/frameworks/native/libs/ui/ |
H A D | FrameStats.cpp | 41 size_t frameCount = desiredPresentTimesNano.size(); local 46 memcpy(timestamps, desiredPresentTimesNano.array(), frameCount * timestampSize); 47 timestamps += frameCount; 49 memcpy(timestamps, actualPresentTimesNano.array(), frameCount * timestampSize); 50 timestamps += frameCount; 52 memcpy(timestamps, frameReadyTimesNano.array(), frameCount * timestampSize); 65 size_t frameCount = (size - timestampSize) / (3 * timestampSize); local 70 desiredPresentTimesNano.resize(frameCount); 71 memcpy(desiredPresentTimesNano.editArray(), timestamps, frameCount * timestampSize); 72 timestamps += frameCount; [all...] |
/frameworks/av/media/libeffects/testlibs/ |
H A D | AudioBiquadFilter.h | 27 // The filter works on fixed sized blocks of data (frameCount multi-channel 72 // Process a buffer of data. Always processes frameCount multi-channel 75 // in The input buffer. Should be of size frameCount * nChannels. 76 // out The output buffer. Should be of size frameCount * nChannels. 77 // frameCount Number of multi-channel samples to process. 79 int frameCount); 98 int frameCount); 154 bool updateCoefs(const audio_coef_t coefs[NUM_COEFS], int frameCount); 158 int frameCount); 161 int frameCount); [all...] |
H A D | AudioBiquadFilter.cpp | 66 int frameCount) { 67 (this->*mCurProcessFunc)(in, out, frameCount); 121 int frameCount) { 122 int64_t maxDelta = mMaxDelta * frameCount; 141 int frameCount) { 144 memcpy(out, in, frameCount * mNumChannels * sizeof(audio_sample_t)); 150 int frameCount) { 151 size_t nFrames = frameCount; 184 int frameCount) { 185 if (updateCoefs(mTargetCoefs, frameCount)) { 65 process(const audio_sample_t in[], audio_sample_t out[], int frameCount) argument 120 updateCoefs(const audio_coef_t coefs[NUM_COEFS], int frameCount) argument 139 process_bypass(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument 148 process_normal_mono(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument 182 process_transition_normal_mono(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument 191 process_transition_bypass_mono(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument 200 process_normal_multi(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument 240 process_transition_normal_multi(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument 249 process_transition_bypass_multi(const audio_sample_t * in, audio_sample_t * out, int frameCount) argument [all...] |
H A D | AudioShelvingFilter.h | 93 // frameCount * nChannels interlaced samples. Processing can be done 97 // frameCount Number of frames to produce. 99 int frameCount) { mBiquad.process(in, out, frameCount); } 98 process(const audio_sample_t in[], audio_sample_t out[], int frameCount) argument
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H A D | AudioPeakingFilter.h | 99 // frameCount * nChannels interlaced samples. Processing can be done 103 // frameCount Number of frames to produce. 105 int frameCount) { mBiquad.process(in, out, frameCount); } 104 process(const audio_sample_t in[], audio_sample_t out[], int frameCount) argument
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/frameworks/av/media/libnbaio/ |
H A D | SourceAudioBufferProvider.cpp | 50 ALOG_ASSERT(buffer != NULL && buffer->frameCount > 0 && mGetCount == 0); 54 if (mRemaining < buffer->frameCount) { 55 buffer->frameCount = mRemaining; 58 mGetCount = buffer->frameCount; 62 if (buffer->frameCount > mSize) { 64 mAllocated = malloc(buffer->frameCount * mFrameSize); 65 mSize = buffer->frameCount; 68 ssize_t actual = mSource->read(mAllocated, buffer->frameCount, pts); 70 ALOG_ASSERT((size_t) actual <= buffer->frameCount); 74 buffer->frameCount [all...] |
H A D | AudioBufferProviderSource.cpp | 46 return mBuffer.raw != NULL ? mBuffer.frameCount - mConsumed : 0; 57 mBuffer.frameCount = count; 65 size_t available = mBuffer.frameCount - mConsumed; 72 if (CC_UNLIKELY((mConsumed += count) >= mBuffer.frameCount)) { 104 mBuffer.frameCount = count; 107 ALOG_ASSERT(mBuffer.raw != NULL && mBuffer.frameCount <= count); 118 size_t available = mBuffer.frameCount - mConsumed; 134 if (CC_LIKELY((mConsumed += ret) < mBuffer.frameCount)) {
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/frameworks/av/include/media/ |
