| /device/samsung/manta/ |
| media_profiles.xml | 37 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
91 sampleRate="44100"
104 sampleRate="44100"
118 sampleRate="44100"
133 sampleRate="44100"
147 sampleRate="44100"
161 sampleRate="44100"
183 sampleRate="44100"
196 sampleRate="44100"
211 sampleRate="44100
[all...] |
| /hardware/libhardware_legacy/audio/ |
| A2dpAudioInterface.cpp | 64 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
68 return mHardwareInterface->openOutputStream(devices, format, channels, sampleRate, status);
82 if ((err = out->set(devices, format, channels, sampleRate)) == NO_ERROR) {
107 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
110 return mHardwareInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
202 size_t A2dpAudioInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
204 return mHardwareInterface->getInputBufferSize(sampleRate, format, channelCount);
247 if (lRate == 0) lRate = sampleRate();
252 (lRate != sampleRate())){
255 if (pRate) *pRate = sampleRate();
[all...] |
| AudioHardwareGeneric.cpp | 68 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
82 status_t lStatus = out->set(this, mFd, devices, format, channels, sampleRate);
102 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
122 status_t lStatus = in->set(this, mFd, devices, format, channels, sampleRate, acoustics);
207 if (lRate == 0) lRate = sampleRate();
212 (lRate != sampleRate())) {
215 if (pRate) *pRate = sampleRate();
252 snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
325 (*pRate != sampleRate())) {
329 *pRate = sampleRate();
[all...] |
| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| AsyncAudioDecoder.h | 47 void decodeAsync(ArrayBuffer* audioData, float sampleRate, PassOwnPtr<AudioBufferCallback> successCallback, PassOwnPtr<AudioBufferCallback> errorCallback);
50 static void decode(ArrayBuffer* audioData, float sampleRate, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback);
|
| AudioSourceNode.h | 38 AudioSourceNode(AudioContext* context, float sampleRate)
39 : AudioNode(context, sampleRate)
|
| ChannelMergerNode.h | 41 static PassRefPtr<ChannelMergerNode> create(AudioContext*, float sampleRate, unsigned numberOfInputs);
56 ChannelMergerNode(AudioContext*, float sampleRate, unsigned numberOfInputs);
|
| ChannelSplitterNode.h | 37 static PassRefPtr<ChannelSplitterNode> create(AudioContext*, float sampleRate, unsigned numberOfOutputs);
47 ChannelSplitterNode(AudioContext*, float sampleRate, unsigned numberOfOutputs);
|
| DelayProcessor.cpp | 35 DelayProcessor::DelayProcessor(AudioContext* context, float sampleRate, unsigned numberOfChannels, double maxDelayTime)
36 : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
|
| AudioScheduledSourceNode.cpp | 45 AudioScheduledSourceNode::AudioScheduledSourceNode(AudioContext* context, float sampleRate)
46 : AudioSourceNode(context, sampleRate)
68 double sampleRate = this->sampleRate();
76 size_t startFrame = AudioUtilities::timeToSampleFrame(m_startTime, sampleRate);
77 size_t endFrame = m_endTime == UnknownTime ? 0 : AudioUtilities::timeToSampleFrame(m_endTime, sampleRate);
|
| AudioParam.cpp | 158 double sampleRate = context()->sampleRate();
160 double endTime = startTime + numberOfValues / sampleRate;
164 m_value = m_timeline.valuesForTimeRange(startTime, endTime, narrowPrecisionToFloat(m_value), values, numberOfValues, sampleRate, sampleRate);
|
| ConvolverNode.cpp | 48 ConvolverNode::ConvolverNode(AudioContext* context, float sampleRate)
49 : AudioNode(context, sampleRate)
139 bufferBus->setSampleRate(buffer->sampleRate());
163 return m_reverb ? m_reverb->impulseResponseLength() / static_cast<double>(sampleRate()) : 0;
173 return m_reverb ? m_reverb->latencyFrames() / static_cast<double>(sampleRate()) : 0;
|
| MediaElementAudioSourceNode.cpp | 49 : AudioSourceNode(context, context->sampleRate())
86 if (sourceSampleRate != sampleRate()) {
87 double scaleFactor = sourceSampleRate / sampleRate();
120 ASSERT(m_sourceSampleRate != sampleRate());
124 ASSERT(m_sourceSampleRate == sampleRate());
|
| /external/chromium_org/third_party/WebKit/Source/testing/runner/ |
| MockWebAudioDevice.h | 42 explicit MockWebAudioDevice(double sampleRate);
47 virtual double sampleRate();
|
| /external/srec/srec/cfront/ |
| ca_wave.c | 72 ASSERT(hFrontend->config->waveobj->samplerate);
80 hWave->data.samplerate = hFrontend->config->waveobj->samplerate;
84 hFrontend->config->waveobj->samplerate / FRAMERATE,
132 hWave->data.samplerate = 0;
161 int samplerate,
179 hWave->data.samplerate = samplerate;
|
| /frameworks/av/cmds/stagefright/ |
| SineSource.cpp | 12 SineSource::SineSource(int32_t sampleRate, int32_t numChannels)
14 mSampleRate(sampleRate),
|
| /frameworks/av/media/libnbaio/ |
| AudioStreamInSource.cpp | 47 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common);
50 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
|
| /frameworks/base/core/java/android/speech/srec/ |
| WaveHeader.java | 29 * <li> sampleRate - usually 8000, 11025, 16000, 22050, or 44100 hz.
