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Diffstat (limited to 'services/audioflinger/AudioResamplerSinc.cpp')
| -rw-r--r-- | services/audioflinger/AudioResamplerSinc.cpp | 358 |
1 files changed, 358 insertions, 0 deletions
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp new file mode 100644 index 0000000..9e5e254 --- /dev/null +++ b/services/audioflinger/AudioResamplerSinc.cpp @@ -0,0 +1,358 @@ +/* + * Copyright (C) 2007 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#include <string.h> +#include "AudioResamplerSinc.h" + +namespace android { +// ---------------------------------------------------------------------------- + + +/* + * These coeficients are computed with the "fir" utility found in + * tools/resampler_tools + * TODO: A good optimization would be to transpose this matrix, to take + * better advantage of the data-cache. + */ +const int32_t AudioResamplerSinc::mFirCoefsUp[] = { + 0x7fffffff, 0x7f15d078, 0x7c5e0da6, 0x77ecd867, 0x71e2e251, 0x6a6c304a, 0x61be7269, 0x58170412, 0x4db8ab05, 0x42e92ea6, 0x37eee214, 0x2d0e3bb1, 0x22879366, 0x18951e95, 0x0f693d0d, 0x072d2621, + 0x00000000, 0xf9f66655, 0xf51a5fd7, 0xf16bbd84, 0xeee0d9ac, 0xed67a922, 0xece70de6, 0xed405897, 0xee50e505, 0xeff3be30, 0xf203370f, 0xf45a6741, 0xf6d67d53, 0xf957db66, 0xfbc2f647, 0xfe00f2b9, + 0x00000000, 0x01b37218, 0x0313a0c6, 0x041d930d, 0x04d28057, 0x053731b0, 0x05534dff, 0x05309bfd, 0x04da440d, 0x045c1aee, 0x03c1fcdd, 0x03173ef5, 0x02663ae8, 0x01b7f736, 0x0113ec79, 0x007fe6a9, + 0x00000000, 0xff96b229, 0xff44f99f, 0xff0a86be, 0xfee5f803, 0xfed518fd, 0xfed521fd, 0xfee2f4fd, 0xfefb54f8, 0xff1b159b, 0xff3f4203, 0xff6539e0, 0xff8ac502, 0xffae1ddd, 0xffcdf3f9, 0xffe96798, + 0x00000000, 0x00119de6, 0x001e6b7e, 0x0026cb7a, 0x002b4830, 0x002c83d6, 0x002b2a82, 0x0027e67a, 0x002356f9, 0x001e098e, 0x001875e4, 0x0012fbbe, 0x000de2d1, 0x00095c10, 0x00058414, 0x00026636, + 0x00000000, 0xfffe44a9, 0xfffd206d, 0xfffc7b7f, 0xfffc3c8f, 0xfffc4ac2, 0xfffc8f2b, 0xfffcf5c4, 0xfffd6df3, 0xfffdeab2, 0xfffe6275, 0xfffececf, 0xffff2c07, 0xffff788c, 0xffffb471, 0xffffe0f2, + 0x00000000, 0x000013e6, 0x00001f03, 0x00002396, 0x00002399, 0x000020b6, 0x00001c3c, 0x00001722, 0x00001216, 0x00000d81, 0x0000099c, 0x0000067c, 0x00000419, 0x0000025f, 0x00000131, 0x00000070, + 0x00000000, 0xffffffc7, 0xffffffb3, 0xffffffb3, 0xffffffbe, 0xffffffcd, 0xffffffdb, 0xffffffe7, 0xfffffff0, 0xfffffff7, 0xfffffffb, 0xfffffffe, 0xffffffff, 0x00000000, 0x00000000, 0x00000000, + 0x00000000 // this one is needed for lerping the last coefficient +}; + +/* + * These coefficients are optimized for 48KHz -> 44.1KHz (stop-band at 22.050KHz) + * It's possible to use the above coefficient for any down-sampling + * at the expense of a slower processing loop (we can interpolate + * these coefficient from the above by "Stretching" them in time). + */ +const int32_t AudioResamplerSinc::mFirCoefsDown[] = { + 0x7fffffff, 0x7f55e46d, 0x7d5b4c60, 0x7a1b4b98, 0x75a7fb14, 0x7019f0bd, 0x698f875a, 0x622bfd59, 0x5a167256, 0x5178cc54, 0x487e8e6c, 0x3f53aae8, 0x36235ad4, 0x2d17047b, 0x245539ab, 0x1c00d540, + 0x14383e57, 0x0d14d5ca, 0x06aa910b, 0x0107c38b, 0xfc351654, 0xf835abae, 0xf5076b45, 0xf2a37202, 0xf0fe9faa, 0xf00a3bbd, 0xefb4aa81, 0xefea2b05, 0xf0959716, 0xf1a11e83, 