| /external/chromium_org/media/audio/ |
| sample_rates.h | 33 MEDIA_EXPORT bool ToAudioSampleRate(int sample_rate, AudioSampleRate* asr);
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| audio_power_monitor.h | 38 // |sample_rate| is the audio signal sample rate (Hz). |time_constant|
42 AudioPowerMonitor(int sample_rate, const base::TimeDelta& time_constant);
68 // |sample_rate| and |time_constant|.
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| /external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
| unittest.proto | 9 optional int32 sample_rate = 4;
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| /external/webrtc/src/modules/audio_processing/test/ |
| unittest.proto | 9 optional int32 sample_rate = 4;
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| /external/chromium_org/content/shell/renderer/test_runner/ |
| web_test_interfaces.cc | 77 WebAudioDevice* WebTestInterfaces::CreateAudioDevice(double sample_rate) {
78 return new MockWebAudioDevice(sample_rate);
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| /external/chromium_org/media/base/ |
| audio_converter.cc | 48 if (input_params.sample_rate() != output_params.sample_rate()) {
49 DVLOG(1) << "Resampling from " << input_params.sample_rate() << " to "
50 << output_params.sample_rate();
54 input_params.sample_rate() /
55 static_cast<double>(output_params.sample_rate());
65 static_cast<double>(input_params.sample_rate()));
68 static_cast<double>(output_params.sample_rate()));
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| audio_buffer_converter_unittest.cc | 21 static scoped_refptr<AudioBuffer> MakeTestBuffer(int sample_rate,
28 sample_rate,
59 (static_cast<double>(output_params_.sample_rate()) /
60 in->sample_rate());
70 EXPECT_EQ(out->sample_rate(), output_params_.sample_rate());
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| audio_buffer.h | 40 int sample_rate,
50 int sample_rate,
57 int sample_rate,
97 int sample_rate() const { return sample_rate_; } function in class:media::AudioBuffer
124 int sample_rate,
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| audio_hardware_config.cc | 48 return output_params_.sample_rate();
63 return input_params_.sample_rate();
104 const double twenty_ms_size = 2.0 * output_params_.sample_rate() / 100;
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| /external/chromium_org/media/formats/mpeg/ |
| mpeg1_audio_stream_parser.h | 50 int sample_rate; member in struct:media::MPEG1AudioStreamParser::Header
78 int* sample_rate,
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| /hardware/libhardware/modules/usbaudio/ |
| alsa_device_profile.h | 87 unsigned profile_calc_min_period_size(alsa_device_profile* profile, unsigned sample_rate);
88 unsigned int profile_get_period_size(alsa_device_profile* profile, unsigned sample_rate);
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| /external/chromium_org/media/audio/alsa/ |
| alsa_util.cc | 18 int sample_rate,
31 sample_rate, 1, latency_us);
98 int sample_rate,
102 sample_rate, pcm_format, latency_us);
108 int sample_rate,
112 sample_rate, pcm_format, latency_us);
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| /external/chromium_org/content/renderer/media/ |
| webrtc_audio_capturer.cc | 150 ", channel_layout=%d, sample_rate=%d, buffer_size=%d"
155 device_info_.device.input.sample_rate,
158 device_info_.device.matched_output.sample_rate,
207 << device_info_.device.input.sample_rate;
209 if (media::ToAudioSampleRate(device_info_.device.input.sample_rate, &asr)) {
214 device_info_.device.input.sample_rate);
220 static_cast<float>(device_info_.device.input.sample_rate));
250 device_info_.device.input.sample_rate,
309 float sample_rate) {
312 << "sample_rate=" << sample_rate << ")" local
[all...] |
| webrtc_audio_device_impl.cc | 57 int sample_rate,
83 const int frames_per_10_ms = (sample_rate / 100);
103 sample_rate,
128 int sample_rate,
140 int frames_per_10_ms = (sample_rate / 100);
163 sample_rate,
176 sample_rate,
