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