0xf2f6f7a0, 0xf481fff4, + 0xf62e48ce, 0xf7e98ca5, 0xf9a38b4c, 0xfb4e4bfa, 0xfcde456f, 0xfe4a6d30, 0xff8c2fdf, 0x009f5555, 0x0181d393, 0x0233940f, 0x02b62f06, 0x030ca07d, 0x033afa62, 0x03461725, 0x03334f83, 0x030835fa, + 0x02ca59cc, 0x027f12d1, 0x022b570d, 0x01d39a49, 0x017bb78f, 0x0126e414, 0x00d7aaaf, 0x008feec7, 0x0050f584, 0x001b73e3, 0xffefa063, 0xffcd46ed, 0xffb3ddcd, 0xffa29aaa, 0xff988691, 0xff949066, + 0xff959d24, 0xff9a959e, 0xffa27195, 0xffac4011, 0xffb72d2b, 0xffc28569, 0xffcdb706, 0xffd85171, 0xffe20364, 0xffea97e9, 0xfff1f2b2, 0xfff80c06, 0xfffcec92, 0x0000a955, 0x00035fd8, 0x000532cf, + 0x00064735, 0x0006c1f9, 0x0006c62d, 0x000673ba, 0x0005e68f, 0x00053630, 0x000475a3, 0x0003b397, 0x0002fac1, 0x00025257, 0x0001be9e, 0x0001417a, 0x0000dafd, 0x000089eb, 0x00004c28, 0x00001f1d, + 0x00000000, 0xffffec10, 0xffffe0be, 0xffffdbc5, 0xffffdb39, 0xffffdd8b, 0xffffe182, 0xffffe638, 0xffffeb0a, 0xffffef8f, 0xfffff38b, 0xfffff6e3, 0xfffff993, 0xfffffba6, 0xfffffd30, 0xfffffe4a, + 0xffffff09, 0xffffff85, 0xffffffd1, 0xfffffffb, 0x0000000f, 0x00000016, 0x00000015, 0x00000012, 0x0000000d, 0x00000009, 0x00000006, 0x00000003, 0x00000002, 0x00000001, 0x00000000, 0x00000000, + 0x00000000 // this one is needed for lerping the last coefficient +}; + +// ---------------------------------------------------------------------------- + +static inline +int32_t mulRL(int left, int32_t in, uint32_t vRL) +{ +#if defined(__arm__) && !defined(__thumb__) + int32_t out; + if (left) { + asm( "smultb %[out], %[in], %[vRL] \n" + : [out]"=r"(out) + : [in]"%r"(in), [vRL]"r"(vRL) + : ); + } else { + asm( "smultt %[out], %[in], %[vRL] \n" + : [out]"=r"(out) + : [in]"%r"(in), [vRL]"r"(vRL) + : ); + } + return out; +#else + if (left) { + return int16_t(in>>16) * int16_t(vRL&0xFFFF); + } else { + return int16_t(in>>16) * int16_t(vRL>>16); + } +#endif +} + +static inline +int32_t mulAdd(int16_t in, int32_t v, int32_t a) +{ +#if defined(__arm__) && !defined(__thumb__) + int32_t out; + asm( "smlawb %[out], %[v], %[in], %[a] \n" + : [out]"=r"(out) + : [in]"%r"(in), [v]"r"(v), [a]"r"(a) + : ); + return out; +#else + return a + in * (v>>16); + // improved precision + // return a + in * (v>>16) + ((in * (v & 0xffff)) >> 16); +#endif +} + +static inline +int32_t mulAddRL(int left, uint32_t inRL, int32_t v, int32_t a) +{ +#if defined(__arm__) && !defined(__thumb__) + int32_t out; + if (left) { + asm( "smlawb %[out], %[v], %[inRL], %[a] \n" + : [out]"=r"(out) + : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a) + : ); + } else { + asm( "smlawt %[out], %[v], %[inRL], %[a] \n" + : [out]"=r"(out) + : [inRL]"%r"(inRL), [v]"r"(v), [a]"r"(a) + : ); + } + return out; +#else + if (left) { + return a + (int16_t(inRL&0xFFFF) * (v>>16)); + //improved precision + // return a + (int16_t(inRL&0xFFFF) * (v>>16)) + ((int16_t(inRL&0xFFFF) * (v & 0xffff)) >> 16); + } else { + return a + (int16_t(inRL>>16) * (v>>16)); + } +#endif +} + +// ---------------------------------------------------------------------------- + +AudioResamplerSinc::AudioResamplerSinc(int bitDepth, + int inChannelCount, int32_t sampleRate) + : AudioResampler(bitDepth, inChannelCount, sampleRate), + mState(0) +{ + /* + * Layout of the state buffer for 32 tap: + * + * "present" sample beginning of 