/frameworks/av/media/libaudioprocessing/ |
D | RecordBufferConverter.cpp | 78 AudioBufferProvider *provider, size_t frames) in convert() argument 90 for (size_t i = frames; i > 0; ) { in convert() 94 frames -= i; // cannot fill request. in convert() 109 if (mBufFrameSize != 0 && mBufFrames < frames) { in convert() 111 mBufFrames = frames; in convert() 115 memset(mBuf, 0, frames * mBufFrameSize); in convert() 116 frames = mResampler->resample((int32_t*)mBuf, frames, provider); in convert() 118 convertResampler(dst, mBuf, frames); in convert() 120 return frames; in convert() 220 void *dst, const void *src, size_t frames) in convertNoResampler() argument [all …]
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D | BufferProviders.cpp | 276 void DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames) in copyFrames() argument 279 mInBuffer->setFrameCount(frames); in copyFrames() 280 mInBuffer->update(mInFrameSize * frames); in copyFrames() 281 mOutBuffer->setFrameCount(frames); in copyFrames() 286 mOutBuffer->update(mOutFrameSize * frames); in copyFrames() 291 mOutBuffer->commit(mOutFrameSize * frames); in copyFrames() 353 void RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames) in copyFrames() argument 356 src, mInputChannels, mIdxAry, mSampleSize, frames); in copyFrames() 374 void ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames) in copyFrames() argument 376 memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCount); in copyFrames()
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/frameworks/av/media/libstagefright/webm/ |
D | WebmFrameThread.cpp | 113 List<const sp<WebmFrame> >& frames, in initCluster() 116 CHECK(!frames.empty() && children.empty()); in initCluster() 118 const sp<WebmFrame> f = *(frames.begin()); in initCluster() 139 void WebmFrameSinkThread::flushFrames(List<const sp<WebmFrame> >& frames, bool last) { in flushFrames() argument 140 if (frames.empty()) { in flushFrames() 146 initCluster(frames, clusterTimecodeL, children); in flushFrames() 150 size_t n = frames.size(); in flushFrames() 163 const sp<WebmFrame> f = *(frames.begin()); in flushFrames() 170 initCluster(frames, clusterTimecodeL, children); in flushFrames() 173 frames.erase(frames.begin()); in flushFrames() [all …]
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/frameworks/av/services/mediaanalytics/ |
D | MetricsSummarizerPlayer.cpp | 73 int64_t frames = 0; in mergeRecord() local 74 if (item.getInt64("android.media.mediaplayer.frames", &frames)) in mergeRecord() 75 ALOGV("found framess of %" PRId64, frames); in mergeRecord() 76 if (frames >= 0) { in mergeRecord() 77 summation.addInt64("android.media.mediaplayer.frames",frames); in mergeRecord()
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/frameworks/av/media/libaaudio/src/utility/ |
D | LinearRamp.cpp | 31 bool LinearRamp::nextSegment(int32_t frames, float *levelFrom, float *levelTo) { in nextSegment() argument 36 if (frames >= mRemaining) { in nextSegment() 41 level = mLevelFrom + (frames * (mLevelTo - mLevelFrom) / mRemaining); in nextSegment() 42 mRemaining -= frames; in nextSegment()
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D | LinearRamp.h | 37 void setLengthInFrames(int32_t frames) { in setLengthInFrames() argument 38 mLengthInFrames = frames; in setLengthInFrames() 82 bool nextSegment(int32_t frames, float *levelFrom, float *levelTo);
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/frameworks/av/media/libaudioprocessing/tests/ |
D | test_utils.h | 101 TestProvider(void* addr, size_t frames, size_t frameSize, 104 mNumFrames(frames), 193 static void createSine(void *vbuffer, size_t frames, 198 for (size_t i = 0; i < frames; ++i) { 217 static void createChirp(void *vbuffer, size_t frames, 223 double k = (maxfreq - minfreq) / (2. * tscale * frames); 224 for (size_t i = 0; i < frames; ++i) { 280 createBufferByFrames<T>(info.channels, info.samplerate, info.frames); 290 void createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames) 292 mNumFrames = frames;
