Currently, when we generate the list of supported video codecs that will
be signaled in SDP, we start with the internal video codecs and then
append the external video codecs. When we create a video encoder for a
given codec, we prefer an external encoder over an internal encoder.
This CL lists the external video codecs first in SDP instead, so that we
consistently prefer external video codecs over internal.
The reason for doing this is that we will otherwise prefer an internal
SW H264 encoder over an external HW H264 encoder if the H264 profiles
differs.
BUG=chromium:688541
Review-Url: https://codereview.webrtc.org/2974383002
Cr-Commit-Position: refs/heads/master@{#19026}
Replace the use of webrtc::VideoEncoderFactory with
cricket::WebRtcVideoEncoderFactory and remove the adapter classes
between these two factory types.
Some code changes were necessary in order to accomplish this:
* Move SimulcastEncoderAdapter from
webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's
where it's used).
* Rename simulcast_unittest.h to simulcast_test_utility.h and make it
into it's own target, because it's used from both
simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc.
* Remove ownership of the encoder factory from SimulcastEncoderAdapter,
and make the necessary changes in surrounding code.
The goal with this CL is to clean up the code, and also to free up
the name webrtc::VideoEncoderFactory for future use.
BUG=webrtc:7925
Review-Url: https://codereview.webrtc.org/2964953002
Cr-Commit-Position: refs/heads/master@{#18945}
This CL removes code that supported the now removed
downstream dependencies in the support for using an
external audio processing module.
BUG=webrtc:7939
Review-Url: https://codereview.webrtc.org/2969213002
Cr-Commit-Position: refs/heads/master@{#18929}
Lower then bitrate so that the delta between the highest layer of the
lower stream and lowest layer of the higher stream is not too large.
BUG=webrtc:4172
This is a reland of the following CL:
Review-Url: https://codereview.webrtc.org/2791273002
Cr-Commit-Position: refs/heads/master@{#18232}
Committed: dceb42da3e
https: //codereview.webrtc.org/2883963002
Review-Url: https://codereview.webrtc.org/2966833002
Cr-Commit-Position: refs/heads/master@{#18913}
Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.
The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
Can be enabled by setting "enable_encrypted_rtp_header_extensions" in
"crypto_options" of "PeerConnectionFactoryInterface::Options" and will
only be used if both peers support it.
BUG=webrtc:3411
Review-Url: https://codereview.webrtc.org/2761143002
Cr-Commit-Position: refs/heads/master@{#18842}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
Patch set 1 is a reland + trivial rebase.
Patch set >= 2 contains bug fixes.
> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6
BUG=webrtc:7664
Review-Url: https://codereview.webrtc.org/2953053002
Cr-Commit-Position: refs/heads/master@{#18782}
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).
The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.
The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.
Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
This eliminates a thread hop in PeerConnectionFactory initialization,
and will allow some code to be simplified.
BUG=None
Review-Url: https://codereview.webrtc.org/2934103002
Cr-Commit-Position: refs/heads/master@{#18613}
WebRtcVideoChannel and and WebRtcVideoEngine seem to have been removed, and only WebRtcVideoChannel2 and WebRtcVideoEngine2 remain, which removes the need for the "2" postfix.
BUG=None
Review-Url: https://codereview.webrtc.org/2932073002
Cr-Commit-Position: refs/heads/master@{#18531}
Reason for revert:
Looks like there's still one failing perf test:
RampUpTest.UpDownUpTransportSequenceNumberPacketLoss
Original issue's description:
> Reland of Periodically update codec bit/frame rate settings. (patchset #1 id:1 of https://codereview.webrtc.org/2923993002/ )
>
> Reason for revert:
> Create reland cl that we can patch with fix.
>
> Original issue's description:
> > Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )
> >
> > Reason for revert:
> > Breaks some Call perf tests that are not run by the try bots....
