6b2919ed789b89ca38882cd5d44efe0349e41fc8: Bug 1376873 - Fix fake h.264 encoder; r=pehrsons
Dan Minor <dminor@mozilla.com> - Mon, 16 Jul 2018 11:50:40 -0400 - rev 444250
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Fix fake h.264 encoder; r=pehrsons Upstream webrtc.org has added more stringent checks for SPS and PPS identifiers. This breaks out the SPS and PPS NALUs into separate binary blobs to ensure that the identifiers are zero. It is also necessary to set pragma pack so that the structure does not contain values which trick the parser into thinking a new NALU has begun. The ENCODED_FRAME_MAGIC is changed for a similar reason. All of the constants were determined by running mochitests with the actual h.264 encoder. For similar reasons, this also changes things so that only IFrames are sent, as the upstream code is now checking for previous IFrame identifiers and will drop frames if they are not sent. The SPS and PPS NALUs are now embedded into a single frame rather than being sent as separate frames as was previously done. This is consistent with the real H.264 plugin, and fixes a problem with intermittent failures due to occasionally bad timestamps. Differential Revision: https://phabricator.services.mozilla.com/D7447
da8d3eb163bdd7f836723c434bf3a3cf0b76a02c: Bug 1376873 - Update videoconduit_unittests gtests; r=pehrsons
Dan Minor <dminor@mozilla.com> - Fri, 16 Feb 2018 09:47:19 -0500 - rev 444249
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Update videoconduit_unittests gtests; r=pehrsons Differential Revision: https://phabricator.services.mozilla.com/D7446
0fc8c65e163ed5ccfc09c55ec906973cf35a79b4: Bug 1376873 - Use audio/video sync groups; r=bwc
Dan Minor <dminor@mozilla.com> - Fri, 27 Apr 2018 07:25:19 -0400 - rev 444248
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Use audio/video sync groups; r=bwc This uses sync groups in the receive stream configs for the conduits rather than establishing sync through direct calls. When the streams are created in call.cc, ConfigureSync is called, which results in SetSync being called on the video stream which is the replacement for SetSyncChannel in branch 57 of webrtc.org. With the current code, a video stream can only be synchronized to a single audio stream. Using sync groups enforces a stronger constraint that only one pair of audio and video streams can be synchronized for each sync group. The comments in call.cc imply this is what is supported by webrtc.org, it seems safer to also follow this constraint rather than circumvent it by calling directly into the underlying code. Differential Revision: https://phabricator.services.mozilla.com/D7445
de3f5bd6846bc9aca8ae1ebb47cd9d365fe52565: Bug 1376873 - Fix GetRTCPSenderReport; r=ng
Dan Minor <dminor@mozilla.com> - Wed, 07 Feb 2018 15:00:17 -0500 - rev 444247
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Fix GetRTCPSenderReport; r=ng Differential Revision: https://phabricator.services.mozilla.com/D7431
2951112ade46b0daba1d6c95bd44f1711ba6575a: Bug 1376873 - Remove CPULoadStateObserver; r=ng
Dan Minor <dminor@mozilla.com> - Thu, 19 Apr 2018 13:37:51 -0400 - rev 444246
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Remove CPULoadStateObserver; r=ng We went through a lot of trouble to plumb the CPULoadState down to MediaOptimization, but the value is not actually used for anything, at least since the Branch 57 update. This removes the plumbing, since it seems we are getting along ok without it. Differential Revision: https://phabricator.services.mozilla.com/D7443
9fe47f8ea1b861a0d994988edd977b39852ab738: Bug 1376873 - Disable Mid support in RtpDemuxer; r=mjf
Dan Minor <dminor@mozilla.com> - Tue, 27 Mar 2018 15:43:30 -0400 - rev 444245
