| /frameworks/base/core/java/android/bluetooth/ |
| D | BluetoothCodecConfig.java | 67 private final int mSampleRate; field in BluetoothCodecConfig
82 mSampleRate = sampleRate; in BluetoothCodecConfig()
97 other.mSampleRate == mSampleRate && in equals()
110 return Objects.hash(mCodecType, mCodecPriority, mSampleRate, in hashCode()
122 return (mSampleRate != SAMPLE_RATE_NONE) && in isValid()
146 if (mSampleRate == SAMPLE_RATE_NONE) { in toString()
149 if ((mSampleRate & SAMPLE_RATE_44100) != 0) { in toString()
152 if ((mSampleRate & SAMPLE_RATE_48000) != 0) { in toString()
155 if ((mSampleRate & SAMPLE_RATE_88200) != 0) { in toString()
158 if ((mSampleRate & SAMPLE_RATE_96000) != 0) { in toString()
[all …]
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| D | BluetoothAudioConfig.java | 31 private final int mSampleRate; field in BluetoothAudioConfig
36 mSampleRate = sampleRate; in BluetoothAudioConfig()
45 return (bac.mSampleRate == mSampleRate && in equals()
54 return mSampleRate | (mChannelConfig << 24) | (mAudioFormat << 28); in hashCode()
59 return "{mSampleRate:" + mSampleRate + ",mChannelConfig:" + mChannelConfig in toString()
81 out.writeInt(mSampleRate); in writeToParcel()
91 return mSampleRate; in getSampleRate()
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| /frameworks/av/media/libaaudio/src/core/ |
| D | AAudioStreamParameters.cpp | 39 mSampleRate = other.mSampleRate; in copyFrom()
80 if (mSampleRate != AAUDIO_UNSPECIFIED in validate()
81 && (mSampleRate < SAMPLE_RATE_HZ_MIN || mSampleRate > SAMPLE_RATE_HZ_MAX)) { in validate()
82 ALOGE("AAudioStreamParameters: sampleRate out of range = %d", mSampleRate); in validate()
106 ALOGD("AAudioStreamParameters mSampleRate = %d", mSampleRate); in dump()
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| D | AAudioStreamParameters.h | 41 return mSampleRate; in getSampleRate()
45 mSampleRate = sampleRate; in setSampleRate()
107 int32_t mSampleRate = AAUDIO_UNSPECIFIED; variable
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| /frameworks/av/services/audioflinger/ |
| D | FastCapture.cpp | 36 mReadBuffer(NULL), mReadBufferState(-1), mFormat(Format_Invalid), mSampleRate(0), in FastCapture()
103 mSampleRate = 0; in onStateChange()
106 mSampleRate = Format_sampleRate(mFormat); in onStateChange()
112 dumpState->mSampleRate = mSampleRate; in onStateChange()
132 if (frameCount > 0 && mSampleRate > 0) { in onStateChange()
139 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 in onStateChange()
140 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 in onStateChange()
141 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 in onStateChange()
142 mForceNs = (frameCount * 950000000LL) / mSampleRate; // 0.95 in onStateChange()
143 mWarmupNsMin = (frameCount * 750000000LL) / mSampleRate; // 0.75 in onStateChange()
[all …]
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| D | FastCaptureDumpState.cpp | 28 mReadSequence(0), mFramesRead(0), mReadErrors(0), mSampleRate(0), mFrameCount(0) in FastCaptureDumpState()
44 double periodSec = (double) mFrameCount / mSampleRate; in dump()
49 mReadErrors, mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles, in dump()
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| D | FastMixer.cpp | 64 mSampleRate(0), in FastMixer()
163 mSampleRate = 0; in onStateChange()
168 mSampleRate = Format_sampleRate(mFormat); in onStateChange()
176 dumpState->mSampleRate = mSampleRate; in onStateChange()
187 if (frameCount > 0 && mSampleRate > 0) { in onStateChange()
195 mMixer = new AudioMixer(frameCount, mSampleRate, FastMixerState::sMaxFastTracks); in onStateChange()
207 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 in onStateChange()
208 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 in onStateChange()
209 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 in onStateChange()
210 mForceNs = (frameCount * 950000000LL) / mSampleRate; // 0.95 in onStateChange()
[all …]
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| /frameworks/base/media/java/android/media/ |
| D | AudioFormat.java | 632 mSampleRate = sampleRate; in AudioFormat()
653 private int mSampleRate; field in AudioFormat
678 return mSampleRate; in getSampleRate()
735 getChannelCount(), mSampleRate, toLogFriendlyEncoding(mEncoding)); in toLogFriendlyString()
