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| Name: live555 | Distribution: openSUSE:Factory:zSystems |
| Version: 2023.05.10 | Vendor: openSUSE |
| Release: 1.1 | Build date: Tue Jun 6 20:12:22 2023 |
| Group: System/Libraries | Build host: s390zp23 |
| Size: 536456 | Source RPM: live555-2023.05.10-1.1.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
* Mon May 29 2023 Dirk Müller <dmueller@suse.com>
- update to 2023.5.10:
* Fixed a minor memory leak in the "RTSPServer" code.
* Calls to "send()" and "sendto()" now explicitly take
"MSG_NOSIGNAL" rather than 0 as the 'flags' parameter.
In most systems, 0 seems to work, but apparently not in
Debian Linux.
* 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.
/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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