- Nov 19, 2018
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Adds ability to retrieve the shm renderer info from a call id, in case the client is launched mid call (else it can't show the call). SinkClient now keeps track of its width and height. Change-Id: Ie43c196c60de5e22825fc71ff404e99bbfbe9402 Gitlab: #59 Reviewed-by:
Sebastien Blin <sebastien.blin@savoirfairelinux.com>
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Change-Id: I5f8fb9802290974a9cc74f535ff22365b56e50c6
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- Nov 16, 2018
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Adrien Béraud authored
Change-Id: Iab6a12cca382c628dc7d3e983259fe2ce1010851
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Adrien Béraud authored
Improve loading performance when having more than one Ring account by loading Ring accounts configuration in parallel. Change-Id: Ic39bc24551a20be2e98ee89221e8eb6e66cb09fd
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Andreas Traczyk authored
Change-Id: Ieb4821822ecfb1bacfee5ad4abec96f861b63950
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- Nov 08, 2018
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Philippe Gorley authored
Adds support for single- and double-precision floating point samples in the WavWriter. Change-Id: I62f7dac3989b176a39d77882bafcfcae38a733f1
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Philippe Gorley authored
Can be used to call setInterruptCallback on encoders or decoders. Change-Id: I7b17aa93f211936f67ca237e5fea3f266ae6ca83
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Philippe Gorley authored
Audio time bases are always the inverse of the sample rate, meaning the increment is always the number of samples in a frame. Video usually has a time base inverse of its framerate, but this is not always the case. Change-Id: I50d2d84d073052f8b3a832e8b99725b9d66b12a8
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Philippe Gorley authored
Needed for audio streaming because the audio input needs to be accessed at call level so it can tell the input to switch to a file or a different input. Getting a shared pointer to an AudioInput should only be done during setup, so as to not walk the whole map in audio processing loops. Change-Id: I49be1cb3c641b50e6f70356f330d40e1c27bef61
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Most of the time, the frame resolution is unchanged, and the scaler only needs to convert pixel formats. Also avoids this crash when size is the same: https://trac.ffmpeg.org/ticket/5356 Change-Id: I4c1c3e0a8b5bba8c937317074e8a9c5652fca407 Gitlab: #58 Reviewed-by:
Sebastien Blin <sebastien.blin@savoirfairelinux.com>
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Don't bump version, as API is not broken. Breaks ABI, but we do not set the SONAME tag, meaning we don't have an ABI version. Change-Id: I5bc436f00e20eca7a789b0d8f5863d6622f82d0f Reviewed-by:
Sebastien Blin <sebastien.blin@savoirfairelinux.com>
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- Nov 01, 2018
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Philippe Gorley authored
AudioInput now manages audio frames, not AudioSender. Makes AudioInput observable. AudioSender subscribes to AudioInput. LocalRecorder does not need to observe AudioInput or VideoInput because these already record frames. Will be done in future work on the recording system. Change-Id: I011d742063386498d59b2962f7c333b999d0921c
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Philippe Gorley authored
Change-Id: I951cc45a6dda6c13c10e2dedb41fb2d19e8b0206
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- Oct 30, 2018
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There is currently no distinction between the "peer busy" and "we replied busy after timeout" states, since both end in the BUSY state. Add a new PEER_BUSY state allowing such a distinction: * PEER_BUSY is set when peer replied busy * BUSY is set when we replied busy to an incoming call Bump daemon API number to major 7.0.0 since this is breaking the current API. In fact, these changes should not break anything in any well implemented client because unknown states should be properly handled, but better check. Change-Id: Id83f6db3d4524a91951b9945797f5fd2c019ff2f Reviewed-by:
Sebastien Blin <sebastien.blin@savoirfairelinux.com>
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- Oct 26, 2018
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Adrien Béraud authored
Change-Id: Id0fab10de383e71920959f5c025d0f2710f99d86
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- Oct 24, 2018
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Change-Id: I047f0f1a87466e440cafdb0984b311b8bcb5e5ed Reviewed-by:
Andreas Traczyk <andreas.traczyk@savoirfairelinux.com>
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Change-Id: I86f08805d93c85eafa88c2a4fc61bcd092e1ea57
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- Oct 23, 2018
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Adds unit tests for VideoScaler. Change-Id: I38fc5edbd37ca11196c72db8aed7927d6371b678
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- Oct 22, 2018
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Philippe Gorley authored
Returns AudioFrame instead of raw AVFRame pointer when calling AudioBuffer.toAVFrame, and take AudioFrame as parameter when appending an AVFrame to an AudioBuffer. No longer need to free AVFrame during encoding/conversions. Change-Id: I28aa992a5483f84f6cb1a5157718c11c3a69518c
