Commit graph

32 commits

Author SHA1 Message Date
Peter Thatcher
54cfdf3754
Add control for choosing between AEC3 and AECM (#67) 2022-02-23 18:59:14 -07:00
Tommi
87f7090fd9 Replace more instances of rtc::RefCountedObject with make_ref_counted.
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.

Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
2021-04-27 17:01:59 +00:00
Artem Titov
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
Artem Titov
c8421c4c3e Replace rtc::ThreadChecker with webrtc::SequenceChecker
Bug: webrtc:12419
Change-Id: I825c014cc1c4b1dcba5ef300409d44859e971144
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33136}
2021-02-02 14:56:27 +00:00
Mirko Bonadei
3b68aa346a Move some RTC_LOG to RTC_DLOG.
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}
2021-01-28 10:05:00 +00:00
Karl Wiberg
c95b939667 Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.

Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
2020-11-09 10:47:55 +00:00
Mirko Bonadei
f9e5248f5d Inclusive language in //sdk/android.
Bug: webrtc:11680
Change-Id: I80f6b3c2ba21f49b0c05ebc27aecfc27a7be5836
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178392
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31590}
2020-06-30 13:34:05 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
Alex Narest
bbeb10925e Reporting audio device underrun counter
Bug: webrtc:10884
Change-Id: I35636fcbc1e2a19a89242379cdff6ec5c12fd21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28874}
2019-08-16 11:49:55 +00:00
Danil Chapovalov
896f4b666c Use Default instead of GlobalTaskQueueFactory to create AudioDeviceBuffer for android
Bug: webrtc:10284
Change-Id: I979eab78e1841e2b6900d7729159ee69274af8e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144031
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28425}
2019-07-01 09:43:06 +00:00
Eric Stevenson
0d65fb5451 Mass refactoring: Change JNI #includes to use full paths (webrtc/).
Using relative paths for JNI includes is causing build failures in chromium.

WebRTC already uses full include paths for generated JNI headers, so this CL
just removes the "jni_package" parameter from WebRTC generate_jni() targets
and removes the "jni/" portion of includes. The "jni_package" variable will be
removed from the generate_jni() template shortly.

To fix includes:
find . -name *.cc -exec sed -i -E 's@(#include.+generated.+jni)/jni/(.+_jni.h)@\1/\2@' {} \;

See https://groups.google.com/a/chromium.org/forum/?#!topic/java/MEovGrAwbqI
for discussion on naming scheme.

No-Try: True
TBR: kwiberg@webrtc.org
Bug: chromium:964169
Change-Id: I758c1b41bf6f5005587e55b82f14065fe251baad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143521
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28380}
2019-06-26 08:23:14 +00:00
Sebastian Jansson
c01367db40 Deprecating ThreadChecker specific interface.
All changes outside thread_checker.h are by:
s/CalledOnValidThread/IsCurrent/
s/DetachFromThread/Detach/

Bug: webrtc:9883
Change-Id: Idbb1086bff0817db58e770116acf4c9d60fae8b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27494}
2019-04-08 16:58:07 +00:00
Danil Chapovalov
1c41be6e05 Propagate TaskQueueFactory to AudioDeviceBuffer
keep using GlobalTaskQueueFactory in android/ios bindings.
Switch to DefaultTaskQueueFactory in tests.

Bug: webrtc:10284
Change-Id: I034c70542be5eeb830be86527830d51204fb2855
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130223
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27380}
2019-04-01 08:00:49 +00:00
Paulina Hensman
1ca30a7e41 Decouple input and output sample rate overrides.
We may sometimes want to override only input or only output, or
override them with different values. This change allows setting the
overrides separately.


Change-Id: Ib0c44cb7a3cfa834f997fb6cd54f7cad68705f41
Bug: webrtc:10441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128763
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27236}
2019-03-22 11:56:01 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Steve Anton
40d55331d7 Include absl/memory/memory.h if absl::make_unique is used
Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Iaf4533d2ce0e80b351a8a664ef8cf7ba0e5ec583
Reviewed-on: https://webrtc-review.googlesource.com/c/115746
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26168}
2019-01-08 20:08:32 +00:00
Paulina Hensman
17d57c7c13 Reintroduce division by two for audio playout delay
When migrating the audio device, we accidentally dropped a /2 for
PlayoutDelay. This meant we would estimate a delay of 150ms instead of
75ms for JavaAudioDeviceModules. This change fixes that.

Bug: webrtc:7452
Change-Id: I20b70ebf141410209953243ae665644b92e480f5
Reviewed-on: https://webrtc-review.googlesource.com/c/113946
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25986}
2018-12-12 11:43:14 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
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}
2018-10-23 11:32:56 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Magnus Jedvert
f4aeb891b7 Android: Handle StartRecording() failure gracefully
This CL also adds a test to test the behavior when StartRecording()
fails, which is the case when e.g. the microphone is already in use.

Bug: webrtc:9491
Change-Id: Ifce60ce5e9b7fa7521ca5c9fe20794233456b9ce
Reviewed-on: https://webrtc-review.googlesource.com/87105
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23842}
2018-07-04 14:08:54 +00:00
Paulina Hensman
7a29426142 Detach audio devices from thread on terminate.
To allow the AudioDeviceModule to be reinitialized on a different thread
after termination, detach AudioDeviceModule and the input/output devices
when Terminate is called. Also destroy the AudioDeviceBuffer.

