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| author | xians <xians@chromium.org> | 2014-09-23 08:32:23 -0700 |
|---|---|---|
| committer | Commit bot <commit-bot@chromium.org> | 2014-09-23 15:32:36 +0000 |
| commit | e5d4e40f138c70432e05101b2ebea7b305aa0341 (patch) | |
| tree | 469c90b23f5e4c27983a2d2bdb865d51ab38f334 /content | |
| parent | 131ad350c467be1fe8e0c87acc85ae5c2339d21e (diff) | |
| download | chromium_src-e5d4e40f138c70432e05101b2ebea7b305aa0341.zip chromium_src-e5d4e40f138c70432e05101b2ebea7b305aa0341.tar.gz chromium_src-e5d4e40f138c70432e05101b2ebea7b305aa0341.tar.bz2 | |
Revert of Fix the way how we create webrtc::AudioProcessing in Chrome (patchset #5 id:100001 of https://codereview.chromium.org/588523002/)
Reason for revert:
It broke some internal webrtc bots, revert it for now and will reland it after fixing the problems.
http://chromegw.corp.google.com/i/internal.chromium.webrtc/builders/Mac%20Tester/builds/22092
Original issue's description:
> Fix the way how we create webrtc::AudioProcessing in Chrome.
>
> BUG=415935
> TEST=all webrtc tests in all bots + manual test to verify the agc loggings exist.
>
> Committed: https://crrev.com/a5e9fc62b7bf25931ffe6153cc738098d8119c28
> Cr-Commit-Position: refs/heads/master@{#295990}
TBR=tommi@chromium.org
NOTREECHECKS=true
NOTRY=true
BUG=415935
Review URL: https://codereview.chromium.org/594883002
Cr-Commit-Position: refs/heads/master@{#296191}
Diffstat (limited to 'content')
| -rw-r--r-- | content/renderer/media/media_stream_audio_processor.cc | 3 |
1 files changed, 1 insertions, 2 deletions
diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc index ac41187..4efc507 100644 --- a/content/renderer/media/media_stream_audio_processor.cc +++ b/content/renderer/media/media_stream_audio_processor.cc @@ -19,7 +19,6 @@ #include "media/base/audio_fifo.h" #include "media/base/channel_layout.h" #include "third_party/WebKit/public/platform/WebMediaConstraints.h" -#include "third_party/libjingle/overrides/init_webrtc.h" #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" #include "third_party/webrtc/modules/audio_processing/typing_detection.h" @@ -424,7 +423,7 @@ void MediaStreamAudioProcessor::InitializeAudioProcessingModule( #endif // Create and configure the webrtc::AudioProcessing. - audio_processing_.reset(CreateWebRtcAudioProcessing(config)); + audio_processing_.reset(webrtc::AudioProcessing::Create(config)); // Enable the audio processing components. if (echo_cancellation) { |