755 private int mSampleRate = SAMPLE_RATE_UNSPECIFIED; field in AudioFormat.Builder
772 mSampleRate = af.mSampleRate; in Builder()
788 af.mSampleRate = mSampleRate; in build()
944 mSampleRate = sampleRate; in setSampleRate()
963 && (mSampleRate != that.mSampleRate)) in equals()
972 return Objects.hash(mPropertySetMask, mSampleRate, mEncoding, mChannelMask, in hashCode()
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| /frameworks/av/media/libaudioprocessing/ |
| D | AudioResamplerDyn.cpp | 187 mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better in AudioResamplerDyn()
303 !isClose(inSampleRate, oldSampleRate, mFilterSampleRate, mSampleRate)) { in setSampleRate()
323 if (inSampleRate >= mSampleRate * 4) { in setSampleRate()
325 } else if (inSampleRate >= mSampleRate * 2) { in setSampleRate()
334 if (inSampleRate >= mSampleRate * 4) { in setSampleRate()
336 } else if (inSampleRate >= mSampleRate * 2) { in setSampleRate()
341 if (inSampleRate <= mSampleRate) { in setSampleRate()
351 if (inSampleRate >= mSampleRate * 4) { in setSampleRate()
353 } else if (inSampleRate >= mSampleRate * 2) { in setSampleRate()
358 if (inSampleRate <= mSampleRate) { in setSampleRate()
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| /frameworks/av/media/libstagefright/ |
| D | WAVExtractor.cpp | 84 int32_t mSampleRate; member
214 mSampleRate = U32_LE_AT(&formatSpec[4]); in init()
216 if (mSampleRate == 0) { in init()
313 mTrackMeta->setInt32(kKeySampleRate, mSampleRate); in init()
329 if (!mSampleRate) in init()
333 1000000LL * num_samples / mSampleRate; in init()
359 mSampleRate(0), in WAVSource()
366 CHECK(mMeta->findInt32(kKeySampleRate, &mSampleRate)); in WAVSource()
432 int64_t samplenumber = (seekTimeUs * mSampleRate) / 1000000; in read()
436 pos = (seekTimeUs * mSampleRate) / 1000000 * mNumChannels * (mBitsPerSample >> 3); in read()
[all …]
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| D | AudioSource.cpp | 57 mSampleRate(sampleRate), in AudioSource()
200 meta->setInt32(kKeySampleRate, mSampleRate); in getFormat()
272 … ((int64_t)kAutoRampDurationUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting in read()
275 … ((int64_t)kAutoRampStartUs * mSampleRate + 500000LL) / 1000000LL; //Need type casting in read()
288 if (mSampleRate != mOutSampleRate) { in read()
289 timeUs *= (int64_t)mSampleRate / (int64_t)mOutSampleRate; in read()
334 mNumFramesReceived + mNumFramesLost) * usPerSec / mSampleRate; in dataCallback()
431 (mSampleRate >> 1)) / mSampleRate; in queueInputBuffer_l()
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| D | AudioPlayer.cpp | 43 mSampleRate(0), in AudioPlayer()
126 success = format->findInt32(kKeySampleRate, &mSampleRate); in start()
182 offloadInfo.sample_rate = mSampleRate; in start()
192 mSampleRate, numChannels, channelMask, audioFormat, in start()
240 AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT, audioMask, in start()
569 / mSampleRate; in fillBuffer()
629 mSampleRate = sampleRate; in getOutputPlayPositionUs_l()
633 if (mSampleRate != 0) { in getOutputPlayPositionUs_l()
634 playedUs = (static_cast<int64_t>(playedSamples) * 1000000 ) / mSampleRate; in getOutputPlayPositionUs_l()
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| D | AACWriter.cpp | 49 mSampleRate(-1), in AACWriter()
87 CHECK(meta->findInt32(kKeySampleRate, &mSampleRate)); in addSource()
117 mFrameDurationUs = (kSamplesPerFrame * 1000000LL + (mSampleRate >> 1)) in start()
118 / mSampleRate; in start()
253 CHECK(getSampleRateTableIndex(mSampleRate, &kSampleFreqIndex)); in writeAdtsHeader()
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| /frameworks/av/media/libaaudio/src/client/ |
| D | IsochronousClockModel.cpp | 33 , mSampleRate(48000) in IsochronousClockModel()
128 mSampleRate = sampleRate; in setSampleRate()
143 return (AAUDIO_NANOS_PER_SECOND * framesDelta) / mSampleRate; in convertDeltaPositionToTime()
147 return (mSampleRate * nanosDelta) / AAUDIO_NANOS_PER_SECOND; in convertDeltaTimeToPosition()
185 ALOGD("IsochronousClockModel::mSampleRate = %6d", mSampleRate); in dump()
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| D | IsochronousClockModel.h | 51 return mSampleRate; in getSampleRate()
105 int32_t mSampleRate; variable
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| /frameworks/av/cmds/stagefright/ |
