Searched defs:sample_rate (Results 1 - 25 of 129) sorted by relevance

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/external/chromium_org/media/audio/
H A Dsample_rates.cc11 bool ToAudioSampleRate(int sample_rate, AudioSampleRate* asr) { argument
13 switch (sample_rate) {
H A Daudio_power_monitor.cc19 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
H A Daudio_parameters.cc23 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
/external/chromium_org/remoting/host/
H A Daudio_capturer.cc13 bool AudioCapturer::IsValidSampleRate(int sample_rate) { argument
14 switch (sample_rate) {
/external/chromium_org/third_party/webrtc/modules/audio_device/android/
H A Dopensles_common.cc21 SLDataFormat_PCM CreatePcmConfiguration(int sample_rate) { argument
29 configuration.samplesPerSec = sample_rate * 1000;
H A Dfine_audio_buffer.cc23 int sample_rate)
26 sample_rate_(sample_rate),
21 FineAudioBuffer(AudioDeviceBuffer* device_buffer, int desired_frame_size_bytes, int sample_rate) argument
/external/chromium_org/media/base/android/
H A Dwebaudio_media_codec_info.h15 unsigned long sample_rate; member in struct:media::WebAudioMediaCodecInfo
/external/chromium_org/content/shell/renderer/test_runner/
H A Dmock_web_audio_device.cc9 MockWebAudioDevice::MockWebAudioDevice(double sample_rate) argument
10 : sample_rate_(sample_rate) {}
/external/chromium_org/ppapi/cpp/
H A Daudio_config.cc31 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
H A Daudio_config.h69 /// @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
/external/flac/libFLAC/include/protected/
H A Dstream_decoder.h45 unsigned sample_rate; /* in Hz */ member in struct:FLAC__StreamDecoderProtected
/external/chromium_org/content/renderer/media/webrtc/
H A Dwebrtc_audio_sink_adapter.cc26 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
/external/chromium_org/media/filters/
H A Daudio_file_reader.h32 // Open() reads the audio data format so that the sample_rate(),
49 int sample_rate() const { return sample_rate_; } function in class:media::AudioFileReader
/external/chromium_org/ppapi/tests/
H A Dtest_audio_config.cc29 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));
/external/chromium_org/third_party/webrtc/common_audio/resampler/
H A Dsinusoidal_linear_chirp_source.cc20 SinusoidalLinearChirpSource::SinusoidalLinearChirpSource(int sample_rate, argument
22 : sample_rate_(sample_rate),
/external/chromium_org/third_party/webrtc/common_audio/
H A Dwav_writer.h27 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,
/external/chromium_org/content/common/media/
H A Dmedia_param_traits.cc23 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);
/external/chromium_org/content/renderer/media/
H A Daudio_renderer_mixer_manager.cc65 int sample_rate = params.sample_rate(); local
67 int sample_rate = hardware_config_->GetOutputSampleRate(); local
75 sample_rate, 16, hardware_config_->GetHighLatencyBufferSize());
H A Dmedia_stream_audio_sink_owner.cc17 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
H A Dpeer_connection_audio_sink_owner.cc17 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
H A Dwebaudio_capturer_source.cc33 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
H A Dwebrtc_local_audio_source_provider.cc36 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
/external/chromium_org/ppapi/thunk/
H A Dppb_audio_config_thunk.cc17 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
/external/chromium_org/third_party/webrtc/examples/android/opensl_loopback/
H A Dfake_audio_device_buffer.cc32 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;
/external/chromium_org/content/browser/speech/endpointer/
H A Dendpointer.cc18 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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