Index: content/renderer/media/media_stream_audio_processor_options.cc |
diff --git a/content/renderer/media/media_stream_audio_processor_options.cc b/content/renderer/media/media_stream_audio_processor_options.cc |
index d3299929056c0395a514ff5b128972af1b5b556e..9d2f5ee58c0c27d5a5ba77c7d51458332cafbf9d 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()); |