In the AudioMixerImpl implementation, removing a source never fails
and the return value is always true (see audio_mixer/audio_mixer_impl.cc).
A return value of |false| signaled that removing a source failed for
some reason. We have come to the conclusion that
* we don't know how to handle a return value of |false|
* we can't think of why an alternative implementation would need to
signal failure when removing a stream.
To avoid having a status code that is never read, never acted upon and
probably never set to anything but |true|, we change ::RemoveSource to
not have a return value.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2506173003
Cr-Commit-Position: refs/heads/master@{#15150}
Reason for revert:
The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change:
function removeVideoCodec(offerSdp) {
- offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n',
- 'a=rtpmap:100 XVP8/90000\r\n');
+ offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/,
+ 'a=rtpmap:$1 XVP8/90000\r\n');
return offerSdp;
}
Original issue's description:
> Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ )
>
> Reason for revert:
> Breaks chromium.fyi test:
> WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
>
> Original issue's description:
> > Stop using hardcoded payload types for video codecs
> >
> > This CL stops using hardcoded payload types for different video codecs
> > and will dynamically assign them payload types incrementally from 96 to
> > 127 instead.
> >
> > This CL:
> > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
> > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
> > internally supported software codecs instead. The purpose is to
> > streamline the payload type assignment in webrtcvideoengine2.cc which
> > will now have two encoder factories of the same
> > WebRtcVideoEncoderFactory type; one internal and one external.
> > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
> > instead.
> > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
> > moves the create function to the internal encoder factory instead.
> > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
> > interface without any static functions.
> > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
> > the internal and external codecs and assigns them payload types
> > incrementally from 96 to 127.
> > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
> > what payload types will be used.
> >
> > BUG=webrtc:6677,webrtc:6705
> > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
> >
> > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b
> > Cr-Commit-Position: refs/heads/master@{#15135}
>
> TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6677,webrtc:6705
>
> Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591
> Cr-Commit-Position: refs/heads/master@{#15140}
TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6677,webrtc:6705
Review-Url: https://codereview.webrtc.org/2511933002
Cr-Commit-Position: refs/heads/master@{#15148}
Payload type -> RtpCodecParameters maps added for sender and receiver.
This is a follow-up to https://codereview.webrtc.org/2484193002/ which
did the same thing for VideoMediaInfo. This information will be used to
produce RTCCodecStats[1].
Voice[Sender/Receiver]Info is updated with current codec payload type
for every stream which can be used to look up the codec in
VoiceMediaInfo.
[1] https://w3c.github.io/webrtc-stats/#codec-dict*
BUG=chromium:659117
Review-Url: https://codereview.webrtc.org/2503383002
Cr-Commit-Position: refs/heads/master@{#15144}
Added the callback in native c++ API.
The callback function is called when a track is added and a new RtpReceiver is created.
The application can tell when tracks are added to the streams by listening to this callback.
BUG=webrtc:6112
Review-Url: https://codereview.webrtc.org/2505173002
Cr-Commit-Position: refs/heads/master@{#15142}
Reason for revert:
Breaks chromium.fyi test:
WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
Original issue's description:
> Stop using hardcoded payload types for video codecs
>
> This CL stops using hardcoded payload types for different video codecs
> and will dynamically assign them payload types incrementally from 96 to
> 127 instead.
>
> This CL:
> * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
> webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
> internally supported software codecs instead. The purpose is to
> streamline the payload type assignment in webrtcvideoengine2.cc which
> will now have two encoder factories of the same
> WebRtcVideoEncoderFactory type; one internal and one external.
> * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
> instead.
> * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
> moves the create function to the internal encoder factory instead.
> * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
> interface without any static functions.
> * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
> the internal and external codecs and assigns them payload types
> incrementally from 96 to 127.
> * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
> what payload types will be used.
