/external/chromium_org/media/audio/ |
H A D | sample_rates.cc | 11 bool ToAudioSampleRate(int sample_rate, AudioSampleRate* asr) { argument 13 switch (sample_rate) {
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H A D | audio_power_monitor.cc | 19 int sample_rate, const base::TimeDelta& time_constant) 21 1.0f - expf(-1.0f / (sample_rate * time_constant.InSecondsF()))) { 18 AudioPowerMonitor( int sample_rate, const base::TimeDelta& time_constant) argument
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H A D | audio_parameters.cc | 23 int sample_rate, int bits_per_sample, 27 sample_rate_(sample_rate), 35 int sample_rate, int bits_per_sample, 39 sample_rate_(sample_rate), 47 int channels, int sample_rate, 52 sample_rate_(sample_rate), 62 int channels, int sample_rate, 70 sample_rate_ = sample_rate; 110 sample_rate_ == other.sample_rate() && 22 AudioParameters(Format format, ChannelLayout channel_layout, int sample_rate, int bits_per_sample, int frames_per_buffer) argument 34 AudioParameters(Format format, ChannelLayout channel_layout, int sample_rate, int bits_per_sample, int frames_per_buffer, int effects) argument 46 AudioParameters(Format format, ChannelLayout channel_layout, int channels, int sample_rate, int bits_per_sample, int frames_per_buffer, int effects) argument 61 Reset(Format format, ChannelLayout channel_layout, int channels, int sample_rate, int bits_per_sample, int frames_per_buffer) argument
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/external/chromium_org/remoting/host/ |
H A D | audio_capturer.cc | 13 bool AudioCapturer::IsValidSampleRate(int sample_rate) { argument 14 switch (sample_rate) {
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/external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
H A D | opensles_common.cc | 21 SLDataFormat_PCM CreatePcmConfiguration(int sample_rate) { argument 29 configuration.samplesPerSec = sample_rate * 1000;
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H A D | fine_audio_buffer.cc | 23 int sample_rate) 26 sample_rate_(sample_rate), 21 FineAudioBuffer(AudioDeviceBuffer* device_buffer, int desired_frame_size_bytes, int sample_rate) argument
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/external/chromium_org/media/base/android/ |
H A D | webaudio_media_codec_info.h | 15 unsigned long sample_rate; member in struct:media::WebAudioMediaCodecInfo
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/external/chromium_org/content/shell/renderer/test_runner/ |
H A D | mock_web_audio_device.cc | 9 MockWebAudioDevice::MockWebAudioDevice(double sample_rate) argument 10 : sample_rate_(sample_rate) {}
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/external/chromium_org/ppapi/cpp/ |
H A D | audio_config.cc | 31 PP_AudioSampleRate sample_rate, 33 : sample_rate_(sample_rate), 38 instance.pp_instance(), sample_rate, sample_frame_count)); 42 instance.pp_instance(), sample_rate, sample_frame_count)); 59 PP_AudioSampleRate sample_rate, 64 sample_rate, 69 RecommendSampleFrameCount(sample_rate, 30 AudioConfig(const InstanceHandle& instance, PP_AudioSampleRate sample_rate, uint32_t sample_frame_count) argument 57 RecommendSampleFrameCount( const InstanceHandle& instance, PP_AudioSampleRate sample_rate, uint32_t requested_sample_frame_count) argument
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H A D | audio_config.h | 69 /// @param[in] sample_rate A <code>PP_AudioSampleRate</code> which is either 76 PP_AudioSampleRate sample_rate, 100 /// @param[in] sample_rate A <code>PP_AudioSampleRate</code> which is either 110 PP_AudioSampleRate sample_rate, 117 PP_AudioSampleRate sample_rate() const { return sample_rate_; } function in class:pp::AudioConfig
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/external/flac/libFLAC/include/protected/ |
H A D | stream_decoder.h | 45 unsigned sample_rate; /* in Hz */ member in struct:FLAC__StreamDecoderProtected
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/external/chromium_org/content/renderer/media/webrtc/ |
H A D | webrtc_audio_sink_adapter.cc | 26 int sample_rate, 29 sink_->OnData(audio_data, 16, sample_rate, number_of_channels, 25 OnData(const int16* audio_data, int sample_rate, int number_of_channels, int number_of_frames) argument
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/external/chromium_org/media/filters/ |
H A D | audio_file_reader.h | 32 // Open() reads the audio data format so that the sample_rate(), 49 int sample_rate() const { return sample_rate_; } function in class:media::AudioFileReader
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/external/chromium_org/ppapi/tests/ |
