| Index: content/renderer/media/webrtc_audio_capturer.cc
|
| diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
|
| index 3299987d67b8eacb9a7eb84427a58d33ee147069..bad0e49176ffef6300c5cf58e8a7241991649a13 100644
|
| --- a/content/renderer/media/webrtc_audio_capturer.cc
|
| +++ b/content/renderer/media/webrtc_audio_capturer.cc
|
| @@ -47,20 +47,10 @@ class WebRtcAudioCapturer::TrackOwner
|
| explicit TrackOwner(WebRtcLocalAudioTrack* track)
|
| : delegate_(track) {}
|
|
|
| - void Capture(const int16* audio_data,
|
| - base::TimeDelta delay,
|
| - double volume,
|
| - bool key_pressed,
|
| - bool need_audio_processing,
|
| - bool force_report_nonzero_energy) {
|
| + void Capture(const int16* audio_data, bool force_report_nonzero_energy) {
|
| base::AutoLock lock(lock_);
|
| if (delegate_) {
|
| - delegate_->Capture(audio_data,
|
| - delay,
|
| - volume,
|
| - key_pressed,
|
| - need_audio_processing,
|
| - force_report_nonzero_energy);
|
| + delegate_->Capture(audio_data, force_report_nonzero_energy);
|
| }
|
| }
|
|
|
| @@ -170,7 +160,6 @@ bool WebRtcAudioCapturer::Initialize() {
|
| // layout that includes the keyboard mic.
|
| if ((device_info_.device.input.effects &
|
| media::AudioParameters::KEYBOARD_MIC) &&
|
| - MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() &&
|
| audio_constraints.GetProperty(
|
| MediaAudioConstraints::kGoogExperimentalNoiseSuppression)) {
|
| if (channel_layout == media::CHANNEL_LAYOUT_STEREO) {
|
| @@ -236,8 +225,6 @@ WebRtcAudioCapturer::WebRtcAudioCapturer(
|
| device_info_(device_info),
|
| volume_(0),
|
| peer_connection_mode_(false),
|
| - key_pressed_(false),
|
| - need_audio_processing_(false),
|
| audio_device_(audio_device),
|
| audio_source_(audio_source) {
|
| DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()";
|
| @@ -334,9 +321,6 @@ void WebRtcAudioCapturer::SetCapturerSource(
|
| // Notify the |audio_processor_| of the new format.
|
| audio_processor_->OnCaptureFormatChanged(params);
|
|
|
| - MediaAudioConstraints audio_constraints(constraints_,
|
| - device_info_.device.input.effects);
|
| - need_audio_processing_ = audio_constraints.NeedsAudioProcessing();
|
| // Notify all tracks about the new format.
|
| tracks_.TagAll();
|
| }
|
| @@ -468,7 +452,6 @@ void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source,
|
| TrackList::ItemList tracks_to_notify_format;
|
| int current_volume = 0;
|
| base::TimeDelta audio_delay;
|
| - bool need_audio_processing = true;
|
| {
|
| base::AutoLock auto_lock(lock_);
|
| if (!running_)
|
| @@ -480,16 +463,8 @@ void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source,
|
| volume_ = static_cast<int>((volume * MaxVolume()) + 0.5);
|
| current_volume = volume_ > MaxVolume() ? MaxVolume() : volume_;
|
| audio_delay = base::TimeDelta::FromMilliseconds(audio_delay_milliseconds);
|
| - audio_delay_ = audio_delay;
|
| - key_pressed_ = key_pressed;
|
| tracks = tracks_.Items();
|
| tracks_.RetrieveAndClearTags(&tracks_to_notify_format);
|
| -
|
| - // Set the flag to turn on the audio processing in PeerConnection level.
|
| - // Note that, we turn off the audio processing in PeerConnection if the
|
| - // processor has already processed the data.
|
| - need_audio_processing = need_audio_processing_ ?
|
| - !MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() : false;
|
| }
|
|
|
| DCHECK(audio_processor_->InputFormat().IsValid());
|
| @@ -525,8 +500,7 @@ void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source,
|
| // Feed the post-processed data to the tracks.
|
| for (TrackList::ItemList::const_iterator it = tracks.begin();
|
| it != tracks.end(); ++it) {
|
| - (*it)->Capture(output, audio_delay, current_volume, key_pressed,
|
| - need_audio_processing, force_report_nonzero_energy);
|
| + (*it)->Capture(output, force_report_nonzero_energy);
|
| }
|
|
|
| if (new_volume) {
|
| @@ -589,14 +563,6 @@ int WebRtcAudioCapturer::GetBufferSize(int sample_rate) const {
|
| return (sample_rate / 100);
|
| }
|
|
|
| -void WebRtcAudioCapturer::GetAudioProcessingParams(
|
| - base::TimeDelta* delay, int* volume, bool* key_pressed) {
|
| - base::AutoLock auto_lock(lock_);
|
| - *delay = audio_delay_;
|
| - *volume = volume_;
|
| - *key_pressed = key_pressed_;
|
| -}
|
| -
|
| void WebRtcAudioCapturer::SetCapturerSourceForTesting(
|
| const scoped_refptr<media::AudioCapturerSource>& source,
|
| media::AudioParameters params) {
|
|
|