| /frameworks/opt/net/voip/src/jni/rtp/ |
| GsmCodec.cpp | 42 int set(int sampleRate, const char */* fmtp */) {
43 return (sampleRate == 8000 && mEncode && mDecode) ? 160 : -1;
|
| /cts/tests/tests/nativemedia/aaudio/src/ |
| utils.cpp | 56 aaudio_direction_t direction, int32_t sampleRate,
60 mRequested{sampleRate, channelCount, dataFormat, sharingMode, perfMode},
77 AAudioStreamBuilder_setSampleRate(mBuilder, mRequested.sampleRate);
108 mActual.sampleRate = AAudioStream_getSampleRate(mStream);
109 ASSERT_GE(mActual.sampleRate, 44100);
110 ASSERT_LE(mActual.sampleRate, 96000); // TODO what is min/max?
|
| /external/libmtp/logs/ |
| mtp-detect-samsung-yh-925.txt | 68 de93: SampleRate
90 de93: SampleRate
106 de93: SampleRate
120 de93: SampleRate
134 de93: SampleRate
156 de93: SampleRate
|
| mtp-detect-creative-zen-vision-W.txt | 77 de93: SampleRate
100 de93: SampleRate
121 de93: SampleRate
145 de93: SampleRate
222 de93: SampleRate
248 de93: SampleRate
274 de93: SampleRate
300 de93: SampleRate
326 de93: SampleRate
|
| mtp-detect-toshiba-gigabeat-p10.txt | 75 de93: SampleRate
92 de93: SampleRate
109 de93: SampleRate
|
| mtp-detect-nokia.txt | 187 de93: SampleRate
213 de93: SampleRate
239 de93: SampleRate
265 de93: SampleRate
291 de93: SampleRate
317 de93: SampleRate
343 de93: SampleRate
395 de93: SampleRate
|
| /frameworks/av/media/libeffects/lvm/lib/StereoWidening/src/ |
| LVCS_Equaliser.c | 77 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
83 Offset = (LVM_UINT16)(pParams->SampleRate + (pParams->SpeakerType * (1 + LVM_FS_48000)));
137 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
143 Offset = (LVM_UINT16)(pParams->SampleRate + (pParams->SpeakerType * (1+LVM_FS_48000)));
|
| LVCS_BypassMix.c | 129 LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[0],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
134 LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
145 LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[1],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
153 LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
205 LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[0],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
207 LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[1],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
240 LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
243 LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
|
| LVCS_ReverbGenerator.c | 86 if(pInstance->Params.SampleRate != pParams->SampleRate ) /* Sample rate change test */
92 Delay = (LVM_UINT16)LVCS_StereoDelayCS[(LVM_UINT16)pParams->SampleRate];
104 Offset = (LVM_UINT16)pParams->SampleRate;
170 if(pInstance->Params.SampleRate != pParams->SampleRate ) /* Sample rate change test */
176 Delay = (LVM_UINT16)LVCS_StereoDelayCS[(LVM_UINT16)pParams->SampleRate];
188 Offset = (LVM_UINT16)pParams->SampleRate;
|
| LVCS_StereoEnhancer.c | 76 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
83 Offset = (LVM_UINT16)pParams->SampleRate;
107 Offset = (LVM_UINT16)(pParams->SampleRate);
165 if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
172 Offset = (LVM_UINT16)pParams->SampleRate;
195 Offset = (LVM_UINT16)(pParams->SampleRate);
|
| /hardware/interfaces/automotive/vehicle/2.0/default/common/src/ |
| VehicleHalManager.cpp | 154 ops.sampleRate = checkSampleRate(*config, ops.sampleRate);
167 mHal->subscribe(opt.propId, opt.vehicleAreas, opt.sampleRate);
267 float sampleRate) {
269 if (std::abs(sampleRate) > std::numeric_limits<float>::epsilon()) {
275 if (sampleRate > config.maxSampleRate) {
277 "to max.", sampleRate, config.maxSampleRate);
280 if (sampleRate < config.minSampleRate) {
282 "to min.", sampleRate, config.minSampleRate);
286 return sampleRate; // Provided sample rate was good, no changes
[all...] |
| /frameworks/av/media/libstagefright/rtsp/ |
| APacketSource.cpp | 471 int32_t sampleRate, numChannels;
473 desc.c_str(), &sampleRate, &numChannels);
