/frameworks/av/media/libaaudio/src/binding/ |
H A D | AAudioStreamConfiguration.h | 81 void setBufferCapacity(int32_t frames) { argument 82 mBufferCapacity = frames;
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/frameworks/av/media/libaaudio/src/client/ |
H A D | AudioStreamInternalCapture.cpp | 70 //ALOGD("AudioStreamInternalCapture::processDataNow() - tried to read %d frames, read %d", 166 int64_t frames = local 170 if (frames < mLastFramesWritten) { 171 frames = mLastFramesWritten; 173 mLastFramesWritten = frames; 175 //ALOGD("AudioStreamInternalCapture::getFramesWritten() returns %lld", (long long)frames); 176 return frames; 181 int64_t frames = mAudioEndpoint.getDataWriteCounter() local 183 //ALOGD("AudioStreamInternalCapture::getFramesRead() returns %lld", (long long)frames); 184 return frames; [all...] |
/frameworks/av/media/libaaudio/src/core/ |
H A D | AAudioAudio.cpp | 182 int32_t frames) 185 streamBuilder->setBufferCapacity(frames); 207 int32_t frames) 210 streamBuilder->setFramesPerDataCallback(frames); 181 AAudioStreamBuilder_setBufferCapacityInFrames(AAudioStreamBuilder* builder, int32_t frames) argument 206 AAudioStreamBuilder_setFramesPerDataCallback(AAudioStreamBuilder* builder, int32_t frames) argument
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H A D | AudioStream.h | 235 virtual int64_t incrementFramesWritten(int32_t frames) { argument 236 return mFramesWritten.increment(frames); 239 virtual int64_t incrementFramesRead(int32_t frames) { argument 240 return mFramesRead.increment(frames); 311 int32_t mFramesPerDataCallback = AAUDIO_UNSPECIFIED; // frames
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H A D | AudioStreamBuilder.h | 98 AudioStreamBuilder* setBufferCapacity(int32_t frames) { argument 99 mBufferCapacity = frames; 181 int32_t mFramesPerDataCallback = AAUDIO_UNSPECIFIED; // frames
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/frameworks/av/media/libaaudio/src/fifo/ |
H A D | FifoBuffer.cpp | 78 int32_t FifoBuffer::convertFramesToBytes(fifo_frames_t frames) { argument 79 return frames * mBytesPerFrame;
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/frameworks/av/media/libaaudio/src/utility/ |
H A D | LinearRamp.cpp | 31 bool LinearRamp::nextSegment(int32_t frames, float *levelFrom, float *levelTo) { argument 36 if (frames >= mRemaining) { 41 level = mLevelFrom + (frames * (mLevelTo - mLevelFrom) / mRemaining); 42 mRemaining -= frames;
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H A D | LinearRamp.h | 37 void setLengthInFrames(int32_t frames) { argument 38 mLengthInFrames = frames; 77 * @param frames number of frames in the segment 82 bool nextSegment(int32_t frames, float *levelFrom, float *levelTo);
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/frameworks/av/media/libaudioclient/ |
H A D | AudioTrack.cpp | 55 static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed) argument 57 return ((double)frames * 1000000000) / ((double)sampleRate * speed); 623 // force refresh of remaining frames by processAudioBuffer() as last 686 "stop() called with %u frames delivered", mReleased.value()); 1912 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END 2179 // There could be enough non-contiguous frames available to satisfy the remaining request 2243 // take the frames that will be lost by track recreation into account in saved position 2297 // This is the sole place to read server consumed frames 2306 // unless the server has more than 2^31 frames in its buffer, 2409 // apply server offset (frames flushe 2481 const int64_t frames = local [all...] |
/frameworks/av/media/libaudiohal/ |
H A D | StreamHalLocal.cpp | 221 status_t StreamOutHalLocal::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) { argument 223 return mStream->get_presentation_position(mStream, frames, timestamp); 281 status_t StreamInHalLocal::getCapturePosition(int64_t *frames, int64_t *time) { argument 283 return mStream->get_capture_position(mStream, frames, time);
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/frameworks/av/media/libaudioprocessing/ |
