
* Moved some more generic 'media server' functionality from
"RTSPServer" to its parent class "GenericMediaServer".
* Added a new pure virtual function "getRTPSinkandRTCP()"
to "ServerMediaSubsession" to allow callers to get ('const')
access to a stream's "RTPSink" and/or "RTCPInstance" (and
thus their corresponding "Groupsock" objects) after the
stream has been created (using "getStreamParameters()".
* Updated "Groupsock" to allow for the possibility of there
being more than one 'destRecord' for each sessionId.
(This is something that doesn't happen in the normal case;
it's only a special case for WebRTC.)
- 2015.08.07:
* If a "RTCPInstance" happens to have both a source and a sink
(an unusual situation), we now include both "SR" and "RR"
reports in each outgoing RTCP report packet.
* When a "RTPSink" is being closed, we no longer turn off
background reading on its 'groupsock' (because, being a
"RTPSink", we never turned it on), just in case the
'groupsock' is also being shared with something else
(e.g., a "RTPSource") that does background read handling).
- 2015.08.06:
* Fixed a bug that would cause the destruction of a
"RTCPInstance" that was sharing a 'groupsock'
with a "RTPSource" (i.e., for multiplexed RTP and RTCP) to
stop the "RTPSource" from continuing to receive incoming RTP
packets. This normally wasn't a major problem, because the
destruction of the "RTCPInstance" was usually followed
immediately by the destruction of the "RTPSource".
However, it's also possible for the "RTPSource" to stay alive
long after the "RTCPInstance" is deleted; in this case things
will now work correctly.
- 2015.07.31:
* Fixed a minor memory leak in the "ProxyServerMediaSession"
code ("PresentationTimeSessionNormalizer"s and
"PresentationTimeSubsessionNormalizer"s weren't being deleted
properly). (Thanks to Dnyanesh Gate for reporting this.)
* Made the "ProxyServerMediaSession" code a bit more
bullet-proof, by resetting the 'back-end' connection if a
"SETUP" command fails. (Thanks to Craig Matsuura for providing
a real-world example of "SETUP" failing.)
* Fixed the 'estimated bitrate' values in
"testMPEG1or2VideoReceiver.cpp" and
"testMPEG2TransportReceiver.cpp" to match those in the
corresponding "test*Streamer.cpp" files.
(Thanks to Alex Anderson for reporting this.)
- 2015.07.23:
* Fixed a potential buffer overflow bug in "RTSPServer".
(Thanks to "an anonymous researcher working with Beyond
Security's SecuriTeam Secure Disclosure" for discovering this.)
- 2015.07.19:
* Fixed a bug in "RTPInterface::sendDataOverTCP()"; it was
disabling transmission on its socket if the "send()" call
failed. We now do this only if the error was not "EAGAIN".
(Thanks to Erik Oomen for bringing this to our attention.)
* Changed "QuickTimeFileSink" to try to work around an issue with
QuickTime sometimes complaining about the frame number in the
last 'sync frame' being 'out of range'.
* Changed the parameter signature for
"ProxyServerMediaSession::createNew()" (and the
"ProxyServerMediaSession" constructor) to take a
"GenericMediaServer*" rather than a "RTSPServer" as parameter.
This makes it possible to create proxy servers that use protocols
other than RTSP at the 'front-end'. (The 'back-end' protocol will
still be RTSP, however.)
* Defined a new class "MediaTranscodingTable" that can be used to
generate "FramedFilter" (subclass) objects that perform media
transcoding. Added a parameter of this type (with default value
NULL) to the "ProxyServerMediaSession" constructor and
"createNew()" function. This makes it possible to - if you wish
- add transcoding functionality to a proxy server. (This feature
is still experimental, and might be changed in the future.)
* Added optional "initialPortNum" and "multiplexRTCPWithRTP"
parameters to the "ProxyServerMediaSession" constructor - to be
passed to the "ProxyServerMediaSubsession" objects that it creates.
This allows subclasses to change these parameters if they wish.
* Updated "ProxyServerMediaSession" to make it possible for
subclasses to create subclasses of "Groupsock" and/or
"RTCPInstance", if they wish.
