summaryrefslogtreecommitdiffstats
path: root/content
diff options
context:
space:
mode:
authorajm@chromium.org <ajm@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2014-08-15 17:16:21 +0000
committerajm@chromium.org <ajm@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2014-08-15 17:16:21 +0000
commit419761e17cd2ae7e9bb1d3301f3e6e28c189136c (patch)
tree42b386e7ef6806217033b1368115e12845970f02 /content
parentf16411e8a20fed3f17f7406b9a91462dd846585f (diff)
downloadchromium_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')
-rw-r--r--content/renderer/media/media_stream_audio_processor.cc15
-rw-r--r--content/renderer/media/media_stream_audio_processor_options.cc12
-rw-r--r--content/renderer/media/media_stream_audio_processor_options.h6
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);