| /frameworks/base/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_Equaliser.c | 72 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
78 Offset = (LVM_UINT16)(pParams->SampleRate + (pParams->SpeakerType * (1+LVM_FS_48000)));
|
| LVCS_StereoEnhancer.c | 69 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
76 Offset = (LVM_UINT16)pParams->SampleRate;
99 Offset = (LVM_UINT16)(pParams->SampleRate);
|
| /frameworks/base/media/libstagefright/rtsp/ |
| ARawAudioAssembler.cpp | 134 int32_t sampleRate, numChannels;
136 desc, &sampleRate, &numChannels);
138 format->setInt32(kKeySampleRate, sampleRate);
|
| /frameworks/media/libvideoeditor/vss/src/ |
| VideoEditorResampler.cpp | 78 M4OSA_Int32 sampleRate, M4OSA_Int32 quality) {
82 bitDepth, inChannelCount, sampleRate, AudioResampler::DEFAULT);
89 context->outSamplingRate = sampleRate;
|
| /hardware/libhardware_legacy/audio/ |
| AudioHardwareInterface.cpp | 110 size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
112 if (sampleRate != 8000) {
113 LOGW("getInputBufferSize bad sampling rate: %d", sampleRate);
|
| /external/webkit/Source/WebCore/platform/audio/mac/ |
| AudioBusMac.mm | 44 PassOwnPtr<AudioBus> AudioBus::loadPlatformResource(const char* name, double sampleRate)
54 OwnPtr<AudioBus> bus(createBusFromInMemoryAudioFile([audioData bytes], [audioData length], false, sampleRate));
|
| AudioDestinationMac.cpp | 43 PassOwnPtr<AudioDestination> AudioDestination::create(AudioSourceProvider& provider, double sampleRate)
45 return adoptPtr(new AudioDestinationMac(provider, sampleRate));
70 AudioDestinationMac::AudioDestinationMac(AudioSourceProvider& provider, double sampleRate)
74 , m_sampleRate(sampleRate)
|
| /external/webkit/Source/WebCore/webaudio/ |
| AudioChannelSplitter.cpp | 40 AudioChannelSplitter::AudioChannelSplitter(AudioContext* context, double sampleRate)
41 : AudioNode(context, sampleRate)
|
| JavaScriptAudioNode.cpp | 45 PassRefPtr<JavaScriptAudioNode> JavaScriptAudioNode::create(AudioContext* context, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs)
47 return adoptRef(new JavaScriptAudioNode(context, sampleRate, bufferSize, numberOfInputs, numberOfOutputs));
50 JavaScriptAudioNode::JavaScriptAudioNode(AudioContext* context, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs)
51 : AudioNode(context, sampleRate)
99 double sampleRate = context()->sampleRate();
104 m_inputBuffers.append(AudioBuffer::create(2, bufferSize(), sampleRate));
105 m_outputBuffers.append(AudioBuffer::create(2, bufferSize(), sampleRate));
|
| OfflineAudioDestinationNode.h | 52 double sampleRate() const { return m_renderTarget->sampleRate(); }
|
| RealtimeAnalyserNode.cpp | 36 RealtimeAnalyserNode::RealtimeAnalyserNode(AudioContext* context, double sampleRate)
37 : AudioNode(context, sampleRate)
|
| /external/webkit/Source/WebKit/chromium/public/ |
| WebAudioBus.h | 50 WEBKIT_API void initialize(unsigned numberOfChannels, size_t length, double sampleRate);
57 WEBKIT_API double sampleRate() const;
|
| /external/webkit/Source/WebKit/chromium/src/ |
| AudioDestinationChromium.h | 45 AudioDestinationChromium(AudioSourceProvider&, double sampleRate);
52 double sampleRate() const { return m_sampleRate; }
|
| /frameworks/base/media/libeffects/lvm/lib/Common/src/ |
| LVC_Mixer_SetTimeConstant.c | 29 /* Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) */
