Original change's description:
> Update RTCStatsCollector to work with RtpTransceivers
>
> Bug: webrtc:8764
> Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d
> Reviewed-on: https://webrtc-review.googlesource.com/49580
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22026}
Bug: webrtc:8764
Change-Id: I6a682824febf3f4f41397fc1a8dd7396c4ffa8e3
Reviewed-on: https://webrtc-review.googlesource.com/54160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22064}
This reverts commit 71439a60e7915179be96dd42dc732dc51c279884.
Reason for revert: https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Mac%20Tester/47796
Original change's description:
> Set session error if SetLocal/RemoteDescription ever fails
>
> This changes SetLocalDescription/SetRemoteDescription to set a
> session error which will cause any future calls to fail early if
> there is an error when applying a session description.
>
> This is needed since until better error recovery is implemented
> failing a call to SetLocalDescription or SetRemoteDescription
> could leave the PeerConnection in an inconsistent state.
>
> Bug: chromium:800775
> Change-Id: If06fd73d6e902af15d072dc562bbe830d3b11ad5
> Reviewed-on: https://webrtc-review.googlesource.com/54061
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22061}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org
Change-Id: I8af271f2b6dd6a896e390a6fe736e809329b4f4a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:800775
Reviewed-on: https://webrtc-review.googlesource.com/54700
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22063}
This changes SetLocalDescription/SetRemoteDescription to set a
session error which will cause any future calls to fail early if
there is an error when applying a session description.
This is needed since until better error recovery is implemented
failing a call to SetLocalDescription or SetRemoteDescription
could leave the PeerConnection in an inconsistent state.
Bug: chromium:800775
Change-Id: If06fd73d6e902af15d072dc562bbe830d3b11ad5
Reviewed-on: https://webrtc-review.googlesource.com/54061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22061}
Due to flakiness. See bug.
TBR=stefan@webrtc.org
Bug: webrtc:8878
Change-Id: Iece487ad9883fbe6209d25291fb70bb52c8afbd3
Reviewed-on: https://webrtc-review.googlesource.com/54601
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22059}
Apparently left over since 2015 refactorings, see cl
https://codereview.webrtc.org/1417283007
Bug: webrtc:5095
Change-Id: I899f2c018d1906b4336b2e80b511f7398bac4947
Reviewed-on: https://webrtc-review.googlesource.com/53200
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22055}
This CL adds the Level Estimator of the new gain controller. The Level
Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub
frames. We take the maximal sample values in every sub frame. We then
apply attack/decay smoothing. This is the final level estimate.
The results will be used with InterpolatedGainCurve (see this CL
https://webrtc-review.googlesource.com/c/src/+/51920). For every level
estimate value, we look up a gain with
InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to
the signal.
Bug: webrtc:7949
Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1
Reviewed-on: https://webrtc-review.googlesource.com/52381
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22054}
It turns out that some headers were not owned by any targets.
These were:
RTCVideoCodec.h
RTCVideoCodecFactory.h
RTCVideoCodecH264.h
RTCVideoEncoderVP8.h
RTCVideoDecoderVP8.h
RTCVideoEncoderVP9.h
RTCVideoDecoderVP9.h
And some were owned by multiple targets, namely:
RTCPeerConnectionFactory+Native.h
RTCPeerConnectionFactory+Private.h
RTCVideoFrameBuffer.h
These have all been moved to their appropriate homes.
This CL also fixes a lot of cyclic interdependencies in the iOS sdk build files.
Bug: webrtc:8855
Change-Id: I1b7ddb6c2a93868d1510ccf0a64bd3dd169ec3e7
Reviewed-on: https://webrtc-review.googlesource.com/49060
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22052}
Moves AndroidNetworkMonitor out of pc folder. Even clients not using
PeerConnection seem to be using it and it doesn't have any dependencies
to the PeerConnection API.
Bug: webrtc:8769
Change-Id: I2bdeff9f5c9925e13388fbc77aa9b264a7583548
Reviewed-on: https://webrtc-review.googlesource.com/53260
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22051}
This CL creates empty placeholders for EchoCanceller3Factory. This
allows for moving the factory of AEC3 as soon as downstream has been
updated to include echo_canceller3_factory.h.
Bug: webrtc:8844
Change-Id: I77c53d8257291f189c637e1c9ed76c4e74be1858
Reviewed-on: https://webrtc-review.googlesource.com/53862
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22050}
The FixedGainController (FGC) applies a fixed gain. It will also
control the limiter. The limiter will be landed over the next several
CLs.
The GainController2 is a 'private submodule' of APM. It will control
the new automatic gain controller (AGC). It controls the AGC through
Initialize() and ApplyConfig().
