| /external/chromium_org/third_party/WebKit/Source/platform/audio/ |
| HRTFDatabase.cpp | 47 PassOwnPtr<HRTFDatabase> HRTFDatabase::create(float sampleRate)
49 OwnPtr<HRTFDatabase> hrtfDatabase = adoptPtr(new HRTFDatabase(sampleRate));
53 HRTFDatabase::HRTFDatabase(float sampleRate)
55 , m_sampleRate(sampleRate)
59 OwnPtr<HRTFElevation> hrtfElevation = HRTFElevation::createForSubject("Composite", elevation, sampleRate);
78 m_elevations[i + jj] = HRTFElevation::createByInterpolatingSlices(m_elevations[i].get(), m_elevations[j].get(), x, sampleRate);
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| AudioDelayDSPKernel.h | 35 AudioDelayDSPKernel(double maxDelayTime, float sampleRate);
52 virtual double delayTime(float sampleRate);
64 size_t bufferLengthForDelay(double delayTime, double sampleRate) const;
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| AudioUtilities.h | 39 // discreteTimeConstantForSampleRate() will return the discrete time-constant for the specific sampleRate.
40 PLATFORM_EXPORT double discreteTimeConstantForSampleRate(double timeConstant, double sampleRate);
43 PLATFORM_EXPORT size_t timeToSampleFrame(double time, double sampleRate);
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| /external/jmonkeyengine/engine/src/core/com/jme3/audio/ |
| AudioData.java | 47 protected int sampleRate;
92 return sampleRate;
99 * @param sampleRate Sample rate, 44100, 22050, etc.
101 public void setupFormat(int channels, int bitsPerSample, int sampleRate){
107 this.sampleRate = sampleRate;
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| /frameworks/av/media/libeffects/lvm/lib/Reverb/src/ |
| LVREV_ApplyNewSettings.c | 76 (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
82 Omega = LVM_GetOmega(pPrivate->NewParams.HPF, pPrivate->NewParams.SampleRate);
95 (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
105 if(pPrivate->NewParams.LPF <= (LVM_FsTable[pPrivate->NewParams.SampleRate] >> 1))
107 Omega = LVM_GetOmega(pPrivate->NewParams.LPF, pPrivate->NewParams.SampleRate);
142 (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
148 LVM_INT32 Fs = LVM_GetFsFromTable(pPrivate->NewParams.SampleRate);
[all...] |
| /frameworks/av/media/libeffects/testlibs/ |
| AudioPeakingFilter.cpp | 44 AudioPeakingFilter::AudioPeakingFilter(int nChannels, int sampleRate)
45 : mBiquad(nChannels, sampleRate) {
46 configure(nChannels, sampleRate);
50 void AudioPeakingFilter::configure(int nChannels, int sampleRate) {
51 mNiquistFreq = sampleRate * 500;
53 mBiquad.configure(nChannels, sampleRate);
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| AudioShelvingFilter.cpp | 50 int sampleRate)
52 mBiquad(nChannels, sampleRate) {
53 configure(nChannels, sampleRate);
56 void AudioShelvingFilter::configure(int nChannels, int sampleRate) {
57 mNiquistFreq = sampleRate * 500;
59 mBiquad.configure(nChannels, sampleRate);
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| AudioEqualizer.cpp | 39 int nChannels, int sampleRate,
43 "sampleRate=%d, nPresets=%d)",
44 pMem, nBands, nChannels, sampleRate, nPresets);
54 return new (pMem) AudioEqualizer(pMem, nBands, nChannels, sampleRate,
58 void AudioEqualizer::configure(int nChannels, int sampleRate) {
59 ALOGV("AudioEqualizer::configure(nChannels=%d, sampleRate=%d)", nChannels,
60 sampleRate);
61 mpLowShelf->configure(nChannels, sampleRate);
63 mpPeakingFilters[i].configure(nChannels, sampleRate);
65 mpHighShelf->configure(nChannels, sampleRate);
[all...] |
| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| DelayNode.cpp | 39 DelayNode::DelayNode(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionState& exceptionState)
40 : AudioBasicProcessorNode(context, sampleRate)
51 m_processor = adoptPtr(new DelayProcessor(context, sampleRate, 1, maxDelayTime));
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| DelayNode.h | 39 static PassRefPtr<DelayNode> create(AudioContext* context, float sampleRate, double maxDelayTime, ExceptionState& exceptionState)
41 return adoptRef(new DelayNode(context, sampleRate, maxDelayTime, exceptionState));
47 DelayNode(AudioContext*, float sampleRate, double maxDelayTime, ExceptionState&);
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| AudioBuffer.cpp | 53 PassRefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
55 if (sampleRate < minAllowedSampleRate() || sampleRate > maxAllowedSampleRate() || numberOfChannels > AudioContext::maxNumberOfChannels() || !numberOfFrames)
58 RefPtr<AudioBuffer> buffer = adoptRef(new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate));
65 PassRefPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate)
67 RefPtr<AudioBus> bus = createBusFromInMemoryAudioFile(data, dataSize, mixToMono, sampleRate);