- 2015.06.25:
* Changed the definition of the "doEventLoop()" "watchVariable" to
make it 'volatile'. (Ditto for the "fTriggersAwaitingHandling"
field in the "BasicTaskScheduler" implementation.) This is to
alleviate a concern about aggressive optimizing compilers
possibly generating incorrect code. (Thanks to Remi
Denis-Courmont for bringing this issue to our attention.)
- 2015.06.24:
* Updated the implementation of "GenericMediaServer" to move the
code that removes and deletes all "ClientConnection",
"ClientSession", and "ServerMediaS(ubs)ession" objects from the
"GenericMediaServer" destructor to a member function "cleanup()".
This member function MUST be called from the destructor of any
subclass of "GenericMediaServer". (Putting this code in the
destructor of "GenericMediaServer" itself was a bug, because the
"ClientConnection", "ClientSession", and
"ServerMediaS(ubs)ession" objects may themselves have been
subclassed, and there may be a problem deleting them after the
"GenericMediaServer" subclass destructor has already been called.
(Thanks to Christopher Benne for noting this.)
* Fixed the way that "RTSPClient" handles responses to
"GET_PARAMETER" to properly allow for possible additional
pipelined responses appearing afterwards.
(Thanks to Paul Clark for identifying this problem.)
* Moved the "ClientSession" liveness checking/timeout mechanism
from "RTSPServer" to its new abstract base class
"GenericMediaServer". (The API and functionality of the
"RTSPServer" class remains unchanged.)
* Updated the "OnDemandServerMediaSubsession" code to make it
possible for subclasses to create and use subclasses of
"RTCPInstance".
* Undid the change that we made to "RTSPClient.hh" in the
previous version. There is no longer a demonstrated need to
make "RTSPClient::connectToServer()" virtual.
* Made a syntactic change to "MatroskaFile.cpp" to eliminate
some compiler warnings.
- 2015.06.21:
* Updated "RTSPClient" to put "port=" rather than "client_port="
in "Transport:" headers when requesting a multicast stream,
in accordance with RFC 2326.
(Thanks to Julian Scheel for noting this.)
* Updated "MultiFramedRTPSource" so that it doesn't deliver
0-length frames to the downstream object - in case the
downstream object interprets this as being an error.
(Thanks to Julian Scheel for the suggestion.)
* Made the member function "RTSPClient::connectToServer()"
virtual, in response to a request from a developer who wanted
to reimplement this in their "RTSPClient" subclass.
* Changed the "Groupsock::output()" function to no longer
take a 'TTL' parameter. (Instead, we now use the TTL (usually
255) that was provided when the "Groupsock" object was
created.)
* Cleaned up the "GroupEId" class that's used by "Groupsock".
(Previously, that class had some extra, experimental
functionality that turned out not to be useful.)
* Cleaned up the "destRecord" structure that's used in
"Groupsock" to represent the (possibly multiple) destinations
for each "Groupsock" object.
* Updated the "groupsock" library and
"OnDemandServerMediaSubsession" to better support (in some
future release) sockets whose destination endpoints are set
via STUN packet exchanges.
- 2015.06.11:
* Fixed a bug in "RTSPClient" that had accidentally been
introduced in version - 2015.06.04 that prevented "Session:"
headers from being included in some requests.
- 2015.06.10:
* Fixed the return type of the "createNewClientConnection()"
virtual function, redefined in
"RTSPServerSupportingHTTPStreaming".
* More changes to satisfy anal-retentive compilers.
* Removed the "DarwinInjector" code; that functionality has
not been supported for some time.
- 2015.06.09a:
* More changes to supposedly satisfy anal-retentive compilers.
- 2015.06.09:
* Added some "friend" declarations to "GenericMediaServer.hh"
and "RTSPServer.hh" in an attempt to placate an anal-retentive
Windows compiler. (Issue reported by Deanna Earley.)
- 2015.06.07:
* Restructured the "RTSPServer" class into an abstract base
class "GenericMediaServer" and a subclass "RTSPServer".
This makes it possible to develop other kinds of media server
that use the same "ServerMediaSession"/"ServerMediaSubsession"
objects to represent the stream(s) that they serve, but using
protocols other than RTSP.