199 (*it)->OnPlayoutData(audio_bus, sample_rate, audio_delay_milliseconds);
434 uint32_t* sample_rate) const {
440 *sample_rate = static_cast<uint32_t>
[all...] |
| webrtc_audio_capturer_unittest.cc | 44 virtual int OnData(const int16* audio_data, int sample_rate,
49 EXPECT_EQ(sample_rate, params_.sample_rate());
96 "", "", params_.sample_rate(),
184 "", "", params_.sample_rate(),
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| media_stream_audio_processor.cc | 323 int sample_rate,
334 InitializeRenderFifoIfNeeded(sample_rate, audio_bus->channels(),
343 sample_rate,
467 kAudioProcessingSampleRate : input_format.sample_rate();
482 int processing_frames = input_format.sample_rate() / 100;
510 int sample_rate, int number_of_channels, int frames_per_buffer) {
513 render_format_.sample_rate() == sample_rate &&
523 sample_rate,
527 const int analysis_frames = sample_rate / 100; // 10 ms chunks
[all...] |
| /external/chromium_org/ppapi/api/ |
| ppb_audio_config.idl | 56 * <code>sample_rate</code> should be the result of calling
80 * @param[in] sample_rate A <code>PP_AudioSampleRate</code> which is either
92 [in] PP_AudioSampleRate sample_rate,
112 * @param[in] sample_rate A <code>PP_AudioSampleRate</code> which is either
123 [in] PP_AudioSampleRate sample_rate,
143 * @param[in] sample_rate A <code>PP_AudioSampleRate</code> which is either
155 [in] PP_AudioSampleRate sample_rate,
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| /external/chromium_org/ppapi/cpp/ |
| 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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| /frameworks/ex/variablespeed/jni/ |
| sola_time_scaler.h | 44 // @param sample_rate sample rate of the audio signal.
46 void Init(int sample_rate, int num_channels) {
47 sample_rate_ = sample_rate;
84 // @param sample_rate sample rate of the signal to process
89 void Init(double sample_rate, int num_channels, double initial_speed,
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| /external/chromium_org/components/domain_reliability/ |
| config.cc | 53 double sample_rate = success ? success_sample_rate : failure_sample_rate; local
54 DCHECK(IsValidSampleRate(sample_rate));
55 return base::RandDouble() < sample_rate;
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| /external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
| fine_audio_buffer_unittest.cc | 60 void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
61 const int kSamplesPer10Ms = sample_rate * 10 / 1000;
81 sample_rate);
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| /external/chromium_org/content/public/common/ |
| media_stream_request.h | 106 int sample_rate,
134 : sample_rate(), channel_layout(), frames_per_buffer(), effects() {
137 AudioDeviceParameters(int sample_rate, int channel_layout,
139 : sample_rate(sample_rate),
146 int sample_rate; member in struct:content::MediaStreamDevice::AudioDeviceParameters
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| /external/chromium_org/content/renderer/media/webrtc/ |
| webrtc_local_audio_track_adapter_unittest.cc | 27 int sample_rate,
75 OnData(_, 16, params_.sample_rate(), params_.channels(),
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| /external/chromium_org/content/renderer/pepper/ |
| pepper_platform_audio_output.h | 30 static PepperPlatformAudioOutput* Create(int sample_rate,
66 bool Initialize(int sample_rate,
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| /external/chromium_org/media/audio/android/ |
| audio_manager_android.cc | 274 int sample_rate = GetNativeOutputSampleRate(); local
275 int buffer_size = GetOptimalOutputFrameSize(sample_rate, 2);
279 sample_rate = input_params.sample_rate();
283 sample_rate, ChannelLayoutToChannelCount(channel_layout));
292 sample_rate, bits_per_sample, buffer_size, AudioParameters::NO_EFFECTS);
365 int AudioManagerAndroid::GetOptimalOutputFrameSize(int sample_rate,
373 sample_rate, channels));
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