>
> BUG=webrtc:6677,webrtc:6705
> R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b
> Cr-Commit-Position: refs/heads/master@{#15135}
TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6677,webrtc:6705
Review-Url: https://codereview.webrtc.org/2513633002
Cr-Commit-Position: refs/heads/master@{#15140}
This CL stops using hardcoded payload types for different video codecs
and will dynamically assign them payload types incrementally from 96 to
127 instead.
This CL:
* Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
internally supported software codecs instead. The purpose is to
streamline the payload type assignment in webrtcvideoengine2.cc which
will now have two encoder factories of the same
WebRtcVideoEncoderFactory type; one internal and one external.
* Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
instead.
* Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
moves the create function to the internal encoder factory instead.
* Removes video_encoder.cc. webrtc::VideoEncoder is now just an
interface without any static functions.
* The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
the internal and external codecs and assigns them payload types
incrementally from 96 to 127.
* Updates webrtcvideoengine2_unittest.cc and removes assumptions about
what payload types will be used.
BUG=webrtc:6677,webrtc:6705
R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2493133002 .
Cr-Commit-Position: refs/heads/master@{#15135}
We support multiple payload types, and one which matches the audio codec the closest, is picked (or the one with lowest clock rate, if no perfect match is found).
The exact clock rate is then ignored and DTMF packets are time stamped with the rate of the current audio codec. This is exactly the way the code has worked up to this point, but until now we have been under the impression that we were in fact sending 8k DTMF.
In other words, this is an improvement over the current situation, since we will most likely find a payload type which matches the codec clock rate.
This CL also does a little cleaning of the DTMFQueue and RTPSenderAudio classes.
BUG=webrtc:2795
Review-Url: https://codereview.webrtc.org/2392883002
Cr-Commit-Position: refs/heads/master@{#15129}
Just returns the configuration the PC was constructed with, or the last
one passed into SetConfiguration.
BUG=chromium:587453
Review-Url: https://codereview.webrtc.org/2504103002
Cr-Commit-Position: refs/heads/master@{#15111}
This is yet another reland of https://codereview.webrtc.org/2434073003/
including two fixes:
1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that.
2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams.
Please review only the changes after patch set 1.
Original description:
Extract bitrate allocation of spatial/temporal layers out of codec impl.
This CL makes a number of intervowen changes:
* Add BitrateAllocation struct, that contains a codec independent view
of how the target bitrate is distributed over spatial and temporal
layers.
* Adds the BitrateAllocator interface, which takes a bitrate and frame
rate and produces a BitrateAllocation.
* A default (non layered) implementation is added, and
SimulcastRateAllocator is extended to fully handle VP8 allocation.
This includes capturing TemporalLayer instances created by the
encoder.
* ViEEncoder now owns both the bitrate allocator and the temporal layer
factories for VP8. This allows allocation to happen fully outside of
the encoder implementation.
This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.
BUG=webrtc:6301
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2510583002 .
Cr-Commit-Position: refs/heads/master@{#15105}
The peer connection loopback test could catch regressions too, but it's
too slow to run on downstream ARM emulators. I'm adding a test here
that just makes sure we can load the JNI and init audio/video engines
in WebRTC.
This test overlaps in functionality with the existing tests,
but we need it anyway since all existing tests are too timing-sensitive.
Removes resources from the test; they're awkward downstream and we
don't really need them anyway.
BUG=b/32820229
Review-Url: https://codereview.webrtc.org/2506603002
Cr-Commit-Position: refs/heads/master@{#15101}
DataChannel.SignalOpened and unittests added.
PeerConnection.SignalDataChannelCreated added and wired up to
RTCStatsCollector.OnDataChannelCreated on RTCStatsCollector
construction.
RTCStatsCollector.OnSignalOpened/Closed added and wired up on
OnDataChannelCreated.
rtcstatscollector_unittest.cc updated, faking that channels are opened
and closed.
I did not want to use DataChannelObserver because it is used for more
than state changes and there can only be one observer (unless code is
updated). Since DataChannel already had a SignalClosed it made sense to
add a SignalOpened.
