Normally, packet/frame info is delivered to AudioReceiveStream's
source_tracker_ when an audio frame is pulled out of the stream (as a
side-effect of GetAudioFrameWithInfo). When playout is muted, though,
packets are thrown away in ChannelReceive::OnReceivedPayloadData, so
AudioRtpReceiver stops seeing updates to its RtpSources and any related
information (e.g. CSRCs and associated timestamps, levels).
Skipping the playout path here has a downside of being misaligned with
whatever playout delay would normally be, but it allows clients that
want to consume RtpSource information to be able to do so while playout
is muted.
Bug: None
Change-Id: Id00566b645de4196c2341611cd9e8b94b35aa157
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203500
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Ranveer Aggarwal <ranvr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33236}
This is a reland of 51e5c4b0f47926e2586d809e47dc60fe4812b782
It may happen that user will pass config with min bitrate > max bitrate.
In such case we can't generate cached_constraints and will crash before.
The reland will handle this situation gracefully.
Original change's description:
> Fix data race for config_ in AudioSendStream
>
> config_ was written and read on different threads without sync. This CL
> moves config access on worker_thread_ with all other required fields.
> It keeps only bitrate allocator accessed from worker_queue_, because
> it is used from it in other classes and supposed to be single threaded.
>
> Bug: None
> Change-Id: I23ece4dc8b09b41a8c589412bedd36d63b76cbc5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203267
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33125}
Bug: None
Change-Id: I274ff15208d69c25fb25a0f1dd0a0e37b72480b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33162}
This is a reland of d48a2b14e7545d0a0778df753e062075c044e2a1
The diff of the reland (what caused the tsan error) can be seen
by diffing patch sets 2 and 3. Essentially I missed keeping the calls
to the transport controller on the worker thread. Note to self to add
thread/sequence checks to that code so that we won't have to rely on
tsan :)
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
Bug: webrtc:11993, webrtc:12430
Change-Id: I4fccaa418d22c2087a55bbb3ddbb25fac3b4dfcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205580
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33153}
This reverts commit d48a2b14e7545d0a0778df753e062075c044e2a1.
Reason for revert: TSan tests started to fail constantly after this CL (it looks like it is flaky and the CQ was lucky to get green). See https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25042/overview.
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
TBR=nisse@webrtc.org,tommi@webrtc.org
Change-Id: Id87cf9cbcc8ed58e74d755a110f0ef9dd980e298
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205525
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33145}
This reverts commit 51e5c4b0f47926e2586d809e47dc60fe4812b782.
Reason for revert: Speculatively reverting because WebRTC fails to
roll due to a DCHECK in audio_send_stream.cc in a web platform test
and this is the only CL on the blamelist that touches that file.
Original change's description:
> Fix data race for config_ in AudioSendStream
>
> config_ was written and read on different threads without sync. This CL
> moves config access on worker_thread_ with all other required fields.
> It keeps only bitrate allocator accessed from worker_queue_, because
> it is used from it in other classes and supposed to be single threaded.
>
> Bug: None
> Change-Id: I23ece4dc8b09b41a8c589412bedd36d63b76cbc5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203267
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33125}
TBR=danilchap@webrtc.org,peah@webrtc.org,nisse@webrtc.org,hta@webrtc.org,titovartem@webrtc.org
# Initially not skipping CQ checks because original CL landed > 1 day
# ago. Adding NOTRY now because of ios_sim_x64_dbg_ios12 issues.
NOTRY=True
Bug: None
Change-Id: I33355198fca96faad7ac77538c7bd31425f46ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205340
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33142}
This moves the thread hop for network events, from BaseChannel and
into Call. The reason for this is to move the control over those hops
(including DeliverPacket[Async]) into the same class where the state
is held that is affected by those hops. Once that's done, we can start
moving the relevant network state over to the network thread and
eventually remove the hops.
I'm also adding several TODOs for tracking future steps and give
developers a heads up.
Bug: webrtc:11993
Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33138}
config_ was written and read on different threads without sync. This CL
moves config access on worker_thread_ with all other required fields.
It keeps only bitrate allocator accessed from worker_queue_, because
it is used from it in other classes and supposed to be single threaded.
Bug: None
Change-Id: I23ece4dc8b09b41a8c589412bedd36d63b76cbc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203267
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33125}
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.
`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.
Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl".
Bug: webrtc:12404
Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33108}
This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
This reverts commit 69241a93fb14f6527a26d5c94dde879013012d2a.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
This is a reland of 444e04be6988fbdcc039d775481ac22481ff9ff4
Reason for reland: resolved the breaks from downstream project
Original change's description:
> ChannelStatistics used for RTP stats in VoipStatistics.
>
> - Added local and remote RTP statistics query API.
> - Change includes simplifying remote SSRC change handling
> via received RTP and RTCP packets.