> >
> > Original issue's description:
> > > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> > >
> > > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > > * Fix test
> > >
> > > BUG=7664
> > >
> > > Review-Url: https://codereview.webrtc.org/2883963002
> > > Cr-Commit-Position: refs/heads/master@{#18473}
> > > Committed: 6431e21da6
> >
> > TBR=stefan@webrtc.org,holmer@google.com
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2923993002
> > Cr-Commit-Position: refs/heads/master@{#18475}
> > Committed: 5390c4814d
>
> TBR=stefan@webrtc.org,holmer@google.com
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=7664
>
> Review-Url: https://codereview.webrtc.org/2924023002
> Cr-Commit-Position: refs/heads/master@{#18497}
> Committed: cdafeda1cbTBR=stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7664
Review-Url: https://codereview.webrtc.org/2926283002
Cr-Commit-Position: refs/heads/master@{#18500}
Reason for revert:
Create reland cl that we can patch with fix.
Original issue's description:
> Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )
>
> Reason for revert:
> Breaks some Call perf tests that are not run by the try bots....
>
> Original issue's description:
> > Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
> >
> > That however exposes a bunch of failed test, so this CL also fixed a few other things:
> > * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> > * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> > * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> > * Fix test
> >
> > BUG=7664
> >
> > Review-Url: https://codereview.webrtc.org/2883963002
> > Cr-Commit-Position: refs/heads/master@{#18473}
> > Committed: 6431e21da6
>
> TBR=stefan@webrtc.org,holmer@google.com
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=7664
>
> Review-Url: https://codereview.webrtc.org/2923993002
> Cr-Commit-Position: refs/heads/master@{#18475}
> Committed: 5390c4814dTBR=stefan@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7664
Review-Url: https://codereview.webrtc.org/2924023002
Cr-Commit-Position: refs/heads/master@{#18497}
Reason for revert:
Breaks some Call perf tests that are not run by the try bots....
Original issue's description:
> Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended.
>
> That however exposes a bunch of failed test, so this CL also fixed a few other things:
> * FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
> * FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
> * Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
> * Fix test
>
> BUG=7664
>
> Review-Url: https://codereview.webrtc.org/2883963002
> Cr-Commit-Position: refs/heads/master@{#18473}
> Committed: 6431e21da6TBR=stefan@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7664
Review-Url: https://codereview.webrtc.org/2923993002
Cr-Commit-Position: refs/heads/master@{#18475}
That however exposes a bunch of failed test, so this CL also fixed a few other things:
* FakeEncoder should trust the configured FPS value rather than guesstimating itself based on the realtime clock, so as not to completely undershoot targets in offline mode. Also, compensate for key-frame overshoots when outputting delta frames.
* FrameDropper should not assuming incoming frame rate is 0 if no frames have been seen.
* Fix a bunch of test cases that started failing because they were relying on the fake encoder undershooting.
* Fix test
BUG=7664
Review-Url: https://codereview.webrtc.org/2883963002
Cr-Commit-Position: refs/heads/master@{#18473}
Reason for revert:
Broken downstream project.
Original issue's description:
> Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
>
> BUG=webrtc:7395
>
> Review-Url: https://codereview.webrtc.org/2888303005
> Cr-Commit-Position: refs/heads/master@{#18417}
> Committed: 9641c13327TBR=deadbeef@webrtc.org,stefan@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,holmer@google.com,zstein@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7395
Review-Url: https://codereview.webrtc.org/2914413002
Cr-Commit-Position: refs/heads/master@{#18420}
Reason for revert:
Broken downstream projects
Original issue's description:
> Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.
>
> Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
>
> BUG=webrtc:5079
> R=deadbeef@webrtc.org, hbos@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2863123002 .
> Cr-Commit-Position: refs/heads/master@{#18384}
> Committed: e80f4c91d0TBR=hbos@webrtc.org,deadbeef@webrtc.org,holmer@google.com,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2916793003
Cr-Commit-Position: refs/heads/master@{#18386}
Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
BUG=webrtc:5079
R=deadbeef@webrtc.org, hbos@webrtc.org
Review-Url: https://codereview.webrtc.org/2863123002 .