2nd buffer + * v v + * 0 01 2 23 3 + * 0 F0 0 F0 F + * [pppppppppppppppInnnnnnnnnnnnnnnnpppppppppppppppInnnnnnnnnnnnnnnn] + * ^ ^ head + * + * p = past samples, convoluted with the (p)ositive side of sinc() + * n = future samples, convoluted with the (n)egative side of sinc() + * r = extra space for implementing the ring buffer + * + */ + + const size_t numCoefs = 2*halfNumCoefs; + const size_t stateSize = numCoefs * inChannelCount * 2; + mState = new int16_t[stateSize]; + memset(mState, 0, sizeof(int16_t)*stateSize); + mImpulse = mState + (halfNumCoefs-1)*inChannelCount; + mRingFull = mImpulse + (numCoefs+1)*inChannelCount; +} + +AudioResamplerSinc::~AudioResamplerSinc() +{ + delete [] mState; +} + +void AudioResamplerSinc::init() { +} + +void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, + AudioBufferProvider* provider) +{ + mFirCoefs = (mInSampleRate <= mSampleRate) ? mFirCoefsUp : mFirCoefsDown; + + // select the appropriate resampler + switch (mChannelCount) { + case 1: + resample<1>(out, outFrameCount, provider); + break; + case 2: + resample<2>(out, outFrameCount, provider); + break; + } +} + + +template<int CHANNELS> +void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, + AudioBufferProvider* provider) +{ + int16_t* impulse = mImpulse; + uint32_t vRL = mVolumeRL; + size_t inputIndex = mInputIndex; + uint32_t phaseFraction = mPhaseFraction; + uint32_t phaseIncrement = mPhaseIncrement; + size_t outputIndex = 0; + size_t outputSampleCount = outFrameCount * 2; + size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; + + AudioBufferProvider::Buffer& buffer(mBuffer); + while (outputIndex < outputSampleCount) { + // buffer is empty, fetch a new one + while (buffer.frameCount == 0) { + buffer.frameCount = inFrameCount; + provider->getNextBuffer(&buffer); + if (buffer.raw == NULL) { + goto resample_exit; + } + const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; + if (phaseIndex == 1) { + // read one frame + read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex); + } else if (phaseIndex == 2) { + // read 2 frames + read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex); + inputIndex++; + if (inputIndex >= mBuffer.frameCount) { + inputIndex -= mBuffer.frameCount; + provider->releaseBuffer(&buffer); + } else { + read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex); + } + } + } + int16_t *in = buffer.i16; + const size_t frameCount = buffer.frameCount; + + // Always read-in the first samples from the input buffer + int16_t* head = impulse + halfNumCoefs*CHANNELS; + head[0] = in[inputIndex*CHANNELS + 0]; + if (CHANNELS == 2) + head[1] = in[inputIndex*CHANNELS + 1]; + + // handle boundary case + int32_t l, r; + while (outputIndex < outputSampleCount) { + filterCoefficient<CHANNELS>(l, r, phaseFraction, impulse); + out[outputIndex++] += 2 * mulRL(1, l, vRL); + out[outputIndex++] += 2 * mulRL(0, r, vRL); + + phaseFraction += phaseIncrement; + const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; + if (phaseIndex == 1) { + inputIndex++; + if (inputIndex >= frameCount) + break; // need a new buffer + read<CHANNELS>(impulse, phaseFraction, in, inputIndex); + } else if(phaseIndex == 2) { // maximum value + inputIndex++; + if (inputIndex >= frameCount) + break; // 0 frame