/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...] |
/frameworks/base/tests/Camera2Tests/SmartCamera/SimpleCamera/src/androidx/media/filterfw/decoder/ |
AudioSample.java | 21 public final int sampleRate; 25 public AudioSample(int sampleRate, int channelCount, byte[] bytes) { 26 this.sampleRate = sampleRate;
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/external/chromium_org/third_party/WebKit/Source/core/platform/audio/chromium/ |
AudioBusChromium.cpp | 36 PassRefPtr<AudioBus> decodeAudioFileData(const char* data, size_t size, double sampleRate) 39 if (WebKit::Platform::current()->loadAudioResource(&webAudioBus, data, size, sampleRate)) 44 PassRefPtr<AudioBus> AudioBus::loadPlatformResource(const char* name, float sampleRate) 50 // FIXME: the sampleRate parameter is ignored. It should be removed from the API. 51 RefPtr<AudioBus> audioBus = decodeAudioFileData(resource.data(), resource.size(), sampleRate); 57 if (audioBus->sampleRate() == sampleRate) 60 return AudioBus::createBySampleRateConverting(audioBus.get(), false, sampleRate); 63 PassRefPtr<AudioBus> createBusFromInMemoryAudioFile(const void* data, size_t dataSize, bool mixToMono, float sampleRate) 65 // FIXME: the sampleRate parameter is ignored. It should be removed from the API [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)) ||
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/external/chromium_org/third_party/WebKit/Source/core/platform/audio/ |
HRTFKernel.h | 54 static PassRefPtr<HRTFKernel> create(AudioChannel* channel, size_t fftSize, float sampleRate) 56 return adoptRef(new HRTFKernel(channel, fftSize, sampleRate)); 59 static PassRefPtr<HRTFKernel> create(PassOwnPtr<FFTFrame> fftFrame, float frameDelay, float sampleRate) 61 return adoptRef(new HRTFKernel(fftFrame, frameDelay, sampleRate)); 72 float sampleRate() const { return m_sampleRate; } 73 double nyquist() const { return 0.5 * sampleRate(); } 80 HRTFKernel(AudioChannel*, size_t fftSize, float sampleRate); 82 HRTFKernel(PassOwnPtr<FFTFrame> fftFrame, float frameDelay, float sampleRate) 85 , m_sampleRate(sampleRate)
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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 PassRefPtr<AudioBus> createBusFromInMemoryAudioFile(const void* data, size_t dataSize, bool mixToMono, float sampleRate); 47 PassRefPtr<AudioBus> createBusFromAudioFile(const char* filePath, bool mixToMono, float sampleRate); 49 // May pass in 0.0 for sampleRate in which case it will use the AudioBus's sampleRate
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AudioUtilities.cpp | 52 double discreteTimeConstantForSampleRate(double timeConstant, double sampleRate) 54 return 1 - exp(-1 / (sampleRate * timeConstant)); 57 size_t timeToSampleFrame(double time, double sampleRate) 59 return static_cast<size_t>(round(time * sampleRate));
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AudioDSPKernel.h | 44 , m_sampleRate(kernelProcessor->sampleRate()) 48 AudioDSPKernel(float sampleRate) 50 , m_sampleRate(sampleRate) 60 float sampleRate() const { return m_sampleRate; } 61 double nyquist() const { return 0.5 * sampleRate(); }
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HRTFDatabaseLoader.cpp | 43 PassRefPtr<HRTFDatabaseLoader> HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(float sampleRate) 52 loader = s_loaderMap->get(sampleRate); 54 ASSERT(sampleRate == loader->databaseSampleRate()); 58 loader = adoptRef(new HRTFDatabaseLoader(sampleRate)); 59 s_loaderMap->add(sampleRate, loader.get()); 66 HRTFDatabaseLoader::HRTFDatabaseLoader(float sampleRate) 68 , m_databaseSampleRate(sampleRate)
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/external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
AudioBuffer.cpp | 43 PassRefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) 45 if (sampleRate < 22050 || sampleRate > 96000 || numberOfChannels > AudioContext::maxNumberOfChannels() || !numberOfFrames) 48 return adoptRef(new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate)); 51 PassRefPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float sampleRate) 53 RefPtr<AudioBus> bus = createBusFromInMemoryAudioFile(data, dataSize, mixToMono, sampleRate); 60 AudioBuffer::AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) 62 , m_sampleRate(sampleRate) 77 , m_sampleRate(bus->sampleRate())
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DelayDSPKernel.cpp | 46 ASSERT(processor && processor->sampleRate() > 0); 47 if (!(processor && processor->sampleRate() > 0)) 55 m_buffer.allocate(bufferLengthForDelay(m_maxDelayTime, processor->sampleRate())); 58 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, processor->sampleRate()); 61 DelayDSPKernel::DelayDSPKernel(double maxDelayTime, float sampleRate) 62 : AudioDSPKernel(sampleRate) 71 size_t bufferLength = bufferLengthForDelay(maxDelayTime, sampleRate); 79 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate); 82 size_t DelayDSPKernel::bufferLengthForDelay(double maxDelayTime, double sampleRate) const 86 return 1 + AudioUtilities::timeToSampleFrame(maxDelayTime, sampleRate); [all...] |
