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Searched
defs:sampleRate
(Results
1 - 25
of
211
) sorted by null
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/external/chromium_org/content/shell/renderer/test_runner/
mock_web_audio_device.cc
18
double MockWebAudioDevice::
sampleRate
() {
/frameworks/av/media/libstagefright/codecs/aacenc/inc/
bitenc.h
35
Word32
sampleRate
;
47
Word16
samplerate
/external/aac/libMpegTPEnc/src/
tpenc_adif.cpp
109
INT
sampleRate
= adif->samplingRate;
147
transportEnc_writePCE(hBs, adif->cm,
sampleRate
, adif->instanceTag, adif->profile, 0, 0, alignAnchor);
/external/chromium_org/third_party/WebKit/Source/platform/audio/
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
(); }
AudioProcessor.h
47
AudioProcessor(float
sampleRate
, unsigned numberOfChannels)
50
, m_sampleRate(
sampleRate
)
72
float
sampleRate
() const { return m_sampleRate; }
AudioDestination.h
50
AudioDestination(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float
sampleRate
);
55
static PassOwnPtr<AudioDestination> create(AudioIOCallback&, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float
sampleRate
);
61
float
sampleRate
() const { return m_sampleRate; }
HRTFDatabase.h
46
static PassOwnPtr<HRTFDatabase> create(float
sampleRate
);
57
float
sampleRate
() const { return m_sampleRate; }
63
explicit HRTFDatabase(float
sampleRate
);
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,
90
HRTFElevation(PassOwnPtr<HRTFKernelList> kernelListL, PassOwnPtr<HRTFKernelList> kernelListR, int elevation, float
sampleRate
)
94
, m_sampleRate(
sampleRate
)
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
)
AudioBus.h
90
float
sampleRate
() const { return m_sampleRate; }
91
void setSampleRate(float
sampleRate
) { m_sampleRate =
sampleRate
; }
147
static PassRefPtr<AudioBus> loadPlatformResource(const char* name, float
sampleRate
);
/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/content/renderer/media/
renderer_webaudiodevice_impl.cc
74
double RendererWebAudioDeviceImpl::
sampleRate
() {
/external/chromium_org/third_party/WebKit/Source/modules/webaudio/
AudioBuffer.h
46
static AudioBuffer* create(unsigned numberOfChannels, size_t numberOfFrames, float
sampleRate
);
47
static AudioBuffer* create(unsigned numberOfChannels, size_t numberOfFrames, float
sampleRate
, ExceptionState&);
50
static AudioBuffer* createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, float
sampleRate
);
56
double duration() const { return length() / static_cast<double>(
sampleRate
()); }
57
float
sampleRate
() const { return m_sampleRate; }
70
AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, float
sampleRate
);
AudioScheduledSourceNode.cpp
44
AudioScheduledSourceNode::AudioScheduledSourceNode(AudioContext* context, float
sampleRate
)
45
: AudioSourceNode(context,
sampleRate
)
66
double
sampleRate
= this->
sampleRate
();
74
size_t startFrame = AudioUtilities::timeToSampleFrame(m_startTime,
sampleRate
);
75
size_t endFrame = m_endTime == UnknownTime ? 0 : AudioUtilities::timeToSampleFrame(m_endTime,
sampleRate
);
AudioParam.cpp
158
double
sampleRate
= context()->
sampleRate
();
160
double endTime = startTime + numberOfValues /
sampleRate
;
164
m_value = m_timeline.valuesForTimeRange(startTime, endTime, narrowPrecisionToFloat(m_value), values, numberOfValues,
sampleRate
,
sampleRate
);
ScriptProcessorNode.cpp
59
ScriptProcessorNode* ScriptProcessorNode::create(AudioContext* context, float
sampleRate
, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
87
return adoptRefCountedGarbageCollectedWillBeNoop(new ScriptProcessorNode(context,
sampleRate
, bufferSize, numberOfInputChannels, numberOfOutputChannels));
90
ScriptProcessorNode::ScriptProcessorNode(AudioContext* context, float
sampleRate
, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
91
: AudioNode(context,
sampleRate
)
130
float
sampleRate
= context()->
sampleRate
();
135
AudioBuffer* inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(),
sampleRate
) : 0;
136
AudioBuffer* outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(),
sampleRate
) : 0;
259
double playbackTime = (context()->currentSampleFrame() + m_bufferSize) / static_cast<double>(context()->
sampleRate
());
/external/chromium_org/third_party/WebKit/Source/platform/exported/
WebAudioBus.cpp
47
void WebAudioBus::initialize(unsigned numberOfChannels, size_t length, double
sampleRate
)
51
audioBus->setSampleRate(
sampleRate
);
112
double WebAudioBus::
sampleRate
() const
117
return m_private->
sampleRate
();
/frameworks/av/media/libnbaio/
AudioStreamInSource.cpp
46
uint32_t
sampleRate
= mStream->common.get_sample_rate(&mStream->common);
49
mFormat = Format_from_SR_C(
sampleRate
,
AudioStreamOutSink.cpp
43
uint32_t
sampleRate
= mStream->common.get_sample_rate(&mStream->common);
46
mFormat = Format_from_SR_C(
sampleRate
,
/frameworks/av/media/libstagefright/codecs/common/include/
voAAC.h
45
int
sampleRate
; /*! audio file sample rate */
/external/sonivox/arm-fm-22k/lib_src/
eas_pcm.h
45
EAS_U32
sampleRate
;
/external/sonivox/arm-hybrid-22k/lib_src/
eas_pcm.h
45
EAS_U32
sampleRate
;
/external/sonivox/arm-wt-22k/lib_src/
eas_pcm.h
45
EAS_U32
sampleRate
;
/frameworks/av/media/libstagefright/rtsp/
ARawAudioAssembler.cpp
136
int32_t
sampleRate
, numChannels;
138
desc, &
sampleRate
, &numChannels);
140
format->setInt32(kKeySampleRate,
sampleRate
);
/hardware/libhardware_legacy/audio/
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
,
Completed in 1146 milliseconds
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