| /external/chromium_org/content/renderer/media/ |
| webrtc_local_audio_source_provider_unittest.cc | 22 source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
29 source_params_.frames_per_buffer() * source_params_.channels();
50 source_provider_->OnSetFormat(source_params_);
53 media::AudioParameters source_params_; member in class:content::WebRtcLocalAudioSourceProviderTest
69 // source_params_.frames_per_buffer() of zero into the resampler since there
76 source_params_.frames_per_buffer() * source_params_.channels();
81 source_params_.sample_rate(),
82 source_params_.channels()
[all...] |
| webrtc_local_audio_renderer.cc | 96 // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match
100 if (source_params_ == params)
103 source_params_ = params;
105 sink_params_ = media::AudioParameters(source_params_.format(),
106 source_params_.channel_layout(), source_params_.channels(),
107 source_params_.input_channels(), source_params_.sample_rate(),
108 source_params_.bits_per_sample(),
116 2 * source_params_.frames_per_buffer()
[all...] |
| webrtc_local_audio_renderer.h | 96 // we have received valid |source_params_| &&
101 // Sets new |source_params_| and then re-initializes and restarts |sink_|.
138 media::AudioParameters source_params_; member in class:content::WebRtcLocalAudioRenderer
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| webrtc_local_audio_source_provider.h | 93 media::AudioParameters source_params_; member in class:content::WebRtcLocalAudioSourceProvider
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| media_stream_audio_processor.cc | 59 : source_params_(source_params),
73 int max_frame_size = std::max(source_params_.frames_per_buffer(),
77 kMaxNumberOfBuffersInFifo * source_params_.sample_rate() / 100);
78 fifo_.reset(new media::AudioFifo(source_params_.channels(), buffer_size));
104 // fifo_->frames() / source_params_.sample_rate() is less than
107 sink_params_.frames_per_buffer() * source_params_.sample_rate()) {
139 return source_params_;
162 const media::AudioParameters source_params_; member in class:content::MediaStreamAudioProcessor::MediaStreamAudioConverter
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| webrtc_local_audio_source_provider.cc | 67 source_params_ = params;
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