| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| OscillatorNode.cpp | 46 PassRefPtr<OscillatorNode> OscillatorNode::create(AudioContext* context, float sampleRate)
48 return adoptRef(new OscillatorNode(context, sampleRate));
51 OscillatorNode::OscillatorNode(AudioContext* context, float sampleRate)
52 : AudioScheduledSourceNode(context, sampleRate)
117 float sampleRate = this->sampleRate();
121 DEFINE_STATIC_REF(PeriodicWave, periodicWaveSine, (PeriodicWave::createSine(sampleRate)));
126 DEFINE_STATIC_REF(PeriodicWave, periodicWaveSquare, (PeriodicWave::createSquare(sampleRate)));
131 DEFINE_STATIC_REF(PeriodicWave, periodicWaveSawtooth, (PeriodicWave::createSawtooth(sampleRate)));
136 DEFINE_STATIC_REF(PeriodicWave, periodicWaveTriangle, (PeriodicWave::createTriangle(sampleRate)));
[all...] |
| AudioContext.cpp | 81 bool AudioContext::isSampleRateRangeGood(float sampleRate)
85 return sampleRate >= 44100 && sampleRate <= 96000;
107 PassRefPtr<AudioContext> AudioContext::create(Document& document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
110 return OfflineAudioContext::create(&document, numberOfChannels, numberOfFrames, sampleRate, exceptionState);
136 m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(sampleRate());
140 AudioContext::AudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
155 m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(sampleRate);
158 m_renderTarget = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate);
297 PassRefPtr<AudioBuffer> AudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState
[all...] |
| AnalyserNode.h | 38 static PassRefPtr<AnalyserNode> create(AudioContext* context, float sampleRate)
40 return adoptRef(new AnalyserNode(context, sampleRate));
72 AnalyserNode(AudioContext*, float sampleRate);
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| BiquadFilterNode.h | 49 static PassRefPtr<BiquadFilterNode> create(AudioContext* context, float sampleRate)
51 return adoptRef(new BiquadFilterNode(context, sampleRate));
70 BiquadFilterNode(AudioContext*, float sampleRate);
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| ConvolverNode.h | 40 static PassRefPtr<ConvolverNode> create(AudioContext* context, float sampleRate)
42 return adoptRef(new ConvolverNode(context, sampleRate));
61 ConvolverNode(AudioContext*, float sampleRate);
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| DynamicsCompressorNode.h | 38 static PassRefPtr<DynamicsCompressorNode> create(AudioContext* context, float sampleRate)
40 return adoptRef(new DynamicsCompressorNode(context, sampleRate));
65 DynamicsCompressorNode(AudioContext*, float sampleRate);
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| GainNode.h | 42 static PassRefPtr<GainNode> create(AudioContext* context, float sampleRate)
44 return adoptRef(new GainNode(context, sampleRate));
61 GainNode(AudioContext*, float sampleRate);
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| PeriodicWave.cpp | 48 PassRefPtr<PeriodicWave> PeriodicWave::create(float sampleRate, Float32Array* real, Float32Array* imag)
53 RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
61 PassRefPtr<PeriodicWave> PeriodicWave::createSine(float sampleRate)
63 RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
68 PassRefPtr<PeriodicWave> PeriodicWave::createSquare(float sampleRate)
70 RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
75 PassRefPtr<PeriodicWave> PeriodicWave::createSawtooth(float sampleRate)
77 RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
82 PassRefPtr<PeriodicWave> PeriodicWave::createTriangle(float sampleRate)
84 RefPtr<PeriodicWave> periodicWave = adoptRef(new PeriodicWave(sampleRate));
[all...] |
| AsyncAudioDecoder.cpp | 50 void AsyncAudioDecoder::decodeAsync(ArrayBuffer* audioData, float sampleRate, PassOwnPtr<AudioBufferCallback> successCallback, PassOwnPtr<AudioBufferCallback> errorCallback)
61 m_thread->postTask(new Task(WTF::bind(&AsyncAudioDecoder::decode, audioDataRef.release().leakRef(), sampleRate, successCallback.leakPtr(), errorCallback.leakPtr())));
64 void AsyncAudioDecoder::decode(ArrayBuffer* audioData, float sampleRate, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback)
67 RefPtr<AudioBuffer> audioBuffer = AudioBuffer::createFromAudioFileData(audioData->data(), audioData->byteLength(), false, sampleRate);
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| AudioContext.idl | 30 Constructor(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate),
42 readonly attribute float sampleRate;
50 [RaisesException] AudioBuffer createBuffer(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate);
82 // void prepareOfflineBufferRendering(unsigned long numberOfChannels, unsigned long numberOfFrames, float sampleRate);
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| ChannelSplitterNode.cpp | 37 PassRefPtr<ChannelSplitterNode> ChannelSplitterNode::create(AudioContext* context, float sampleRate, unsigned numberOfOutputs)
42 return adoptRef(new ChannelSplitterNode(context, sampleRate, numberOfOutputs));
45 ChannelSplitterNode::ChannelSplitterNode(AudioContext* context, float sampleRate, unsigned numberOfOutputs)
46 : AudioNode(context, sampleRate)
|
| /device/asus/flo/ |
| media_profiles.xml | 38 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
100 sampleRate="8000"
112 sampleRate="48000"
124 sampleRate="48000"
136 sampleRate="48000"
148 sampleRate="48000"
161 sampleRate="8000"
