| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| tns_func.h | 32 Word32 samplerate,
39 Word32 samplerate,
|
| /frameworks/av/media/libstagefright/codecs/aacenc/src/ |
| aacenc_core.c | 89 config.sampleRate,
110 qcInit.averageBits = (Word16) ((config.bitRate * FRAME_LEN_LONG) / config.sampleRate);
112 qcInit.padding.paddingRest = config.sampleRate;
115 (config.sampleRate>>1));
129 hAacEnc->bseInit.sampleRate = config.sampleRate;
171 aacEnc->config.sampleRate);
176 aacEnc->config.sampleRate);
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| /frameworks/av/services/oboeservice/ |
| TimestampScheduler.h | 52 int32_t sampleRate) {
53 mBurstPeriod = AAUDIO_NANOS_PER_SECOND * framesPerBurst / sampleRate;
|
| /frameworks/av/media/libeffects/lvm/lib/Eq/src/ |
| LVEQNB_Control.c | 150 LVM_UINT32 fs = (LVM_UINT32)LVEQNB_SampleRateTab[(LVM_UINT16)pParams->SampleRate]; /* Sample rate */
248 LVEQNB_SinglePrecCoefs((LVM_UINT16)pInstance->Params.SampleRate,
267 LVEQNB_DoublePrecCoefs((LVM_UINT16)pInstance->Params.SampleRate,
287 LVEQNB_SinglePrecCoefs((LVM_UINT16)pInstance->Params.SampleRate,
401 if (pParams->SampleRate != pInstance->Params.SampleRate)
403 LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->BypassMixer.MixerStream[0],LVEQNB_BYPASS_MIXER_TC,(LVM_Fs_en)pParams->SampleRate,2);
404 LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->BypassMixer.MixerStream[1],LVEQNB_BYPASS_MIXER_TC,(LVM_Fs_en)pParams->SampleRate,2);
411 (pInstance->Params.SampleRate != pParams->SampleRate ) ||
[all...] |
| /cts/tests/tests/media/src/android/media/cts/ |
| NonBlockingAudioTrack.java | 48 public NonBlockingAudioTrack(int sampleRate, int channelCount, boolean hwAvSync,
67 sampleRate,
76 sampleRate,
91 .setSampleRate(sampleRate)
97 mSampleRate = sampleRate;
|
| AudioHelper.java | 35 public static byte[] createSoundDataInByteArray(int bufferSamples, final int sampleRate,
37 final double rad = 2 * Math.PI * frequency / sampleRate;
39 sweep = Math.PI * sweep / ((double)sampleRate * vai.length);
48 public static short[] createSoundDataInShortArray(int bufferSamples, final int sampleRate,
50 final double rad = 2 * Math.PI * frequency / sampleRate;
52 sweep = Math.PI * sweep / ((double)sampleRate * vai.length);
59 public static float[] createSoundDataInFloatArray(int bufferSamples, final int sampleRate,
61 final double rad = 2 * Math.PI * frequency / sampleRate;
63 sweep = Math.PI * sweep / ((double)sampleRate * vaf.length);
231 public AudioRecordAudit(int audioSource, int sampleRate, int channelMask
[all...] |
| AudioNativeTest.java | 47 2 /* numChannels */, 48000 /* sampleRate */, false /* useFloat */,
56 2 /* numChannels */, 48000 /* sampleRate */, false /* useFloat */,
208 doRecordTest(record, 4 /* numChannels */, 44100 /* sampleRate */, false /* useFloat */,
277 int numChannels, int sampleRate, boolean useFloat,
281 // Log.d(TEST_NAME, "open numChannels:" + numChannels + " sampleRate:" + sampleRate);
282 assertTrue(TEST_NAME, record.open(numChannels, sampleRate, useFloat,
287 (int)((long)sampleRate * segmentDurationMs * numChannels / 1000);
|
| /frameworks/av/media/libaaudio/tests/ |
| test_open_params.cpp | 59 int32_t sampleRate,
77 direction, channelCount, sampleRate, format);
84 AAudioStreamBuilder_setSampleRate(aaudioBuilder, sampleRate);
113 if (sampleRate != AAUDIO_UNSPECIFIED) {
114 EXPECT_EQ(sampleRate, actualSampleRate);
|
| /hardware/libhardware_legacy/audio/ |
| 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; }
90 virtual uint32_t latency() const { return ((1000*bufferSize())/frameSize())/sampleRate() + 200; }
|
| AudioHardwareInterface.cpp | 113 size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
115 if (sampleRate != 8000) {
116 ALOGW("getInputBufferSize bad sampling rate: %d", sampleRate);
156 uint32_t *sampleRate,
159 return openOutputStream(devices, format, channels, sampleRate, status);
|
| 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();
132 audio_channel_count_from_out_mask(channels()) / sampleRate());
147 snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
179 audio_channel_count_from_in_mask(channels()) / sampleRate());
191 snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
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| AudioHardwareStub.h | 33 virtual uint32_t sampleRate() const { return 44100; }
50 virtual uint32_t sampleRate() const { return 8000; }
83 uint32_t *sampleRate=0,