82 AudioBuffer::AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
84 , m_sampleRate(sampleRate)
105 , m_sampleRate(bus->sampleRate())
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| DelayDSPKernel.cpp | 43 ASSERT(processor && processor->sampleRate() > 0);
44 if (!(processor && processor->sampleRate() > 0))
52 m_buffer.allocate(bufferLengthForDelay(m_maxDelayTime, processor->sampleRate()));
55 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, processor->sampleRate());
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| OfflineAudioContext.h | 36 static PassRefPtr<OfflineAudioContext> create(ExecutionContext*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState&);
41 OfflineAudioContext(Document*, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
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| AudioBuffer.h | 46 static PassRefPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
49 static PassRefPtr<AudioBuffer> createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate);
53 double duration() const { return length() / sampleRate(); }
54 float sampleRate() const { return m_sampleRate; }
74 AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate);
|
| /external/chromium_org/third_party/WebKit/Source/testing/runner/ |
| MockWebAudioDevice.cpp | 37 MockWebAudioDevice::MockWebAudioDevice(double sampleRate)
38 : m_sampleRate(sampleRate)
54 double MockWebAudioDevice::sampleRate()
|
| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| psy_configuration.h | 91 Word32 GetSRIndex(Word32 sampleRate);
95 Word32 samplerate,
100 Word32 samplerate,
|
| psy_main.h | 50 Word32 sampleRate,
67 Word32 sampleRate);
|
| tns_func.h | 32 Word32 samplerate,
39 Word32 samplerate,
|
| /frameworks/av/media/libstagefright/codecs/aacenc/src/ |
| aacenc.c | 142 config.sampleRate = 44100;
274 pOutInfo->Format.SampleRate = hAacEnc->config.sampleRate;
334 config.sampleRate = pAAC_param->sampleRate;
341 /* check the samplerate */
345 if(config.sampleRate == sampRateTab[i])
357 if(config.sampleRate%8000 == 0)
362 (config.bitRate > config.sampleRate*6*config.nChannelsOut)))
364 config.bitRate = 640*config.sampleRate/tmp*config.nChannelsOut
[all...] |
| aacenc_core.c | 90 config.sampleRate,
111 qcInit.averageBits = (Word16) ((config.bitRate * FRAME_LEN_LONG) / config.sampleRate);
113 qcInit.padding.paddingRest = config.sampleRate;
116 (config.sampleRate>>1));
130 hAacEnc->bseInit.sampleRate = config.sampleRate;
172 aacEnc->config.sampleRate);
177 aacEnc->config.sampleRate);
|
| /device/generic/goldfish/camera/ |
| media_profiles.xml | 38 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
100 sampleRate="8000"
113 sampleRate="8000"
134 sampleRate="8000"
147 sampleRate="8000"
168 sampleRate="8000"
181 sampleRate="8000"
202 sampleRate="8000"
215 sampleRate="8000"
236 sampleRate="8000
[all...] |
| /hardware/qcom/audio/legacy/alsa_sound/ |
| AudioHardwareALSA.cpp | 724 uint32_t *sampleRate,
728 ALOGV("openOutputStream: devices 0x%x channels %d sampleRate %d",
729 devices, *channels, *sampleRate);
747 ((*sampleRate == VOIP_SAMPLING_RATE_8K) || (*sampleRate == VOIP_SAMPLING_RATE_16K))) {
762 if(*sampleRate == VOIP_SAMPLING_RATE_8K) {
765 else if(*sampleRate == VOIP_SAMPLING_RATE_16K) {
769 ALOGE("unsupported samplerate %d for voip",*sampleRate);
783 alsa_handle.sampleRate = *sampleRate
[all...] |
| /frameworks/av/media/libstagefright/codecs/aacenc/SampleCode/ |
| AAC_E_SAMPLES.c | 38 "voAACEncTest -if <inputfile.pcm> -of <outputfile.aac> -sr <samplerate> -ch <channel> -br <bitrate> -adts <adts> \n"
41 "-sr input pcm samplerate, default 44100 \n"
43 "-br encoded aac bitrate, default 64000 * (samplerate/100)*channel/441(480)\n"
53 // bitRate/nChannels < sampleRate*6
57 param->sampleRate = 44100;
84 param->sampleRate = atoi(*argv);
116 if(param->sampleRate%8000 == 0)
118 param->bitRate = 640*param->nChannels*param->sampleRate/scale;
|
| /frameworks/base/core/java/android/speech/srec/ |
| MicrophoneInputStream.java | 42 * @param sampleRate sample rate of the microphone, typically 11025 or 8000.
43 * @param fifoDepth depth of the real time fifo, measured in sampleRate clock ticks.
46 public MicrophoneInputStream(int sampleRate, int fifoDepth) throws IOException {
47 mAudioRecord = AudioRecordNew(sampleRate, fifoDepth);
105 private static native int AudioRecordNew(int sampleRate, int fifoDepth);
|
| /hardware/libhardware_legacy/audio/ |
| A2dpAudioInterface.h | 52 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
59 uint32_t *sampleRate=0,
67 uint32_t *sampleRate,
85 virtual uint32_t sampleRate() const { return 44100; }
90 virtual uint32_t latency() const { return ((1000*bufferSize())/frameSize())/sampleRate() + 200; }
|