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/frameworks/av/media/libstagefright/rtsp/ |
D | ARTPAssembler.cpp | 76 const List<sp<ABuffer> > &frames) { in MakeADTSCompoundFromAACFrames() 78 for (List<sp<ABuffer> >::const_iterator it = frames.begin(); in MakeADTSCompoundFromAACFrames() 79 it != frames.end(); ++it) { in MakeADTSCompoundFromAACFrames() 86 for (List<sp<ABuffer> >::const_iterator it = frames.begin(); in MakeADTSCompoundFromAACFrames() 87 it != frames.end(); ++it) { in MakeADTSCompoundFromAACFrames() 116 CopyTimes(accessUnit, *frames.begin()); in MakeADTSCompoundFromAACFrames()
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D | ARTPAssembler.h | 53 const List<sp<ABuffer> > &frames); 56 const List<sp<ABuffer> > &frames);
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/frameworks/av/services/audioflinger/ |
D | AudioStreamOut.cpp | 54 status_t AudioStreamOut::getRenderPosition(uint64_t *frames) in getRenderPosition() argument 75 *frames = mRenderPosition / mRateMultiplier; in getRenderPosition() 81 status_t AudioStreamOut::getRenderPosition(uint32_t *frames) in getRenderPosition() argument 86 *frames = (uint32_t)position64; in getRenderPosition() 91 status_t AudioStreamOut::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) in getPresentationPosition() argument 109 *frames = adjustedPosition / mRateMultiplier; in getPresentationPosition() 112 *frames = halPosition; in getPresentationPosition()
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D | AudioStreamOut.h | 57 status_t getRenderPosition(uint32_t *frames); 59 virtual status_t getRenderPosition(uint64_t *frames); 61 virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
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/frameworks/av/media/libaaudio/src/client/ |
D | AudioStreamInternalCapture.cpp | 166 int64_t frames = in getFramesWritten() local 170 if (frames < mLastFramesWritten) { in getFramesWritten() 171 frames = mLastFramesWritten; in getFramesWritten() 173 mLastFramesWritten = frames; in getFramesWritten() 176 return frames; in getFramesWritten() 181 int64_t frames = mAudioEndpoint.getDataWriteCounter() in getFramesRead() local 184 return frames; in getFramesRead()
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/frameworks/base/core/java/android/speech/tts/ |
D | SynthesisPlaybackQueueItem.java | 189 public final int frames; field in SynthesisPlaybackQueueItem.ProgressMarker 195 public ProgressMarker(int frames, int start, int end) { in ProgressMarker() argument 196 this.frames = frames; in ProgressMarker() 208 int markerInFrames = marker.frames == 0 ? 1 : marker.frames; in updateMarker() 227 getDispatcher().dispatchOnRangeStart(marker.start, marker.end, marker.frames); in onMarkerReached()
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/frameworks/av/media/libstagefright/omx/tests/ |
D | FrameDropper_test.cpp | 99 void RunTest(const TestFrame* frames, size_t size) { in RunTest() argument 102 int64_t testTimeUs = frames[i].timeUs + jitter; in RunTest() 104 (long long)frames[i].timeUs, (long long)testTimeUs, jitter); in RunTest() 105 EXPECT_EQ(frames[i].shouldDrop, mFrameDropper->shouldDrop(testTimeUs)); in RunTest()
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/frameworks/wilhelm/tools/permute/ |
D | permute.c | 202 switch (sfinfo_in.frames) { in permute() 205 fprintf(stderr, "%s: unsupported frames %d\n", path_in, (int) sfinfo_in.frames); in permute() 212 double durationSeconds = (double) sfinfo_in.frames / (double) sfinfo_in.samplerate; in permute() 224 used = split(&s, 0, sfinfo_in.frames, s.mSegmentMax); in permute() 241 void *ptr = malloc(sfinfo_in.frames * frameSizeRead); in permute() 244 count = sf_readf_short(sf_in, ptr, sfinfo_in.frames); in permute() 245 if (count != sfinfo_in.frames) { in permute() 247 (int) sfinfo_in.frames, (int) count); in permute() 279 assert(permutedStart == sfinfo_in.frames); in permute()
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/frameworks/native/opengl/tests/hwc/ |
