deviceAnnounced_ should be cleared if unregistered, to be valid for
next annoucement

Change-Id: I35d62e01d11780d4b30c312ec695b9cb0321d618

Mohamed Chibani authored 3 years ago and Sébastien Blin committed 3 years ago

When processing an incoming or outgoing re-invite, the ice media
session and its associated RTP sessions are kept until the
new ice media session is successfully negotiated.

Gitlab: #666
Change-Id: I608d241d82b346b2c190ada046eed86dd86a7178

Olivier Dion authored 3 years ago and Sébastien Blin committed 3 years ago

Change-Id: I028985ac90d07199cfa30d33e21f55ebef228ff3

Olivier Dion authored 3 years ago and Sébastien Blin committed 3 years ago

Change-Id: Ic25a41cb28ae1e8b8dc51a149c304edc7f04ef50

Olivier Dion authored 3 years ago and Sébastien Blin committed 3 years ago

Change-Id: I3aae03770f302bfe043f7bf6070a99a0d56fcebc

Olivier Dion authored 3 years ago and Sébastien Blin committed 3 years ago

GOOPS is Guile Object Oriented Programming System.

It's easier to extend and break less the scenarios using GOOPS.  e.g., a
scenario published on Gitlab should work two weeks later.  This is not the case
right now and the goal of using GOOPS is to offer a stable API for the agent,
while keeping it flexible.

Change-Id: If6a038b6d8c371f8e74849749770f1dec8559b91

Change-Id: I1e8e88f993d8fa66224ea1bcd48d6e9fbcb39d5e

callbacks, such as transaction_request_cb uses the sipTransportBroker
object. So, sipTransportBroker MUST be destroyed after that every
callbacks are called, so after the destruction of endpt_

Change-Id: I795e02544abf9188db4971d7465ac537b1a568d9
GitLab: #681

Olivier Dion authored 3 years ago and Sébastien Blin committed 3 years ago

Change-Id: I3f7b3fd9e5bea986f5b600273ae4ddb07bdd787a

Change-Id: I02ccb3c2d05aaa14a7f42dd0c6e2854f9905ccf2

Change-Id: I2496bc42192ad65135c781476f6f5d4bcc795d19

Change-Id: I564d9d68f73c20a3105bfd706627b6e3db02aaf5
GitLab: #238

Add a media negotiation unit test for auto-answer mode

Gitlab: #645

Change-Id: I37f768ce48e078fbd95a2c9b28997877a4dc468d

if videoRtp is not initialized, the information is dropped. So,
it should be stored to avoid any drop of informations.
Note: in the future the container should change, because rotation
is currently applied to all video streams.

Change-Id: I036ffe7b6a248a3d8d7defeff1be78e008447fbd
GitLab: #678

Mohamed Chibani authored 3 years ago and Sébastien Blin committed 3 years ago

Currently, the media attribute in the SDP is always set to
'sendrecv' regardless of the mute state of the media.
In this patch, media direction will be set according to mute
state of the media.
Note that this only applies if the mute/unmute requires media
renegotiation (SIP re-invite with new SDP session). Currently,
this only the case for video media. For audio, mute/unmute is
done locally without SIP re-invite.

References:
RFC-3264
RFC-4317 (non-normative)

Gitlab: #645

Change-Id: I604331255bd25dfe732e192039a673a0980105fa

This code is unused since we introduced processRtcpChecker()

Change-Id: If86295d885997c415e8f1681c6b9ad049c93de9e

getVideoInput is called in two different place:
+ openVideoInput from the client
+ VideoRtpSession::startSender() from the daemon
Nothing is blocking both sides to do it at the same time.
In this case, two VideoInputs can be created, causing the second
one to fail because the device is busy (used by the first one)
and causing weird results (i.e. sometimes the video is not shown).

Change-Id: I23e34a95efe4972fbe111c19f0f0989f5e8ae8a9

Sometimes, buffer with incorrect capacity were injected into
the audio, causing the daemon to crash.
Before the stream's split, stopStream was flushing the buffers.
In this patch, stopStream will correctly re-init the buffers and
avoid to inject incorrect buffer into the audio.
Also, remove unused value in buf_manager.h and clean some warnings
for log.
Finally, modify engineServiceRing to avoid many push operations
and make it similar to engineServicePlay

Change-Id: I589cea814452d3dbf5172af6369e448dcd9d5c72
GitLab: #676

Kateryna Kostiuk authored 3 years ago and Kateryna Kostiuk committed 3 years ago

Because media encoder flush() involves encoding we should change
initialized flag after calling flush(), otherwise, it could lead to
new stream initializing.

jami-client-macos#299

Change-Id: I88d5648a6f252bd78776f505f891f09a4e5b12e1

Aline Gondim Santos authored 3 years ago and Sébastien Blin committed 3 years ago

- tranport -> transport

Change-Id: I1991ba7324c22625e8b6cdf568357e3ff6540557

Change-Id: I3f0015d5f492137592af5d6c247ca439dd75dd38

Using JAMI_TIMESTAMP_FMT, developers can now format the log's timestamp
according to their needs and taste.  The formatting is the same as strftime(3).

Change-Id: Ibea56852b2efc37f66aeeeda857e307130099720

Mohamed Chibani authored 3 years ago and Sébastien Blin committed 3 years ago

Decouple the creation and the initialization of media coding
sessions to ease the re-initialization of encoding session
(when resolution changes for instance).

