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author | ajm@chromium.org <ajm@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2014-08-15 17:16:21 +0000 |
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committer | ajm@chromium.org <ajm@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2014-08-15 17:16:21 +0000 |
commit | 419761e17cd2ae7e9bb1d3301f3e6e28c189136c (patch) | |
tree | 42b386e7ef6806217033b1368115e12845970f02 /content | |
parent | f16411e8a20fed3f17f7406b9a91462dd846585f (diff) | |
download | chromium_src-419761e17cd2ae7e9bb1d3301f3e6e28c189136c.zip chromium_src-419761e17cd2ae7e9bb1d3301f3e6e28c189136c.tar.gz chromium_src-419761e17cd2ae7e9bb1d3301f3e6e28c189136c.tar.bz2 |
Provide experimental AudioProcessing options at creation.
Remove now unneeded MediaStreamAudioProcessorOptions setters. Removes
all calls to AudioProcessing::SetExtraOptions() from Chromium.
TESTED=verified that options are set correctly in AudioProcessing.
BUG=webrtc:3702
Review URL: https://codereview.chromium.org/470403002
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@289923 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'content')
3 files changed, 8 insertions, 25 deletions
diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc index 16f43db..76ce92c 100644 --- a/content/renderer/media/media_stream_audio_processor.cc +++ b/content/renderer/media/media_stream_audio_processor.cc @@ -408,16 +408,20 @@ void MediaStreamAudioProcessor::InitializeAudioProcessingModule( return; } + // Experimental options provided at creation. + webrtc::Config config; + if (goog_experimental_aec) + config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true)); + if (goog_experimental_ns) + config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(true)); + // Create and configure the webrtc::AudioProcessing. - audio_processing_.reset(webrtc::AudioProcessing::Create()); + audio_processing_.reset(webrtc::AudioProcessing::Create(config)); // Enable the audio processing components. if (echo_cancellation) { EnableEchoCancellation(audio_processing_.get()); - if (goog_experimental_aec) - EnableExperimentalEchoCancellation(audio_processing_.get()); - if (playout_data_source_) playout_data_source_->AddPlayoutSink(this); } @@ -425,9 +429,6 @@ void MediaStreamAudioProcessor::InitializeAudioProcessingModule( if (goog_ns) EnableNoiseSuppression(audio_processing_.get()); - if (goog_experimental_ns) - EnableExperimentalNoiseSuppression(audio_processing_.get()); - if (goog_high_pass_filter) EnableHighPassFilter(audio_processing_.get()); diff --git a/content/renderer/media/media_stream_audio_processor_options.cc b/content/renderer/media/media_stream_audio_processor_options.cc index d329992..9d2f5ee 100644 --- a/content/renderer/media/media_stream_audio_processor_options.cc +++ b/content/renderer/media/media_stream_audio_processor_options.cc @@ -233,12 +233,6 @@ void EnableNoiseSuppression(AudioProcessing* audio_processing) { CHECK_EQ(err, 0); } -void EnableExperimentalNoiseSuppression(AudioProcessing* audio_processing) { - webrtc::Config config; - config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(true)); - audio_processing->SetExtraOptions(config); -} - void EnableHighPassFilter(AudioProcessing* audio_processing) { CHECK_EQ(audio_processing->high_pass_filter()->Enable(true), 0); } @@ -254,12 +248,6 @@ void EnableTypingDetection(AudioProcessing* audio_processing, typing_detector->SetParameters(0, 0, 0, 0, 0, 100); } -void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing) { - webrtc::Config config; - config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true)); - audio_processing->SetExtraOptions(config); -} - void StartEchoCancellationDump(AudioProcessing* audio_processing, base::File aec_dump_file) { DCHECK(aec_dump_file.IsValid()); diff --git a/content/renderer/media/media_stream_audio_processor_options.h b/content/renderer/media/media_stream_audio_processor_options.h index 4683555..6f6526f 100644 --- a/content/renderer/media/media_stream_audio_processor_options.h +++ b/content/renderer/media/media_stream_audio_processor_options.h @@ -94,9 +94,6 @@ void EnableEchoCancellation(AudioProcessing* audio_processing); // Enables the noise suppression in |audio_processing|. void EnableNoiseSuppression(AudioProcessing* audio_processing); -// Enables the experimental noise suppression in |audio_processing|. -void EnableExperimentalNoiseSuppression(AudioProcessing* audio_processing); - // Enables the high pass filter in |audio_processing|. void EnableHighPassFilter(AudioProcessing* audio_processing); @@ -104,9 +101,6 @@ void EnableHighPassFilter(AudioProcessing* audio_processing); void EnableTypingDetection(AudioProcessing* audio_processing, webrtc::TypingDetection* typing_detector); -// Enables the experimental echo cancellation in |audio_processing|. -void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing); - // Starts the echo cancellation dump in |audio_processing|. void StartEchoCancellationDump(AudioProcessing* audio_processing, base::File aec_dump_file); |