/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/ |
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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H A D | OggExtractor.cpp | 617 uint32_t frames = getNumSamplesInPacket(*out); local 618 mCurGranulePosition += frames;
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/frameworks/av/services/audioflinger/ |
H A D | AudioStreamOut.cpp | 48 status_t AudioStreamOut::getRenderPosition(uint64_t *frames) argument 69 *frames = mRenderPosition / mRateMultiplier; 75 status_t AudioStreamOut::getRenderPosition(uint32_t *frames) argument 80 *frames = (uint32_t)position64; 85 status_t AudioStreamOut::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) argument 97 // Adjust for standby using HAL rate frames. 98 // Only apply this correction if the HAL is getting PCM frames. 103 *frames = adjustedPosition / mRateMultiplier; 106 *frames = halPosition;
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H A D | BufferProviders.cpp | 235 void DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames) argument 237 mDownmixConfig.inputCfg.buffer.frameCount = frames; 239 mDownmixConfig.outputCfg.buffer.frameCount = frames; 298 void RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames) argument 301 src, mInputChannels, mIdxAry, mSampleSize, frames); 319 void ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames) argument 321 memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCount); 446 LOG_ALWAYS_FATAL("Releasing more frames(%zu) than available(%zu)", 475 // Note dstFrames is the required number of frames. 479 //rate, actual frames processe [all...] |
H A D | test-resample.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 | Tracks.cpp | 580 // framesReady() may return an approximation of the number of frames if called 586 // Static tracks return zero frames immediately upon stopping (for FastTracks). 601 // The server side frames are already translated to client frames. 946 // a track is considered presented when the total number of frames written to audio HAL 947 // corresponds to the number of frames written when presentationComplete() is called for the 950 // to detect when all frames have been played. In this case framesWritten isn't 1119 // Set correction for flushed frames that are not accounted for in released. 1188 bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames) argument 1193 inBuffer.frameCount = frames; [all...] |
H A D | AudioFlinger.cpp | 1233 size_t frames; local 1238 frames = dev->get_input_buffer_size(dev, &config); 1239 if (frames != 0) { 1252 break; // retries failed, break out of loop with frames == 0. 1256 if (frames > 0 && config.sample_rate != sampleRate) { 1257 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1259 return frames; // may be converted to bytes at the Java level. 3069 #define TEE_SINK_READ 1024 // frames per I/O operation
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H A D | Threads.cpp | 130 // minimum normal sink buffer size, expressed in milliseconds rather than frames 2237 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2288 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2319 // round up to nearest 16 frames to satisfy AudioMixer 2323 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2332 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2386 // return an estimation of rendered frames when the output is suspended 2393 uint32_t frames; local 6775 convert(void *dst, AudioBufferProvider *provider, size_t frames) argument 6917 convertNoResampler( void *dst, const void *src, size_t frames) argument 6957 convertResampler( void *dst, void *src, size_t frames) argument [all...] |
/frameworks/av/services/audioflinger/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_utils.h | 95 TestProvider(void* addr, size_t frames, size_t frameSize, argument 98 mNumFrames(frames), 132 ALOGV("getNextBuffer() requested %zu frames out of %zu frames available" 133 " and returned %zu frames", 148 ALOGE("releaseBuffer() released %zu frames but only %zu available " 154 ALOGV("releaseBuffer() released %zu frames out of %zu frames available " 176 size_t mNumFrames; // total frames 179 size_t mUnrel; // number of frames no 187 createSine(void *vbuffer, size_t frames, size_t channels, double sampleRate, double freq) argument 211 createChirp(void *vbuffer, size_t frames, size_t channels, double sampleRate, double minfreq, double maxfreq) argument 284 createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames) argument [all...] |
/frameworks/base/cmds/bootanimation/ |
H A D | BootAnimation.h | 71 int pause; // The number of frames to pause for at the end of this part 75 SortedVector<Frame> frames; member in struct:android::BootAnimation::Animation::Part
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/frameworks/av/media/libstagefright/webm/ |
H A D | WebmFrameThread.cpp | 94 // frames: 95 // sequence of input audio/video frames received from the source. 99 // frame since frames are ordered by timestamp. 107 List<const sp<WebmFrame> >& frames, 110 CHECK(!frames.empty() && children.empty()); 112 const sp<WebmFrame> f = *(frames.begin()); 129 // Write out (possibly multiple) webm cluster(s) from frames split on video key frames. 133 void WebmFrameSinkThread::flushFrames(List<const sp<WebmFrame> >& frames, bool last) { argument 134 if (frames 106 initCluster( List<const sp<WebmFrame> >& frames, uint64_t& clusterTimecodeL, List<sp<WebmElement> >& children) argument [all...] |
/frameworks/native/opengl/tests/hwc/ |
H A D | hwcStress.cpp | 199 static vector <vector <sp<GraphicBuffer> > > frames; variable 396 // Regenerate a new set of test frames when this pass is 413 list = hwcTestCreateLayerList(testRandMod(frames.size()) + 1); 419 // Prandomly select a subset of frames to be used by this pass. 421 selectedFrames = vectorRandSelect(frames, list->numHwLayers); 545 * named frames. The graphic buffers are contained within a vector of 562 frames.clear(); 563 frames.resize(rows); 566 // All frames within a row have to have the same format and 591 frames[ro [all...] |
/frameworks/wilhelm/tests/sandbox/ |
H A D | playbq.c | 249 fprintf(stderr, " -f# frames per buffer (default 512)\n"); 498 sf_count_t frames = framesPerBuffer; local 502 count = sf_readf_float(sndfile, (float *) buffer, frames); 505 count = sf_readf_int(sndfile, (int *) buffer, frames); 508 count = sf_readf_int(sndfile, (int *) buffer, frames); 512 count = sf_readf_short(sndfile, (short *) buffer, frames);
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H A D | playbq.cpp | 249 fprintf(stderr, " -f# frames per buffer (default 512)\n"); 498 sf_count_t frames = framesPerBuffer; local 502 count = sf_readf_float(sndfile, (float *) buffer, frames); 505 count = sf_readf_int(sndfile, (int *) buffer, frames); 508 count = sf_readf_int(sndfile, (int *) buffer, frames); 512 count = sf_readf_short(sndfile, (short *) buffer, frames);
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/frameworks/av/media/libmedia/ |
H A D | AudioTrack.cpp | 53 static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed) argument 55 return ((double)frames * 1000000000) / ((double)sampleRate * speed); 597 // force refresh of remaining frames by processAudioBuffer() as last 1876 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END 2143 // There could be enough non-contiguous frames available to satisfy the remaining request 2205 // take the frames that will be lost by track recreation into account in saved position 2239 // This is the sole place to read server consumed frames 2248 // unless the server has more than 2^31 frames in its buffer, 2312 // apply server offset (frames flushed is ignored 2357 // To avoid a race, read the presented frames firs 2380 const int64_t frames = local [all...] |
/frameworks/data-binding/prebuilds/1.0-rc0/ |
H A D | databinding-studio-bundle.jar | META-INF/ META-INF/MANIFEST.MF android/ android/databinding/ android/databinding/Bindable.class Bindable. ... |