Cross Reference: /hardware/qcom/audio/legacy/alsa

Lines Matching refs:sampleRate
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;
828       err = out->set(format, channels, sampleRate, devices);
868         if (0 == *sampleRate) {
869             alsa_handle.sampleRate = info.AudioBlocksArray[info.nAudioBlocks-1].nSamplingFreq;
870             *sampleRate = alsa_handle.sampleRate;
872             alsa_handle.sampleRate = *sampleRate;
877         ALOGD("alsa_handle.channels %d alsa_handle.sampleRate %d",alsa_handle.channels,alsa_handle.sampleRate);
904             err = out->set(format, channels, sampleRate, devices);
923       alsa_handle.sampleRate = DEFAULT_SAMPLING_RATE;
981           err = out->set(format, channels, sampleRate, devices);
1022     alsa_handle.sampleRate = DEFAULT_SAMPLING_RATE;
1095                                    uint32_t *sampleRate,
1108     ALOGD("openInputStream: devices 0x%x channels %d sampleRate %d", devices, *channels, *sampleRate);
1115        ((*sampleRate == VOIP_SAMPLING_RATE_8K) || (*sampleRate == VOIP_SAMPLING_RATE_16K))) {
1130            if(*sampleRate == VOIP_SAMPLING_RATE_8K) {
1133            else if(*sampleRate == VOIP_SAMPLING_RATE_16K) {
1137                ALOGE("unsupported samplerate %d for voip",*sampleRate);
1151            alsa_handle.sampleRate = *sampleRate;
1188            if(sampleRate) {
1189                it->sampleRate = *sampleRate;
1200         err = in->set(format, channels, sampleRate, devices);
1219         alsa_handle.sampleRate = android::AudioRecord::DEFAULT_SAMPLE_RATE;
1361         if(sampleRate) {
1362             it->sampleRate = *sampleRate;
1367             it->bufferSize = getInputBufferSize(it->sampleRate,*format,it->channels);
1374            err = in->set(format, channels, sampleRate, devices);
1410 size_t AudioHardwareALSA::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
1414         if(sampleRate == 8000 || sampleRate == 16000 || sampleRate == 32000) {
1415             bufferSize = (sampleRate * channelCount * 20 * sizeof(int16_t)) / 1000;
1416         } else if (sampleRate == 11025 || sampleRate == 12000) {
1418         } else if (sampleRate == 22050 || sampleRate == 24000) {
1420         } else if (sampleRate == 44100 || sampleRate == 48000) {
1456         alsa_handle.sampleRate = DEFAULT_SAMPLING_RATE;
1552     alsa_handle.sampleRate = VOICE_SAMPLING_RATE;