This CL contains
* build changes to make modules/audio_processing/agc2 an independent
target
* a new MutableFloatAudioFrame which is the audio interface between
AGC2 and APM
* move of the fixed gain application from GainController2 to
FixedGainController.
If you are a googler, there is more information in this doc:
https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit#
Bug: webrtc:7949
Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a
Reviewed-on: https://webrtc-review.googlesource.com/50440
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22046}
The functions replace some existing code and will be used in the
the new AutomaticGainController.
Bug: webrtc:7949
Change-Id: I9a32132d4a4699a507b8548a2eac10972a2f3fd6
Reviewed-on: https://webrtc-review.googlesource.com/53141
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22045}
This CL removes direct access to SendSideCongestionController (SSCC) via
the RtpTransportControllerSend interface and replaces all usages with
calls on RtpTransportControllerSend which will in turn calls SSCC. This
prepares for later refactor of RtpTransportControllerSend.
Bug: webrtc:8415
Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5
Reviewed-on: https://webrtc-review.googlesource.com/53860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22044}
The functions "memcpy" and "memset" are defined in "string.h" which
was not included. Found this when compiling with g++ 5.4 on Ubuntu
Xenial.
Bug: None
Change-Id: Ife9a9ce2a168ecc24d983afcfc0a39784cbedf9f
Reviewed-on: https://webrtc-review.googlesource.com/54121
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22043}
Bug: webrtc:7600
Change-Id: Ic4e5560fdeb9848c65c59e0f45ca3a2a4a22a2ad
Reviewed-on: https://webrtc-review.googlesource.com/53401
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22042}
RTCConfiguration.
This bug holds IceConfig unchanged in PeerConnection::SetConfiguration
when the update of IceConfig is necessary, unless ice_check_min_interval
is part of the update.
TBR=deadbeef@webrtc.org
Bug: webrtc:8898
Change-Id: I87774863bfedd7c05408fb22937d7322e53417c3
Reviewed-on: https://webrtc-review.googlesource.com/54201
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#22041}
This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3.
Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java
Original change's description:
> Enables PeerConnectionFactory using external fec controller
>
> Bug: webrtc:8799
> Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8
> Reviewed-on: https://webrtc-review.googlesource.com/43961
> Commit-Queue: Ying Wang <yinwa@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22038}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org
Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8799
Reviewed-on: https://webrtc-review.googlesource.com/54080
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22040}
There's currently a race while deleting an instance of the
class if frame delivery hasn't been explicitly stopped.
Bug: webrtc:8894
Change-Id: I1c60e6e3f9a3e51b16a21a610d21e33fcf58cc0e
Tbr: kthelgason@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/53980
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22039}
As part of this, I'm moving global thread related functions over to
platform_thread_types.* and introducing platform_thread_types.cc
for the implementation.
Change-Id: I4624877fb379e77d215f4ecd60f20b06d8df3ce5
Bug: webrtc:8893
Reviewed-on: https://webrtc-review.googlesource.com/53940
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22037}
Move headers used only in implementation of TaskQueue to .cc files
Bug: None
Change-Id: I6efc9279ae2fef4693b91e992c66236cb9d3d27c
Reviewed-on: https://webrtc-review.googlesource.com/51763
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22035}
This method used to just wrap frame when passed a native frame and
create a new one when passed non-native frame. This caused a memory
leak when a new frame was returned because the caller didn't release
the frame. Now the method always returns a new frame and the caller is
responsible for releasing it.
Bug: webrtc:8892, b/72675429
Change-Id: I06d67a6ed4c059cae1d709c51b0266f9c72fef1a
Reviewed-on: https://webrtc-review.googlesource.com/53840
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22033}
Avoid including audio_processing.h from within AEC3.
Bug: webrtc:8844
Change-Id: I02c475c2fb84e2c24eac86baac3c7edaa08bebc0
Reviewed-on: https://webrtc-review.googlesource.com/53065
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22029}
This is one of several small steps of separating APM and AEC3.
Bug: webrtc:8844
Change-Id: Ib6e518fec5f7566cab3823ab35fcede8433f8f4e
Reviewed-on: https://webrtc-review.googlesource.com/53142
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22028}
The max bitrate is already constrained to the bitrate specified by
ScreenshareLayerConfig, it should not be boosted past that.
Bug: webrtc:8785
Change-Id: If400e829b6bf209e3052e908fcabd65ba2c9457e
Reviewed-on: https://webrtc-review.googlesource.com/53320
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22027}
Bug: webrtc:8764
Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d
Reviewed-on: https://webrtc-review.googlesource.com/49580
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22026}
This change includes updates to the sdp logic, and transceiver
dissociation and also tests these updates. The sdp validation for
unified plan is updated to consider both the stored remote and local
descriptions for an offer, because either could be the most up to date.