74 pInstance->Delta=Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) in Q 0.31 format
|
| LVC_Mixer_VarSlope_SetTimeConstant.c | 30 /* Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) */
94 pInstance->Delta=Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) in Q 0.31 format
|
| /system/media/wilhelm/tools/permute/ |
| permute.c | 173 switch (sfinfo_in.samplerate) {
183 fprintf(stderr, "%s: unsupported sample rate %d\n", path_in, sfinfo_in.samplerate);
212 double durationSeconds = (double) sfinfo_in.frames / (double) sfinfo_in.samplerate;
214 s.mMinSegmentLengthFrames = minSegmentLengthSeconds * sfinfo_in.samplerate;
259 sfinfo_out.samplerate = sfinfo_in.samplerate;
294 sfinfo_in.samplerate), (unsigned) ((s.mSegmentArray[i].mFrameLength * 1000.0) /
295 sfinfo_in.samplerate));
|
| /frameworks/base/voip/jni/rtp/ |
| AudioGroup.cpp | 100 AudioCodec *codec, int sampleRate, int sampleCount,
104 bool mix(int32_t *output, int head, int tail, int sampleRate);
166 AudioCodec *codec, int sampleRate, int sampleCount,
178 mSampleRate = sampleRate / 1000;
234 bool AudioStream::mix(int32_t *output, int head, int tail, int sampleRate)
253 if (sampleRate == mSampleRate) {
476 bool set(int sampleRate, int sampleCount);
572 bool AudioGroup::set(int sampleRate, int sampleCount)
580 mSampleRate = sampleRate;
594 sampleRate, sampleCount, -1, -1))
[all...] |
| /cts/apps/CtsVerifier/jni/audioquality/ |
| GenerateSinusoid.h | 28 void generateSinusoid(float freq, float duration, float sampleRate,
|
| /device/generic/goldfish/audio/ |
| AudioHardwareGeneric.h | 49 virtual uint32_t sampleRate() const { return 44100; }
83 virtual uint32_t sampleRate() const { return 8000; }
123 uint32_t *sampleRate=0,
131 uint32_t *sampleRate,
|
| /external/webkit/Source/WebCore/platform/audio/ |
| AudioBus.h | 79 double sampleRate() const { return m_sampleRate; }
80 void setSampleRate(double sampleRate) { m_sampleRate = sampleRate; }
133 static PassOwnPtr<AudioBus> loadPlatformResource(const char* name, double sampleRate);
|
| /frameworks/base/include/media/ |
| IAudioFlinger.h | 50 uint32_t sampleRate,
63 uint32_t sampleRate,
74 virtual uint32_t sampleRate(int output) const = 0;
112 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
|
| /frameworks/base/media/jni/soundpool/ |
| SoundPool.cpp | 497 uint32_t sampleRate;
503 p = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format);
505 p = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format);
514 LOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d",
515 p->pointer(), p->size(), sampleRate, numChannels);
517 if (sampleRate > kMaxSampleRate) {
518 LOGE("Sample rate (%u) out of range", sampleRate);
533 mSampleRate = sampleRate;
581 uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5)
[all...] |
| /frameworks/base/media/libeffects/lvm/lib/Common/lib/ |
| Filter.h | 54 LVM_Fs_en SampleRate);
|
| /frameworks/base/media/libeffects/testlibs/ |
| AudioPeakingFilter.h | 43 // sampleRate The input/output sample rate, in Hz.
44 AudioPeakingFilter(int nChannels, int sampleRate);
49 // sampleRate The input/output sample rate, in Hz.
50 void configure(int nChannels, int sampleRate);
|
| AudioShelvingFilter.h | 50 // sampleRate The input/output sample rate, in Hz.
51 AudioShelvingFilter(ShelfType type, int nChannels, int sampleRate);
56 // sampleRate The input/output sample rate, in Hz.
57 void configure(int nChannels, int sampleRate);
|