Having OnSignalBlah in RTCStatsCollector is new in this CL but will
likely be needed to correctly handle RTPMediaStreamTracks being added
and detached independently of getStats. This CL establishes this
pattern.
(An integration test will be needed for this and all the other stats to
make sure everything is wired up correctly and test outside of a
mock/fake environment, but this is not news.)
BUG=chromium:636818, chromium:627816
Review-Url: https://codereview.webrtc.org/2472113002
Cr-Commit-Position: refs/heads/master@{#15059}
This CL will start to distinguish H264 profiles during SDP negotiation.
We currently don't look at the H264 profile at all and assume they are
all Constrained Baseline Level 3.1. This CL will start to check profiles
for equality when matching, and will generate the correct answer H264
level.
Each local supported H264 profile needs to be listed explicitly in the
list of local supported codecs, even if they are redundant. For example,
Baseline profile should be listed explicitly even though another profile
that is a superset of Baseline is also listed. The reason for this is to
simplify the code and avoid profile intersection during matching. So
VideoCodec::Matches will check for profile equality, and not check if
one codec is a subset of the other. This also leads to the nice property
that VideoCodec::Matches is symmetric, i.e. iif a.Matches(b) then
b.Matches(a).
BUG=webrtc:6337
TBR=tkchin@webrtc.org
Review-Url: https://codereview.webrtc.org/2483173002
Cr-Commit-Position: refs/heads/master@{#15051}
Reason for revert:
Broke a lot of tests in chromium.webrtc browser_tests. See e.g. https://build.chromium.org/p/chromium.webrtc/builders/Mac%20Tester/builds/62228 and https://build.chromium.org/p/chromium.webrtc/builders/Win8%20Tester/builds/30102.
[ RUN ] WebRtcVideoQualityBrowserTests/WebRtcVideoQualityBrowserTest.MANUAL_TestVideoQualityH264/1
...
#
# Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\modules\rtp_rtcp\source\rtp_format_h264.cc, line 170
# last system error: 0
# Check failed: packetization_mode_ == kH264PacketizationMode1 (0 vs. 2)
#
Original issue's description:
> Implement H.264 packetization mode 0.
>
> This approach extends the H.264 specific information with
> a packetization mode enum.
>
> Status: Parameter is in code. No way to set it yet.
>
> Rebase of CL 2009213002
>
> BUG=600254
>
> Committed: https://crrev.com/3bba101f36483b8030a693dfbc93af736d1dba68
> Cr-Commit-Position: refs/heads/master@{#15032}
TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=600254
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2500743002
Cr-Commit-Position: refs/heads/master@{#15050}
Reason for revert:
Relanding, now that downstream issues have been fixed.
Original issue's description:
> Revert of Declare VideoCodec.codec_specific_info private (patchset #5 id:80001 of https://codereview.webrtc.org/2452963002/ )
>
> Reason for revert:
> Broke a google3 build
>
> Original issue's description:
> > Declare VideoCodec.codec_specific_info private
> >
> > This completes the privatization of the codec specific
> > information in VideoCodec.
> >
> > BUG=webrtc:6603
> >
> > Committed: https://crrev.com/792738640234d81c916ac4458ac72286cb2548a4
> > Cr-Commit-Position: refs/heads/master@{#15013}
>
> TBR=tommi@chromium.org,magjed@chromium.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6603
>
> Committed: https://crrev.com/7fe6db91d99cf6d43874651bcca56092cf869e2f
> Cr-Commit-Position: refs/heads/master@{#15027}
TBR=tommi@chromium.org,magjed@chromium.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6603
Review-Url: https://codereview.webrtc.org/2494683006
Cr-Commit-Position: refs/heads/master@{#15041}
These functions currently copy cricket::Codec classes by value which is
expensive since they contain e.g. std::map<std::string, std::string>
containers with parameters. This CL avoids copying them altogether.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2493733003
Cr-Commit-Position: refs/heads/master@{#15040}
Reason for revert:
Relanding after known downstream breakages have been fixed.
