Second round of the new Windows ADM is now ready for review. Main
changes are:
Supports internal (automatic) restart of audio streams when an active
audio stream disconnects (happens when a device is removed).
Adds support for IAudioClient3 and IAudioClient2 for platforms which
supports it (>Win8 and >Win10).
Modifies the threading model to support restart "from the inside" on
the native audio thread.
Adds two new test methods for the ADM to emulate restart events or
stream-switch events.
Adds two new test methods to support rate conversion to ensure that
audio can be tested in loopback even if devices runs at different
sample rates.
Added initial components for low-latency support. Verified that it works
but disabled it with a flag for now.
Bug: webrtc:9265
Change-Id: Ia8e577daabea6b433f2c2eabab4e46ce8added6a
Reviewed-on: https://webrtc-review.googlesource.com/86020
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24578}
First version was reverted in https://webrtc-review.googlesource.com/c/src/+/97941.
The issue is now fixed.
TBR=ivoc
Bug: b/113648245
Change-Id: If631fdea95aa963952f15e48e9d2d678797dc225
Reviewed-on: https://webrtc-review.googlesource.com/97942
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24573}
This reverts commit cd87e014f34069fd5a73c1ed5b74ddf251a95c2d.
Reason for revert: Somehow introduces a race where rtc::Thread auto-wrapping may be applied to the TestAudioDevice thread rather than the main thread. This causes failures when running video_engine_tests without any test filter.
Original change's description:
> Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper."
>
> This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a
>
> Original change's description:
> > Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
> >
> > In addition, let the processing thread loop explicitly, and not use
> > the deprecated builtin looping in PlatformThread.
> >
> > Bug: webrtc:3380
> > Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> > Reviewed-on: https://webrtc-review.googlesource.com/96544
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24492}
>
> Bug: webrtc:3380
> Change-Id: I671e3a60ace6ade765a8537b7e20e36f1782a60d
> Reviewed-on: https://webrtc-review.googlesource.com/97320
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24528}
TBR=henrika@webrtc.org,nisse@webrtc.org,titovartem@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
# Skip anyway, needed for a two-step revert.
No-try: True
Bug: webrtc:3380
Change-Id: Ia7c8cfab36b8259f150b5ccd0c28defd0e7237f6
Reviewed-on: https://webrtc-review.googlesource.com/97682
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24552}
Also ensures that audio parameters are accessed atomically.
Bug: b/113648245
Change-Id: Ic812bfe2b2c4cfb3b00d9d411bb4986dfeda1028
Reviewed-on: https://webrtc-review.googlesource.com/97331
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24550}
This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a
Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
>
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
>
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}
Bug: webrtc:3380
Change-Id: I671e3a60ace6ade765a8537b7e20e36f1782a60d
Reviewed-on: https://webrtc-review.googlesource.com/97320
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24528}
This reverts commit 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a.
Reason for revert: Makes the perf test RampUpTest.AudioTransportSequenceNumber fail on windows, almost every time.
Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
>
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
>
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}
TBR=henrika@webrtc.org,nisse@webrtc.org,titovartem@webrtc.org
Change-Id: I8867a22d695494bd5abfda6a97f0719cb3ff3d66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3380
Reviewed-on: https://webrtc-review.googlesource.com/96840
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24496}
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.
A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.
The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.
The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.
Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
In addition, let the processing thread loop explicitly, and not use
the deprecated builtin looping in PlatformThread.
Bug: webrtc:3380
Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
Reviewed-on: https://webrtc-review.googlesource.com/96544
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24492}
The old test supported audio but only in combination with a fake ADM.
The new version allows the user to run real video and audio.
Now possible to do:
./out/Debug/video_loopback.exe --audio --use_real_adm
To run the test in loopback using real default audio devices.
By default:
./out/Debug/video_loopback.exe --audio
runs with fake audio devices as before.
Bug: webrtc:9265
Change-Id: Id89924ec0276f929487c71fc6321dcd9cb92693d
Reviewed-on: https://webrtc-review.googlesource.com/96161
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24463}
This doesn't cause problems because seems like GN just ignores these attempts and confines everything to the root.
