| /external/jmonkeyengine/engine/src/terrain/com/jme3/terrain/noise/modulator/ |
| CatRom2.java | 39 private int sampleRate = 100;
45 public CatRom2(final int sampleRate) {
46 this.sampleRate = sampleRate;
47 this.table = new float[4 * sampleRate + 1];
48 for (int i = 0; i < 4 * sampleRate + 1; i++) {
49 float x = i / (float) sampleRate;
59 public static CatRom2 getInstance(final int sampleRate) {
60 if (!CatRom2.instances.containsKey(sampleRate)) {
61 CatRom2.instances.put(sampleRate, new CatRom2(sampleRate));
[all...] |
| /external/webkit/Source/WebCore/platform/audio/chromium/ |
| AudioBusChromium.cpp | 37 PassOwnPtr<AudioBus> AudioBus::loadPlatformResource(const char* name, double sampleRate)
39 // FIXME: the sampleRate parameter is ignored. It should be removed from the API.
40 OwnPtr<AudioBus> audioBus = PlatformBridge::loadPlatformAudioResource(name, sampleRate);
45 if (audioBus->sampleRate() == sampleRate)
48 return AudioBus::createBySampleRateConverting(audioBus.get(), false, sampleRate);
51 PassOwnPtr<AudioBus> createBusFromInMemoryAudioFile(const void* data, size_t dataSize, bool mixToMono, double sampleRate)
53 // FIXME: the sampleRate parameter is ignored. It should be removed from the API.
54 OwnPtr<AudioBus> audioBus = PlatformBridge::decodeAudioFileData(static_cast<const char*>(data), dataSize, sampleRate);
59 if ((!mixToMono || audioBus->numberOfChannels() == 1) && audioBus->sampleRate() == sampleRate
[all...] |
| /frameworks/av/media/libeffects/lvm/lib/Reverb/src/ |
| LVREV_SetControlParameters.c | 64 ((pNewParams->SampleRate != LVM_FS_8000) && (pNewParams->SampleRate != LVM_FS_11025) && (pNewParams->SampleRate != LVM_FS_12000) &&
65 (pNewParams->SampleRate != LVM_FS_16000) && (pNewParams->SampleRate != LVM_FS_22050) && (pNewParams->SampleRate != LVM_FS_24000) &&
66 (pNewParams->SampleRate != LVM_FS_32000) && (pNewParams->SampleRate != LVM_FS_44100) && (pNewParams->SampleRate != LVM_FS_48000)) ||
|
| 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...] |
| /external/webkit/Source/WebCore/webaudio/ |
| DelayDSPKernel.cpp | 47 ASSERT(processor && processor->sampleRate() > 0);
51 m_buffer.resize(static_cast<size_t>(processor->sampleRate() * DefaultMaxDelayTime));
54 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, processor->sampleRate());
57 DelayDSPKernel::DelayDSPKernel(double maxDelayTime, double sampleRate)
58 : AudioDSPKernel(sampleRate)
67 size_t bufferLength = static_cast<size_t>(sampleRate * maxDelayTime);
75 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate);
91 double sampleRate = this->sampleRate();
92 double delayTime = delayProcessor() ? delayProcessor()->delayTime()->value() : m_desiredDelayFrames / sampleRate;
[all...] |
| DelayNode.cpp | 33 DelayNode::DelayNode(AudioContext* context, double sampleRate)
34 : AudioBasicProcessorNode(context, sampleRate)
36 m_processor = adoptPtr(new DelayProcessor(sampleRate, 1));
|
| HighPass2FilterNode.cpp | 33 HighPass2FilterNode::HighPass2FilterNode(AudioContext* context, double sampleRate)
34 : AudioBasicProcessorNode(context, sampleRate)
36 m_processor = adoptPtr(new BiquadProcessor(BiquadProcessor::HighPass2, sampleRate, 1, false));
|
| LowPass2FilterNode.cpp | 33 LowPass2FilterNode::LowPass2FilterNode(AudioContext* context, double sampleRate)
34 : AudioBasicProcessorNode(context, sampleRate)
36 m_processor = adoptPtr(new BiquadProcessor(BiquadProcessor::LowPass2, sampleRate, 1, false));
|
| AudioBuffer.h | 44 static PassRefPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate);
47 static PassRefPtr<AudioBuffer> createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, double sampleRate);
51 double duration() const { return length() / sampleRate(); }
52 double sampleRate() const { return m_sampleRate; }
69 AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate);
|
| AudioBuffer.cpp | 42 PassRefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate)
44 return adoptRef(new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate));
47 PassRefPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, double sampleRate)
49 OwnPtr<AudioBus> bus = createBusFromInMemoryAudioFile(data, dataSize, mixToMono, sampleRate);
56 AudioBuffer::AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate)
58 , m_sampleRate(sampleRate)
71 , m_sampleRate(bus->sampleRate())
|
| /external/webkit/Source/WebCore/platform/audio/ |
| HRTFKernel.h | 54 static PassRefPtr<HRTFKernel> create(AudioChannel* channel, size_t fftSize, double sampleRate, bool bassBoost)
56 return adoptRef(new HRTFKernel(channel, fftSize, sampleRate, bassBoost));
59 static PassRefPtr<HRTFKernel> create(PassOwnPtr<FFTFrame> fftFrame, double frameDelay, double sampleRate)
61 return adoptRef(new HRTFKernel(fftFrame, frameDelay, sampleRate));
72 double sampleRate() const { return m_sampleRate; }
73 double nyquist() const { return 0.5 * sampleRate(); }
80 HRTFKernel(AudioChannel* channel, size_t fftSize, double sampleRate, bool bassBoost);
82 HRTFKernel(PassOwnPtr<FFTFrame> fftFrame, double frameDelay, double sampleRate)
85 , m_sampleRate(sampleRate)
|
| AudioFileReader.h | 41 // Pass in 0.0 for sampleRate to use the file's sample-rate, otherwise a sample-rate conversion to the requested
42 // sampleRate will be made (if it doesn't already match the file's sample-rate).