* Added a new virtual function "createGroupsock" to
"OnDemandServerMediaSubsession". This makes it possible for
subclasses of "OnDemandServerMediaSubsession" to automatically
use subclasses of "Groupsock" (e.g., those that implement
STUN/DTLS).
* Moved the "ignoreSigPipeOnSocket()" function from
"RTSPCommon.hh" ("liveMedia" library) to "GroupsockHelper.hh"
("groupsock" library), because the function is not specific to
RTSP.
- 2015.06.04:
* Added optional support for including the RTSP "Speed:" header
in "PLAY" requests. (Thanks to Sarma Kolavasi.)
* Updated the implementation of "setResultErrMsg()" in
"BasicUsageEnvironment" to work properly in Windows.
(Thanks to Stas Tsymbalov.)
- 2015.05.31:
* Updated the "ProxyServerMediaSession" code to recover better
if a back-end RTSP "PLAY" command fails (for whatever reason).
Should this happen, we now reset the connection to the
'back-end' server. (This will cause the initial 'front-end'
client connection (that caused the "PLAY" command to be sent)
to fail, but subsequent 'front-end' client requests will now
have a better chance of succeeding.)
- 2015.05.28:
* Fixed a bug in error reporting in the "groupsock" library.
In a couple of places, we were using the result of
"getResultMsg()" directly in a call to "setResultMsg()", but
unfortunately those functions are implemented (at least in
"BasicUsageEnvironment") using the same buffer.
(Thanks to Stas Tsymbalov for reporting this.)
* Updated the "MPEGVideoStreamFramer" class (and thereby its
subclasses, including "H264VideoStreamFramer" to implement
the "doStopGettingFrames()" virtual function by calling
"flushInput()". This should fix a potential problem whereby
these classes might not work correctly if the downstream
reader calls "stopPlaying()", and then resumes reading.
(Thanks to Stas Tsymbalov for bringing this issue to our
attention.)
- 2015.05.25:
* Fixed a bug in "StreamReplicator::removeStreamReplica()":
It should have been calling "deactivateStreamReplica()"
* before* possibly deleting the "StreamReplicator" object
(if this was the last replica, and
"fDeleteWhenLastReplicaDies" was True).
(Thanks to Stas Tsymbalov for reporting this.)
* Fixed some potential problems with "StreamReplica"
deactivation. (Thanks to Stas Tsymbalov.)
* Updated the "RTSPServer" implementation to call
"ignoreSigPipeOnSocket()" on 'client connection' sockets,
rather than just on the main server socket. This is to
ensure that the server doesn't get killed if a client -
running on the same host - gets killed. (Note that, because
of this fix, it should never be necessarily to set the
"MSG_NOSIGNAL" flag on any of our calls to "send()".)
- 2015.05.12:
* Updated the previous revision to change the order in which
fields are deleted in the "RTSPServer" destructor, to avoid
a possible crash if "RTSPServer" objects are deleted.
(Thanks to ChaSeop Im for noting the problem.)
- 2015.05.03:
* Updated the "RTSPServer" implementation to fix a bug in
RTP/RTCP-over-TCP streaming. Before, if the
"RTSPClientConnection" object closed before the
"RTSPClientSession" object, and the TCP connection was also
being used for RTP/RTCP-over-TCP streaming, then the streaming
state (in the "RTSPClientSession") would stay alive, even
though the TCP socket had closed (and the socket number
possibly reused for a subsequent connection). This could cause
a problem when the "RTSPClientSession" was later reclaimed
(due to inactivity). Now, whenever a "RTSPClientConnection"
object is closed (due to the RTSP TCP connection closing), we
make sure that we also close any stream that had been using
the same TCP connection for RTP/RTCP-over-TCP streaming.
(Thanks to Kirill Zhegulev for noting this issue.)
* Removed extraneous comments near the top of
"testProgs/registerRTSPStream".
- 2015.04.22:
* Updated "config.iphone" and "config.iphone-simulator" to work
with the latest Xcode. (Thanks to Braden Ackerman.)