Original issue's description:
> Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ )
>
> Reason for revert:
> Breaks chrome, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/19019/steps/compile/logs/stdio
>
> Analysis: Chrome uses cricket::VideoFrame, without explicitly including webrtc/media/base/videoframe.h, and breaks when that file is no longer included by any other webrtc headers. Will reland after updating Chrome.
>
> Original issue's description:
> > Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
> >
> > Replaced with webrtc::VideoFrame.
> >
> > TBR=mflodman@webrtc.org
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/45c8b8940042bd2574c39920804ade8343cefdba
> > Cr-Commit-Position: refs/heads/master@{#14885}
>
> TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/7341ab8e2505c9763d208e069bda269018357e7d
> Cr-Commit-Position: refs/heads/master@{#14886}
TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2487633002
Cr-Commit-Position: refs/heads/master@{#15039}
This approach extends the H.264 specific information with
a packetization mode enum.
Status: Parameter is in code. No way to set it yet.
Rebase of CL 2009213002
BUG=600254
Review-Url: https://codereview.webrtc.org/2337453002
Cr-Commit-Position: refs/heads/master@{#15032}
Reason for revert:
Broke a google3 build
Original issue's description:
> Declare VideoCodec.codec_specific_info private
>
> This completes the privatization of the codec specific
> information in VideoCodec.
>
> BUG=webrtc:6603
>
> Committed: https://crrev.com/792738640234d81c916ac4458ac72286cb2548a4
> Cr-Commit-Position: refs/heads/master@{#15013}
TBR=tommi@chromium.org,magjed@chromium.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6603
Review-Url: https://codereview.webrtc.org/2491933002
Cr-Commit-Position: refs/heads/master@{#15027}
Reason for revert:
Seems to be causing flakiness in perf test:
FullStackTest.ScreenshareSlidesVP8_2TL_LossyNet
Original issue's description:
> Reland of Issue 2434073003: Extract bitrate allocation ...
>
> This is a reland of https://codereview.webrtc.org/2434073003/ including
> some fixes for failing test cases.
>
> Original description:
>
> Extract bitrate allocation of spatial/temporal layers out of codec impl.
>
> This CL makes a number of intervowen changes:
>
> * Add BitrateAllocation struct, that contains a codec independent view
> of how the target bitrate is distributed over spatial and temporal
> layers.
>
> * Adds the BitrateAllocator interface, which takes a bitrate and frame
> rate and produces a BitrateAllocation.
>
> * A default (non layered) implementation is added, and
> SimulcastRateAllocator is extended to fully handle VP8 allocation.
> This includes capturing TemporalLayer instances created by the
> encoder.
>
> * ViEEncoder now owns both the bitrate allocator and the temporal layer
> factories for VP8. This allows allocation to happen fully outside of
> the encoder implementation.
>
> This refactoring will make it possible for ViEEncoder to signal the
> full picture of target bitrates to the RTCP module.
>
> BUG=webrtc:6301
>
> Committed: https://crrev.com/647bf43dcb2fd16fccf276bd94dc4400728bb405
> Cr-Commit-Position: refs/heads/master@{#15023}
TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2491393002
Cr-Commit-Position: refs/heads/master@{#15026}
This is a reland of https://codereview.webrtc.org/2434073003/ including
some fixes for failing test cases.
Original description:
Extract bitrate allocation of spatial/temporal layers out of codec impl.
This CL makes a number of intervowen changes:
* Add BitrateAllocation struct, that contains a codec independent view
of how the target bitrate is distributed over spatial and temporal
layers.
* Adds the BitrateAllocator interface, which takes a bitrate and frame
rate and produces a BitrateAllocation.
* A default (non layered) implementation is added, and
SimulcastRateAllocator is extended to fully handle VP8 allocation.
This includes capturing TemporalLayer instances created by the
encoder.
* ViEEncoder now owns both the bitrate allocator and the temporal layer
factories for VP8. This allows allocation to happen fully outside of
the encoder implementation.
