| /external/chromium_org/third_party/WebKit/Source/modules/webaudio/ |
| AudioBasicProcessorNode.h | 42 AudioBasicProcessorNode(AudioContext*, float sampleRate);
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| BiquadProcessor.h | 52 BiquadProcessor(AudioContext*, float sampleRate, size_t numberOfChannels, bool autoInitialize);
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| MediaElementAudioSourceNode.h | 55 virtual void setFormat(size_t numberOfChannels, float sampleRate);
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| MediaStreamAudioSourceNode.h | 55 virtual void setFormat(size_t numberOfChannels, float sampleRate);
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| PannerNode.cpp | 49 PannerNode::PannerNode(AudioContext* context, float sampleRate)
50 : AudioNode(context, sampleRate)
147 m_panner = Panner::create(m_panningModel, sampleRate(), context()->hrtfDatabaseLoader());
202 OwnPtr<Panner> newPanner = Panner::create(model, sampleRate(), context()->hrtfDatabaseLoader());
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| WaveShaperProcessor.h | 47 WaveShaperProcessor(float sampleRate, size_t numberOfChannels);
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| AudioBufferSourceNode.h | 47 static PassRefPtr<AudioBufferSourceNode> create(AudioContext*, float sampleRate);
97 AudioBufferSourceNode(AudioContext*, float sampleRate);
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| /external/chromium_org/third_party/WebKit/Source/platform/audio/ |
| AudioDSPKernelProcessor.h | 54 AudioDSPKernelProcessor(float sampleRate, unsigned numberOfChannels);
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| Panner.h | 52 static PassOwnPtr<Panner> create(PanningModel, float sampleRate, HRTFDatabaseLoader*);
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| AudioDSPKernelProcessor.cpp | 43 AudioDSPKernelProcessor::AudioDSPKernelProcessor(float sampleRate, unsigned numberOfChannels)
44 : AudioProcessor(sampleRate, numberOfChannels)
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| EqualPowerPanner.cpp | 43 EqualPowerPanner::EqualPowerPanner(float sampleRate)
49 m_smoothingConstant = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate);
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| /external/chromium_org/third_party/WebKit/public/testing/ |
| WebTestInterfaces.h | 78 blink::WebAudioDevice* createAudioDevice(double sampleRate);
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| /external/srec/srec/include/ |
| frontpar.h | 35 int samplerate; member in struct:__anon26483
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| front.h | 130 int samplerate; member in struct:__anon26477
149 int samplerate; member in struct:__anon26478
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| /frameworks/av/include/media/stagefright/ |
| AudioSource.h | 38 uint32_t sampleRate,
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| /frameworks/av/media/libeffects/lvm/lib/Bass/lib/ |
| LVDBE.h | 164 /* Capabilities.SampleRate = LVDBE_CAP_32000 + LVCS_DBE_44100; */
244 LVDBE_Fs_en SampleRate;
258 LVM_UINT16 SampleRate; /* Sampling rate capabilities */
405 /* SampleRate: Changing the sample rate may cause pops and clicks. */
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| /frameworks/av/media/libstagefright/ |
| AACWriter.cpp | 210 static bool getSampleRateTableIndex(int sampleRate, uint8_t* tableIndex) {
220 if (sampleRate == kSampleRateTable[index]) {
222 sampleRate, index);
228 ALOGE("Sampling rate %d bps is not supported", sampleRate);
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| /hardware/libhardware_legacy/include/hardware_legacy/ |
| AudioHardwareBase.h | 48 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
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| /hardware/qcom/audio/legacy/alsa_sound/ |
| ALSAStreamOps.cpp | 135 if (mHandle->sampleRate != *rate)
138 *rate = mHandle->sampleRate;
295 uint32_t ALSAStreamOps::sampleRate() const
297 return mHandle->sampleRate;
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| /external/libmtp/examples/ |
| tracks.c | 51 if (track->samplerate != 0) {
52 printf(" Sample rate: %u Hz\n", track->samplerate);
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| /external/srec/srec_jni/ |
| android_speech_srec_MicrophoneInputStream.cpp | 69 (JNIEnv *env, jclass clazz, jint sampleRate, jint fifoFrames) {
72 AUDIO_SOURCE_VOICE_RECOGNITION, sampleRate,
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| /frameworks/av/media/libnbaio/ |
| NBAIO.cpp | 96 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount)
99 switch (sampleRate) {
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| /frameworks/av/media/libstagefright/codecs/aacenc/src/ |
| tns.c | 134 Word32 sampleRate, /*!< Sampling frequency */
162 tC->tnsStartBand = FreqToBandWithRounding(tC->tnsStartFreq, sampleRate,
166 sampleRate,
171 sampleRate,
201 Word32 sampleRate, /*!< Sampling frequency */
228 tC->tnsStartBand=FreqToBandWithRounding(tC->tnsStartFreq, sampleRate,
232 sampleRate,
237 sampleRate,
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| /frameworks/av/services/audioflinger/ |
| AudioResampler.h | 47 int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);
86 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
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| /frameworks/base/core/java/android/speech/tts/ |
| SynthesisPlaybackQueueItem.java | 66 SynthesisPlaybackQueueItem(int streamType, int sampleRate,
78 mAudioTrack = new BlockingAudioTrack(streamType, sampleRate, audioFormat,
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