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Philippe Gorley authored
RingBufferPool::getData overrides the format of the AudioBuffer, so resampling may be required. Protects AudioFormat with a mutex to avoid data races. Change-Id: I0a9d8686f142c192f912887b175a42bacb0c1a57
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Philippe Gorley authored
Resolution/codecs won't change without a SIP renegotiation. Change-Id: Ie6d393caa5a3e631c7be14356a7929c0f11e781c
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Adrien Béraud authored
* Update restbed to latest master * Removed dropped Kashmir dependency * Removed merged/obsolete patches (locale, async read, strand) * Use branch including necessary findopenssl changes for contrib Change-Id: I6cdf7de1005f82abeac77eb6bfb0002df83c8017 Reviewed-by:
Philippe Gorley <philippe.gorley@savoirfairelinux.com>
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Adrien Béraud authored
Change-Id: I93cc9a7e2c365bf84f64ef6ffbd05131b76ccfae
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- Oct 19, 2018
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Change-Id: I6d01dc501cad775d7561eaf1857393c79dfa2365
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Change-Id: I0f8f9571a715fe26a7db61a43675efc50ce71564
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Change-Id: I7af6bc775b08fc478704dfc5b595677a40c13a8f
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SIP and TURN tests fail sporadically, and all recent additions to the test folder have been under the unitTest folder. Change-Id: Icac4827f53da3fbd72492cdfb024a003decbbbec
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Change-Id: I4c70799ce157e97001e85052987835781d5152a7
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Time base for audio should always be the inverse of the sampling rate. Change-Id: I335a4bab1c29b5f0411dcb38a3bab379b48fa433
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- Oct 16, 2018
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Philippe Gorley authored
Makes it only visible to audio_input's translation unit during compilation, instead of a class member. Change-Id: Id4d5d6d134b244aca83d88825c939e10840d2131
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Philippe Gorley authored
Audio will soon be implemented using Observer/Observable, so move these classes somewhere more sensible. Change-Id: Iff0ca61a4c73c1ed7fa82ed91f8ca2d11afd65d9
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Philippe Gorley authored
Also avoids race conditions when stopping the recorder Change-Id: I498b2f9802e647a05410bd5bcb5cf4248a02bcf4
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Philippe Gorley authored
RingBufferPool has all the methods necessary to interface with RingBuffer. Change-Id: I04b7cb0ac9b7033e1f1047d7f00708eb987bfdc7
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- Oct 13, 2018
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Makes use of AVFrame's reference counting if possible instead of copying the data. Only bumps minor version because MediaFrame et al weren't in the ABI until now. Change-Id: I692e76230ed057c1ad4e46ab59ea5cfd163fb98d
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- Oct 10, 2018
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Philippe Gorley authored
AVFrame.linesize may contain some padding, use AVFrame.nb_samples instead. Change-Id: I5e46e89ac102b8dfd939c9ccbc3f51e73d995c6e
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- Oct 02, 2018
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CMake build fails on MinGW when trying to find argon2 Change-Id: Ibe9451cfd324cbfd196434e3cbf8f01e1ae41a1d Reviewed-by:
Andreas Traczyk <andreas.traczyk@savoirfairelinux.com>
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- Sep 30, 2018
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Adrien Béraud authored
Change-Id: I9bf15b7bffb89773a95963f662e05df7040f5f1a
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- Sep 27, 2018
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Andreas Traczyk authored
- critical for building with iPhoneOS 12.0 SDK Change-Id: I89f012d83d0403fb15b9071a28d1bb60a5ffb681
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Hugo Lefeuvre authored
Busy tone is broken by design: since there is no way to play a tone for a given amount of time, callFailure() was calling stopTone() just after playATone(), meaning that the tone was only played for a few milliseconds. callBusy() was even more broken because it was calling checkAudio() just after playATone(), meaning that (1) the tone would only play for a few ms and (2) the tone would start again when the audio layer is restarted. Even worse, callBusy was not calling stopTone(), meaning that if an incoming call tone was being playing played then it would never be stopped. This feature might be reintroduced for accessibility purposes at some point in the future, along with a refactoring of the tone system in the daemon. Change-Id: I15957e050688bfe5f5ce84f971d5a14b875028b9 Reviewed-by:
Sebastien Blin <sebastien.blin@savoirfairelinux.com>
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- Sep 26, 2018
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Change-Id: I21e3a2fade975f03fbdbde84f6fe6e94b8f51926 Reviewed-by:
Philippe Gorley <philippe.gorley@savoirfairelinux.com>
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