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Disable Mid support in RtpDemuxer; r=mjf The only use of Mid in the current webrtc.org code is in the unit tests. RtpStreamReceiverController only allows adding sinks using SSRCs. Because of this, we'll end up dropping packets in the RtpDemuxer with the current code as none of our Mids will be recognized. Tip of webrtc.org fully supports using Mids, so we'll be able to enable this code again after the next update. Differential Revision: https://phabricator.services.mozilla.com/D7442
b949356616223660996acf39e1e353e7bbd2a60d: Bug 1376873 - Use Call interface in AudioConduit; r=padenot
Dan Minor <dminor@mozilla.com> - Tue, 21 Aug 2018 13:39:53 -0400 - rev 444244
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Use Call interface in AudioConduit; r=padenot Differential Revision: https://phabricator.services.mozilla.com/D7441
6eca45c8313156fdf460e516281885ed7cbf2306: Bug 1376873 - Disable or replace thread check assertions; r=pehrsons
Dan Minor <dminor@mozilla.com> - Mon, 26 Mar 2018 16:19:23 -0400 - rev 444243
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Disable or replace thread check assertions; r=pehrsons The changes to Call are required because we create the Call object on the main thread, but deliver packets and query stats from the socket thread. The changes to ChannelProxy are required because we query stats from the socket thread rather than the main thread. For RtpVideoStreamReceiver, this removes the worker_task_checker_ assertions and replaces them with a critical section. This is how the code worked prior to this update. We create the Call object (and thus eventually the RtpVideoStreamReceiver) on the main thread, but we want to deliver packets on the socket thread. To retain these assertions we'd either have to dispatch calls to deliver packets from the socket thread to the main, which seems pretty bad from a performance point of view, or we'd have to refactor the code to create the Call object on the socket thread, which seems like a major refactoring best done outside of a branch update. Going back to the previous behaviour seemed like the least bad alternative. Differential Revision: https://phabricator.services.mozilla.com/D7440
909f472c2e402f81a8326e93e839666d2aba1fc5: Bug 1376873 - Add RtpPacketQueue; r=pehrsons
Dan Minor <dminor@mozilla.com> - Thu, 22 Mar 2018 09:44:49 -0400 - rev 444242
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Add RtpPacketQueue; r=pehrsons We'll need to queue rtp packets in both the AudioConduit and the VideoConduit. This adds a class to manage the packet queue to reduce duplicated code between the conduits. Differential Revision: https://phabricator.services.mozilla.com/D7439
19d9a30b300c4653088bffa1598c26f6f16c427d: Bug 1376873 - Updates to dom/media/, dom/media/systemservices and dom/media/webrtc; r=pehrsons
Dan Minor <dminor@mozilla.com> - Tue, 20 Feb 2018 15:23:09 -0500 - rev 444241
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Updates to dom/media/, dom/media/systemservices and dom/media/webrtc; r=pehrsons Differential Revision: https://phabricator.services.mozilla.com/D7438
6a52b92ff03731a1e2ef13aa7f7349a1e7e48bee: Bug 1376873 - Mediapipeline updates; r=pehrsons
Dan Minor <dminor@mozilla.com> - Fri, 16 Feb 2018 09:15:44 -0500 - rev 444240
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Mediapipeline updates; r=pehrsons Support for native_handle() has been removed by upstream. Differential Revision: https://phabricator.services.mozilla.com/D7437
6a6771656fe550f2bef1f87dcf2ccea8c52a03c5: Bug 1376873 - Update mtransport sigslot.h; r=bwc
Dan Minor <dminor@mozilla.com> - Thu, 15 Feb 2018 15:14:58 -0500 - rev 444239
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Update mtransport sigslot.h; r=bwc This updates the copy of sigslot used by mtransport to match that used by webrtc. Differential Revision: https://phabricator.services.mozilla.com/D7436
ffdbeba93fa632a830138cb79c53cfc15bf1f13d: Bug 1376873 - WebrtcMediaDataDecoderCodec updates; r=pehrsons