H A D | test_audio_config.cc | 29 PP_AudioSampleRate sample_rate = audio_config_interface_->RecommendSampleRate( local 31 ASSERT_TRUE(sample_rate == PP_AUDIOSAMPLERATE_NONE || 32 sample_rate == PP_AUDIOSAMPLERATE_44100 || 33 sample_rate == PP_AUDIOSAMPLERATE_48000); 53 PP_AudioSampleRate sample_rate = kSampleRates[i]; local 63 instance_->pp_instance(), sample_rate, request_frame_count); 68 instance_->pp_instance(), sample_rate, frame_count); 71 ASSERT_EQ(sample_rate, audio_config_interface_->GetSampleRate(ac));
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/external/chromium_org/third_party/webrtc/common_audio/resampler/ |
H A D | sinusoidal_linear_chirp_source.cc | 20 SinusoidalLinearChirpSource::SinusoidalLinearChirpSource(int sample_rate, argument 22 : sample_rate_(sample_rate),
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/external/chromium_org/third_party/webrtc/common_audio/ |
H A D | wav_writer.h | 27 WavFile(const std::string& filename, int sample_rate, int num_channels); 38 int sample_rate() const { return sample_rate_; } function in class:webrtc::WavFile 58 int sample_rate,
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/external/chromium_org/content/common/media/ |
H A D | media_param_traits.cc | 23 m->WriteInt(p.sample_rate()); 33 int format, channel_layout, sample_rate, bits_per_sample, local 38 !m->ReadInt(iter, &sample_rate) || 47 sample_rate, bits_per_sample, frames_per_buffer, effects);
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/external/chromium_org/content/renderer/media/ |
H A D | audio_renderer_mixer_manager.cc | 65 int sample_rate = params.sample_rate(); local 67 int sample_rate = hardware_config_->GetOutputSampleRate(); local 75 sample_rate, 16, hardware_config_->GetHighLatencyBufferSize());
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H A D | media_stream_audio_sink_owner.cc | 17 int sample_rate, 30 sample_rate, 16 OnData(const int16* audio_data, int sample_rate, int number_of_channels, int number_of_frames, const std::vector<int>& channels, int audio_delay_milliseconds, int current_volume, bool need_audio_processing, bool key_pressed) argument
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H A D | peer_connection_audio_sink_owner.cc | 17 int sample_rate, 28 sample_rate, 16 OnData(const int16* audio_data, int sample_rate, int number_of_channels, int number_of_frames, const std::vector<int>& channels, int audio_delay_milliseconds, int current_volume, bool need_audio_processing, bool key_pressed) argument
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H A D | webaudio_capturer_source.cc | 33 size_t number_of_channels, float sample_rate) { 35 DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate=" 36 << sample_rate << ")"; 51 channel_layout, number_of_channels, sample_rate, 16, 52 sample_rate / 100); 32 setFormat( size_t number_of_channels, float sample_rate) argument
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H A D | webrtc_local_audio_source_provider.cc | 36 int sample_rate = hardware_config->GetOutputSampleRate(); local 39 media::CHANNEL_LAYOUT_STEREO, 2, sample_rate, 16, 90 int sample_rate, 88 OnData( const int16* audio_data, int sample_rate, int number_of_channels, int number_of_frames) argument
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/external/chromium_org/ppapi/thunk/ |
H A D | ppb_audio_config_thunk.cc | 17 PP_AudioSampleRate sample_rate, 23 return enter.functions()->CreateAudioConfig(instance, sample_rate, 27 uint32_t RecommendSampleFrameCount_1_0(PP_AudioSampleRate sample_rate, argument 30 return PPB_AudioConfig_Shared::RecommendSampleFrameCount_1_0(sample_rate, 35 PP_AudioSampleRate sample_rate, 42 sample_rate, requested_sample_frame_count); 16 CreateStereo16bit(PP_Instance instance, PP_AudioSampleRate sample_rate, uint32_t sample_frame_count) argument 34 RecommendSampleFrameCount_1_1(PP_Instance instance, PP_AudioSampleRate sample_rate, uint32_t requested_sample_frame_count) argument
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/external/chromium_org/third_party/webrtc/examples/android/opensl_loopback/ |
H A D | fake_audio_device_buffer.cc | 32 assert(static_cast<int>(fsHz) == sample_rate()); 37 assert(static_cast<int>(fsHz) == sample_rate()); 86 int FakeAudioDeviceBuffer::sample_rate() const { function in class:webrtc::FakeAudioDeviceBuffer 92 return sample_rate() * 10 / 1000;
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/external/chromium_org/content/browser/speech/endpointer/ |
H A D | endpointer.cc | 18 Endpointer::Endpointer(int sample_rate) argument 22 sample_rate_(sample_rate), 26 frame_size_ = static_cast<int>(sample_rate / static_cast<float>(kFrameRate)); 52 ep_config.set_sample_rate(static_cast<float>(sample_rate));
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