91 uint32_t *sampleRate,
|
| A2dpAudioInterface.cpp | 65 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
69 return mHardwareInterface->openOutputStream(devices, format, channels, sampleRate, status);
83 if ((err = out->set(devices, format, channels, sampleRate)) == NO_ERROR) {
108 uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
111 return mHardwareInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
203 size_t A2dpAudioInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
205 return mHardwareInterface->getInputBufferSize(sampleRate, format, channelCount);
248 if (lRate == 0) lRate = sampleRate();
253 (lRate != sampleRate())){
256 if (pRate) *pRate = sampleRate();
[all...] |
| 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();
[all...] |
| /system/media/audio_utils/spdif/ |
| DTSFrameScanner.cpp | 109 int32_t sampleRate = kDTSSampleRateTable[sfreq];
110 if (sampleRate < 0) {
111 ALOGE("DTSFrameScanner: ERROR - invalid sampleRate[%u] = %d", sfreq, sampleRate);
114 mSampleRate = (uint32_t) sampleRate;
|
| /external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/ |
| AACTrackImpl.java | 67 int samplerate; field in class:AACTrackImpl
105 double packetsPerSecond = (double)samplerate / 1024.0;
136 audioSampleEntry.setSampleRate(samplerate);
158 audioSpecificConfig.setSamplingFrequencyIndex(samplingFrequencyIndexMap.get(samplerate));
172 trackMetaData.setTimescale(samplerate); // Audio tracks always use samplerate as timescale
232 samplerate = samplingFrequencyIndexMap.get(brb.readBits(4));
285 "samplerate=" + samplerate +
|
| /external/sonic/ |
| wave.c | 21 int sampleRate;
169 int sampleRate)
181 writeInt(file, sampleRate); /* 24 - samples per second (numbers per second) */
182 writeInt(file, sampleRate * 2); /* 28 - bytes per second */
210 file->sampleRate = readInt(file); /* 24 - samples per second (numbers per second) */
242 int *sampleRate,
259 *sampleRate = file->sampleRate;
267 int sampleRate,
279 file->sampleRate = sampleRate
[all...] |
| /frameworks/av/media/libaudiohal/ |
| StreamPowerLog.h | 42 void init(uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format) {
50 (long long)sampleRate * kPowerLogSamplingIntervalMs / 1000;
52 sampleRate,
|
| /frameworks/av/media/libnbaio/ |
| AudioStreamInSource.cpp | 50 uint32_t sampleRate;
52 result = mStream->getAudioProperties(&sampleRate, &channelMask, &streamFormat);
54 mFormat = Format_from_SR_C(sampleRate,
|
| AudioStreamOutSink.cpp | 48 uint32_t sampleRate;
50 result = mStream->getAudioProperties(&sampleRate, &channelMask, &streamFormat);
52 mFormat = Format_from_SR_C(sampleRate,
|
| /packages/apps/Test/connectivity/PMC/src/com/android/pmc/ |
| A2dpReceiver.java | 197 int sampleRate = BluetoothCodecConfig.SAMPLE_RATE_NONE;
273 if (!extras.containsKey("SampleRate")) {
277 tmpStr = extras.getString("SampleRate");
279 sampleRate = Integer.valueOf(tmpStr);
320 || sampleRate == BluetoothCodecConfig.SAMPLE_RATE_NONE
330 if (!setCodecValue(codecType, sampleRate, bitsPerSample, channelMode,
434 * @param sampleRate - Sample Rate
441 private boolean setCodecValue(int codecType, int sampleRate, int bitsPerSample,
444 Log.d(TAG, "SetCodecValue: Codec Type: " + codecType + " sampleRate: " + sampleRate
[all...] |
| /cts/apps/CtsVerifier/src/com/android/cts/verifier/audio/ |
| WavAnalyzer.java | 13 private final int sampleRate; // Recording sampling rate.
25 public WavAnalyzer(byte[] byteData, int sampleRate, Listener listener) {
27 this.sampleRate = sampleRate;
135 + Util.toLength(Common.PREFIX_LENGTH_S + Common.PAUSE_AFTER_PREFIX_DURATION_S, sampleRate);
137 listener.sendMessage("Prefix starts at " + (double) dataStartI / sampleRate + " s \n");
138 if (dataStartI > Math.round(sampleRate * (Common.PREFIX_LENGTH_S
161 -2.0 * Math.PI * freq / sampleRate).exp();
189 + Util.toLength(i * (Common.PIP_DURATION_S + Common.PAUSE_DURATION_S), sampleRate);
198 -2.0 * Math.PI * Common.FREQUENCIES[i] / sampleRate).exp()
[all...] |
| /cts/tests/tests/nativemedia/aaudio/src/ |
| utils.h | 28 int32_t sampleRate;
62 StreamBuilderHelper(aaudio_direction_t direction, int32_t sampleRate,
|
| /frameworks/av/cmds/stagefright/ |
| SineSource.cpp | 12 SineSource::SineSource(int32_t sampleRate, int32_t numChannels)
14 mSampleRate(sampleRate),
|
| /frameworks/av/media/libaaudio/src/core/ |
| AAudioStreamParameters.h | 44 void setSampleRate(int32_t sampleRate) {
45 mSampleRate = sampleRate;
|