45 PassOwnPtr<AudioBus> createBusFromInMemoryAudioFile(const void* data, size_t dataSize, bool mixToMono, double sampleRate);
47 PassOwnPtr<AudioBus> createBusFromAudioFile(const char* filePath, bool mixToMono, double sampleRate);
49 // May pass in 0.0 for sampleRate in which case it will use the AudioBus's sampleRate
|
| AudioDSPKernel.h | 44 , m_sampleRate(kernelProcessor->sampleRate())
48 AudioDSPKernel(double sampleRate)
50 , m_sampleRate(sampleRate)
60 double sampleRate() const { return m_sampleRate; }
61 double nyquist() const { return 0.5 * sampleRate(); }
|
| HRTFDatabase.cpp | 48 PassOwnPtr<HRTFDatabase> HRTFDatabase::create(double sampleRate)
50 OwnPtr<HRTFDatabase> hrtfDatabase = adoptPtr(new HRTFDatabase(sampleRate));
54 HRTFDatabase::HRTFDatabase(double sampleRate)
56 , m_sampleRate(sampleRate)
60 OwnPtr<HRTFElevation> hrtfElevation = HRTFElevation::createForSubject("Composite", elevation, sampleRate);
79 m_elevations[i + jj] = HRTFElevation::createByInterpolatingSlices(m_elevations[i].get(), m_elevations[j].get(), x, sampleRate);
|
| /frameworks/av/media/libstagefright/codecs/aacenc/src/ |
| psy_configuration.c | 39 Word32 sampleRate;
69 Word32 GetSRIndex(Word32 sampleRate)
71 if (92017 <= sampleRate) return 0;
72 if (75132 <= sampleRate) return 1;
73 if (55426 <= sampleRate) return 2;
74 if (46009 <= sampleRate) return 3;
75 if (37566 <= sampleRate) return 4;
76 if (27713 <= sampleRate) return 5;
77 if (23004 <= sampleRate) return 6;
78 if (18783 <= sampleRate) return 7
[all...] |
| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| bitenc.h | 35 Word32 sampleRate;
47 Word16 samplerate
|
| /frameworks/av/cmds/stagefright/ |
| SineSource.h | 12 SineSource(int32_t sampleRate, int32_t numChannels);
|
| /external/srec/doc/logs/uapi/ |
| run_parameters.log | 16 CREC.Frontend.samplerate = 11025
24 CREC.Frontend.samplerate = 8000
32 CREC.Frontend.samplerate = 8000
|
| /hardware/libhardware_legacy/audio/ |
| AudioDumpInterface.cpp | 60 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
68 outFinal = mFinalInterface->openOutputStream(devices, format, channels, sampleRate, status);
72 lRate = outFinal->sampleRate();
88 if (sampleRate != 0) {
89 if (*sampleRate != 0) {
90 lRate = *sampleRate;
92 *sampleRate = lRate;
127 uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics)
134 inFinal = mFinalInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
138 lRate = inFinal->sampleRate();
[all...] |
| AudioDumpInterface.h | 42 uint32_t sampleRate);
46 virtual uint32_t sampleRate() const;
84 uint32_t sampleRate);
87 virtual uint32_t sampleRate() const;
125 uint32_t *sampleRate=0,
149 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
152 uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics);
|
| /frameworks/av/media/libeffects/lvm/lib/Bass/src/ |
| LVDBE_Control.c | 113 LVM_UINT16 Offset = (LVM_UINT16)((LVM_UINT16)pParams->SampleRate + (LVM_UINT16)(pParams->CentreFrequency * (1+LVDBE_FS_48000)));
163 pInstance->pData->AGCInstance.AGC_Attack = LVDBE_AGC_ATTACK_Table[(LVM_UINT16)pParams->SampleRate]; /* Attack multiplier */
164 pInstance->pData->AGCInstance.AGC_Decay = LVDBE_AGC_DECAY_Table[(LVM_UINT16)pParams->SampleRate]; /* Decay multipler */
247 pInstance->pData->AGCInstance.VolumeTC = LVDBE_VolumeTCTable[(LVM_UINT16)pParams->SampleRate]; /* Volume update time constant */
265 (LVM_Fs_en)pInstance->Params.SampleRate,
283 /* SampleRate: Changing the sample rate may cause pops and clicks. */
318 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
329 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
[all...] |
| /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;
|
| /external/webkit/Source/WebCore/bindings/js/ |
| JSAudioContextCustom.cpp | 64 // new AudioContext(in unsigned long numberOfChannels, in unsigned long numberOfFrames, in float sampleRate);
70 float sampleRate = exec->argument(2).toFloat(exec);
72 audioContext = AudioContext::createOfflineContext(document, numberOfChannels, numberOfFrames, sampleRate);
107 // AudioBuffer createBuffer(in unsigned long numberOfChannels, in unsigned long numberOfFrames, in float sampleRate);
113 float sampleRate = exec->argument(2).toFloat(exec);
115 RefPtr<AudioBuffer> audioBuffer = audioContext->createBuffer(numberOfChannels, numberOfFrames, sampleRate);
|
| /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);
|
| 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);
|
