/external/chromium_org/ppapi/cpp/ |
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,
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/external/chromium_org/media/audio/ |
audio_parameters.cc | 23 int sample_rate, int bits_per_sample, 27 sample_rate_(sample_rate), 36 int sample_rate, int bits_per_sample, 40 sample_rate_(sample_rate), 49 int sample_rate, int bits_per_sample, 58 sample_rate_ = sample_rate;
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sample_rates.cc | 9 AudioSampleRate AsAudioSampleRate(int sample_rate) { 10 switch (sample_rate) {
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audio_parameters.h | 47 int sample_rate, int bits_per_sample, 51 int sample_rate, int bits_per_sample, 55 int sample_rate, int bits_per_sample, 73 int sample_rate() const { return sample_rate_; } function in class:media::AudioParameters 85 sample_rate_ == other.sample_rate() && 115 if (a.sample_rate() != b.sample_rate()) 116 return a.sample_rate() < b.sample_rate();
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audio_util.h | 18 // Computes a buffer size based on the given |sample_rate|. Must be used in 20 MEDIA_EXPORT size_t GetHighLatencyOutputBufferSize(int sample_rate);
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/external/chromium_org/ppapi/thunk/ |
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, 30 return PPB_AudioConfig_Shared::RecommendSampleFrameCount_1_0(sample_rate, 35 PP_AudioSampleRate sample_rate, 42 sample_rate, requested_sample_frame_count);
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/external/chromium_org/ppapi/tests/ |
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/remoting/host/ |
audio_capturer.cc | 13 bool AudioCapturer::IsValidSampleRate(int sample_rate) { 14 switch (sample_rate) {
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/external/chromium_org/content/public/common/ |
media_stream_request.cc | 32 sample_rate(0), 40 int sample_rate, 45 sample_rate(sample_rate),
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/external/chromium_org/ppapi/shared_impl/ |
ppb_audio_config_shared.h | 29 PP_AudioSampleRate sample_rate, 31 static uint32_t RecommendSampleFrameCount_1_0(PP_AudioSampleRate sample_rate, 34 PP_AudioSampleRate sample_rate, 51 bool Init(PP_AudioSampleRate sample_rate, uint32_t sample_frame_count);
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ppb_audio_config_shared.cc | 32 PP_AudioSampleRate sample_rate, 36 if (!object->Init(sample_rate, sample_frame_count)) 43 PP_AudioSampleRate sample_rate, 57 PP_AudioSampleRate sample_rate, 86 if (hardware_sample_rate == sample_rate) { 116 if (hardware_sample_rate == 44100 && sample_rate == 48000 && 149 bool PPB_AudioConfig_Shared::Init(PP_AudioSampleRate sample_rate, 155 if (sample_rate != PP_AUDIOSAMPLERATE_44100 && 156 sample_rate != PP_AUDIOSAMPLERATE_48000) 165 sample_rate_ = sample_rate; [all...] |
/prebuilts/python/darwin-x86/2.7.5/lib/python2.7/ |
sunaudio.py | 26 sample_rate = get_long_be(fp.read(4)) 35 return (data_size, encoding, sample_rate, channels, info) 41 data_size, encoding, sample_rate, channels, info = hdr 47 print 'Sample rate:', sample_rate
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/prebuilts/python/linux-x86/2.7.5/lib/python2.7/ |
sunaudio.py | 26 sample_rate = get_long_be(fp.read(4)) 35 return (data_size, encoding, sample_rate, channels, info) 41 data_size, encoding, sample_rate, channels, info = hdr 47 print 'Sample rate:', sample_rate
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/external/chromium/webkit/glue/media/ |
audio_decoder.cc | 24 const char* data, size_t data_size, double sample_rate) { 37 double file_sample_rate = reader.sample_rate(); 42 // For now, we're ignoring the requested 'sample_rate' and returning 45 // (sample_rate != 0.0) ? sample_rate : file_sample_rate;
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audio_decoder.h | 16 size_t data_size, double sample_rate);
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/external/chromium_org/content/renderer/media/ |
webrtc_audio_capturer_sink_owner.cc | 16 int sample_rate, 24 return delegate_->CaptureData(channels, audio_data, sample_rate,
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audio_decoder.h | 16 size_t data_size, double sample_rate);
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webaudio_capturer_source.cc | 32 size_t number_of_channels, float sample_rate) { 33 DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate=" 34 << sample_rate << ")"; 39 capturer_->SetCapturerSource(this, channel_layout, sample_rate);
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webrtc_audio_capturer.cc | 37 int GetBufferSizeForSampleRate(int sample_rate) { 41 buffer_size = (sample_rate / 100); 46 buffer_size = 2 * sample_rate / 100; 49 buffer_size = 2 * sample_rate / 100; 63 bool Initialize(int sample_rate, 65 int buffer_size = GetBufferSizeForSampleRate(sample_rate); 75 sample_rate, bits_per_sample, buffer_size); 162 bool WebRtcAudioCapturer::Reconfigure(int sample_rate, 165 if (!new_buffer->Initialize(sample_rate, channel_layout)) 186 int sample_rate, 314 << "sample_rate=" << sample_rate << ")"; local [all...] |
/external/chromium_org/media/audio/linux/ |
alsa_util.h | 22 int sample_rate, 29 int sample_rate,
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/external/chromium_org/content/common/media/ |
media_stream_options.cc | 48 int sample_rate, 51 : device(service_param, device_param, name_param, sample_rate, 63 first.device.sample_rate == second.device.sample_rate &&
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/external/chromium_org/content/renderer/media/android/ |
audio_decoder_android.h | 19 double sample_rate,
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/external/chromium_org/media/base/android/ |
webaudio_media_codec_info.h | 15 unsigned long sample_rate; member in struct:media::WebAudioMediaCodecInfo
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/external/chromium_org/media/audio/mac/ |
audio_auhal_mac_unittest.cc | 52 sample_rate_ = preferred_params.sample_rate(); 72 AudioOutputStream* CreateWithSampleRate(int sample_rate) { 73 sample_rate_ = sample_rate; 88 int sample_rate() const { return sample_rate_; } function in class:media::AudioOutputStreamWrapper 125 int sample_rate = aosw.sample_rate(); local 126 EXPECT_GE(sample_rate, 16000); 127 EXPECT_LE(sample_rate, 192000); 195 SineWaveAudioSource source(1, 200.0, aosw.sample_rate()); 210 int non_default_sample_rate = aosw.sample_rate() == 44100 [all...] |
/external/webrtc/src/modules/audio_processing/ |
debug.proto | 6 optional int32 sample_rate = 1;
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