/external/webkit/Source/WebCore/webaudio/ |
AudioBasicProcessorNode.h | 42 AudioBasicProcessorNode(AudioContext*, double sampleRate);
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BiquadProcessor.h | 50 BiquadProcessor(FilterType, double sampleRate, size_t numberOfChannels, bool autoInitialize = true);
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AudioBasicProcessorNode.cpp | 39 AudioBasicProcessorNode::AudioBasicProcessorNode(AudioContext* context, double sampleRate) 40 : AudioNode(context, sampleRate)
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AudioBufferSourceNode.h | 48 static PassRefPtr<AudioBufferSourceNode> create(AudioContext*, double sampleRate); 84 AudioBufferSourceNode(AudioContext*, double sampleRate);
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AudioNode.h | 51 AudioNode(AudioContext*, double sampleRate); 116 double sampleRate() const { return m_sampleRate; }
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ConvolverNode.cpp | 47 ConvolverNode::ConvolverNode(AudioContext* context, double sampleRate) 48 : AudioNode(context, sampleRate)
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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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/hardware/msm7k/libaudio-qsd8k/ |
AudioHardware.cpp | 224 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) 239 status_t lStatus = out->set(this, devices, format, channels, sampleRate); 264 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, 275 status_t lStatus = in->set(this, devices, format, channels, sampleRate, acoustic_flags); 557 size_t AudioHardware::getBufferSize(uint32_t sampleRate, int channelCount) 561 if (sampleRate < 11025) { 563 } else if (sampleRate < 22050) { 565 } else if (sampleRate < 32000) { 567 } else if (sampleRate < 44100) { 577 size_t AudioHardware::getInputBufferSize(uint32_t sampleRate, int format, int channelCount [all...] |
/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/include/ |
front.h | 130 int samplerate; member in struct:__anon15425 149 int samplerate; member in struct:__anon15426
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/external/srec/srec_jni/ |
android_speech_srec_MicrophoneInputStream.cpp | 57 (JNIEnv *env, jclass clazz, jint sampleRate, jint fifoFrames) { 60 AUDIO_SOURCE_VOICE_RECOGNITION, sampleRate,
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/frameworks/av/include/media/ |
AudioTrack.h | 119 uint32_t sampleRate = 0); 137 * sampleRate: Track sampling rate in Hz. 158 uint32_t sampleRate = 0, 181 uint32_t sampleRate = 0, 200 * - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...) 206 uint32_t sampleRate = 0, 299 status_t setSampleRate(uint32_t sampleRate); 521 uint32_t 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/ |
TrackBase.h | 42 uint32_t sampleRate, 77 uint32_t sampleRate() const; // FIXME inline after cblk sr moved
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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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/external/aac/libAACenc/src/ |
aacenc.cpp | 114 INT sampleRate); 401 switch (config->sampleRate) 429 config->sampleRate, 478 config->sampleRate); 484 config->ancDataBitRate += ( (hAacEnc->ancillaryBitsPerFrame * config->sampleRate) / config->framelength ); 492 FIXP_DBL tmp = fDivNorm(config->framelength, config->sampleRate, &q_res); 516 config->sampleRate, 540 config->sampleRate, 575 qcInit.padding.paddingRest = config->sampleRate; 578 bw_ratio = fDivNorm((FIXP_DBL)hAacEnc->bandwidth90dB, (FIXP_DBL)(config->sampleRate>>1), &qbw) [all...] |
pnsparam.cpp | 188 int FDKaacEnc_lookUpPnsUse (int bitRate, int sampleRate, int numChan, const int isLC) { 213 switch (sampleRate) { 242 INT sampleRate, 267 hUsePns = FDKaacEnc_lookUpPnsUse (bitRate, sampleRate, numChan, isLC); 279 sampleRate,
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/external/aac/libSBRdec/include/ |
sbrdecoder.h | 171 * \param sampleRateIn Input samplerate of the SBR decoder instance. 172 * \param sampleRateOut Output samplerate of the SBR decoder instance. 299 * \param sampleRate Output samplerate. 310 int *sampleRate,
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/external/aac/libSBRenc/include/ |
sbr_encoder.h | 135 USHORT sampleRate; /*!< */ 296 * \param sampleRate Input: Encoder samplerate. output core encoder samplerate. 310 INT *sampleRate,
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/frameworks/wilhelm/tests/examples/ |
slesTestFeedback.cpp | 38 static SLuint32 sampleRate = 44100; // -s# 214 sampleRate = atoi(&arg[2]); 215 switch (sampleRate) { 228 (unsigned) sampleRate); 303 pcm.samplesPerSec = sampleRate * 1000;
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