The logcat is spammed with warnings from timestamp aligner.
This CL make a workaround to use the timestamp aligner to get capture times,
without generating the warnings.
Bug: webrtc:14970
Change-Id: Idab4b298e0484a57841a214db9440f9ac6faaa4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296324
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#39486}
The logcat is spammed with warnings from timestamp aligner.
This CL make a workaround to use the timestamp aligner to get capture times,
without generating the warnings.
Bug: webrtc:14970
Change-Id: Idab4b298e0484a57841a214db9440f9ac6faaa4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296324
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#39486}
This is to avoid using 0 as a default value.
Also fix a bug in audio_device_buffer where the timestamp aligner used the wrong input timestamp.
Bug: webrtc:13609
Change-Id: I00016e68ab50d052990c2b9f80aa1e2d7e167b93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291118
Reviewed-by: Olov Brändström <brandstrom@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39177}
Add timestamps to audio_record_jni DataIsRecorded() function, and make
WebRtcAudioRecord find and send the time stamp to that function.
This CL is an continuation of
https://webrtc-review.googlesource.com/c/src/+/249085
Bug: webrtc:13609
Change-Id: I63ab89f1215893cbe1d11d9d8948f5639fc5cdfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249951
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35933}
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will
be used to store audio timestaps in future changes.
This is a part of the A/V sync metric metric feature for mobile. The metric
have already launched for web clients.
Bug: webrtc:13609
Change-Id: I0031843476ff1b573b262308fca52d587fae30b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35851}
Some locations in the WebRTC codebase RTC_LOG the value of the
__FUNCTION__ macro which probably is useful in debug mode. Moving
these instances to RTC_DLOG saves ~10 KiB on arm64.
Bug: webrtc:11986
Change-Id: I5d81cc459d2850657a712b9aed80c187edf49a3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203981
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33086}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
The last run-time logic for selecting function pointers was removed in
May 2016, here: https://codereview.webrtc.org/1955413003
It would be even better if we could eliminate the function pointers
entirely and just have different implementations that we select at
compile time; I've left a TODO asking for this.
Bug: webrtc:9553
Change-Id: Ica71d71e19759da00967168f6479b7eb8b46c590
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144053
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28414}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
This CL removes a set of DCHECKs in AudioDeviceBuffer (ADB) where the goal has been
to ensure that some methods are called on one and the same native I/O thread.
The implementation of the ADB is platform independent but the underlying (driving)
audio components differ between platforms. This combination has shown to generate complex
corner cases such as:
- OS dependent I/O-thread(s) changes while audio is active
- OS dependent audio device changes and it leads to restart of native I/O threads
- Start/Stop of audio has different timing depending on platform and possibly also usage of
JNI and/or emulators.
To summarize: the gain of maintaining the current strict thread checking (in Debug mode)
is not worth all the efforts trying to resolve complex dynamic cases where the native
I/O threads changes ID.
TBR=glaznev
Bug: b/115385789
Change-Id: I681c89adec497a18b97d2a40421c04ea218fd919
Reviewed-on: https://webrtc-review.googlesource.com/100200
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24723}
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}
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}
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}
...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}
Removes current_mic_level_, new_mic_level_ and clock_drift_, together
with APIs for accessing them.
Bug: webrtc:8598
Change-Id: I8e07396fcafd2a719e204730e2c7d26797bed762
Reviewed-on: https://webrtc-review.googlesource.com/39783
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21632}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/audio_device/audio_device_buffer.cc (Browse further)