Gitlab: jami-project#1294

Change-Id: Id5927d4aeb29bc9d4fe806bde6dc9bd28a0e6fe2

Sébastien Blin authored 3 years ago and Adrien Béraud committed 3 years ago

Change-Id: I547dfeaf1734873ac9f8db82a181769de11a2bb6

Olivier Dion authored 3 years ago and Sébastien Blin committed 3 years ago

Change-Id: I8a02ae82cb2cf1494a13b263aca0555ab84ed98f

Olivier Dion authored 3 years ago and Sébastien Blin committed 3 years ago

Change-Id: I2fca281e9144d06243ca09879a1d86551ccd1996

Olivier Dion authored 3 years ago and Sébastien Blin committed 3 years ago

Change-Id: If91bed7badd8229aee216a91445b758e8901e573

This patch fix 3 little issues:
+ "device" was always empty. Uses the transport to fill the device
field. This will be used in the next version of the protocol.
+ raiseHand was bugguy in two ways:
    + It was only usable by moderators
    + Somebody was able to change the raiseHand for other peers

Change-Id: I32bf37c5063973a5a5962dd8984c87fb05d33a3d

Change-Id: I82bde67fed7a295e888c2ea8f96c6e4646f51e64

Kateryna Kostiuk authored 3 years ago and Kateryna Kostiuk committed 3 years ago

Change-Id: I2a22e80a8ea826641afef9a82bf3b17ca35fcaed

Sébastien Blin authored 3 years ago and Mohamed Chibani committed 3 years ago

Caught by ut_media_encoder:

READ of size 8 at 0x602000029818 thread T0
    #0 0x5649e914d6e0 in jami::MediaEncoder::send(AVPacket&, int) media/media_encoder.cpp:504
    #1 0x5649e914d05c in jami::MediaEncoder::encode(AVFrame*, int) media/media_encoder.cpp:480
    #2 0x5649e8cac119 in jami::test::MediaEncoderTest::testMultiStream() media/test_media_encoder.cpp:183

outputCtx_->nb_streams can be different from encoders.size(), and if
not checked can lead to a crash

Change-Id: I3c0217ec3e1d50950bf46989e0dd6797fee8972e

Lock was:
    frame #3: 0x00000074f22d0b74 libc++_shared.so`std::__ndk1::mutex::lock()
    frame #5: 0x00000074f2f91b10 libring.so`std::__ndk1::lock_guard<std::__ndk1::mutex>::lock_guard()
    frame #6: 0x00000074f3156d3c libring.so`jami::video::SinkClient::registerTarget() at sinkclient.h:80:37
    frame #7: 0x00000074f3154f74 libring.so`DRing::registerSinkTarget(sinkId="2920319278288091") at videomanager.cpp:527:15
    frame #8: 0x00000074f2f1dccc libring.so`Java_net_jami_daemon_JamiServiceJNI_unregisterVideoCallback()
    frame #45: 0x00000074f2f3491c libring.so`SwigDirector_VideoCallback::decodingStopped()
    frame #48: 0x00000074f319a480 libring.so`void jami::emitSignal<DRing::VideoSignal::DecodingStopped>() at ring_signal.h:66:13
    frame #49: 0x00000074f31993ec libring.so`jami::video::SinkClient::setFrameSize(this=0x00000074c298caa0, width=0, height=0) at sinkclient.cpp:482:9
    frame #50: 0x00000074f3199f74 libring.so`jami::video::SinkClient::update() at sinkclient.cpp:427:13
    frame #51: 0x00000074f316c69c libring.so`jami::Observable<std::__ndk1::shared_ptr<DRing::MediaFrame> >::notify() at observer.h:138:23
    frame #52: 0x00000074f31a7094 libring.so`jami::video::VideoGenerator::publishFrame() at video_base.cpp:56:5


Change-Id: I1824bad767543a3e789e13af1489613a044a7473

Adrien Béraud authored 3 years ago and Sébastien Blin committed 3 years ago

Change-Id: I274ed5b9b6f97b06e7a355b941f81f86c7816c24

This allows us to build with webrtc-audio-processing distro packages
currently available across supported Debian releases.

Change-Id: I25584a0564ceb78e8eefec78e5e903356c37ddab

This fix ut_media_encoder
Also avoid crash when outputCtx_ is nullptr and the format is unrecognised.

Change-Id: I6df5a2e229e0313ea2b3c34b37f9f2900058c75f

This fixes Android' accelerated video encoding

Change-Id: Ie0cc7f3699f8c74eb205a395ae4235b1b1a7af89

This allow to generate jamid for dbus usage and client-qt with
libwrap.

Change-Id: Ia23a0cec2efb4347166e0123c662da6c134c2238
GitLab: jami-project#1312

+ If in screen sharing, the hardware acceleration encoding is disabled

GitLab: jami-project#1294

Change-Id: I24982b4454d724bb86059a50a3c2d01fe6fab147

Because some callbacks can take a long time (accepting a SIP channel
will starts to sync profiles, ask for new channel, etc), control packets
are handled in their own thread. However, if the peer accepts a request,
it can starts to use it and send data on it. When finished, it will close
the socket and both peer will remove the channel.
In some case, the channel can be removed before
MultiplexedSocket::Impl::onAccept, causing upper layers to handle a
bad channel. In this patch, we removes the channel only when upper layers
know that the channel is accepted and ready to use. Also, onAccept()
doesn't create wrong channels (which was a bug, where isInitiator() was
in the wrong state).

GitLab: #659
Change-Id: I4f45feacc2624ce0135ace09f7e0188b56fcf81f

Else, the module can be created twice and one will be invalidated
causing some crashes

Change-Id: I1403fc27e3baa24c1ad6e4c8ae5a352e328374ee