| /cts/tests/tests/media/src/android/media/cts/ |
| CodecState.java | 282 int sampleRate =
289 " sampleRate:" + sampleRate + " channels:" + channelCount);
293 // are a few cases where ch=0 and samplerate=0 were returned by MediaExtractor.
295 sampleRate < 8000 || sampleRate > 128000) {
298 mAudioTrack = new NonBlockingAudioTrack(sampleRate, channelCount,
|
| /cts/tests/tests/nativemedia/aaudio/src/ |
| test_aaudio.cpp | 74 const int32_t framesPerMsec = actual().sampleRate / MILLIS_PER_SECOND;
86 const int32_t framesToRecord = actual().sampleRate; // 1 second
110 const int32_t framesToRecord = actual().sampleRate / 8;
129 const int32_t framesToRecord = actual().sampleRate / 10; // 1/10 second
222 writeLoops = 1 * actual().sampleRate / framesPerBurst(); // 1 second
225 actual().sampleRate); // N bursts
263 ASSERT_NEAR(actual().sampleRate, measuredRate, rateTolerance);
|
| /frameworks/av/media/libaudioprocessing/tests/ |
| test-mixer.cpp | 50 fprintf(stderr, " <command> can be 'sine:[(i|f),]<channels>,<frequency>,<samplerate>'\n");
51 fprintf(stderr, " 'chirp:[(i|f),]<channels>,<samplerate>'\n");
55 uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) {
62 info.samplerate = sampleRate;
65 printf("saving file:%s channels:%u samplerate:%u frames:%zu\n",
66 filename, info.channels, info.samplerate, frames);
|
| /external/aac/libPCMutils/src/ |
| limiter.cpp | 100 unsigned int sampleRate, maxSampleRate;
164 limiter->sampleRate = maxSampleRate;
406 TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate)
414 if (sampleRate > limiter->maxSampleRate) return TDLIMIT_INVALID_PARAMETER;
417 attack = (unsigned int)(limiter->attackMs * sampleRate / 1000);
418 release = (unsigned int)(limiter->releaseMs * sampleRate / 1000);
433 limiter->sampleRate = sampleRate;
453 attack = (unsigned int)(attackMs * limiter->sampleRate / 1000);
477 release = (unsigned int)(releaseMs * limiter->sampleRate / 1000)
[all...] |
| /external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/ |
| AC3TrackImpl.java | 21 int samplerate; field in class:AC3TrackImpl
62 audioSampleEntry.setSampleRate(samplerate);
81 trackMetaData.setTimescale(samplerate); // Audio tracks always use samplerate as timescale
149 samplerate = 48000;
153 samplerate = 44100;
157 samplerate = 32000;
161 samplerate = 0;
165 if (samplerate == 0) {
183 samplerate /= 2
[all...] |
| /external/libmtp/logs/ |
| mtp-detect-creative-zen-V.txt | 79 de93: SampleRate
101 de93: SampleRate
121 de93: SampleRate
143 de93: SampleRate
166 de93: SampleRate
|
| /frameworks/av/media/libstagefright/rtsp/ |
| AMPEG4ElementaryAssembler.cpp | 89 static bool GetSampleRateIndex(int32_t sampleRate, size_t *tableIndex) {
99 if (sampleRate == kSampleRateTable[index]) {
189 int32_t sampleRate, numChannels;
191 desc.c_str(), &sampleRate, &numChannels);
194 CHECK(GetSampleRateIndex(sampleRate, &mSampleRateIndex));
|
| /frameworks/base/media/tests/NativeMidiDemo/java/com/example/android/nativemididemo/ |
| NativeMidi.java | 281 String sampleRate = am.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
282 if (sampleRate == null) sampleRate = "48000";
286 String audioInitResult = initAudio(Integer.parseInt(sampleRate), mFramesPerBuffer);
340 private native String initAudio(int sampleRate, int playSamples);
|
| /frameworks/opt/net/voip/src/jni/rtp/ |
| AmrCodec.cpp | 53 int set(int sampleRate, const char *fmtp);
67 int AmrCodec::set(int sampleRate, const char *fmtp)
97 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1;
211 int set(int sampleRate, const char */* fmtp */) {
212 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1;
|
| /external/aac/libAACenc/src/ |
| psy_configuration.cpp | 100 LONG sampleRate;
211 static AAC_ENCODER_ERROR FDKaacEnc_initSfbTable(LONG sampleRate, INT blockType, INT granuleLength, INT *sfbOffset, INT *sfbCnt)
240 if(sfbInfo[i].sampleRate == sampleRate){
459 const LONG samplerate,
484 barcFactor = fDivNorm(fixMin(FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfbActive], samplerate), MAX_BARC),
489 pePerWindow = fDivNorm(bitrate, samplerate, &qperwin);
520 barcWidth = FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb+1], samplerate) -
521 FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb], samplerate);
569 INT samplerate,
[all...] |
| bandwidth.cpp | 198 const INT sampleRate,
213 switch (sampleRate) {
289 INT sampleRate,
334 *bandWidth = FDKmin(proposedBandWidth, FDKmin(20000, sampleRate>>1));