| D | SineSource.cpp | 14 mSampleRate(sampleRate), in SineSource()
54 meta->setInt32(kKeySampleRate, mSampleRate); in getFormat()
77 const double k = kFrequency / mSampleRate * (2.0 * M_PI); in read()
92 kKeyTime, ((int64_t)mPhase * 1000000) / mSampleRate); in read()
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| /frameworks/opt/net/voip/src/jni/rtp/ |
| D | AudioGroup.cpp | 127 int mSampleRate; member in __anon74d586cf0111::AudioStream
180 mSampleRate = sampleRate / 1000; in set()
182 mInterval = mSampleCount / mSampleRate; in set()
185 for (mBufferMask = 8; mBufferMask < mSampleRate; mBufferMask <<= 1); in set()
224 (codec ? codec->name : "RAW"), mSampleRate, mInterval, mMode); in set()
252 head *= mSampleRate; in mix()
253 tail *= mSampleRate; in mix()
255 if (sampleRate == mSampleRate) { in mix()
286 if (duration >= 0 && duration < mSampleRate * DTMF_PERIOD) { in encode()
294 if (duration >= mSampleRate * DTMF_PERIOD) { in encode()
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| /frameworks/av/media/libnbaio/ |
| D | NBAIO.cpp | 37 return format.mSampleRate; in Format_sampleRate()
55 ret.mSampleRate = sampleRate; in Format_from_SR_C()
160 return format.mSampleRate != 0 && format.mChannelCount != 0 && in Format_isValid()
166 return format1.mSampleRate == format2.mSampleRate && in Format_isEqual()
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| /frameworks/av/media/libstagefright/codecs/aacenc/ |
| D | SoftAACEncoder.cpp | 50 mSampleRate(44100), in SoftAACEncoder()
197 aacParams->nSampleRate = mSampleRate; in internalGetParameter()
225 pcmParams->nSamplingRate = mSampleRate; in internalGetParameter()
294 mSampleRate = aacParams->nSampleRate; in internalSetParameter()
317 mSampleRate = pcmParams->nSamplingRate; in internalSetParameter()
337 mSampleRate, mNumChannels, mBitRate); in setAudioParams()
347 params.sampleRate = mSampleRate; in setAudioParams()
384 status_t err = getSampleRateTableIndex(mSampleRate, index); in setAudioSpecificConfigData()
386 ALOGE("Unsupported sample rate (%u Hz)", mSampleRate); in setAudioSpecificConfigData()
482 (copy * 1000000ll / mSampleRate) in onQueueFilled()
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| /frameworks/av/include/media/ |
| D | AudioResampler.h | 103 const int32_t mSampleRate; variable
144 + (mSampleRate - 1))/mSampleRate; in getInFrameCountRequired()
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| /frameworks/av/media/libaudioprocessing/include/media/ |
| D | AudioResampler.h | 103 const int32_t mSampleRate; variable
144 + (mSampleRate - 1))/mSampleRate; in getInFrameCountRequired()
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| /frameworks/av/media/libaudioprocessing/tests/ |
| D | test_utils.h | 244 : mSampleRate(0),
259 createChirp<T>(mAddr, mNumFrames, mChannels, mSampleRate, minfreq, maxfreq);
267 createSine<T>(mAddr, mNumFrames, mChannels, mSampleRate, freq);
297 mSampleRate = sampleRate;
301 return mSampleRate;
309 uint32_t mSampleRate;
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| /frameworks/av/media/libstagefright/codecs/mp3dec/test/ |
| D | mp3reader.h | 37 uint32_t getSampleRate() { return mSampleRate;} in getSampleRate()
45 uint32_t mSampleRate; variable
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| /frameworks/av/media/libaaudio/examples/utils/ |
| D | AAudioArgsParser.h | 46 return mSampleRate; in getSampleRate()
50 mSampleRate = sampleRate; in setSampleRate()
108 AAudioStreamBuilder_setSampleRate(builder, mSampleRate); in applyParameters()
118 int32_t mSampleRate = AAUDIO_UNSPECIFIED; variable
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| /frameworks/av/media/libstagefright/codecs/flac/enc/ |
| D | SoftFlacEncoder.cpp | 55 mSampleRate(44100), in SoftFlacEncoder()
203 pcmParams->nSamplingRate = mSampleRate; in internalGetParameter()
222 flacParams->nSampleRate = mSampleRate; in internalGetParameter()
276 mSampleRate = pcmParams->nSamplingRate; in internalSetParameter()
277 ALOGV("will encode %d channels at %dHz", mNumChannels, mSampleRate); in internalSetParameter()
497 mNumChannels, mSampleRate); in configureEncoder()
506 ok = ok && FLAC__stream_encoder_set_sample_rate(mFlacStreamEncoder, mSampleRate); in configureEncoder()
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