/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) { 67 mAllocated = calloc(buffer->frameCount, mFrameSize); 72 mSize = buffer->frameCount; 76 ssize_t actual = mSource->read(mAllocated, buffer->frameCount); 78 ALOG_ASSERT((size_t) actual <= buffer->frameCount); 82 buffer->frameCount [all...] |
H A D | AudioBufferProviderSource.cpp | 46 return mBuffer.raw != NULL ? mBuffer.frameCount - mConsumed : 0; 55 mBuffer.frameCount = count; 63 size_t available = mBuffer.frameCount - mConsumed; 70 if (CC_UNLIKELY((mConsumed += count) >= mBuffer.frameCount)) { 101 mBuffer.frameCount = count; 104 ALOG_ASSERT(mBuffer.raw != NULL && mBuffer.frameCount <= count); 115 size_t available = mBuffer.frameCount - mConsumed; 131 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 46 // buffer->frameCount maximum number of desired frames 49 // buffer->raw non-NULL pointer to buffer->frameCount contiguous available frames 50 // buffer->frameCount number of contiguous available frames at buffer->raw, 51 // 0 < buffer->frameCount <= entry value 55 // buffer->frameCount 0 61 // buffer->frameCount number of frames to release, must be <= number of frames 65 // 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); 157 private static native long nativeConstructor(int valueCount, int frameCount); argument 159 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) { 76 // this, pBuffer, pBuffer->frameCount); 80 copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount); 84 if (mBuffer.frameCount == 0) { 85 mBuffer.frameCount = pBuffer->frameCount; 88 // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014. 90 // By API spec, if res != OK, then mBuffer.frameCount == 0. 92 ALOG_ASSERT(res == OK || mBuffer.frameCount [all...] |
H A D | FastCapture.cpp | 92 const size_t frameCount = current->mFrameCount; local 128 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) { 132 if (frameCount > 0 && mSampleRate > 0) { 136 size_t bufferSize = frameCount * Format_frameSize(mFormat); 139 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 140 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 141 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 142 mForceNs = (frameCount * 950000000LL) / mSampleRate; // 0.95 143 mWarmupNsMin = (frameCount * 750000000LL) / mSampleRate; // 0.75 144 mWarmupNsMax = (frameCount * 125000000 164 const size_t frameCount = current->mFrameCount; local [all...] |
H A D | AudioResampler.cpp | 280 mBuffer.frameCount = 0; 311 mBuffer.frameCount = 0; 353 while (mBuffer.frameCount == 0) { 354 mBuffer.frameCount = inFrameCount; 360 // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount); 361 if (mBuffer.frameCount > inputIndex) break; 363 inputIndex -= mBuffer.frameCount; 364 mX0L = mBuffer.i16[mBuffer.frameCount*2-2]; 365 mX0R = mBuffer.i16[mBuffer.frameCount*2-1]; 367 // mBuffer.frameCount [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); 95 if (inputIndex == mBuffer.frameCount) { 98 mBuffer.frameCount = inFrameCount; 104 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); 134 if (mBuffer.frameCount == 0) { 135 mBuffer.frameCount = inFrameCount; 140 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount); 163 if (inputIndex == mBuffer.frameCount) { [all...] |
H A D | FastMixer.cpp | 141 const size_t frameCount = current->mFrameCount; local 174 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) { 182 if (frameCount > 0 && mSampleRate > 0) { 190 mMixer = new AudioMixer(frameCount, mSampleRate, FastMixerState::sMaxFastTracks); 193 mMixerBufferSize = mixerFrameSize * frameCount; 198 mSinkBufferSize = sinkFrameSize * frameCount; 201 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 202 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 203 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 204 mForceNs = (frameCount * 95000000 336 const size_t frameCount = current->mFrameCount; local [all...] |
H A D | Tracks.cpp | 72 size_t frameCount, 94 mFrameCount(frameCount), 116 size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize; 251 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 256 buf.mFrameCount = buffer->frameCount; 258 buffer->frameCount = 0; 343 size_t frameCount, 350 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 384 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 66 TrackBase( ThreadBase *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, audio_session_t sessionId, int clientUid, IAudioFlinger::track_flags_t flags, bool isOut, alloc_type alloc, track_type type) argument 336 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, audio_session_t sessionId, int uid, IAudioFlinger::track_flags_t flags, track_type type) argument 1126 OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, int uid) argument 1325 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 1461 RecordTrack( RecordThread *thread, const sp<Client>& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, audio_session_t sessionId, int uid, IAudioFlinger::track_flags_t flags, track_type type) argument 1661 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 | test-resample.cpp | 276 size_t requestedFrames = buffer->frameCount; 278 buffer->frameCount = mNumFrames - mNextFrame; 282 printf("mPvalue[%zu]=%zu not %zu\n", mNextPidx-1, provided, buffer->frameCount); 283 if (provided < buffer->frameCount) { 284 buffer->frameCount = provided; 293 requestedFrames, (size_t) (mNumFrames - mNextFrame), buffer->frameCount); 295 mUnrel = buffer->frameCount; 296 if (buffer->frameCount > 0) { 305 if (buffer->frameCount > mUnrel) { 307 "to release\n", buffer->frameCount, mUnre [all...] |
