| a6e4328680e263a0b5f423828ae653816f2fac70 |
|
09-Dec-2015 |
Tommi <tommi@webrtc.org> |
Remove unnecessary test code on Windows.
BUG=chromium:568266,chromium:567264
R=niklas.enbom@webrtc.org
Review URL: https://codereview.webrtc.org/1506203006 .
Cr-Commit-Position: refs/heads/master@{#10961}
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 2515af28e97213b4a4b89269f6b855378d31e153 |
|
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Removing some unnecessary string manipulation code from VoEBase::GetVersion().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1493663002
Cr-Commit-Position: refs/heads/master@{#10868}
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 3e6db2321ccdc8738c9cecbe9bdab13d4f0f658d |
|
26-Nov-2015 |
kjellander <kjellander@webrtc.org> |
audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 13725089ef91f932b37b2447c3f05d9cd9f89984 |
|
25-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID.
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1459083007
Cr-Commit-Position: refs/heads/master@{#10788}
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 74640895fafbb90a6630a6a91b80da0a7cff229c |
|
29-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
audio_coding: rename interface -> include
BUG=webrtc:5095
R=henrik.lundin@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417173004 .
Cr-Commit-Position: refs/heads/master@{#10444}
/external/webrtc/webrtc/voice_engine/voice_engine_impl.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/voice_engine/voice_engine_impl.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/voice_engine/voice_engine_impl.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/voice_engine/voice_engine_impl.cc
|
| 0d266054acece70259fc1e85026194154f41e5a0 |
|
04-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: apply new style guide on VoE interfaces and their implementations
Changes:
1. Ran clang-format on VoE interfaces and their implementations.
2. Replaced virtual with override in derived classes.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49239004
Cr-Commit-Position: refs/heads/master@{#9130}
/external/webrtc/webrtc/voice_engine/voice_engine_impl.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/voice_engine/voice_engine_impl.cc
|
| 34fe0153b90c3e616a0fd5aba8aa8118ae600760 |
|
22-Apr-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reland "Stop using ACM factory in VoiceEngine"
This change was originally landed as r5954, but had to be reverted in
r5955 due to bots failing. The failures should be fixed in r5956,
so the original change is now relanded.
BUG=2996
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5958 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 0c108d0b4d47c573ccb8ccb6fc13fa0d96f2e2b1 |
|
22-Apr-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Stop using ACM factory in VoiceEngine"
Some of the bots where breaking.
TBR=henrika@webrtc.org
BUG=2996
Review URL: https://webrtc-codereview.appspot.com/12319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5955 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 139706ec0bfc0cf70e0734a92ecba4251f9bb936 |
|
22-Apr-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Stop using ACM factory in VoiceEngine
The factory injection was introduces in order to facilitate switching
between ACM1 and ACM2. Now, ACM1 is being deprecated, and this switching
mechanism is no longer needed.
BUG=2996
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5954 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 372ae83228cba221771740301d67bb07ad58da81 |
|
22-Apr-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reland "Make VoiceEngine choose ACM2 by default""
This cl was originally committed as r5923, but was reverted in r5926
due to a blocking bug (issue 3143). The blocking bug was resolved in
r5936.
BUG=2996
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5950 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| e2e9abb3bcb2b2c92b3205d45d0b141811e336c0 |
|
17-Apr-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Make VoiceEngine choose ACM2 by default"
The reason for reverting is that Issue 3143 should be resolved
first.
TBR=henrika@webrtc.org
BUG=3143
Review URL: https://webrtc-codereview.appspot.com/12119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5926 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 6cec07f6a7ff5d90a55ba18c90cfad8da295aaa7 |
|
17-Apr-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make VoiceEngine choose ACM2 by default
The use of a factory for ACM will be removed in later CLs.
BUG=2996
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5923 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 8883a0f47fa22fa8194c8afb23006610e2c647ab |
|
09-Apr-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds
Landing https://webrtc-codereview.appspot.com/11419004/ manually.
TBR=niklase
BUG=none
Review URL: https://webrtc-codereview.appspot.com/11439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5872 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 944cbeb2926feb86a687e5fda9e2ac88ea8e3001 |
|
18-Mar-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Resolves TSan v2 warnings in voe_auto_test.
See bug report for details.
BUG=1590
R=tommi@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5714 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 573a1b45b5b7638605d9727be57c73e838d6ee45 |
|
10-Jan-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Android: Fixes crash when exiting WebRTCDemo.
BUG=2738
R=fischman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5365 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 9ee75e9c77b467e74e470905822d0279b0e8a639 |
|
11-Dec-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
BUG=N/A
R=fischman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| a750044396f9b7ae4a08e81924c4ee1e1ed09e66 |
|
20-Nov-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes a crash in VoE when unregistering JNI hooks.
BUG=11695087
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5144 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| e509f943eded156f7a8365b0b001abe73646acfa |
|
12-Sep-2013 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
This issue is related to
https://chromereviews.googleplex.com/9908014/
I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL.
BUG=
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2171004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 82f014aa0bc225076516a3d77ad02deb69cfd809 |
|
10-Sep-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
OpenSL (not default): Enables low latency audio on Android.
BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2032004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 9080518a3928285be9f94684adad064c65d2cdf3 |
|
05-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Restore severity precondition to logging.h.
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| d900e8bea84c474696bf0219aed1353ce65ffd8e |
|
03-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Proper spacing for end-of-namespace comments.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1760006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 956aa7e0874f2e08c335a82a2c32f400fac8b031 |
|
21-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in voice_engine/
BUG=1662
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1434005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 9213521ea98b0977c7cdabd2853060835af226f3 |
|
14-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove const for plain data types in voice_engine/
BUG=1644
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1463004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 6141e13873d0fdea626de08dfec2efa2c9171c76 |
|
09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 -> int32_t in voice_engine/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1305004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 0989fb7bfa482074e0161ea177653a44174ac492 |
|
15-Feb-2013 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make VoiceEngineImpl inherit from VoiceEngine.
This associates the two types instead of incorrectly reinterpret casting
VoiceEngineImpl* to VoiceEngine* (since these types were previously unrelated).
Please see more details in the bug for how this is currently causing problems
with security tools.
BUG=38612
Review URL: https://webrtc-codereview.appspot.com/1099013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3520 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/voice_engine_impl.cc
|