* Fixed a rare memory leak in "MultiFramedRTPSource" that might
occur if it's reading an incoming packet over TCP - requiring
>1 read for the packet - and the "MultiFramedRTPSource" gets
closed or paused while this is happening.
(Thanks to Kirill Zhegulev for noting this.)
- 2015.04.16:
* Added the "f" (force symbolic link) flag to the "ln" command
in the "make install" Makefile rules, in case you're
reinstalling the same version of a library.
(Thanks to Luca Ceresoli for noting the need for this.)
- 2015.04.15:
* Removed the previous (20 kByte) hard-wired limitation in the
size of incoming packets for "MultiFramedRTPSource". (Now,
any size packet up to the maximum size of 65535 can be
handled.)
* Added a (u_int16_t) field "desiredMaxIncomingPacketSize" to
"RTSPClient". If set to a value >0, then a "Blocksize:"
header with this value (minus an allowance for IP, UDP, and
RTP headers) will be sent with each "SETUP" request.
(Thanks to Deanna Earley for noting the optional RTSP
"Blocksize" header.)
- 2015.04.01:
* By default, "H264or5VideoStreamDiscreteFramer" sets
"fPictureEndMarker" (and thus the RTP 'M' bit) if the NAL
unit is VCL. Because this isn't always the right thing to do
(e.g., if we're delivering multiple 'slice' NAL units per
'access unit' (picture)), we now move this test into a virtual
function
"H264or5VideoStreamDiscreteFramer::nalUnitEndsAccessUnit()".
If desired, you can implement a subclass that redefines this
virtual function. (Thanks to Chris Richardson for bringing
this issue to our attention.)
* Made a minor syntactic change to
"ProxyServerMediaSubsession.cpp" to ensure that it compiles
with some old versions of VC++.
- 2015.03.19:
* Updated the "RTSPClient" code for handling a
"WWW-Authenticate:" header in a "401 Unauthorized" response.
We now check for the "stale=TRUE" parameter. If it's set,
then we resend the command, even if we already handled an
earlier "WWW-Authenticate:" header. (Thanks to Deanna Earley
for noting the need to handle "stale=TRUE".)
- 2015.03.16:
* Made a small change to the "BasicTaskScheduler"
implementation to reduce the likelihood of a race condition
with external thread(s) calling "triggerEvent()".
- 2015.03.06a:
* Oops - forgot to add '\0'-termination to the previous fix.
- 2015.03.06:
* Updated "RTSPClient" to decode %-encoded characters, should
they appear in the <username> and/or <password> fields in a
"rtsp://" URL. (Thanks to Deanna Earley for suggesting this.)
- 2015.03.01:
* Updated the "H264or5VideoRTPSink" implementation to make sure
that any stale fragmented data is flushed (discarded) if a
server's stream is paused. This ensures that - after we
resume from the pause - that we never stream data with old
presentation times. (Thanks to Gilles Chanteperdrix for
discovering and reporting this issue.)
- 2015.02.26:
* Fixed a bug in "ProxyServerMediaSubsession" that could cause a
crash if the parent "ProxyServerMediaSession" object is
removed from the RTSP server and deleted. (Thanks to Sergio ?
for first reporting this problem. Thanks to Chiung Ikhwan
for discovering the source of the bug.)
- 2015.02.23:
* Fixed a bug in
"OnDemandServerMediaSubsession::getCurrentNPT()".
(Thanks to Gilles Chanteperdrix for noting this.)
- 2015.02.17:
* Latest version of the "LIVE555 Streaming Media" code
(reinstalled due to a server crash).
- 2015.02.13:
* Oops - removed the "#define DEBUG" that had inadvertently
been left in "RTCP.cpp" in the previous version.
- 2015.02.12:
* Updated the previous release of "RTCP.cpp" to ensure that it
will compile for Windows.
- 2015.02.10:
* Added experimental support for sending RTCP "APP" packets,
and handling incoming RTCP "APP" packets. (Thanks to Nick
Ogden for suggesting this, and providing an example
implementation.)
- 2015.02.05:
* Made the "ProxyServerMediaSession" code a bit more
'bulletproof'.
- 2015.02.04:
* Fixed a bug in "DigestAuthentication" that could cause the
proxy server code to crash if it was given a username and
password for its 'back end' server.