69 * @param sampleRate typically 8000, 11025, 16000, 22050, or 44100 hz.
73 public WaveHeader(short format, short numChannels, int sampleRate, short bitsPerSample, int numBytes) {
75 mSampleRate = sampleRate;
129 * @param sampleRate sample rate, typically 8000, 11025, 16000, 22050, or 44100 hz.
132 public WaveHeader setSampleRate(int sampleRate) {
133 mSampleRate = sampleRate;
272 "WaveHeader format=%d numChannels=%d sampleRate=%d bitsPerSample=%d numBytes=%d",
|
| /frameworks/opt/net/voip/src/jni/rtp/ |
| GsmCodec.cpp | 42 int set(int sampleRate, const char *fmtp) {
43 return (sampleRate == 8000 && mEncode && mDecode) ? 160 : -1;
|
| /packages/apps/VoiceDialer/src/com/android/voicedialer/ |
| RecognizerEngine.java | 110 * @param sampleRate the same rate coming from the mic or micFile
113 File micFile, int sampleRate) {
124 if (mSampleRate != sampleRate) {
131 mSampleRate = sampleRate;
138 if (sampleRate == 8000) {
152 mic = new MicrophoneInputStream(sampleRate, sampleRate * 15);
159 if (mLogger != null) mic = mLogger.logInputStream(mic, sampleRate);
|
| /external/chromium_org/third_party/WebKit/Source/platform/audio/ |
| HRTFElevation.cpp | 99 bool HRTFElevation::calculateSymmetricKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName,
104 bool success = calculateKernelsForAzimuthElevation(azimuth, elevation, sampleRate, subjectName, kernelL1, kernelR1);
113 success = calculateKernelsForAzimuthElevation(symmetricAzimuth, elevation, sampleRate, subjectName, kernelL2, kernelR2);
124 bool HRTFElevation::calculateKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName,
172 RefPtr<AudioBus> response(AudioBus::createBySampleRateConverting(preSampleRateConvertedResponse.get(), false, sampleRate));
178 RefPtr<AudioBus> impulseResponse(AudioBus::loadPlatformResource(resourceName.utf8().data(), sampleRate));
185 size_t expectedLength = static_cast<size_t>(256 * (sampleRate / 44100.0));
198 const size_t fftSize = HRTFPanner::fftSizeForSampleRate(sampleRate);
199 kernelL = HRTFKernel::create(leftEarImpulseResponse, fftSize, sampleRate);
200 kernelR = HRTFKernel::create(rightEarImpulseResponse, fftSize, sampleRate);
[all...] |
| HRTFKernel.cpp | 70 HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate)
72 , m_sampleRate(sampleRate)
86 unsigned numberOfFadeOutFrames = static_cast<unsigned>(sampleRate / 4410); // 10 sample-frames @44.1KHz sample-rate
121 float sampleRate1 = kernel1->sampleRate();
122 float sampleRate2 = kernel2->sampleRate();
|
| /frameworks/av/media/libstagefright/rtsp/ |
| APacketSource.cpp | 473 int32_t sampleRate, numChannels;
475 desc.c_str(), &sampleRate, &numChannels);
477 mFormat->setInt32(kKeySampleRate, sampleRate);
489 int32_t sampleRate, numChannels;
491 desc.c_str(), &sampleRate, &numChannels);
493 mFormat->setInt32(kKeySampleRate, sampleRate);
496 if (sampleRate != 8000 || numChannels != 1) {
502 int32_t sampleRate, numChannels;
504 desc.c_str(), &sampleRate, &numChannels);
506 mFormat->setInt32(kKeySampleRate, sampleRate);
[all...] |
| /device/asus/grouper/ |
| media_profiles.xml | 39 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
101 sampleRate="8000"
114 sampleRate="8000"
127 sampleRate="44100"
144 sampleRate="8000"
161 sampleRate="8000"
178 sampleRate="44100"
|
| /external/chromium_org/media/base/android/java/src/org/chromium/media/ |
| WebAudioMediaCodecBridge.java | 70 int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
150 Log.d(LOG_TAG, "Final: Rate: " + sampleRate +
157 sampleRate,
178 sampleRate = newFormat.getInteger(MediaFormat.KEY_SAMPLE_RATE);
199 int sampleRate,
|
| /external/chromium_org/third_party/WebKit/Source/platform/audio/chromium/ |
| AudioDestinationChromium.cpp | 48 PassOwnPtr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
50 return adoptPtr(new AudioDestinationChromium(callback, inputDeviceId, numberOfInputChannels, numberOfOutputChannels, sampleRate));
53 AudioDestinationChromium::AudioDestinationChromium(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
58 , m_sampleRate(sampleRate)
85 m_audioDevice = adoptPtr(blink::Platform::current()->createAudioDevice(m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, inputDeviceId));
|
