Searched defs:frames (Results 1 - 25 of 35) sorted by path

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/frameworks/av/media/libaaudio/src/binding/
H A DAAudioStreamConfiguration.h81 void setBufferCapacity(int32_t frames) { argument
82 mBufferCapacity = frames;
/frameworks/av/media/libaaudio/src/client/
H A DAudioStreamInternalCapture.cpp70 //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;
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/frameworks/av/media/libaaudio/src/core/
H A DAAudioAudio.cpp182 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
H A DAudioStream.h235 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
H A DAudioStreamBuilder.h98 AudioStreamBuilder* setBufferCapacity(int32_t frames) { argument
99 mBufferCapacity = frames;
181 int32_t mFramesPerDataCallback = AAUDIO_UNSPECIFIED; // frames
/frameworks/av/media/libaaudio/src/fifo/
H A DFifoBuffer.cpp78 int32_t FifoBuffer::convertFramesToBytes(fifo_frames_t frames) { argument
79 return frames * mBytesPerFrame;
/frameworks/av/media/libaaudio/src/utility/
H A DLinearRamp.cpp31 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;
H A DLinearRamp.h37 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);
/frameworks/av/media/libaudioclient/
H A DAudioTrack.cpp55 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
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/frameworks/av/media/libaudiohal/
H A DStreamHalLocal.cpp221 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);
/frameworks/av/media/libaudioprocessing/
H A DBufferProviders.cpp276 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
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H A DRecordBufferConverter.cpp78 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
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/frameworks/av/media/libaudioprocessing/tests/
H A Dtest-mixer.cpp48 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
H A Dtest-resampler.cpp60 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
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H A Dtest_utils.h101 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
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/frameworks/av/media/libstagefright/
H A DOggExtractor.cpp617 uint32_t frames = getNumSamplesInPacket(*out); local
618 mCurGranulePosition += frames;
H A DXINGSeeker.cpp139 int32_t frames = U32_AT(buffer); local
144 if (frames) {
145 seeker->mDurationUs = (int64_t)frames * samples_per_frame * 1000000LL / sampling_rate;
/frameworks/av/media/libstagefright/mpeg2ts/
H A DESQueue.cpp826 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);
/frameworks/av/media/libstagefright/omx/tests/
H A DFrameDropper_test.cpp99 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));
/frameworks/av/media/libstagefright/rtsp/
H A DARTPAssembler.cpp76 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
/frameworks/av/media/libstagefright/webm/
H A DWebmFrameThread.cpp100 // 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
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/frameworks/av/services/audioflinger/
H A DAudioFlinger.cpp1313 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
H A DAudioStreamOut.cpp54 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;
H A DThreads.cpp134 // 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
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H A DTracks.cpp630 // 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;
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