available, 2 frames needed + // read first frame + read<CHANNELS>(impulse, phaseFraction, in, inputIndex); + inputIndex++; + if (inputIndex >= frameCount) + break; // 0 frame available, 1 frame needed + // read second frame + read<CHANNELS>(impulse, phaseFraction, in, inputIndex); + } + } + + // if done with buffer, save samples + if (inputIndex >= frameCount) { + inputIndex -= frameCount; + provider->releaseBuffer(&buffer); + } + } + +resample_exit: + mImpulse = impulse; + mInputIndex = inputIndex; + mPhaseFraction = phaseFraction; +} + +template<int CHANNELS> +/*** +* read() +* +* This function reads only one frame from input buffer and writes it in +* state buffer +* +**/ +void AudioResamplerSinc::read( + int16_t*& impulse, uint32_t& phaseFraction, + int16_t const* in, size_t inputIndex) +{ + const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; + impulse += CHANNELS; + phaseFraction -= 1LU<<kNumPhaseBits; + if (impulse >= mRingFull) { + const size_t stateSize = (halfNumCoefs*2)*CHANNELS; + memcpy(mState, mState+stateSize, sizeof(int16_t)*stateSize); + impulse -= stateSize; + } + int16_t* head = impulse + halfNumCoefs*CHANNELS; + head[0] = in[inputIndex*CHANNELS + 0]; + if (CHANNELS == 2) + head[1] = in[inputIndex*CHANNELS + 1]; +} + +template<int CHANNELS> +void AudioResamplerSinc::filterCoefficient( + int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples) +{ + // compute the index of the coefficient on the positive side and + // negative side + uint32_t indexP = (phase & cMask) >> cShift; + uint16_t lerpP = (phase & pMask) >> pShift; + uint32_t indexN = (-phase & cMask) >> cShift; + uint16_t lerpN = (-phase & pMask) >> pShift; + if ((indexP == 0) && (lerpP == 0)) { + indexN = cMask >> cShift; + lerpN = pMask >> pShift; + } + + l = 0; + r = 0; + int32_t const* coefs = mFirCoefs; + int16_t const *sP = samples; + int16_t const *sN = samples+CHANNELS; + for (unsigned int i=0 ; i<halfNumCoefs/4 ; i++) { + interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); + interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); + sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits; + interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); + interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); + sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits; + interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); + interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); + sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits; + interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); + interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); + sP -= CHANNELS; sN += CHANNELS; coefs += 1<<coefsBits; + } +} + +template<int CHANNELS> +void AudioResamplerSinc::interpolate( + int32_t& l, int32_t& r, + int32_t const* coefs, int16_t lerp, int16_t const* samples) +{ + int32_t c0 = coefs[0]; + int32_t c1 = coefs[1]; + int32_t sinc = mulAdd(lerp, (c1-c0)<<1, c0); + if (CHANNELS == 2) { + uint32_t rl = *reinterpret_cast<uint32_t const*>(samples); + l = mulAddRL(1, rl, sinc, l); + r = mulAddRL(0, rl, sinc, r); + } else { + r = l = mulAdd(samples[0], sinc, l); + } +} + +// ---------------------------------------------------------------------------- +}; // namespace android + |