/hardware/libhardware_legacy/audio/ |
H A D | A2dpAudioInterface.h | 52 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount); 59 uint32_t *sampleRate=0, 67 uint32_t *sampleRate, 85 virtual uint32_t sampleRate() const { return 44100; } function in class:android_audio_legacy::A2dpAudioInterface::A2dpAudioStreamOut 90 virtual uint32_t latency() const { return ((1000*bufferSize())/frameSize())/sampleRate() + 200; }
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H A D | AudioHardwareGeneric.h | 50 virtual uint32_t sampleRate() const { return 44100; } function in class:android_audio_legacy::AudioStreamOutGeneric 84 virtual uint32_t sampleRate() const { return 8000; } function in class:android_audio_legacy::AudioStreamInGeneric 124 uint32_t *sampleRate=0, 132 uint32_t *sampleRate,
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H A D | AudioHardwareStub.h | 33 virtual uint32_t sampleRate() const { return 44100; } function in class:android_audio_legacy::AudioStreamOutStub 50 virtual uint32_t sampleRate() const { return 8000; } function in class:android_audio_legacy::AudioStreamInStub 83 uint32_t *sampleRate=0, 91 uint32_t *sampleRate,
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H A D | AudioHardwareInterface.cpp | 113 size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) argument 115 if (sampleRate != 8000) { 116 ALOGW("getInputBufferSize bad sampling rate: %d", sampleRate);
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H A D | AudioHardwareGeneric.cpp | 68 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) 82 status_t lStatus = out->set(this, mFd, devices, format, channels, sampleRate); 102 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, 122 status_t lStatus = in->set(this, mFd, devices, format, channels, sampleRate, acoustics); 207 if (lRate == 0) lRate = sampleRate(); 212 (lRate != sampleRate())) { 215 if (pRate) *pRate = sampleRate(); 252 snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate()); 325 (*pRate != sampleRate())) { 329 *pRate = sampleRate(); 67 openOutputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) argument 101 openInputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) argument [all...] |
H A D | 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()); 45 openOutputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) argument 64 openInputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) argument
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H A D | A2dpAudioInterface.cpp | 64 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) 68 return mHardwareInterface->openOutputStream(devices, format, channels, sampleRate, status); 82 if ((err = out->set(devices, format, channels, sampleRate)) == NO_ERROR) { 107 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, 110 return mHardwareInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics); 202 size_t A2dpAudioInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) argument 204 return mHardwareInterface->getInputBufferSize(sampleRate, format, channelCount); 247 if (lRate == 0) lRate = sampleRate(); 252 (lRate != sampleRate())){ 255 if (pRate) *pRate = sampleRate(); 63 openOutputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) argument 106 openInputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) argument [all...] |
H A D | AudioDumpInterface.cpp | 60 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) 68 outFinal = mFinalInterface->openOutputStream(devices, format, channels, sampleRate, status); 72 lRate = outFinal->sampleRate(); 88 if (sampleRate != 0) { 89 if (*sampleRate != 0) { 90 lRate = *sampleRate; 92 *sampleRate = lRate; 127 uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) 134 inFinal = mFinalInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics); 138 lRate = inFinal->sampleRate(); 59 openOutputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) argument 126 openInputStream(uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) argument 251 getInputBufferSize(uint32_t sampleRate, int format, int channelCount) argument 258 AudioStreamOutDump(AudioDumpInterface *interface, int id, AudioStreamOut* finalStream, uint32_t devices, int format, uint32_t channels, uint32_t sampleRate) argument 312 uint32_t AudioStreamOutDump::sampleRate() const function in class:android::AudioStreamOutDump 419 AudioStreamInDump(AudioDumpInterface *interface, int id, AudioStreamIn* finalStream, uint32_t devices, int format, uint32_t channels, uint32_t sampleRate) argument 515 uint32_t AudioStreamInDump::sampleRate() const function in class:android::AudioStreamInDump [all...] |
/hardware/qcom/audio/legacy/alsa_sound/ |
H A D | ALSAStreamOps.cpp | 135 if (mHandle->sampleRate != *rate) 138 *rate = mHandle->sampleRate; 295 uint32_t ALSAStreamOps::sampleRate() const function in class:android_audio_legacy::ALSAStreamOps 297 return mHandle->sampleRate;
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H A D | AudioHardwareALSA.cpp | 724 uint32_t *sampleRate, 728 ALOGV("openOutputStream: devices 0x%x channels %d sampleRate %d", 729 devices, *channels, *sampleRate); 747 ((*sampleRate == VOIP_SAMPLING_RATE_8K) || (*sampleRate == VOIP_SAMPLING_RATE_16K))) { 762 if(*sampleRate == VOIP_SAMPLING_RATE_8K) { 765 else if(*sampleRate == VOIP_SAMPLING_RATE_16K) { 769 ALOGE("unsupported samplerate %d for voip",*sampleRate); 783 alsa_handle.sampleRate = *sampleRate; 721 openOutputStream(uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status) argument 1092 openInputStream(uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) argument 1410 getInputBufferSize(uint32_t sampleRate, int format, int channelCount) argument [all...] |
H A D | AudioUsbALSA.cpp | 91 status_t AudioUsbALSA::getCap(char * type, int &channels, int &sampleRate) argument 104 sampleRate = 0; 216 sampleRate = ratesSupported[i]; 220 ALOGD("sampleRate: %d", sampleRate); 278 status_t AudioUsbALSA::setHardwareParams(pcm *txHandle, uint32_t sampleRate, uint32_t channels, int periodBytes) argument 284 unsigned int requestedRate = sampleRate; 299 ALOGV("Setting period size:%d samplerate:%d, channels: %d",periodBytes,sampleRate, channels); 306 param_set_int(params, SNDRV_PCM_HW_PARAM_RATE, sampleRate); 389 uint32_t sampleRate; local 590 configureDevice(unsigned flags, char* hw, int sampleRate, int channelCount, int periodSize, bool playback) argument 778 uint32_t sampleRate; local [all...] |
H A D | AudioHardwareALSA.h | 174 uint32_t sampleRate; member in struct:android_audio_legacy::alsa_handle_t 274 uint32_t sampleRate() const; 298 virtual uint32_t sampleRate() const function in class:android_audio_legacy::AudioStreamOutALSA 300 return ALSAStreamOps::sampleRate(); 354 virtual uint32_t sampleRate() const function in class:android_audio_legacy::AudioStreamInALSA 356 return ALSAStreamOps::sampleRate(); 487 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channels); 507 uint32_t *sampleRate=0, 516 uint32_t *sampleRate,
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/hardware/ti/omap3/omx/audio/src/openmax_il/aac_enc/inc/ |
H A D | OMX_AacEnc_Utils.h | 247 long sampleRate; /* Samplling frequency in Hz */ member in struct:__anon2059
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/hardware/ti/omap3/omx/audio/src/openmax_il/wma_dec/tests/ |
H A D | WmaDecTest.c | 1733 OMX_U16 sampleRate; local 1749 sampleRate = *((OMX_U16 *)(pBuffer + 46)); 1753 APP_DPRINT("sampleRate %d\n", sampleRate); 1755 pcmBytesPerPacket = (long double)playDuration * (long double) sampleRate / (10000000 * (long double) numPackets);
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