H A D | BufferProviders.cpp | 276 void DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames) argument 279 mInBuffer->setFrameCount(frames); 280 mInBuffer->update(mInFrameSize * frames); 281 mOutBuffer->setFrameCount(frames); 286 mOutBuffer->update(mOutFrameSize * frames); 291 mOutBuffer->commit(mOutFrameSize * frames); 353 void RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames) argument 356 src, mInputChannels, mIdxAry, mSampleSize, frames); 374 void ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames) argument 376 memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCoun [all...] |
H A D | RecordBufferConverter.cpp | 78 AudioBufferProvider *provider, size_t frames) 90 for (size_t i = frames; i > 0; ) { 94 frames -= i; // cannot fill request. 109 if (mBufFrameSize != 0 && mBufFrames < frames) { 111 mBufFrames = frames; 115 memset(mBuf, 0, frames * mBufFrameSize); 116 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 118 convertResampler(dst, mBuf, frames); 120 return frames; 77 convert(void *dst, AudioBufferProvider *provider, size_t frames) argument 219 convertNoResampler( void *dst, const void *src, size_t frames) argument 259 convertResampler( void *dst, void *src, size_t frames) argument [all...] |
/frameworks/av/media/libaudioprocessing/tests/ |
H A D | test-mixer.cpp | 48 fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n"); 55 uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) { 61 info.frames = 0; 65 printf("saving file:%s channels:%u samplerate:%u frames:%zu\n", 66 filename, info.channels, info.samplerate, frames); 73 (void) sf_writef_float(sf, (float*)buffer, frames); 75 (void) sf_writef_short(sf, (short*)buffer, frames); 210 // calculate the number of output frames 312 outputFrames = i; // reset output frames to the data actually produced. 54 writeFile(const char *filename, const void *buffer, uint32_t sampleRate, uint32_t channels, size_t frames, bool isBufferFloat) argument
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H A D | test-resampler.cpp | 60 fprintf(stderr," -O # frames output per call to resample() in CSV format\n"); 61 fprintf(stderr," -P # frames provided per call to resample() in CSV format\n"); 218 input_size = info.frames * info.channels * sizeof(short); 220 (void) sf_readf_short(sf, (short *) input_vaddr, info.frames); 262 const size_t mNumFrames; // total frames 265 size_t mUnrel; // number of frames not yet released 266 const Vector<int> mPvalues; // number of frames provided per call 269 Provider(const void* addr, size_t frames, size_t frameSize, const Vector<int>& Pvalues) argument 271 mNumFrames(frames), 291 printf("getNextBuffer() requested %zu frames ou [all...] |
H A D | test_utils.h | 101 TestProvider(void* addr, size_t frames, size_t frameSize, argument 104 mNumFrames(frames), 138 ALOGV("getNextBuffer() requested %zu frames out of %zu frames available" 139 " and returned %zu frames", 154 ALOGE("releaseBuffer() released %zu frames but only %zu available " 160 ALOGV("releaseBuffer() released %zu frames out of %zu frames available " 182 size_t mNumFrames; // total frames 185 size_t mUnrel; // number of frames no 193 createSine(void *vbuffer, size_t frames, size_t channels, double sampleRate, double freq) argument 217 createChirp(void *vbuffer, size_t frames, size_t channels, double sampleRate, double minfreq, double maxfreq) argument 290 createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames) argument [all...] |
/frameworks/av/media/libstagefright/ |
H A D | OggExtractor.cpp | 617 uint32_t frames = getNumSamplesInPacket(*out); local 618 mCurGranulePosition += frames;
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H A D | XINGSeeker.cpp | 139 int32_t frames = U32_AT(buffer); local 144 if (frames) { 145 seeker->mDurationUs = (int64_t)frames * samples_per_frame * 1000000LL / sampling_rate;
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/frameworks/av/media/libstagefright/mpeg2ts/ |
H A D | ESQueue.cpp | 826 Vector<ADTSPosition> frames; local 828 // The idea here is consume all AAC frames starting at offsets before 920 frames.push(frame); 929 for (size_t frameId = 0; frameId < frames.size(); frameId++) { 930 const ADTSPosition &frame = frames.itemAt(frameId);
