/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/av/media/libaudioclient/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/media/libaaudio/tests/ |
H A D | test_block_adapter.cpp | 38 void fillSequence(int32_t *indexBuffer, int32_t frameCount) { argument 39 ASSERT_LE(frameCount, TEST_BUFFER_SIZE); 40 for (int i = 0; i < frameCount; i++) { 45 int checkSequence(const int32_t *indexBuffer, int32_t frameCount) { argument 47 for (int i = 0; i < frameCount; i++) { 75 int32_t frameCount = numBytes / sizeof(int32_t); variable 76 return checkSequence((int32_t *) buffer, frameCount); 102 int32_t frameCount = numBytes / sizeof(int32_t); variable 103 fillSequence((int32_t *) buffer, frameCount);
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/frameworks/av/media/libaudioprocessing/ |
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 | AudioResampler.cpp | 283 mBuffer.frameCount = 0; 314 mBuffer.frameCount = 0; 356 while (mBuffer.frameCount == 0) { 357 mBuffer.frameCount = inFrameCount; 363 // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount); 364 if (mBuffer.frameCount > inputIndex) break; 366 inputIndex -= mBuffer.frameCount; 367 mX0L = mBuffer.i16[mBuffer.frameCount*2-2]; 368 mX0R = mBuffer.i16[mBuffer.frameCount*2-1]; 370 // 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 | AudioMixer.cpp | 138 // t->frameCount 149 // t->buffer.frameCount 1004 int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) 1013 // (vl + vlInc*frameCount)/65536.0f, frameCount); 1031 } while (--frameCount); 1039 } while (--frameCount); 1047 int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) 1063 } while (--frameCount); 1071 } while (--frameCount); 1003 volumeRampStereo( int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 1046 volumeStereo( int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux) argument 1075 track__16BitsStereo( int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument 1167 track__16BitsMono( int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux) argument 1308 const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames); local 1526 volumeRampMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc) argument 1570 volumeMulti(uint32_t channels, TO* out, size_t frameCount, const TI* in, TA* aux, const TV *vol, TAV vola) argument 1747 track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux) argument [all...] |
/frameworks/av/services/audioflinger/ |
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 | FastMixer.cpp | 143 const size_t frameCount = current->mFrameCount; local 176 if ((!Format_isEqual(mFormat, previousFormat)) || (frameCount != previous->mFrameCount)) { 184 if (frameCount > 0 && mSampleRate > 0) { 192 mMixer = new AudioMixer(frameCount, mSampleRate); 196 mMixerBufferSize = mixerFrameSize * frameCount; 201 mSinkBufferSize = sinkFrameSize * frameCount; 204 mPeriodNs = (frameCount * 1000000000LL) / mSampleRate; // 1.00 205 mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate; // 1.75 206 mOverrunNs = (frameCount * 500000000LL) / mSampleRate; // 0.50 207 mForceNs = (frameCount * 95000000 339 const size_t frameCount = current->mFrameCount; local [all...] |
H A D | Tracks.cpp | 70 size_t frameCount, 94 mFrameCount(frameCount), 116 size_t minBufferSize = buffer == NULL ? roundup(frameCount) : frameCount; 118 if (minBufferSize < frameCount // roundup rounds down for values above UINT_MAX / 2 278 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 283 buf.mFrameCount = buffer->frameCount; 285 buffer->frameCount = 0; 381 size_t frameCount, 390 : TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount, 63 TrackBase( ThreadBase *thread, const sp<Client>& client, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, size_t bufferSize, audio_session_t sessionId, uid_t clientUid, bool isOut, alloc_type alloc, track_type type, audio_port_handle_t portId) argument 373 Track( PlaybackThread *thread, const sp<Client>& client, audio_stream_type_t streamType, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, size_t bufferSize, const sp<IMemory>& sharedBuffer, audio_session_t sessionId, uid_t uid, audio_output_flags_t flags, track_type type, audio_port_handle_t portId) argument 1282 OutputTrack( PlaybackThread *playbackThread, DuplicatingThread *sourceThread, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, uid_t uid) argument 1499 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, size_t bufferSize, audio_output_flags_t flags) argument 1643 RecordTrack( RecordThread *thread, const sp<Client>& client, const audio_attributes_t& attr, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, void *buffer, size_t bufferSize, audio_session_t sessionId, uid_t uid, audio_input_flags_t flags, track_type type, audio_port_handle_t portId) argument 1867 PatchRecord(RecordThread *recordThread, uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format, size_t frameCount, void *buffer, size_t bufferSize, audio_input_flags_t flags) argument [all...] |
/frameworks/av/media/libaudioprocessing/tests/ |
H A D | test_utils.h | 123 size_t requestedFrames = buffer->frameCount; 125 buffer->frameCount = mNumFrames - mNextFrame; 130 mNextIdx-1, provided, buffer->frameCount); 131 if (provided < buffer->frameCount) { 132 buffer->frameCount = provided; 140 requestedFrames, mNumFrames - mNextFrame, buffer->frameCount); 141 mUnrel = buffer->frameCount; 142 if (buffer->frameCount > 0) { 153 if (buffer->frameCount > mUnrel) { 155 "to release", buffer->frameCount, mUnre [all...] |
/frameworks/av/media/libaudiohal/2.0/ |
H A D | EffectBufferHalHidl.cpp | 60 mHidlBuffer.frameCount = 0; 106 void EffectBufferHalHidl::setFrameCount(size_t frameCount) { argument 107 mHidlBuffer.frameCount = frameCount; 108 mAudioBuffer.frameCount = frameCount;
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/frameworks/av/media/libaudiohal/4.0/ |
H A D | EffectBufferHalHidl.cpp | 61 mHidlBuffer.frameCount = 0; 107 void EffectBufferHalHidl::setFrameCount(size_t frameCount) { argument 108 mHidlBuffer.frameCount = frameCount; 109 mAudioBuffer.frameCount = frameCount;
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/frameworks/av/media/libaudioclient/tests/ |
H A D | test_create_audiotrack.cpp | 64 size_t frameCount; local 82 &frameCount, ¬ificationFrames, &useSharedBuffer, 92 audio_bytes_per_sample(format) * frameCount; 95 frameCount = 0; 118 frameCount,
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/frameworks/av/include/private/media/ |
H A D | AudioTrackShared.h | 207 Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, bool isOut, 218 size_t frameCount() const { return mFrameCount; } function in class:android::Proxy 240 ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 360 AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 362 : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, 417 StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, 473 AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 475 : ClientProxy(cblk, buffers, frameCount, frameSize, 495 ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize, 570 AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, argument 688 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/native/services/surfaceflinger/tests/fakehwc/ |
H A D | FakeComposerUtils.h | 109 int frameCount = mComposer.getFrameCount(); local 113 mComposer.waitUntilFrame(frameCount + 1); 114 LOG_ALWAYS_FATAL_IF(frameCount + 1 != mComposer.getFrameCount(), 116 mComposer.getFrameCount() - frameCount);
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