>
> Bug: webrtc:11989
> Change-Id: Ia3ee62c1191baaedc67e033ea3c661d8c9301abc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199060
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Tim Na <natim@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32954}
Bug: webrtc:11989
Change-Id: I88620a9f1c037b512821cac9d556905149666870
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201481
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32966}
- Added local and remote RTP statistics query API.
- Change includes simplifying remote SSRC change handling
via received RTP and RTCP packets.
Bug: webrtc:11989
Change-Id: Ia3ee62c1191baaedc67e033ea3c661d8c9301abc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199060
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32954}
This reverts commit f23e2144e86400e2d68097345d4b3dc7a4b7f8a4.
Reason for revert: Need further discussion on appropriate thread/tq requirements.
Original change's description:
> Add task queue to RtpRtcpInterface::Configuration.
>
> Let ModuleRtpRtcpImpl2 use the configured value instead of
> TaskQueueBase::Current().
>
> Intention is to allow construction of RtpRtcpImpl2 on any thread.
> If a task queue is provided (required for periodic rtt updates), the
> destruction of the object must be done on that same task queue.
>
> Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.
>
> Bug: None
> Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32949}
TBR=danilchap@webrtc.org,ilnik@webrtc.org,saza@webrtc.org,nisse@webrtc.org,srte@webrtc.org
Change-Id: I7e5007f524a39a6552973ec9744cd04c13162432
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201420
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32953}
Let ModuleRtpRtcpImpl2 use the configured value instead of
TaskQueueBase::Current().
Intention is to allow construction of RtpRtcpImpl2 on any thread.
If a task queue is provided (required for periodic rtt updates), the
destruction of the object must be done on that same task queue.
Also, delete ModuleRtpRtcpImpl2::Create, callers updated to use std::make_unique.
Bug: None
Change-Id: I412b7b1e1ce24722ffd23d16aa6c48a7214c9bcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199968
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32949}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
- This CL also affects some return type handling in Android Voip demo
app due to changes in return type handling.
Bug: webrtc:12193
Change-Id: Id76faf7c871476ed1f2d08fb587211ae234ae8d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196625
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32821}
This will soon become a compile-time error. Fix class hierarchies that
wrap StrictMock in a NiceMock or vice-versa by removing redundant
wrappings and removing inheritance from Nice/StrictMock and fixing the
call sites as appropriate.
Bug: b/173702213
Change-Id: Ic90b1f270c180f7308f40e52e358a8f6a6baad86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196461
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32783}
Delete unused macros BWE_MIN and BWE_MAX.
Move enum RateControlState: Make it a private enum class in
AimdRateControl, the only user.
Change users of the header file that only need BandwidthUsage, to
instead include api/network_state_predictor.h, the file defining this
class. As a result, fewer dependencies on
modules/remote_bitrate_estimator.
Bug: None
Change-Id: I4450c79dd58e1875d64dd74d1ae2cb7d911a14b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195222
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32704}
This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.
Transport-cc extension still needs to be negotiated.
Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}
This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447
This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
- TaskQueue needs to be destroyed at last to avoid thread race condition.
Bug: webrtc:12111
Change-Id: Ibfc96e2ebd71a2aa8d1ac8c83038d256bac0e600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193780
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32624}
Starting from Android N, mobile app may not be able to access
microphone while in background where it fails the call.
In order to mitigate the issue, delay the ADM initialization
as late as possible.
Bug: webrtc:12120
Change-Id: I0fbf0300299b6c53413dfaaf88f748edc0a06bc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191100
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32598}
Current VoIP Engine logic stops ADM from polling registered audio
channel when caller invokes StopPlay which can leads to incoming
RTP to be flushed and undesirable statistics report.
Instead, VoipBase::StopPlay should silence the decoded audio sample
from NetEq as muted to avoid mixing while allowing it go through prior
process for correct ingress statistic values.
The ADM stop playing logic will be triggered when all audio channels
are released by VoipBase::ReleaseChannel API.
Bug: webrtc:12121
Change-Id: I410eea4ea13f93acb465ab162a3c14c9819e2b92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191140
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32553}
After recently changing .pylintrc (see [1]) we discovered that
the presubmit check always checks all the python files when just
one python file gets updated.
This CL moves all these files one step closer to what the linter
wants.
Autogenerated with:
# Added all the files under pylint control to ~/Desktop/to-reformat
cat ~/Desktop/to-reformat | xargs sed -i '1i\\'
git cl format --python --full
This is part 1 out of 2. The second part will fix function names and
will not be automated.
[1] - https://webrtc-review.googlesource.com/c/src/+/186664
No-Presubmit: True
Bug: webrtc:12114
Change-Id: Idfec4d759f209a2090440d0af2413a1ddc01b841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32530}
This reverts commit f5e261aaf65cdf2eb903cdf40d651846be44f447.
Reason for revert: Breaks downstream projects.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
API to injecting a heavy audio processing operation into WebRTC audio capture pipeline
Bug: webrtc:12003
Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32291}