OfflineAudioContext.cpp | 38 PassRefPtr<OfflineAudioContext> OfflineAudioContext::create(ScriptExecutionContext* context, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& es) 48 if (numberOfChannels > 10 || !isSampleRateRangeGood(sampleRate)) { 53 RefPtr<OfflineAudioContext> audioContext(adoptRef(new OfflineAudioContext(document, numberOfChannels, numberOfFrames, sampleRate))); 58 OfflineAudioContext::OfflineAudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) 59 : AudioContext(document, numberOfChannels, numberOfFrames, sampleRate)
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AsyncAudioDecoder.cpp | 56 void AsyncAudioDecoder::decodeAsync(ArrayBuffer* audioData, float sampleRate, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback) 63 OwnPtr<DecodingTask> decodingTask = DecodingTask::create(audioData, sampleRate, successCallback, errorCallback); 92 PassOwnPtr<AsyncAudioDecoder::DecodingTask> AsyncAudioDecoder::DecodingTask::create(ArrayBuffer* audioData, float sampleRate, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback) 94 return adoptPtr(new DecodingTask(audioData, sampleRate, successCallback, errorCallback)); 97 AsyncAudioDecoder::DecodingTask::DecodingTask(ArrayBuffer* audioData, float sampleRate, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback) 99 , m_sampleRate(sampleRate) 112 m_audioBuffer = AudioBuffer::createFromAudioFileData(m_audioData->data(), m_audioData->byteLength(), false, sampleRate());
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PeriodicWave.h | 45 static PassRefPtr<PeriodicWave> createSine(float sampleRate); 46 static PassRefPtr<PeriodicWave> createSquare(float sampleRate); 47 static PassRefPtr<PeriodicWave> createSawtooth(float sampleRate); 48 static PassRefPtr<PeriodicWave> createTriangle(float sampleRate); 51 static PassRefPtr<PeriodicWave> create(float sampleRate, Float32Array* real, Float32Array* imag); 65 float sampleRate() const { return m_sampleRate; } 68 explicit PeriodicWave(float sampleRate);
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/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...] |
/external/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/apple/ |
AppleLosslessSpecificBox.java | 20 bitrate? 32bit samplerate
32 private long sampleRate; // 32bit
115 return sampleRate;
118 public void setSampleRate(int sampleRate) {
119 this.sampleRate = sampleRate;
136 sampleRate = IsoTypeReader.readUInt32(content);
152 IsoTypeWriter.writeUInt32(byteBuffer, sampleRate);
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/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
bitenc.h | 35 Word32 sampleRate; 47 Word16 samplerate
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/external/chromium_org/ppapi/examples/audio/ |
audio.html | 15 <embed id="plugin" type="application/x-ppapi-example-audio" samplerate="44100">
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/frameworks/av/cmds/stagefright/ |
SineSource.h | 12 SineSource(int32_t sampleRate, int32_t numChannels);
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/external/srec/doc/logs/uapi/ |
run_parameters.log | 16 CREC.Frontend.samplerate = 11025 24 CREC.Frontend.samplerate = 8000 32 CREC.Frontend.samplerate = 8000
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/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);
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/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/ |
EC3TrackImpl.java | 29 int samplerate; field in class:EC3TrackImpl 76 samplerate = entries.get(0).samplerate; 81 audioSampleEntry.setSampleRate(samplerate); 119 trackMetaData.setTimescale(samplerate); // Audio tracks always use samplerate as timescale 350 entry.samplerate = 48000; 354 entry.samplerate = 44100; 358 entry.samplerate = 32000; 364 entry.samplerate = 24000 411 public int samplerate; field in class:EC3TrackImpl.BitStreamInfo [all...] |
/external/mp4parser/isoparser/src/main/java/com/coremedia/iso/boxes/sampleentry/ |
AudioSampleEntry.java | 55 private long sampleRate; 82 return sampleRate; 133 public void setSampleRate(long sampleRate) { 134 this.sampleRate = sampleRate; 198 //sampleRate = in.readFixedPoint1616(); 199 sampleRate = IsoTypeReader.readUInt32(content); 201 sampleRate = sampleRate >>> 16; 242 ", sampleRate=" + sampleRate [all...] |
/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...] |