Bug: None
Change-Id: Ief50b6e10f3950b319d0bde547f91dcc7f237cda
Reviewed-on: https://webrtc-review.googlesource.com/90041
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24063}
WebRTC internal headers are always included starting from the root
(e.g. #include "modules/audio_device/..."), so there is no need to
specify the include_dirs removed by this CL.
Bug: webrtc:9538
Change-Id: If26edecc004c6e8c3bbef3c8185c7e272110c951
Reviewed-on: https://webrtc-review.googlesource.com/89391
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24034}
The remaining suppression flags require some work in order to be
removed.
Bug: webrtc:9251
Change-Id: I506f6c730456a4c030b87dbc7ba23c7b3359e272
Reviewed-on: https://webrtc-review.googlesource.com/87920
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23932}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script passing top level directories except rtc_base and api
find $@ -type f \( -name \*.h -o -name \*.cc -o -name \*.mm \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I9465c172e65ba6e6ed4e4fdc35b0b265038d6f71
Reviewed-on: https://webrtc-review.googlesource.com/84584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23697}
Because thread_ object is created in Init, destructor used to crash when
calling thread_->Stop() because it was referencing a null pointer.
Bug: webrtc:9404
Change-Id: I1c943d0fa50f9341aaa516b32495bb25bf4d664b
Reviewed-on: https://webrtc-review.googlesource.com/84122
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23682}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This allows constructing TestAudioDeviceModule on a different thread
than the worker thread and avoids unnecessary invoke. Before,
thread->Start() would fail in a thread check.
Bug: b/79961243
Change-Id: I5c55d8feada2b0ae12bc121f3f795e76a8d04059
Reviewed-on: https://webrtc-review.googlesource.com/82941
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23574}
Summary of what this CL does:
Existing users can keep using the old ADM for Windows as before.
A new ADM for Windows is created and a dedicated factory method is used
to create it. The old way (using AudioDeviceImpl) is not utilized.
The new ADM is based on a structure where most of the "action" takes
place in new AudioInput/AudioOutput implementations. This is inline
with our mobile platforms and also makes it easier to break out common
parts into a base class.
The AudioDevice unittest has always mainly focused on the "Start/Stop"-
parts of the ADM and not the complete ADM interface. This new ADM supports
all tests in AudioDeviceTest and is therefore tested in combination with
the old version. A value-parametrized test us added for Windows builds.
Improved readability, threading model and makes the code easier to maintain.
Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility.
Bug: webrtc:9265
Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d
Reviewed-on: https://webrtc-review.googlesource.com/78060
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23554}
Made some minor changes to resolve the issue. Only affects Debug builds.
NOTRY=TRUE
Bug: webrtc:9310
Change-Id: Ieeeb57d24b559282b2eefd4d8785f7cfe4f44e40
Reviewed-on: https://webrtc-review.googlesource.com/79624
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23434}
Instead of making multiple calls to the std::stringstream << operator,
collect all the arguments and make a single printf-like variadic call
under the hood.
Besides reducing our reliance on iostreams, this makes each RTC_LOG_*
call site smaller; in aggregate, this reduces the size of
libjingle_peerconnection_so.so by 28-32 kB.
A quick benchmark indicates that this change makes log statements
a few percent slower.
Bug: webrtc:8982, webrtc:9185
Change-Id: I3137a4dd8ac510e8d910acccb0c97ce4fffb61c9
Reviewed-on: https://webrtc-review.googlesource.com/75440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23375}
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.
The unit tests are re-implemented as XCTests.
(was: https://webrtc-review.googlesource.com/c/src/+/67300)
Bug: webrtc:9120
Change-Id: I46c09900246f75ca5285aeb38f7b8b295784ffac
Reviewed-on: https://webrtc-review.googlesource.com/76741
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23238}
Now using attribute to ensure that we avoid error like these when bulding with -Wthread-safety-analysis:
error: mutex '_critSect' is still held at the end of function [-Werror,-Wthread-safety-analysis]
RTC_NO_THREAD_SAFETY_ANALYSIS is an attribute on functions or methods, which turns off thread safety
checking for that method. It provides an escape hatch for functions which are either
(1) deliberately thread-unsafe, or
(2) are thread-safe, but too complicated for the analysis to understand.