H A D | AudioBufferProvider.h | 32 Buffer() : raw(NULL), frameCount(0) { } 38 size_t frameCount; member in struct:android::AudioBufferProvider::Buffer 52 // buffer->frameCount maximum number of desired frames 55 // buffer->raw non-NULL pointer to buffer->frameCount contiguous available frames 56 // buffer->frameCount number of contiguous available frames at buffer->raw, 57 // 0 < buffer->frameCount <= entry value 61 // buffer->frameCount 0 67 // buffer->frameCount number of frames to release, must be <= number of frames 71 // buffer->frameCount 0; implementation MUST set to zero
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/frameworks/base/graphics/java/android/graphics/ |
H A D | Interpolator.java | 29 public Interpolator(int valueCount, int frameCount) { argument 31 mFrameCount = frameCount; 32 native_instance = nativeConstructor(valueCount, frameCount); 49 public void reset(int valueCount, int frameCount) { argument 51 mFrameCount = frameCount; 52 nativeReset(native_instance, valueCount, frameCount); 156 private static native long nativeConstructor(int valueCount, int frameCount); argument 158 private static native void nativeReset(long native_instance, int valueCount, int frameCount); argument
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/frameworks/av/services/audioflinger/ |
H A D | BufferProviders.cpp | 61 mBuffer.frameCount = 0; 67 if (mBuffer.frameCount != 0) { 77 // this, pBuffer, pBuffer->frameCount, pts); 81 copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount); 85 if (mBuffer.frameCount == 0) { 86 mBuffer.frameCount = pBuffer->frameCount; 89 // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014. 91 // By API spec, if res != OK, then mBuffer.frameCount == 0. 93 ALOG_ASSERT(res == OK || mBuffer.frameCount [all...] |
H A D | FastCapture.cpp | 92 const size_t frameCount = current->mFrameCount; local 126 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) { 130 if (frameCount > 0 && mSampleRate > 0) { 134 (void)posix_memalign(&mReadBuffer, 32, frameCount * Format_frameSize(mFormat)); 135 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 136 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 137 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 138 mForceNs = (frameCount * 950000000LL) / mSampleRate; // 0.95 139 mWarmupNsMin = (frameCount * 750000000LL) / mSampleRate; // 0.75 140 mWarmupNsMax = (frameCount * 125000000 160 const size_t frameCount = current->mFrameCount; local [all...] |
H A D | AudioResamplerCubic.cpp | 67 if (mBuffer.frameCount == 0) { 68 mBuffer.frameCount = inFrameCount; 73 // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount); 96 if (inputIndex == mBuffer.frameCount) { 99 mBuffer.frameCount = inFrameCount; 106 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); 136 if (mBuffer.frameCount == 0) { 137 mBuffer.frameCount = inFrameCount; 142 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); 165 if (inputIndex == mBuffer.frameCount) { [all...] |
H A D | Tracks.cpp | 71 size_t frameCount, 93 mFrameCount(frameCount), 117 size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize; 252 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 257 buf.mFrameCount = buffer->frameCount; 259 buffer->frameCount = 0; 381 size_t frameCount, 388 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 420 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 65 TrackBase( ThreadBase *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, int sessionId, int clientUid, IAudioFlinger::track_flags_t flags, bool isOut, alloc_type alloc, track_type type) argument 374 Track( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, const sp<IMemory>& sharedBuffer, int sessionId, int uid, IAudioFlinger::track_flags_t flags, track_type type) argument 1118 create( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, int uid) argument 1138 TimedTrack( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, const sp<IMemory>& sharedBuffer, int sessionId, int uid) argument 1251 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() local 1637 OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, int uid) argument 1823 PatchTrack(PlaybackThread *playbackThread, audio_stream_type_t streamType, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, IAudioFlinger::track_flags_t flags) argument 1934 RecordTrack( RecordThread *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, int sessionId, int uid, IAudioFlinger::track_flags_t flags, track_type type) argument 2116 PatchRecord(RecordThread *recordThread, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, IAudioFlinger::track_flags_t flags) argument [all...] |