Cr-Commit-Position: refs/heads/master@{#18384}
After this CL, reconfiguring the FlexFEC payload type at the
WebRtcVideoChannel2 level will no longer lead to the recreation of
the VideoReceiveStream. This means that the jitter buffer will
not be destroyed and a smoother video playback is achieved during
SDP renegotiation.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2911913002
Cr-Commit-Position: refs/heads/master@{#18318}
This CL removes |default_recv_ssrc_| from DefaultUnsignalledSsrcHandler
and replaces it with calls to a new member function
WebRtcVideoChannel2::GetDefaultReceiveStreamSsrc. The latter checks
the |default_stream_| member on the
WebRtcVideoChannel2::WebRtcVideoReceiveStreams to know which stream
is the current default stream.
This change removes duplicate state and fixes an issue where
incoming unsignaled SSRCs would compete for being the default
receive stream.
BUG=webrtc:7725
Review-Url: https://codereview.webrtc.org/2906893002
Cr-Commit-Position: refs/heads/master@{#18314}
Reason for revert:
Revert of revert of revert of revert of 'Activating..'. Or "reland of reland of 'Activate..'".
*Now* the internal projects are fixed and the fix is verified.
Original issue's description:
> Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2903153005/ )
>
> Reason for revert:
> Reverting again: internal project issues were apparently not completely fixed.
>
> Original issue's description:
> > Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ )
> >
> > Reason for revert:
> > Revert the revert now that internal projects are updated.
> >
> > Original issue's description:
> > > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
> > >
> > > Reason for revert:
> > > Breaks internal project.
> > >
> > > Original issue's description:
> > > > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> > > >
> > > > A low priority task queue is added to WebRTCVoiceEngine. The
> > > > start/stop debug calls make file logging happen on the task queue.
> > > >
> > > > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > > > so that it can be shared for low priority tasks between different
> > > > subcomponents. It will require some changes to MediaEngine,
> > > > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > > > projects.
> > > >
> > > > A task queue must be created and destroyed from the same thread. With
> > > > this CL that will be the worker thread, which creates and destroys
> > > > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > > > the signaling thread.
> > > >
> > > > NOTRY=True # tests just passed
> > > >
> > > > BUG=webrtc:7404
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2896813002
> > > > Cr-Commit-Position: refs/heads/master@{#18252}
> > > > Committed: c61bf947b4
> > >
> > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7404
> > >
> > > Review-Url: https://codereview.webrtc.org/2904893002
> > > Cr-Commit-Position: refs/heads/master@{#18255}
> > > Committed: be68b72cfa
> >
> > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2903153005
> > Cr-Commit-Position: refs/heads/master@{#18270}
> > Committed: d2303a2338
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2910633002
> Cr-Commit-Position: refs/heads/master@{#18272}
> Committed: fe9ecb07eaTBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2904423002
Cr-Commit-Position: refs/heads/master@{#18300}
Reason for revert:
Reverting again: internal project issues were apparently not completely fixed.
Original issue's description:
> Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ )
>
> Reason for revert:
> Revert the revert now that internal projects are updated.
>
> Original issue's description:
> > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
> >
> > Reason for revert:
> > Breaks internal project.
> >
> > Original issue's description:
> > > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> > >
> > > A low priority task queue is added to WebRTCVoiceEngine. The
> > > start/stop debug calls make file logging happen on the task queue.
> > >
> > > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > > so that it can be shared for low priority tasks between different
> > > subcomponents. It will require some changes to MediaEngine,
> > > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > > projects.
> > >
> > > A task queue must be created and destroyed from the same thread. With
> > > this CL that will be the worker thread, which creates and destroys
> > > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > > the signaling thread.