Bug: webrtc:9202
Change-Id: Ie332bca7eb7eb535ed965de5ddc42872c4f30602
Reviewed-on: https://webrtc-review.googlesource.com/76562
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23221}
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.
The unit tests are re-implemented as XCTests.
(was: https://webrtc-review.googlesource.com/c/src/+/67300)
Bug: webrtc:9120
Change-Id: I07340505137b16c2dd487569ad0112f984557bba
Reviewed-on: https://webrtc-review.googlesource.com/75125
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23208}
This CL is part of the effort to remove warning suppression flags from
the WebRTC build.
Bug: webrtc:9251
Change-Id: I45ece25e897a14a6d4ce8a90ba59688f8fc6fe32
Reviewed-on: https://webrtc-review.googlesource.com/75503
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23187}
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.
The unit tests are re-implemented as XCTests.
Bug: webrtc:9120
Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
Reviewed-on: https://webrtc-review.googlesource.com/67300
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23163}
...continuation of review in https://webrtc-review.googlesource.com/c/src/+/70781
This CL ensures that the FineAudioBuffer can support stereo and also adapts
all classes which uses the FineAudioBuffer.
Note that, this CL does NOT enable stereo on mobile platforms by default. All it does is to ensure
that we *can*. As is, the only functional change is that all clients
will now use a FineAudioBuffer implementation which supports stereo (see
separate unittest).
The FineAudioBuffer constructor has been modified since it is better to
utilize the information provided in the injected AudioDeviceBuffer pointer
instead of forcing the user to supply redundant parameters.
The capacity parameter was also removed since it adds no value now when the
more flexible rtc::BufferT is used.
I have also done local changes (not included in the CL) where I switch
all affected audio backends to stereo and verified that it works in real-time
on all affected platforms (Androiod:OpenSL ES, Android:AAudio and iOS).
Also note that, changes in:
sdk/android/src/jni/audio_device/aaudio_player.cc
sdk/android/src/jni/audio_device/aaudio_recorder.cc
sdk/android/src/jni/audio_device/opensles_player.cc
sdk/android/src/jni/audio_device/opensles_recorder.cc
are simply copies of the changes done under modules/audio_device/android since we currently
have two versions of the ADM for Android.
Bug: webrtc:9172
Change-Id: I1ed3798bd1925381d68f0f9492af921f515b9053
Reviewed-on: https://webrtc-review.googlesource.com/71201
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22998}
Currently this warnings prevernt chromium roll into webrtc, because we
consider them as errors. So to unblock roll all warning are suppressed.
All places are documented into bug and will be fixed later.
TBR=henrika@webrtc.org
Bug: webrtc:9175
Change-Id: I0bf5a4b65eb49308e28f71a92d42b5fad6a99b74
Reviewed-on: https://webrtc-review.googlesource.com/71420
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22956}
This work is also done as a preparation for adding stereo support to the
FineAudioBuffer.
Review hints:
Actual changes are in modules/audio_device/fine_audio_buffer.h,cc, the rest is
just adaptations to match these changes.
We do have a forked ADM today, hence, some changes are duplicated.
The changes have been verified on all affected platforms.
Bug: webrtc:6560
Change-Id: I413af41c43809f61455c45ad383fc4b1c65e1fa1
Reviewed-on: https://webrtc-review.googlesource.com/70781
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22938}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
This CL removes PushCaptureData(), which is unused.
The reason I'm removing it is since this method is cauing chromium-style
violations for all files that includes
modules/audio_device/include/audio_device_defines.h, and it's annoying
to suppress it everywhere.
Bug: webrtc:8659
Change-Id: I9133d05259075d8e8ec89b764be934f37b5fa77e
Reviewed-on: https://webrtc-review.googlesource.com/66404
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22717}