H A D | FastMixer.cpp | 142 const size_t frameCount = current->mFrameCount; local 171 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) { 179 if (frameCount > 0 && mSampleRate > 0) { 187 mMixer = new AudioMixer(frameCount, mSampleRate, FastMixerState::kMaxFastTracks); 190 mMixerBufferSize = mixerFrameSize * frameCount; 195 mSinkBufferSize = sinkFrameSize * frameCount; 198 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 199 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 200 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 201 mForceNs = (frameCount * 95000000 333 const size_t frameCount = current->mFrameCount; local [all...] |
H A D | AudioResampler.cpp | 279 mBuffer.frameCount = 0; 327 mBuffer.frameCount = 0; 369 while (mBuffer.frameCount == 0) { 370 mBuffer.frameCount = inFrameCount; 377 // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount); 378 if (mBuffer.frameCount > inputIndex) break; 380 inputIndex -= mBuffer.frameCount; 381 mX0L = mBuffer.i16[mBuffer.frameCount*2-2]; 382 mX0R = mBuffer.i16[mBuffer.frameCount*2-1]; 384 // mBuffer.frameCount [all...] |
H A D | test-resample.cpp | 278 size_t requestedFrames = buffer->frameCount; 280 buffer->frameCount = mNumFrames - mNextFrame; 284 printf("mPvalue[%zu]=%zu not %zu\n", mNextPidx-1, provided, buffer->frameCount); 285 if (provided < buffer->frameCount) { 286 buffer->frameCount = provided; 295 requestedFrames, (size_t) (mNumFrames - mNextFrame), buffer->frameCount); 297 mUnrel = buffer->frameCount; 298 if (buffer->frameCount > 0) { 307 if (buffer->frameCount > mUnrel) { 309 "to release\n", buffer->frameCount, mUnre [all...] |
H A D | AudioMixer.cpp | 103 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) argument 117 mState.frameCount = frameCount; 201 // t->frameCount 212 // t->buffer.frameCount 705 target == RAMP_VOLUME ? mState.frameCount : 0, 716 target == RAMP_VOLUME ? mState.frameCount : 0, 1006 state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 1009 state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 1117 void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_ argument 1160 volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 1189 track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument 1281 track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument 1717 volumeRampMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc) argument 1761 volumeMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, const TV *vol, TAV vola) argument 1928 track__NoResample(track_t* t, TO* out, size_t frameCount, TO* temp __unused, TA* aux) argument [all...] |
H A D | AudioResamplerDyn.cpp | 524 ALOG_ASSERT(inputIndex <= mBuffer.frameCount, "inputIndex%d > frameCount%d", 525 inputIndex, mBuffer.frameCount); 528 while (mBuffer.frameCount == 0 && inFrameCount > 0) { 529 mBuffer.frameCount = inFrameCount; 535 inFrameCount -= mBuffer.frameCount; 543 if (inputIndex >= mBuffer.frameCount) { 557 const size_t frameCount = mBuffer.frameCount; local 584 if (inputIndex >= frameCount) { [all...] |
H A D | AudioMixer.h | 45 AudioMixer(size_t frameCount, uint32_t sampleRate, 193 uint16_t frameCount; member in struct:android::AudioMixer::track_t 287 size_t frameCount; member in struct:android::AudioMixer::state_t 326 static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 328 static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, 366 static void track__Resample(track_t* t, TO* out, size_t frameCount, 369 static void track__NoResample(track_t* t, TO* out, size_t frameCount,
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/frameworks/av/services/audioflinger/tests/ |
H A D | test_utils.h | 117 size_t requestedFrames = buffer->frameCount; 119 buffer->frameCount = mNumFrames - mNextFrame; 124 mNextIdx-1, provided, buffer->frameCount); 125 if (provided < buffer->frameCount) { 126 buffer->frameCount = provided; 134 requestedFrames, mNumFrames - mNextFrame, buffer->frameCount); 135 mUnrel = buffer->frameCount; 136 if (buffer->frameCount > 0) { 147 if (buffer->frameCount > mUnrel) { 149 "to release", buffer->frameCount, mUnre [all...] |
