| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 9359b5b97810238e506350fdd0cb3d60001525f3 |
|
06-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Disabling AudioDeviceTest.StartStopPlayout on Android.
BUG=5046
TBR=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1374963003 .
Cr-Commit-Position: refs/heads/master@{#10178}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 82e20554cbf8152d67915e3ef764bad8064b6541 |
|
25-Sep-2015 |
henrika <henrika@webrtc.org> |
Modifies invalid DCHECK in AudioRecordJni::OnCacheDirectBufferAddress()
Ensures that we can restart audio recording on Android without hitting
a DCHECK. Also adds a symmetric design for the playout side.
BUG=webrtc:5000
TEST=modules_unittests --gtest_filter=AudioDevice*
Review URL: https://codereview.webrtc.org/1373443003
Cr-Commit-Position: refs/heads/master@{#10072}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| b297c5a01f88219da26cffe433804963d1b70f0f |
|
23-Jul-2015 |
pkasting <pkasting@chromium.org> |
Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .
These are mostly trivial changes and are separated out just to reduce the
diff on that change to the minimum possible.
Note explanatory comments on patch set 1.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1235643003
Cr-Commit-Position: refs/heads/master@{#9617}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 1b12cb0ef73dde404f8972cf29dcac2fac5d2306 |
|
24-Jun-2015 |
henrika <henrika@webrtc.org> |
Enabling AudioDeviceTest.StartStopPlayout on Nexus 9
BUG=webrtc:4682
Review URL: https://codereview.webrtc.org/1206733003
Cr-Commit-Position: refs/heads/master@{#9497}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 26b08605e2b99136fcc1cab0800234f469d6e236 |
|
04-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Use one scoped_refptr.
Uses webrtc/base/scoped_ref_ptr.h and removes the copy in
system_wrappers.
BUG=
R=kwiberg@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1152733005
Cr-Commit-Position: refs/heads/master@{#9370}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| ee369e4277e48624bb557f0264644ed19a40dd67 |
|
25-May-2015 |
henrika <henrika@chromium.org> |
Refactoring of AudioTrackJni and AudioRecordJni using new JVM/JNI classes
BUG=NONE
TEST=./webrtc/build/android/test_runner.py gtest -s modules_unittests --gtest_filter=AudioDevice*
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51079004
Cr-Commit-Position: refs/heads/master@{#9271}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 523183b4aa2118b483967fccfdbe5b848c3eb18d |
|
21-May-2015 |
henrika <henrika@chromium.org> |
Disables AudioDeviceTest.StartStopPlayout for Nexus 9 only
BUG=4682
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50019004
Cr-Commit-Position: refs/heads/master@{#9249}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| b26198972c1fcb4aa7abaf3895b007e301e7d5dc |
|
18-May-2015 |
henrika <henrika@chromium.org> |
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 8324b525dce2c502bbd24b3946bbae207645cde9 |
|
27-Mar-2015 |
henrika <henrika@chromium.org> |
Adding playout volume control to WebRtcAudioTrack.java.
Also adds a framework for an AudioManager to be used by both sides (playout and recording).
This initial implementation only does very simple tasks like setting up the correct audio
mode (needed for correct volume behavior). Note that this CL is mainly about modifying
the volume. The added AudioManager is only a place holder for future work. I could have
done the same parts in the WebRtcAudioTrack class but feel that it is better to move these
parts to an AudioManager already at this stage.
The AudioManager supports Init() where actual audio changes are done (set audio mode etc.)
but it can also be used a simple "construct-and-store-audio-parameters" unit, which is the
case here. Hence, the AM now serves as the center for getting audio parameters and then inject
these into playout and recording sides. Previously, both sides acquired their own parameters
and that is more error prone.
BUG=NONE
TEST=AudioDeviceTest
R=perkj@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45829004
Cr-Commit-Position: refs/heads/master@{#8875}
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 80d9aeeda530aaf7e1851b381f1f7545a876d75e |
|
19-Mar-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Adds full-duplex unit test to AudioDeviceTest on Android
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42709004
Cr-Commit-Position: refs/heads/master@{#8795}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8795 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 74d4792af574a90c4a13b3e57195883e2a546cbe |
|
10-Mar-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Fixes issue in RunPlayoutWithFileAsSource related to uninitialized member
BUG=4408
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45609004
Cr-Commit-Position: refs/heads/master@{#8668}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8668 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|
| 474d1eb22376898b36bcd04b0ce3860fa12fd984 |
|
09-Mar-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Adds C++/JNI/Java unit test for audio device module on Android.
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.
It also:
- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40069004
Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_device/android/audio_device_unittest.cc
|