(Thanks to Sergio Andrade for reporting this.)
* Fixed a minor bug in "MatroskaFileParser".
* Did some syntactic cleanup on a few files to avoid compiler
warnings with the newest version of "gcc".
* Sat Jan 31 2015 aloisio@gmx.com
- fixed paths in live555.pc
- update to version 2015.01.27:
* Fixed a bug in "MPEG2TransportStreamFromESSource" that could
sometimes cause an abort if more than one Elementary Stream
Source were multiplexed into a single Transport Stream.
(Thanks to Marc Palau for reporting this issue.)
- version 2015.01.19:
* Fixed an obscure bug in "RTSPClient" that might conceivably
have caused a crash if it received a completely empty RTSP
response.
- version 2015.01.04:
* Updated "config.iphone-simulator" to work with the latest Xcode.
(Thanks to Braden Ackerman.)
* In the "BasicUsageEnvironment" implementation, renamed
"EventTime" to "_EventTime" to avoid a reported naming conflict.
- version 2014.12.17:
* Updated "RTSPServerSupportingHTTPStreaming" to make sure that
the data stream source gets closed when it's no longer needed.
- version 2014.12.16:
* Changed the FD_SETSIZE check (introduced in version 2014.12.11)
so that it's not done in Windows (because in Windows,
FD_SETSIZE has different semantics).
(Thanks to Deanna Earley for reporting this.)
- version 2014.12.13:
* Updated the H.264/H.265 parsing code in "H264or5VideoStreamFramer"
to be a little smarter about how it computes a file's frame rate
(when streaming a 'raw' H.264 or H.265 file).
(Thanks to Michel Promonet for inspiring this.)
* Updated "config.iphoneos" to work with the latest Xcode.
(Thanks to Braden Ackerman.)
- version 2014.12.11:
* Changed our implementation of "setBackgroundHandling()" and
"moveBackgroundHandling()" in "BasicTaskScheduler" to check for
(and disallow) socket numbers >= FD_SETSIZE, because <sys/select.h>
has a bug (at least, in most systems) that causes buffer overflow
in this case. (Thanks to Michel Promonet for pointing this out.)
- version 2014.12.09:
* Needed to make the "QuickTimeFileSink" constructor and destructor
protected: to allow subclassing.
- version 2014.12.08:
* Fixed a bug in parsing 'absolute' RTSP "Range:" headers with no end
time. (Thanks to Ken Chow for reporting this.)
* Added a new option "-K" to "openRTSP, to tell the client to
periodically send "OPTIONS" requests as 'keep-alives' for buggy
servers that don't use incoming RTCP "RR" packets to indicate client
liveness. (Thanks to Peter Schlaile for this suggestion.)
* Added a new 'protected' virtual member function "noteRecordedFrame()"
to "QuickTimeFileSink". This function is called whenever a frame is
recorded to the output file. The default implementation of this
virtual function does nothing, but subclasses can redefine it if
they wish.
- version 2014.11.28:
* When "RTSPClient" parses a RTSP response, we first skip over any
blank lines that may be at the start of the response. This can
happen if the previous response (e.g., to a "DESCRIBE") contained
extra whitespace. (Thanks to ilwoo Nam for giving an example of
a server that exhibited this behavior.)
- version 2014.11.12:
* We had forgotten to initialize the "RTSPClient" member variable
"fAllowBasicAuthentication" that we introduced in the previous
version.
- version 2014.11.07:
* Added a new "RTSPClient" member function "disallowBasicAuthentication()"
that you can call if you don't want a RTSP client to perform 'basic'
authentication (whcih involves sending the username and password over
the network), even if the server asks for this.
(Thanks to Tomasz Pala for this suggestion.)
* Updated the debugging printout code in "RTCP.cpp" to identify all
known RTCP payload types, even if we don't currently handle them.
We also - when doing debugging printout - parse and print out
the contents of SDES RTCP packets.
- version 2014.11.01:
* Updated "RTSPClient" so that it reuses "fCurrentAuthenticator"
if we previously updated it with data from a "WWW-Authenticate:"
response, even if a non_NULL "authenticator" parameter was
passed as a parameter to the command. This reduces the number
of authetication exchanges that take place if the server asks
for authentication on more than one command in a RTSP session.