H A D | AudioMixer.cpp | 101 AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, uint32_t maxNumTracks) argument 115 mState.frameCount = frameCount; 199 // t->frameCount 210 // t->buffer.frameCount 703 target == RAMP_VOLUME ? mState.frameCount : 0, 714 target == RAMP_VOLUME ? mState.frameCount : 0, 1003 state->outputTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 1006 state->resampleTemp = new int32_t[MAX_NUM_CHANNELS * state->frameCount]; 1114 void AudioMixer::volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_ argument 1157 volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 1186 track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument 1278 track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument 1692 volumeRampMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc) argument 1736 volumeMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, const TV *vol, TAV vola) argument 1902 track__NoResample(track_t* t, TO* out, size_t frameCount, TO* temp __unused, TA* aux) argument [all...] |
H A D | AudioResamplerDyn.cpp | 523 ALOG_ASSERT(inputIndex <= mBuffer.frameCount, "inputIndex%d > frameCount%d", 524 inputIndex, mBuffer.frameCount); 527 while (mBuffer.frameCount == 0 && inFrameCount > 0) { 528 mBuffer.frameCount = inFrameCount; 533 inFrameCount -= mBuffer.frameCount; 541 if (inputIndex >= mBuffer.frameCount) { 555 const size_t frameCount = mBuffer.frameCount; local 582 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, 361 static void track__Resample(track_t* t, TO* out, size_t frameCount, 364 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 | 205 Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut, 216 size_t frameCount() const { return mFrameCount; } function in class:android::Proxy 238 ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 356 AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 358 : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, 407 StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, 457 AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 459 : ClientProxy(cblk, buffers, frameCount, frameSize, 479 ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 544 AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 646 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 | 7 static jlong Interpolator_constructor(JNIEnv* env, jobject clazz, jint valueCount, jint frameCount) argument 9 return reinterpret_cast<jlong>(new SkInterpolator(valueCount, frameCount)); 18 static void Interpolator_reset(JNIEnv* env, jobject clazz, jlong interpHandle, jint valueCount, jint frameCount) argument 21 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, 100 mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user, 138 size_t frameCount, 150 ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 153 inputSource, sampleRate, format, channelMask, frameCount, notificationFrames, 225 mReqFrameCount = frameCount; 594 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, audio_session_t sessionId, transfer_type transferType, audio_input_flags_t flags, int uid, pid_t pid, const audio_attributes_t* pAttributes) argument 133 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, audio_session_t sessionId, transfer_type transferType, audio_input_flags_t flags, int uid, pid_t pid, const audio_attributes_t* pAttributes) argument [all...] |
H A D | AudioTrackShared.cpp | 59 Proxy::Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, argument 61 : mCblk(cblk), mBuffers(buffers), mFrameCount(frameCount), mFrameSize(frameSize), 62 mFrameCountP2(roundup(frameCount)), mIsOut(isOut), mClientInServer(clientInServer), 69 ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 71 : Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer) 75 setBufferSizeInFrames(frameCount); 94 const uint32_t maximum = frameCount(); 524 size_t frameCount, size_t frameSize) 525 : AudioTrackClientProxy(cblk, buffers, frameCount, frameSize), 615 ServerProxy::ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 523 StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) argument 829 tallyUnderrunFrames(uint32_t frameCount) argument 861 StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize) argument [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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