| /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;
|
| /external/chromium_org/third_party/WebKit/Source/platform/audio/chromium/ |
| AudioBusChromium.cpp | 37 PassRefPtr<AudioBus> decodeAudioFileData(const char* data, size_t size, double sampleRate)
40 if (blink::Platform::current()->loadAudioResource(&webAudioBus, data, size, sampleRate))
45 PassRefPtr<AudioBus> AudioBus::loadPlatformResource(const char* name, float sampleRate)
51 // FIXME: the sampleRate parameter is ignored. It should be removed from the API.
52 RefPtr<AudioBus> audioBus = decodeAudioFileData(resource.data(), resource.size(), sampleRate);
58 if (audioBus->sampleRate() == sampleRate)
61 return AudioBus::createBySampleRateConverting(audioBus.get(), false, sampleRate);
64 PassRefPtr<AudioBus> createBusFromInMemoryAudioFile(const void* data, size_t dataSize, bool mixToMono, float sampleRate)
66 // 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)) ||
|
| /external/chromium_org/third_party/WebKit/Source/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)
|
| AudioFileReader.h | 40 // Pass in 0.0 for sampleRate to use the file's sample-rate, otherwise a sample-rate conversion to the requested
41 // sampleRate will be made (if it doesn't already match the file's sample-rate).
44 PLATFORM_EXPORT PassRefPtr<AudioBus> createBusFromInMemoryAudioFile(const void* data, size_t dataSize, bool mixToMono, float sampleRate);
46 PLATFORM_EXPORT PassRefPtr<AudioBus> createBusFromAudioFile(const char* filePath, bool mixToMono, float sampleRate);
48 // May pass in 0.0 for sampleRate in which case it will use the AudioBus's sampleRate
|
| AudioUtilities.cpp | 53 double discreteTimeConstantForSampleRate(double timeConstant, double sampleRate)
55 return 1 - exp(-1 / (sampleRate * timeConstant));
58 size_t timeToSampleFrame(double time, double sampleRate)
60 return static_cast<size_t>(round(time * sampleRate));
|
| AudioDelayDSPKernel.cpp | 48 AudioDelayDSPKernel::AudioDelayDSPKernel(double maxDelayTime, float sampleRate)
49 : AudioDSPKernel(sampleRate)
58 size_t bufferLength = bufferLengthForDelay(maxDelayTime, sampleRate);
66 m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate);
69 size_t AudioDelayDSPKernel::bufferLengthForDelay(double maxDelayTime, double sampleRate) const
73 return 1 + AudioUtilities::timeToSampleFrame(maxDelayTime, sampleRate);
86 double AudioDelayDSPKernel::delayTime(float sampleRate)
88 return m_desiredDelayFrames / sampleRate;
104 float sampleRate = this->sampleRate();
[all...] |
| 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(); }
|
| HRTFDatabaseLoader.cpp | 44 PassRefPtr<HRTFDatabaseLoader> HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(float sampleRate)
53 loader = s_loaderMap->get(sampleRate);
55 ASSERT(sampleRate == loader->databaseSampleRate());
59 loader = adoptRef(new HRTFDatabaseLoader(sampleRate));
60 s_loaderMap->add(sampleRate, loader.get());
67 HRTFDatabaseLoader::HRTFDatabaseLoader(float sampleRate)
68 : m_databaseSampleRate(sampleRate)
|
| HRTFPanner.cpp | 48 HRTFPanner::HRTFPanner(float sampleRate, HRTFDatabaseLoader* databaseLoader)
51 , m_sampleRate(sampleRate)
59 , m_convolverL1(fftSizeForSampleRate(sampleRate))
60 , m_convolverR1(fftSizeForSampleRate(sampleRate))
61 , m_convolverL2(fftSizeForSampleRate(sampleRate))
62 , m_convolverR2(fftSizeForSampleRate(sampleRate))
63 , m_delayLineL(MaxDelayTimeSeconds, sampleRate)
64 , m_delayLineR(MaxDelayTimeSeconds, sampleRate)
77 size_t HRTFPanner::fftSizeForSampleRate(float sampleRate)
82 ASSERT(sampleRate >= 44100 && sampleRate <= 96000.0)
[all...] |
| /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/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| OfflineAudioContext.cpp | 39 PassRefPtr<OfflineAudioContext> OfflineAudioContext::create(ExecutionContext* context, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
61 if (!isSampleRateRangeGood(sampleRate)) {
62 exceptionState.throwDOMException(SyntaxError, "sample rate (" + String::number(sampleRate) + ") must be in the range 44100-96000 Hz.");
66 RefPtr<OfflineAudioContext> audioContext(adoptRef(new OfflineAudioContext(document, numberOfChannels, numberOfFrames, sampleRate)));
71 OfflineAudioContext::OfflineAudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
72 : AudioContext(document, numberOfChannels, numberOfFrames, sampleRate)
|
| 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);
|
| AudioParamTimeline.cpp | 130 double sampleRate = context->sampleRate();
132 double endTime = startTime + 1.1 / sampleRate; // time just beyond one sample-frame
133 double controlRate = sampleRate / AudioNode::ProcessingSizeInFrames; // one parameter change per render quantum
134 value = valuesForTimeRange(startTime, endTime, defaultValue, &value, 1, sampleRate, controlRate);
146 double sampleRate,
159 float value = valuesForTimeRangeImpl(startTime, endTime, defaultValue, values, numberOfValues, sampleRate, controlRate);
170 double sampleRate,
193 unsigned fillToFrame = AudioUtilities::timeToSampleFrame(fillToTime - startTime, sampleRate);
223 double sampleFrameTimeIncr = 1 / sampleRate;
[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);
|
| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| bitenc.h | 35 Word32 sampleRate;
47 Word16 samplerate
|
| /external/chromium_org/ppapi/examples/audio/ |
| audio.html | 15 <embed id="plugin" type="application/x-ppapi-example-audio" samplerate="44100">
|
| /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);
|
| /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...] |