Bug: webrtc:7452
Change-Id: I50ef77c531f33d4efa0567d0475dd8280337bed9
Reviewed-on: https://webrtc-review.googlesource.com/86127
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23784}
2018-06-29 12:00:17 +00:00
Danil Chapovalov
196100efa6 Replace rtc::Optional with absl::optional
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}
2018-06-21 09:32:56 +00:00
henrika
ec9c745228 Adds support for new Windows ADM with limited API support.
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}
2018-06-08 14:44:38 +00:00
Magnus Jedvert
c7da266cb8 Android JNI generation: Set JNI namespace in build files
This CL removes the use of the @JNINamespace annotation and instead
sets the correct JNI namespace in the build file.

Bug: webrtc:8278
Change-Id: Ia4490399e45a97d56b02c260fd80df4edfa092bf
Reviewed-on: https://webrtc-review.googlesource.com/76440
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23299}
2018-05-18 10:44:38 +00:00
Magnus Jedvert
8a483be137 Android JNI: Disallow automatic conversion from long to jint
Start using JniIntWrapper from Chromium instead of bypassing
it in jni_generator_helper.h.

Bug: webrtc:8278
Change-Id: I20313e1e610b05f79c210e823ab50cfb2073674e
Reviewed-on: https://webrtc-review.googlesource.com/74841
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23230}
2018-05-15 07:43:05 +00:00
Paulina Hensman
498592d391 Increase logging for Java ADM
The new ADM code removed some redundancies, which led to a decrease in
log output. This especially affected NS and AEC logs. This change
reintroduces these log messages, making debugging easier. "Acoustic
Echo Canceler" has been changed to AEC for easier grepping.

Some new logging is also added.

Bug: webrtc:7452
Change-Id: I9bfb91895931d73d92f3187c8c7c5b7524ac05ba
Reviewed-on: https://webrtc-review.googlesource.com/71401
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23003}
2018-04-24 13:50:06 +00:00
Sami Kalliomäki
7d8f5949b2 Make depending on a specific audio implementation optional.
Splits out audio_java into audio_api_java and
java_audio_device_module_java.

Makes depending on java_audio_device_module_jni optional for clients
that do not use it. It is only necessary to depend on this target if
depending on java_audio_device_module_java.

Also some cleanup.

Bug: webrtc:7452
Change-Id: Ic6c4dbe11db3ed8330802a8e90203acb8ef18e72
Reviewed-on: https://webrtc-review.googlesource.com/70220
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22981}
2018-04-23 14:18:47 +00:00
Paulina Hensman
30d5f6d7f7 Small cleanup in sdk/android ADM
Mainly remove CHECKinitialized_ macro and AGC functionality. Also make
actual behavior clearer in some functions.

Bug: webrtc:7452
Change-Id: I1eac86f4eaff7b14820d3e4192b15c20ab6acb45
Reviewed-on: https://webrtc-review.googlesource.com/69161
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22820}
2018-04-11 13:44:56 +00:00
Magnus Jedvert
66f1e9eb34 Android: Add AudioDeviceModule interface and clean up implementation code
This CL introduces sdk/android/api/org/webrtc/audio/AudioDeviceModule.java,
which is the new interface for audio device modules on Android.

This CL also refactors the main AudioDeviceModule implementation, which
is sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java and makes
it conform to the new interface. The old code used global static methods
to configure the audio device code. This CL gets rid of all that and uses
a builder pattern in JavaAudioDeviceModule instead. The only two dynamic
methods left in the interface are setSpeakerMute() and setMicrophoneMute().
Removing the global static methods allowed a significant cleanup, and e.g.
the file sdk/android/src/jni/audio_device/audio_manager.cc has been
completely removed.

The PeerConnectionFactory interface is also updated to allow passing in
an external AudioDeviceModule. The current built-in ADM is encapsulated
under LegacyAudioDeviceModule.java, which is the default for now to
ensure backwards compatibility.

Bug: webrtc:7452
Change-Id: I64d5f4dba9a004da001f1acb2bd0c1b1f2b64f21
Reviewed-on: https://webrtc-review.googlesource.com/65360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22765}
2018-04-06 10:13:02 +00:00
Magnus Jedvert
e2be7ee2e8 Android: Split out audio device targets
This CL splits out the audio device module Java code into a separate
target, and also splits up the audio device module implementations into
three different build targets, one for OpenSLES, AAudio, and the Java
based implementation.

Bug: webrtc:7452, webrtc:9048
Change-Id: I8ec09c73580b468837223ddd420fb29ca61fdea5
Reviewed-on: https://webrtc-review.googlesource.com/66461
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22727}
2018-04-04 11:50:30 +00:00
Magnus Jedvert
a21090b770 Android: Remove IsCommunicationModeEnabled() from AudioManager
This method is only used for logging and is blocking further refactoring
work. Once the refactoring and cleanup of the external AudioDeviceModule
is complete, we can revisit what logging we want and need and add it in
a cleaner way.

Bug: webrtc:7452
Change-Id: If08bcfb37860e9e7b9b5105cb75f748b53775f69
Reviewed-on: https://webrtc-review.googlesource.com/65460
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22678}
2018-03-29 12:06:17 +00:00
Magnus Jedvert
1a18e0ac46 Android audio code: Replace C++ template with input/output interface
Bug: webrtc:7452
Change-Id: Id816500051e065918bba5c2235d38ad8eb50a8eb
Reviewed-on: https://webrtc-review.googlesource.com/64442
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22660}
2018-03-28 19:19:37 +00:00
Renamed from sdk/android/src/jni/audio_device/audio_device_template_android.h (Browse further)