| /frameworks/av/media/libaudioprocessing/ |
| D | AudioResampler.cpp | 64 size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
67 size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
343 size_t inputIndex = mInputIndex; in resampleStereo16() local
364 if (mBuffer.frameCount > inputIndex) break; in resampleStereo16()
366 inputIndex -= mBuffer.frameCount; in resampleStereo16()
376 while (inputIndex == 0) { in resampleStereo16()
380 Advance(&inputIndex, &phaseFraction, phaseIncrement); in resampleStereo16()
390 if (inputIndex + 2 < mBuffer.frameCount) { in resampleStereo16()
396 AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr, in resampleStereo16()
401 while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) { in resampleStereo16()
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| D | AudioResamplerCubic.cpp | 59 size_t inputIndex = mInputIndex; in resampleStereo16() local
94 inputIndex++; in resampleStereo16()
95 if (inputIndex == mBuffer.frameCount) { in resampleStereo16()
96 inputIndex = 0; in resampleStereo16()
108 advance(&left, in[inputIndex*2]); in resampleStereo16()
109 advance(&right, in[inputIndex*2+1]); in resampleStereo16()
115 mInputIndex = inputIndex; in resampleStereo16()
126 size_t inputIndex = mInputIndex; in resampleMono16() local
162 inputIndex++; in resampleMono16()
163 if (inputIndex == mBuffer.frameCount) { in resampleMono16()
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| D | AudioResamplerDyn.cpp | 128 const TI* const in, const size_t inputIndex) in readAgain() argument
132 head[i] = in[inputIndex*CHANNELS + i]; in readAgain()
140 const TI* const in, const size_t inputIndex) in readAdvance() argument
149 readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex); in readAdvance()
505 size_t inputIndex = 0; in resample() local
533 ALOG_ASSERT(inputIndex <= mBuffer.frameCount, "inputIndex%zu > frameCount%zu", in resample()
534 inputIndex, mBuffer.frameCount); in resample()
550 reinterpret_cast<TI*>(mBuffer.raw), inputIndex); in resample()
551 inputIndex++; in resample()
554 if (inputIndex >= mBuffer.frameCount) { in resample()
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| D | AudioResamplerSinc.cpp | 301 size_t inputIndex = mInputIndex; in resample() local
319 read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex); in resample()
322 read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex); in resample()
323 inputIndex++; in resample()
324 if (inputIndex >= mBuffer.frameCount) { in resample()
325 inputIndex -= mBuffer.frameCount; in resample()
328 read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex); in resample()
338 head[i] = in[inputIndex*CHANNELS + i]; in resample()
349 inputIndex++; in resample()
350 if (inputIndex >= frameCount) { in resample()
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| D | AudioResamplerDyn.h | 93 const TI* const in, const size_t inputIndex);
97 const TI* const in, const size_t inputIndex);
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| D | AudioResamplerSinc.h | 65 const int16_t* in, size_t inputIndex);
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| /frameworks/base/media/mca/filterpacks/java/android/filterpacks/videoproc/ |
| D | BackDropperFilter.java | 714 int inputIndex = mPingPong ? 0 : 1; in process() local
736 copyShaderProgram.process(mVideoInput, mBgMean[inputIndex]); in process()
741 Frame[] distInputs = { mVideoInput, mBgMean[inputIndex], mBgVariance[inputIndex] }; in process()
771 Frame[] maskVerifyInputs = {mMaskVerify[inputIndex], mMask}; in process()
812 Frame[] meanUpdateInputs = { mVideoInput, mBgMean[inputIndex], mMask }; in process()
819 mVideoInput, mBgMean[inputIndex], mBgVariance[inputIndex], mMask in process()
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| /frameworks/av/media/libstagefright/ |
| D | StagefrightMetadataRetriever.cpp | 262 size_t inputIndex = -1; in extractVideoFrame() local
268 err = decoder->dequeueInputBuffer(&inputIndex, kBufferTimeOutUs); in extractVideoFrame()
276 codecBuffer = inputBuffers[inputIndex]; in extractVideoFrame()
316 if (err == OK && inputIndex < inputBuffers.size()) { in extractVideoFrame()
320 inputIndex, in extractVideoFrame()
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