D | hwcStress.cpp | 199 static vector <vector <sp<GraphicBuffer> > > frames; variable 413 list = hwcTestCreateLayerList(testRandMod(frames.size()) + 1); in main() 421 selectedFrames = vectorRandSelect(frames, list->numHwLayers); in main() 562 frames.clear(); in initFrames() 563 frames.resize(rows); in initFrames() 591 frames[row].resize(cols); in initFrames() 596 frames[row][col] = new GraphicBuffer(w, h, format, texUsage); in initFrames() 597 if ((rv = frames[row][col]->initCheck()) != NO_ERROR) { in initFrames() 604 hwcTestFillColor(frames[row][col].get(), color, alpha); in initFrames() 607 frames[row][col].get(), frames[row][col]->handle, in initFrames()
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/frameworks/av/include/media/ |
D | RecordBufferConverter.h | 63 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 83 void convertNoResampler(void *dst, const void *src, size_t frames); 86 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames);
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D | BufferProviders.h | 86 virtual void copyFrames(void *dst, const void *src, size_t frames) = 0; 107 virtual void copyFrames(void *dst, const void *src, size_t frames); 139 virtual void copyFrames(void *dst, const void *src, size_t frames); 156 virtual void copyFrames(void *dst, const void *src, size_t frames);
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/frameworks/av/media/libmedia/include/media/ |
D | RecordBufferConverter.h | 63 size_t convert(void *dst, AudioBufferProvider *provider, size_t frames); 83 void convertNoResampler(void *dst, const void *src, size_t frames); 86 void convertResampler(void *dst, /*not-a-const*/ void *src, size_t frames);
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D | BufferProviders.h | 86 virtual void copyFrames(void *dst, const void *src, size_t frames) = 0; 107 virtual void copyFrames(void *dst, const void *src, size_t frames); 139 virtual void copyFrames(void *dst, const void *src, size_t frames); 156 virtual void copyFrames(void *dst, const void *src, size_t frames);
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/frameworks/base/cmds/bootanimation/ |
D | FORMAT.md | 17 part1 \ directories full of PNG frames 29 * **FPS:** frames per second, e.g. 60 40 * **PATH:** directory in which to find the frames for this part (e.g. `part0`) 47 ## loading and playing frames 51 one frame in that part (at the specified resolution). For this reason it is important that frames be 58 the trim output for each frame in its directory, so the frames may be properly positioned.
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/frameworks/av/media/libaaudio/src/core/ |
D | AudioStream.h | 235 virtual int64_t incrementFramesWritten(int32_t frames) { in incrementFramesWritten() argument 236 return mFramesWritten.increment(frames); in incrementFramesWritten() 239 virtual int64_t incrementFramesRead(int32_t frames) { in incrementFramesRead() argument 240 return mFramesRead.increment(frames); in incrementFramesRead()
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/frameworks/av/media/libstagefright/ |
D | XINGSeeker.cpp | 139 int32_t frames = U32_AT(buffer); in CreateFromSource() local 144 if (frames) { in CreateFromSource() 145 seeker->mDurationUs = (int64_t)frames * samples_per_frame * 1000000LL / sampling_rate; in CreateFromSource()
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/frameworks/av/media/libaaudio/src/legacy/ |
D | AudioStreamRecord.h | 75 int64_t incrementClientFrameCounter(int32_t frames) override { in incrementClientFrameCounter() argument 76 return incrementFramesRead(frames); in incrementClientFrameCounter()
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D | AudioStreamTrack.h | 75 int64_t incrementClientFrameCounter(int32_t frames) override { in incrementClientFrameCounter() argument 76 return incrementFramesWritten(frames); in incrementClientFrameCounter()
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