475 mFormat->setInt32(kKeySampleRate, sampleRate);
487 int32_t sampleRate, numChannels;
489 desc.c_str(), &sampleRate, &numChannels);
491 mFormat->setInt32(kKeySampleRate, sampleRate);
494 if (sampleRate != 8000 || numChannels != 1) {
500 int32_t sampleRate, numChannels;
502 desc.c_str(), &sampleRate, &numChannels);
504 mFormat->setInt32(kKeySampleRate, sampleRate);
[all...] |
| ARawAudioAssembler.cpp | 136 int32_t sampleRate, numChannels;
138 desc, &sampleRate, &numChannels);
140 format->setInt32(kKeySampleRate, sampleRate);
|
| /external/sonic/ |
| Sonic.java | 41 private int sampleRate;
178 int sampleRate,
181 minPeriod = sampleRate/SONIC_MAX_PITCH;
182 maxPeriod = sampleRate/SONIC_MIN_PITCH;
191 this.sampleRate = sampleRate;
200 int sampleRate,
203 allocateStreamBuffers(sampleRate, numChannels);
217 return sampleRate;
222 int sampleRate)
[all...] |
| /frameworks/av/media/libeffects/testlibs/ |
| AudioBiquadFilter.cpp | 28 AudioBiquadFilter::AudioBiquadFilter(int nChannels, int sampleRate) {
29 configure(nChannels, sampleRate);
33 void AudioBiquadFilter::configure(int nChannels, int sampleRate) {
35 assert(sampleRate > 0);
39 / sampleRate;
|
| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| click_annotate.cc | 34 printf("%s PCMfile DATfile chunkSize sampleRate\n\n", argv[0]);
35 printf("Opens the PCMfile with sampleRate in Hertz.\n");
65 printf("\nThe sampleRate must be a positive integer\n\n");
|
| /frameworks/av/media/libeffects/lvm/lib/Common/src/ |
| LVC_Mixer_SetTimeConstant.c | 29 /* Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) */
89 pInstance->Delta = Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec)
118 pInstance->Delta=Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) in Q 0.31 format
|
| LVC_Mixer_VarSlope_SetTimeConstant.c | 30 /* Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) */
111 pInstance->Delta = Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec)
159 pInstance->Delta=Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) in Q 0.31 format
|
| /hardware/qcom/audio/legacy/alsa_sound/ |
| AudioUsbALSA.h | 90 struct pcm * configureDevice(unsigned flags, char* hw, int sampleRate, int channelCount, int periodSize, bool playback);
109 status_t setHardwareParams(pcm *local_handle, uint32_t sampleRate, uint32_t channels, int periodSize);
115 status_t getCap(char * type, int &channels, int &sampleRate);
|
| /external/webrtc/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/ |
| WebRtcAudioRecord.java | 153 private int initRecording(int sampleRate, int channels) {
154 Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" +
166 final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
178 sampleRate,
196 sampleRate,
|
| WebRtcAudioTrack.java | 155 private void initPlayout(int sampleRate, int channels) {
156 Logging.d(TAG, "initPlayout(sampleRate=" + sampleRate + ", channels="
160 bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
172 sampleRate,
187 sampleRate,
|
| /cts/apps/CtsVerifier/src/com/android/cts/verifier/audio/audiolib/ |
| WaveTableFloatFiller.java | 51 public void setSampleRate(float sampleRate) {
52 mSampleRate = sampleRate;
|
| /external/python/cpython2/Lib/plat-mac/Carbon/ |
| MediaDescr.py | 72 'compressionID', 'packetSize', ('sampleRate', _tofixed, _fromfixed)),
81 'compressionID', 'packetSize', ('sampleRate', _tofixed, _fromfixed), 'samplesPerPacket',
|
| /frameworks/av/media/libaaudio/src/client/ |
| IsochronousClockModel.h | 44 * @param sampleRate rate of the stream in frames per second
46 void setSampleRate(int32_t sampleRate);
|
| /frameworks/av/media/libstagefright/ |
| AudioSource.cpp | 54 uint32_t sampleRate, uint32_t channelCount, uint32_t outSampleRate,
57 mSampleRate(sampleRate),
58 mOutSampleRate(outSampleRate > 0 ? outSampleRate : sampleRate),
71 ALOGV("sampleRate: %u, outSampleRate: %u, channelCount: %u",
72 sampleRate, outSampleRate, channelCount);
74 CHECK(sampleRate > 0);
78 sampleRate,
93 inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
|