This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2488833004
Cr-Commit-Position: refs/heads/master@{#15023}
This completes the privatization of the codec specific
information in VideoCodec.
BUG=webrtc:6603
Review-Url: https://codereview.webrtc.org/2452963002
Cr-Commit-Position: refs/heads/master@{#15013}
Reason for revert:
Breaks perf tests.
Original issue's description:
> Extract bitrate allocation of spatial/temporal layers out of codec impl.
>
> This CL makes a number of intervowen changes:
>
> * Add BitrateAllocation struct, that contains a codec independent view
> of how the target bitrate is distributed over spatial and temporal
> layers.
>
> * Adds the BitrateAllocator interface, which takes a bitrate and frame
> rate and produces a BitrateAllocation.
>
> * A default (non layered) implementation is added, and
> SimulcastRateAllocator is extended to fully handle VP8 allocation.
> This includes capturing TemporalLayer instances created by the
> encoder.
>
> * ViEEncoder now owns both the bitrate allocator and the temporal layer
> factories for VP8. This allows allocation to happen fully outside of
> the encoder implementation.
>
> This refactoring will make it possible for ViEEncoder to signal the
> full picture of target bitrates to the RTCP module.
>
> BUG=webrtc:6301
>
> Committed: https://crrev.com/8f46c679d24a05b3f08e02c6d91ec9637f34e24f
> Cr-Commit-Position: refs/heads/master@{#14998}
TBR=stefan@webrtc.org,perkj@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2489843002
Cr-Commit-Position: refs/heads/master@{#15001}
This CL makes a number of intervowen changes:
* Add BitrateAllocation struct, that contains a codec independent view
of how the target bitrate is distributed over spatial and temporal
layers.
* Adds the BitrateAllocator interface, which takes a bitrate and frame
rate and produces a BitrateAllocation.
* A default (non layered) implementation is added, and
SimulcastRateAllocator is extended to fully handle VP8 allocation.
This includes capturing TemporalLayer instances created by the
encoder.
* ViEEncoder now owns both the bitrate allocator and the temporal layer
factories for VP8. This allows allocation to happen fully outside of
the encoder implementation.
This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2434073003
Cr-Commit-Position: refs/heads/master@{#14998}
Using rtc::PacketTransportInterface instead of cricket::TransportChannel
is a preparation for refactoring channel.cc.
BUG=webrtc:6676
Review-Url: https://codereview.webrtc.org/2483093003
Cr-Commit-Position: refs/heads/master@{#14989}
This change changes mixing to be done at the lowest possible
APM-native rate that does not lead to quality loss. An Audio
Processing-native rate is one of 8, 16, 32, or 48 kHz. Mixing at a
lower sampling rate and avoiding resampling can in many cases lead to
big efficiency improvements, as reported by experiments.
This CL also fixes a design issue with the AudioMixer: audio at
non-native rates is no longer fed to the APM instance which is the
limiter.
NOTRY=True
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2458703002
Cr-Commit-Position: refs/heads/master@{#14980}
Not all members are collected by RTCStatsCollector and detached tracks
are not visible in the returned stats. This needs to be addressed before
closing crbug.com/660827 and crbug.com/659137
BUG=chromium:660827, chromium:659137, chromium:627816
Review-Url: https://codereview.webrtc.org/2467873005
Cr-Commit-Position: refs/heads/master@{#14978}
The audio_device_module field was currently unused. The audio_mixer
field is going to be used to pass an AudioMixer to AudioState.
In the hopefully-not-very-far future, the toplevel WebRTC API will allow passing
a custom AudioMixer, e.g. for spatialized audio (audio in space). If no
mixer is passed, a default mixer is created (the one in modules/audio_mixer).
The only object which will have a permanent reference to the mixer is AudioState.
AudioState is created in WebRTCVoiceEngine with a configuration object,
which already contains a VoiceEngine pointer. In this CL, we extend this
config object with a mixer pointer.
In summary: in an upcoming CL, a mixer will be either created in or passed to
WebRTCVoiceEngine. This mixer will be passed to the ctor of AudioState in a
config struct.