> > >
> > > NOTRY=True # tests just passed
> > >
> > > BUG=webrtc:7404
> > >
> > > Review-Url: https://codereview.webrtc.org/2896813002
> > > Cr-Commit-Position: refs/heads/master@{#18252}
> > > Committed: c61bf947b4
> >
> > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2904893002
> > Cr-Commit-Position: refs/heads/master@{#18255}
> > Committed: be68b72cfa
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2903153005
> Cr-Commit-Position: refs/heads/master@{#18270}
> Committed: d2303a2338TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2910633002
Cr-Commit-Position: refs/heads/master@{#18272}
Reason for revert:
Revert the revert now that internal projects are updated.
Original issue's description:
> Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
>
> Reason for revert:
> Breaks internal project.
>
> Original issue's description:
> > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> >
> > A low priority task queue is added to WebRTCVoiceEngine. The
> > start/stop debug calls make file logging happen on the task queue.
> >
> > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > so that it can be shared for low priority tasks between different
> > subcomponents. It will require some changes to MediaEngine,
> > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > projects.
> >
> > A task queue must be created and destroyed from the same thread. With
> > this CL that will be the worker thread, which creates and destroys
> > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > the signaling thread.
> >
> > NOTRY=True # tests just passed
> >
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2896813002
> > Cr-Commit-Position: refs/heads/master@{#18252}
> > Committed: c61bf947b4
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2904893002
> Cr-Commit-Position: refs/heads/master@{#18255}
> Committed: be68b72cfaTBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2903153005
Cr-Commit-Position: refs/heads/master@{#18270}
Reason for revert:
Breaks internal project.
Original issue's description:
> Activate 'offload debug dump recordings from audio thread to TaskQueue'.
>
> A low priority task queue is added to WebRTCVoiceEngine. The
> start/stop debug calls make file logging happen on the task queue.
>
> In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> so that it can be shared for low priority tasks between different
> subcomponents. It will require some changes to MediaEngine,
> CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> projects.
>
> A task queue must be created and destroyed from the same thread. With
> this CL that will be the worker thread, which creates and destroys
> WebRTCVoiceEngine. With the dependent CL, it will probably change to
> the signaling thread.
>
> NOTRY=True # tests just passed
>
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2896813002
> Cr-Commit-Position: refs/heads/master@{#18252}
> Committed: c61bf947b4TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2904893002
Cr-Commit-Position: refs/heads/master@{#18255}
A low priority task queue is added to WebRTCVoiceEngine. The
start/stop debug calls make file logging happen on the task queue.
In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
so that it can be shared for low priority tasks between different
subcomponents. It will require some changes to MediaEngine,
CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
projects.
A task queue must be created and destroyed from the same thread. With
this CL that will be the worker thread, which creates and destroys
WebRTCVoiceEngine. With the dependent CL, it will probably change to
the signaling thread.
NOTRY=True # tests just passed
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2896813002
Cr-Commit-Position: refs/heads/master@{#18252}
Lower then bitrate so that the delta between the highest layer of the
lower stream and lowest layer of the higher stream is not too large.
Also fix a bug in vie_encoder where the codec was not perioducally
updated unless a new bitrate estimate was triggered.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2791273002
Cr-Commit-Position: refs/heads/master@{#18232}
When operating on mobile devices, where hardware support is available
for the AEC and NS functionality, it is desirable to be able to
operate without hardcoded behaviors for the WebRTC AGC and HPF.
This CL adds support to allow a field trial to turn these off
whenever that is possible.
BUG=webrtc:6220, webrtc:6183, webrtc:6181
Review-Url: https://codereview.webrtc.org/2876133002
Cr-Commit-Position: refs/heads/master@{#18226}
This CL reduces the number of VideoSendStream recreations during SDP
renegotiation by checking the FlexFEC field trials before, and not after,
the SDP codec diffing logic.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2882433003
Cr-Commit-Position: refs/heads/master@{#18211}