174 sampleRate="48000"
187 sampleRate="48000"
200 sampleRate="48000
[all...] |
| /device/lge/hammerhead/ |
| media_profiles.xml | 38 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
100 sampleRate="8000"
112 sampleRate="48000"
124 sampleRate="48000"
136 sampleRate="48000"
148 sampleRate="48000"
161 sampleRate="8000"
174 sampleRate="48000"
187 sampleRate="48000"
200 sampleRate="48000
[all...] |
| /device/lge/mako/ |
| media_profiles.xml | 38 <!ATTLIST Audio sampleRate CDATA #REQUIRED>
100 sampleRate="8000"
112 sampleRate="48000"
124 sampleRate="48000"
136 sampleRate="48000"
148 sampleRate="48000"
161 sampleRate="8000"
174 sampleRate="48000"
187 sampleRate="48000"
200 sampleRate="48000
[all...] |
| /frameworks/av/media/libeffects/lvm/lib/Eq/src/ |
| LVEQNB_Control.c | 146 LVM_UINT32 fs = (LVM_UINT32)LVEQNB_SampleRateTab[(LVM_UINT16)pParams->SampleRate]; /* Sample rate */
240 LVEQNB_DoublePrecCoefs((LVM_UINT16)pInstance->Params.SampleRate,
260 LVEQNB_SinglePrecCoefs((LVM_UINT16)pInstance->Params.SampleRate,
357 if (pParams->SampleRate != pInstance->Params.SampleRate)
359 LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->BypassMixer.MixerStream[0],LVEQNB_BYPASS_MIXER_TC,(LVM_Fs_en)pParams->SampleRate,2);
360 LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->BypassMixer.MixerStream[1],LVEQNB_BYPASS_MIXER_TC,(LVM_Fs_en)pParams->SampleRate,2);
367 (pInstance->Params.SampleRate != pParams->SampleRate ) ||
394 if (pInstance->Params.SampleRate != pParams->SampleRate
[all...] |
| /external/chromium_org/third_party/WebKit/Source/platform/audio/ |
| HRTFElevation.h | 53 static PassOwnPtr<HRTFElevation> createForSubject(const String& subjectName, int elevation, float sampleRate);
56 static PassOwnPtr<HRTFElevation> createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x, float sampleRate);
64 float sampleRate() const { return m_sampleRate; }
86 static bool calculateKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName,
92 static bool calculateSymmetricKernelsForAzimuthElevation(int azimuth, int elevation, float sampleRate, const String& subjectName,
96 HRTFElevation(PassOwnPtr<HRTFKernelList> kernelListL, PassOwnPtr<HRTFKernelList> kernelListR, int elevation, float sampleRate)
100 , m_sampleRate(sampleRate)
|
| AudioProcessor.h | 46 AudioProcessor(float sampleRate, unsigned numberOfChannels)
49 , m_sampleRate(sampleRate)
70 float sampleRate() const { return m_sampleRate; }
|
| Panner.cpp | 41 PassOwnPtr<Panner> Panner::create(PanningModel model, float sampleRate, HRTFDatabaseLoader* databaseLoader)
47 panner = adoptPtr(new EqualPowerPanner(sampleRate));
51 panner = adoptPtr(new HRTFPanner(sampleRate, databaseLoader));
|
| /frameworks/av/media/libeffects/lvm/lib/Bundle/src/ |
| LVM_Control.c | 68 ((pParams->SampleRate != LVM_FS_8000) && (pParams->SampleRate != LVM_FS_11025) && (pParams->SampleRate != LVM_FS_12000) &&
69 (pParams->SampleRate != LVM_FS_16000) && (pParams->SampleRate != LVM_FS_22050) && (pParams->SampleRate != LVM_FS_24000) &&
70 (pParams->SampleRate != LVM_FS_32000) && (pParams->SampleRate != LVM_FS_44100) && (pParams->SampleRate != LVM_FS_48000)) ||
279 (pParams->SampleRate >= TrebleBoostMinRate) &
[all...] |
| /frameworks/av/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_Control.c | 87 if (pParams->SampleRate != pInstance->Params.SampleRate)
89 pInstance->TimerParams.SamplingRate = LVCS_SampleRateTable[pParams->SampleRate];
103 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
140 LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
142 LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
|
| /external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/ |
| AACTrackImpl.java | 67 int samplerate; field in class:AACTrackImpl
105 double packetsPerSecond = (double)samplerate / 1024.0;
136 audioSampleEntry.setSampleRate(samplerate);
158 audioSpecificConfig.setSamplingFrequencyIndex(samplingFrequencyIndexMap.get(samplerate));
172 trackMetaData.setTimescale(samplerate); // Audio tracks always use samplerate as timescale
232 samplerate = samplingFrequencyIndexMap.get(brb.readBits(4));
285 "samplerate=" + samplerate +
|
| /hardware/libhardware_legacy/audio/ |
| AudioHardwareStub.cpp | 46 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
49 status_t lStatus = out->set(format, channels, sampleRate);
65 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
74 status_t lStatus = in->set(format, channels, sampleRate, acoustics);
123 if (pRate) *pRate = sampleRate();
131 usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
146 snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
177 usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
189 snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
|
| AudioHardwareStub.h | 33 virtual uint32_t sampleRate() const { return 44100; }
50 virtual uint32_t sampleRate() const { return 8000; }
83 uint32_t *sampleRate=0,
91 uint32_t *sampleRate,
|
| /external/chromium_org/third_party/WebKit/Source/platform/exported/ |
| WebAudioBus.cpp | 49 void WebAudioBus::initialize(unsigned numberOfChannels, size_t length, double sampleRate)
53 audioBus->setSampleRate(sampleRate);
114 double WebAudioBus::sampleRate() const
119 return m_private->sampleRate();
|
| /frameworks/av/media/libstagefright/ |
| VBRISeeker.cpp | 49 int sampleRate;
50 if (!GetMPEGAudioFrameSize(tmp, &frameSize, &sampleRate)) {
70 numFrames * 1000000ll * (sampleRate >= 32000 ? 1152 : 576) / sampleRate;
|