BUG=webrtc:6346
NOTRY=True
Review-Url: https://codereview.webrtc.org/2456363002
Cr-Commit-Position: refs/heads/master@{#14973}
As I was preparing to move some files from the api/ folder, I noticed
that this file was not included in the BUILD.gn file. I've added it back
in and updated it to compile and run successfully again.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2485603002
Cr-Commit-Position: refs/heads/master@{#14965}
The unit was kbps but the one default use of it is in bps. The inconsistency should be fixed.
BUG=webrtc:6670
Review-Url: https://codereview.webrtc.org/2247213005
Cr-Commit-Position: refs/heads/master@{#14955}
Essentially applying the same change as in
https://codereview.webrtc.org/2023413002 in more locations.
There's only one change affecting production code: enabling the warning
for webrtc/media:rtc_media. The rest are test changes.
With these changes, the only place the warning is disabled is in the Windows
implementation of webrtc/modules/video_capture:video_capture_internal_impl,
which is harder to fix, since it relies on sample code from the Windows SDK.
BUG=webrtc:6653
NOTRY=True
Review-Url: https://codereview.webrtc.org/2468093004
Cr-Commit-Position: refs/heads/master@{#14938}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Original-Commit-Position: refs/heads/master@{#14923}
Cr-Commit-Position: refs/heads/master@{#14935}
Before calling StatsCollctor::GetStats() in PeerConnection::GetStats(), check if the track is valid. If not, return false.
A track is invalid if it is not a nullptr and there is no report data for it.
BUG=webrtc:6652
Review-Url: https://codereview.webrtc.org/2470023004
Cr-Commit-Position: refs/heads/master@{#14934}
Reason for revert:
Still breaks internal downstream project.
Sergey: Please update internal project before relanding this.
Original issue's description:
> Remove deprected functions from EncodedImageCallback and RtpRtcp
>
> Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> These methods should no longer be used anywhere and it's safe to remove
> them.
>
> BUG=chromium:621691
>
> Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
> Cr-Commit-Position: refs/heads/master@{#14923}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2479643002
Cr-Commit-Position: refs/heads/master@{#14925}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Commit-Position: refs/heads/master@{#14923}
Reason for revert:
Breaks everything
Original issue's description:
> Revert of Remove deprected functions from EncodedImageCallback and RtpRtcp (patchset #4 id:100001 of https://codereview.webrtc.org/2405173006/ )
>
> Reason for revert:
> This might be breaking projects downstream.
>
> Original issue's description:
> > Remove deprected functions from EncodedImageCallback and RtpRtcp
> >
> > Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> > These methods should no longer be used anywhere and it's safe to remove
> > them.
> >
> > BUG=chromium:621691
> >
> > Committed: https://crrev.com/fa565842718ad178a7562721b25d916fbabc2b92
> > Cr-Commit-Position: refs/heads/master@{#14902}
>
> TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:621691
>
> Committed: https://crrev.com/6c78307a21252c2dbd704f6d5e92a220fb722ed4
> Cr-Commit-Position: refs/heads/master@{#14914}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2467373003
Cr-Commit-Position: refs/heads/master@{#14915}
Reason for revert:
This might be breaking projects downstream.
Original issue's description:
> Remove deprected functions from EncodedImageCallback and RtpRtcp
>
> Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> These methods should no longer be used anywhere and it's safe to remove
> them.
>
> BUG=chromium:621691
>
> Committed: https://crrev.com/fa565842718ad178a7562721b25d916fbabc2b92
> Cr-Commit-Position: refs/heads/master@{#14902}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2474433008
Cr-Commit-Position: refs/heads/master@{#14914}
This CL sets the data channel type of the session options before setting the bundle-enabled flag of the session options, so that bundle-enabled will be correctly set and the bundle group will be created.
BUG=webrtc:6218
Review-Url: https://codereview.webrtc.org/2473603002
Cr-Commit-Position: refs/heads/master@{#14904}