/frameworks/av/include/private/media/ |
H A D | AudioTrackShared.h | 195 Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut, 226 ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 318 AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 320 : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, 364 StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, 414 AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 416 : ClientProxy(cblk, buffers, frameCount, frameSize, 426 ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 472 AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 474 : ServerProxy(cblk, buffers, frameCount, frameSiz 558 AudioRecordServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool clientInServer) argument [all...] |
/frameworks/base/core/jni/android/graphics/ |
H A D | Interpolator.cpp | 8 static jlong Interpolator_constructor(JNIEnv* env, jobject clazz, jint valueCount, jint frameCount) argument 10 return reinterpret_cast<jlong>(new SkInterpolator(valueCount, frameCount)); 19 static void Interpolator_reset(JNIEnv* env, jobject clazz, jlong interpHandle, jint valueCount, jint frameCount) argument 22 interp->reset(valueCount, frameCount);
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/frameworks/av/media/libmedia/ |
H A D | AudioRecord.cpp | 37 size_t* frameCount, 42 if (frameCount == NULL) { 56 if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) * 81 size_t frameCount, 99 mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, 137 size_t frameCount, 149 ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 152 inputSource, sampleRate, format, channelMask, frameCount, notificationFrames, 228 mReqFrameCount = frameCount; 539 size_t frameCount local 36 getMinFrameCount( size_t* frameCount, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask) argument 75 AudioRecord( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const String16& opPackageName, size_t frameCount, callback_t cbf, void* user, uint32_t notificationFrames, int sessionId, transfer_type transferType, audio_input_flags_t flags, int uid, pid_t pid, const audio_attributes_t* pAttributes) argument 132 set( audio_source_t inputSource, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, callback_t cbf, void* user, uint32_t notificationFrames, bool threadCanCallJava, int sessionId, transfer_type transferType, audio_input_flags_t flags, int uid, pid_t pid, const audio_attributes_t* pAttributes) argument [all...] |
H A D | AudioTrack.cpp | 109 size_t* frameCount, 113 if (frameCount == NULL) { 147 *frameCount = calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, 1.0f); 152 if (*frameCount == 0) { 158 *frameCount, afFrameCount, afSampleRate, afLatency); 183 size_t frameCount, 203 frameCount, flags, cbf, user, notificationFrames, 233 0 /*frameCount*/, flags, cbf, user, notificationFrames, 271 size_t frameCount, 286 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount 108 getMinFrameCount( size_t* frameCount, audio_stream_type_t streamType, uint32_t sampleRate) argument 178 AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, audio_output_flags_t flags, callback_t cbf, void* user, uint32_t notificationFrames, int sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid, pid_t pid, const audio_attributes_t* pAttributes, bool doNotReconnect) argument 266 set( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, audio_output_flags_t flags, callback_t cbf, void* user, uint32_t notificationFrames, const sp<IMemory>& sharedBuffer, bool threadCanCallJava, int sessionId, transfer_type transferType, const audio_offload_info_t *offloadInfo, int uid, pid_t pid, const audio_attributes_t* pAttributes, bool doNotReconnect) argument 1189 size_t frameCount = mReqFrameCount; local [all...] |
/frameworks/native/opengl/tests/angeles/ |
H A D | app-linux.cpp | 206 int frameCount = 0; local 220 frameCount++; 229 printf("totalTime=%f s, frameCount=%d, %.2f fps\n", 230 totalTime, frameCount, frameCount/totalTime);
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