This is important when considering a recycled m section. This also
updates to only dissociate a transceiver when we are setting the remote
or local description from an offer. The final small update allows us to
properly create a media description for a transceiver that is not new
but is part of a recycled m section that has only been set locally for
an offer and we are re-offering.
Bug: webrtc:8765
Change-Id: Ia86e54fcd977478824cfa88ebaf992215ed68aae
Reviewed-on: https://webrtc-review.googlesource.com/52080
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22025}
This reverts commit 4e849f6925b2ac44b0957a228d7131fc391fca54.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Reland "Moved congestion controller to task queue.""
>
> This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793.
>
> Reason for revert: Cause increased congestion and deadlocks in downstream project
>
> Original change's description:
> > Reland "Moved congestion controller to task queue."
> >
> > This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
> >
> > Original change's description:
> > > Moved congestion controller to task queue.
> > >
> > > The goal of this work is to make it easier to experiment with the
> > > bandwidth estimation implementation. For this reason network control
> > > functionality is moved from SendSideCongestionController(SSCC),
> > > PacedSender and BitrateController to the newly created
> > > GoogCcNetworkController which implements the newly created
> > > NetworkControllerInterface. This allows the implementation to be
> > > replaced at runtime in the future.
> > >
> > > This is the first part of a split of a larger CL, see:
> > > https://webrtc-review.googlesource.com/c/src/+/39788/8
> > > For further explanations.
> > >
> > > Bug: webrtc:8415
> > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> > > Reviewed-on: https://webrtc-review.googlesource.com/43840
> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21868}
> >
> > Bug: webrtc:8415
> > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
> > Reviewed-on: https://webrtc-review.googlesource.com/48000
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21899}
>
> TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:8415
> Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83
> Reviewed-on: https://webrtc-review.googlesource.com/52980
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22017}
TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/53262
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22023}
This class adds a convenience method that allows *sending* a task
to the queue (as opposed to posting). Sending is essentially
Post+Wait, a pattern that we don't want to encourage use of
in production code, but is convenient to have from a testing
perspective and there are already several places in the
source code where we use it.
Change-Id: I6efd1b2257e6c641294bb6e4eb53b0021d9553ca
Bug: webrtc:8848
Reviewed-on: https://webrtc-review.googlesource.com/50441
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22022}
The reference back to the decoder class in the decode callback
was null. Due to the amazing properties of ObjC this led to the
setError call to silently fail.
Bug: webrtc:8600
Change-Id: I3f70fbe4c9d533c8612d0bc7bc40813252e492fd
Reviewed-on: https://webrtc-review.googlesource.com/52460
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22021}
We want api/peerconnectioninterface.h (and corresponding build target)
to not depend on call.h, and generally we treat Call as an internal,
non-api, class. But we need CallFactoryInterface in the api in order to
enable use of PeerConnection with or without support for media.
Making CallConfig a top-level class makes it possible to forward declare
it, together with Call, for use in callfactoryinterface.h and
peerconnectioninterface.h.
Delete the peerconnection_and_implicit_call_api target, replaced by
new target callfactory_api, to link between Call and Peerconnection.
Bug: webrtc:7504
Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1
Reviewed-on: https://webrtc-review.googlesource.com/46201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22020}
This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793.
Reason for revert: Cause increased congestion and deadlocks in downstream project
Original change's description:
> Reland "Moved congestion controller to task queue."
>
> This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
>
> Original change's description:
> > Moved congestion controller to task queue.
> >
> > The goal of this work is to make it easier to experiment with the
> > bandwidth estimation implementation. For this reason network control
> > functionality is moved from SendSideCongestionController(SSCC),
> > PacedSender and BitrateController to the newly created
> > GoogCcNetworkController which implements the newly created
> > NetworkControllerInterface. This allows the implementation to be
> > replaced at runtime in the future.
> >
> > This is the first part of a split of a larger CL, see:
> > https://webrtc-review.googlesource.com/c/src/+/39788/8
> > For further explanations.
> >
> > Bug: webrtc:8415
> > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> > Reviewed-on: https://webrtc-review.googlesource.com/43840
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21868}
>
> Bug: webrtc:8415
> Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
> Reviewed-on: https://webrtc-review.googlesource.com/48000
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21899}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8415
Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83
Reviewed-on: https://webrtc-review.googlesource.com/52980
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22017}
This breaks the dependency api:audio_mixer_api --> modules:module_api,
and allows peerconnectioninterface.h to include audio_mixer.h, without
introducing a dependency cycle.
In addition, un-inline all AudioFrame methods, moving implementations
to audio_frame.cc, and replace assert by RTC_CHECK_*.
Bug: webrtc:7504
Change-Id: I11e3d3d22716e9b98976bf830103fbb06e7bbb77
Reviewed-on: https://webrtc-review.googlesource.com/51860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22016}