Dan Minor <dminor@mozilla.com> - Tue, 21 Aug 2018 13:40:45 -0400 - rev 444238
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - WebrtcMediaDataDecoderCodec updates; r=pehrsons This makes the ImageBuffer class implement VideoFrameBuffer class and makes it available in MediaPipeline. To avoid exposing MediaPipeline to a lot of internal details of WebrtcMediaDataDecoder, ImageBuffer is moved to its own header file. We're required to implement a ToI420() method. I've re-used the current implementation of NativeToI420Buffer, which requires more development in order to work properly as the Image class only supports RGB readback. Differential Revision: https://phabricator.services.mozilla.com/D7435
d12ad321047ded2be7ed44353e0b531e8d2c5049: Bug 1376873 - VideoConduit updates; r=pehrsons
Dan Minor <dminor@mozilla.com> - Tue, 21 Aug 2018 13:40:10 -0400 - rev 444237
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - VideoConduit updates; r=pehrsons Differential Revision: https://phabricator.services.mozilla.com/D7433
d1f7d44e05a4b9e908373a82776071c0c51a38bb: Bug 1376873 - WebrtcGmpVideoCodec updates; r=pehrsons
Dan Minor <dminor@mozilla.com> - Tue, 21 Aug 2018 13:40:26 -0400 - rev 444236
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - WebrtcGmpVideoCodec updates; r=pehrsons Differential Revision: https://phabricator.services.mozilla.com/D7432
b59e1fca5c13611253969d537588547391e5d1c8: Bug 1376873 - Fix include path in JsepSessionImpl.cpp; r=bwc
Dan Minor <dminor@mozilla.com> - Tue, 21 Aug 2018 13:39:23 -0400 - rev 444235
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Fix include path in JsepSessionImpl.cpp; r=bwc Differential Revision: https://phabricator.services.mozilla.com/D7430
5d7a8864505d56d50534c9e71621aba8de59487b: Bug 1376873 - Fix up logging in WebrtcLog.cpp; r=ng
Dan Minor <dminor@mozilla.com> - Tue, 21 Aug 2018 13:39:05 -0400 - rev 444234
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Fix up logging in WebrtcLog.cpp; r=ng The webrtc::Trace code is removed by this update. We already had support for LOG (now RTC_LOG) in WebrtcLog.cpp. This removes the trace code from WebRtcLog.cpp and moves the aec logging code from webrtc::Trace to rtc::LogMessage. This also disables logging to stderr in rtc_base/logging.cc. We could disable it using the API, but that happens through peerconnection resulting in some logging occuring during getusermedia. The aec logs were testing with --disable-e10s. Rather than trying to work around sandboxing, I think it makes more sense to fix Bug 1404982 and store the logs in memory for retrieval from about:webrtc. Differential Revision: https://phabricator.services.mozilla.com/D7429
e1ea09e84febf9b59a2ea2a013874593388e3ec2: Bug 1376873 - Rollup conflict fixes for rtp_rtcp module; r=ng
Dan Minor <dminor@mozilla.com> - Tue, 30 Jan 2018 15:12:54 -0500 - rev 444233
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Rollup conflict fixes for rtp_rtcp module; r=ng MozReview-Commit-ID: D09534DOVLj Differential Revision: https://phabricator.services.mozilla.com/D7428
bc92eac0522f10b9c0fc675744561edf25dd4975: Bug 1376873 - Rollup conflict fixes for audio/video code; r=pehrsons
Dan Minor <dminor@mozilla.com> - Mon, 22 Jan 2018 15:04:26 -0500 - rev 444232
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Rollup conflict fixes for audio/video code; r=pehrsons MozReview-Commit-ID: 1T8mgqdkzq3 Differential Revision: https://phabricator.services.mozilla.com/D7427
1f86d23e480686bdaa5ae65a958936204c0b99d6: Bug 1376873 - Rollup of local modifications; r=ng
Dan Minor <dminor@mozilla.com> - Mon, 22 Jan 2018 13:31:57 -0500 - rev 444231
Push 34986 by shindli@mozilla.com at Sat, 03 Nov 2018 09:44:53 +0000
Bug 1376873 - Rollup of local modifications; r=ng MozReview-Commit-ID: 2euYzBEvuNb Differential Revision: https://phabricator.services.mozilla.com/D7425
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