362 sampleRate,
376 *bandWidth = FDKmin(*bandWidth, sampleRate/2);
|
| /frameworks/av/media/libaudioprocessing/ |
| AudioResampler.cpp | 45 AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :
46 AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
151 int32_t sampleRate, src_quality quality) {
222 resampler = new AudioResamplerOrder1(inChannelCount, sampleRate);
227 resampler = new AudioResamplerCubic(inChannelCount, sampleRate);
232 resampler = new AudioResamplerSinc(inChannelCount, sampleRate);
237 resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality);
245 sampleRate, quality);
250 sampleRate, quality);
253 sampleRate, quality)
[all...] |
| AudioResamplerCubic.h | 31 AudioResamplerCubic(int inChannelCount, int32_t sampleRate) :
32 AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {
|
| /cts/tests/tests/media/libaudiojni/ |
| audio-record-native.cpp | 80 uint32_t sampleRate,
123 pcm.sampleRate = sampleRate * 1000;
171 mBufferSize = (BUFFER_SIZE_MSEC * sampleRate / 1000)
441 jint numChannels, jint channelMask, jint sampleRate,
445 const size_t framesPerBuffer = msecPerBuffer * sampleRate / 1000;
450 res = record.open(numChannels, channelMask, sampleRate, useFloat, numBuffers);
488 jint numChannels, jint channelMask, jint sampleRate, jboolean useFloat, jint numBuffers)
494 return (jint) record->open(numChannels, channelMask, sampleRate, useFloat == JNI_TRUE,
|
| audio-track-native.cpp | 74 jint sampleRate, jboolean useFloat, jint numBuffers) {
110 pcm.sampleRate = sampleRate * 1000;
379 jint numChannels, jint channelMask, jint sampleRate, jboolean useFloat,
384 const size_t framesPerBuffer = msecPerBuffer * sampleRate / 1000;
389 res = track.open(numChannels, channelMask, sampleRate, useFloat, numBuffers);
428 jint numChannels, jint channelMask, jint sampleRate,
437 sampleRate,
|
| /frameworks/av/media/libstagefright/codecs/aacenc/src/ |
| qc_main.c | 62 Word32 sampleRate,
70 quot = result / sampleRate;
74 result -= quot * sampleRate;
91 Word32 sampleRate,
100 sampleRate,
107 *paddingRest = *paddingRest + sampleRate;
544 Word32 sampleRate) /* output sampling rate */
553 sampleRate,
558 sampleRate,
|
| /hardware/qcom/audio/legacy/alsa_sound/ |
| AudioHardwareALSA.h | 174 uint32_t sampleRate;
274 uint32_t sampleRate() const;
298 virtual uint32_t sampleRate() const
300 return ALSAStreamOps::sampleRate();
354 virtual uint32_t sampleRate() const
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,
|
| /external/webrtc/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/ |
| WebRtcAudioManager.java | 84 private int sampleRate;
100 sampleRate, channels, hardwareAEC, hardwareAGC, hardwareNS,
140 sampleRate = getNativeOutputSampleRate();
147 getMinOutputFrameSize(sampleRate, channels);
149 inputBufferSize = getMinInputFrameSize(sampleRate, channels);
289 int sampleRate, int channels, boolean hardwareAEC, boolean hardwareAGC,
|
| /frameworks/av/media/libeffects/testlibs/ |
| AudioEqualizer.h | 70 // sampleRate The input/output sample rate, in Hz.
81 int sampleRate,
88 // sampleRate The input/output sample rate, in Hz.
89 void configure(int nChannels, int sampleRate);
232 // sampleRate The input/output sample rate, in Hz.
240 AudioEqualizer(void * pMem, int nBands, int nChannels, int sampleRate,
|
| /hardware/libhardware_legacy/include/hardware_legacy/ |
| AudioHardwareInterface.h | 53 virtual uint32_t sampleRate() const = 0;
137 virtual uint32_t sampleRate() const = 0;
251 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
258 uint32_t *sampleRate=0,
265 uint32_t *sampleRate=0,
274 uint32_t *sampleRate,
|
| /frameworks/av/services/audioflinger/ |
| AudioHwDevice.cpp | 48 " sampleRate %d, Format %#x, "
63 " sampleRate %d, Format %#x, "
|
| RecordTracks.h | 27 uint32_t sampleRate,
101 uint32_t sampleRate,
|
| SpdifStreamOut.cpp | 83 " sampleRate %d, format %#x, channelMask %#x",
88 " sampleRate %d, format %#x, channelMask %#x",
|
| /packages/apps/TV/src/com/android/exoplayer/ |
| MediaFormatUtil.java | 43 int sampleRate = getOptionalIntegerV16(format, android.media.MediaFormat.KEY_SAMPLE_RATE);
60 channelCount, sampleRate, language, MediaFormat.OFFSET_SAMPLE_RELATIVE,
|
| /packages/apps/TV/src/com/android/tv/tuner/exoplayer/audio/ |
| AudioTrackWrapper.java | 120 int sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
142 mAudioTrack.configure(mimeType, channelCount, sampleRate, pcmEncoding, audioBufferSize);
|