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/frameworks/av/media/libstagefright/omx/tests/ |
H A D | FrameDropper_test.cpp | 99 void RunTest(const TestFrame* frames, size_t size) { argument 102 int64_t testTimeUs = frames[i].timeUs + jitter; 104 (long long)frames[i].timeUs, (long long)testTimeUs, jitter); 105 EXPECT_EQ(frames[i].shouldDrop, mFrameDropper->shouldDrop(testTimeUs));
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/frameworks/av/media/libstagefright/rtsp/ |
H A D | ARTPAssembler.cpp | 76 const List<sp<ABuffer> > &frames) { 78 for (List<sp<ABuffer> >::const_iterator it = frames.begin(); 79 it != frames.end(); ++it) { 86 for (List<sp<ABuffer> >::const_iterator it = frames.begin(); 87 it != frames.end(); ++it) { 116 CopyTimes(accessUnit, *frames.begin()); 72 MakeADTSCompoundFromAACFrames( unsigned profile, unsigned samplingFreqIndex, unsigned channelConfig, const List<sp<ABuffer> > &frames) argument
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/frameworks/av/media/libstagefright/webm/ |
H A D | WebmFrameThread.cpp | 100 // frames: 101 // sequence of input audio/video frames received from the source. 105 // frame since frames are ordered by timestamp. 113 List<const sp<WebmFrame> >& frames, 116 CHECK(!frames.empty() && children.empty()); 118 const sp<WebmFrame> f = *(frames.begin()); 135 // Write out (possibly multiple) webm cluster(s) from frames split on video key frames. 139 void WebmFrameSinkThread::flushFrames(List<const sp<WebmFrame> >& frames, bool last) { argument 140 if (frames 112 initCluster( List<const sp<WebmFrame> >& frames, uint64_t& clusterTimecodeL, List<sp<WebmElement> >& children) argument [all...] |
/frameworks/av/services/audioflinger/ |
H A D | AudioFlinger.cpp | 1313 size_t frames; local 1318 status_t result = dev->getInputBufferSize(&config, &frames); 1319 if (result == OK && frames != 0) { 1332 break; // retries failed, break out of loop with frames == 0. 1336 if (frames > 0 && config.sample_rate != sampleRate) { 1337 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1339 return frames; // may be converted to bytes at the Java level. 3351 #define TEE_SINK_READ 1024 // frames per I/O operation
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H A D | AudioStreamOut.cpp | 54 status_t AudioStreamOut::getRenderPosition(uint64_t *frames) argument 75 *frames = mRenderPosition / mRateMultiplier; 81 status_t AudioStreamOut::getRenderPosition(uint32_t *frames) argument 86 *frames = (uint32_t)position64; 91 status_t AudioStreamOut::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) argument 103 // Adjust for standby using HAL rate frames. 104 // Only apply this correction if the HAL is getting PCM frames. 109 *frames = adjustedPosition / mRateMultiplier; 112 *frames = halPosition;
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H A D | Threads.cpp | 134 // minimum normal sink buffer size, expressed in milliseconds rather than frames 1803 dprintf(fd, " Suspended frames: %lld\n", (long long)mSuspendedFrames); 1805 dprintf(fd, " PipeSink frames written: %lld\n", (long long)mPipeSink->framesWritten()); 2342 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2393 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2413 // round up to nearest 16 frames to satisfy AudioMixer 2417 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2426 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames 2487 uint32_t frames; local [all...] |
H A D | Tracks.cpp | 630 // framesReady() may return an approximation of the number of frames if called 636 // Static tracks return zero frames immediately upon stopping (for FastTracks). 651 // The server side frames are already translated to client frames. 1048 // a track is considered presented when the total number of frames written to audio HAL 1049 // corresponds to the number of frames written when presentationComplete() is called for the 1052 // to detect when all frames have been played. In this case framesWritten isn't 1221 // Set correction for flushed frames that are not accounted for in released. 1289 bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames) argument 1294 inBuffer.frameCount = frames; [all...] |