(Thanks to Tomasz Pala for this suggestion.)
* Updated "DigestAuthenticator" to allow for the possibility of
"username" or "password" being NULL.
* Updated the "RTSPServer" implementation to add an access check
before the first "SETUP" (the one that doesn't include a
session id), because it's possible, in principle, for a client
to send such a "SETUP" without first sending a "DESCRIBE".
Therefore, we need to perform access checks on both commands.
- version 2014.10.28:
* Added support for the VP9 video RTP payload format (sending and
receiving), including the demultiplexing and streaming of a VP9
video track from a Matroska-format file.
* Made "VP8VideoRTPSource" more robust against a bad first-byte
header field in the payload.
- version 2014.10.21:
* Increased the max output packet size for "MultiFramedRTPSink"
and "RTCPInstance" from 1448 to 1456, because we had a report
of problems when proxying incoming JPEG/RTP packets of this
size (and because 1456 bytes still gives a packet size of no
more than 1500 bytes when we add
in IP, UDP, and UMTP headers).
- version 2014.10.20:
* Increased the RTSP request and response buffer sizes from 10000
to 20000 bytes, because we saw a RTSP stream (VP8 video) that
had an extremely large "configuration=" string that was hiting
the previous limit.
- version 2014.10.16:
* Fixed the "RTSPServer" implementation to handle a rare race
condition that could cause a "ServerMediaSession" object to
be deleted while it was being used to implement "DESCRIBE".
(Thanks to Michel Promonet for reporting this.)
- version 2014.10.07:
* Fixed a bug in the "MultiFramedRTPSource" implementation where
we weren't properly checking the size of incoming RTP packets
that have the "CC" field (i.e., number of "CSRC" fields) non-zero.
* Updated "Groupsock::output()" to be a virtual function.
(This makes it possible to implement "Groupsock" subclasses that
implement 'bump-in-the-stack' protocols (such as SRT(C)P) below
RTP/RTCP.)
- version 2014.10.03:
* Fixed a problem in the "timestampString()" routine that occurs
if "time_t" is 64 bits, but we're on a 32-bit machine.
(Thanks to Deanna Earley for reporting this.)
* Updated the debugging output code in "RTCP.cpp" to make it
clearer that SDES and APP packets are not invalid; just not
(yet) handled by us.
* Wed Oct 29 2014 olaf@aepfle.de
- BuildRequire pkg-config to get rpm Provides/Requires pkgconfig(live555)
/usr/bin/MPEG2TransportStreamIndexer /usr/bin/live555HLSProxy /usr/bin/live555MediaServer /usr/bin/live555ProxyServer /usr/bin/mikeyParse /usr/bin/openRTSP /usr/bin/playSIP /usr/bin/registerRTSPStream /usr/bin/sapWatch /usr/bin/testAMRAudioStreamer /usr/bin/testDVVideoStreamer /usr/bin/testH264VideoStreamer /usr/bin/testH264VideoToHLSSegments /usr/bin/testH264VideoToTransportStream /usr/bin/testH265VideoStreamer /usr/bin/testH265VideoToTransportStream /usr/bin/testMKVSplitter /usr/bin/testMKVStreamer /usr/bin/testMP3Receiver /usr/bin/testMP3Streamer /usr/bin/testMPEG1or2AudioVideoStreamer /usr/bin/testMPEG1or2ProgramToTransportStream /usr/bin/testMPEG1or2Splitter /usr/bin/testMPEG1or2VideoReceiver /usr/bin/testMPEG1or2VideoStreamer /usr/bin/testMPEG2TransportReceiver /usr/bin/testMPEG2TransportStreamSplitter /usr/bin/testMPEG2TransportStreamTrickPlay /usr/bin/testMPEG2TransportStreamer /usr/bin/testMPEG4VideoStreamer /usr/bin/testOggStreamer /usr/bin/testOnDemandRTSPServer /usr/bin/testRTSPClient /usr/bin/testRelay /usr/bin/testReplicator /usr/bin/testWAVAudioStreamer /usr/bin/vobStreamer
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