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      1 /*
      2  * Copyright (C) 2007 The Android Open Source Project
      3  *
      4  * Licensed under the Apache License, Version 2.0 (the "License");
      5  * you may not use this file except in compliance with the License.
      6  * You may obtain a copy of the License at
      7  *
      8  *      http://www.apache.org/licenses/LICENSE-2.0
      9  *
     10  * Unless required by applicable law or agreed to in writing, software
     11  * distributed under the License is distributed on an "AS IS" BASIS,
     12  * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
     13  * See the License for the specific language governing permissions and
     14  * limitations under the License.
     15  */
     16 
     17 #define LOG_TAG "AudioResamplerCubic"
     18 
     19 #include <stdint.h>
     20 #include <string.h>
     21 #include <sys/types.h>
     22 
     23 #include <log/log.h>
     24 
     25 #include "AudioResamplerCubic.h"
     26 
     27 namespace android {
     28 // ----------------------------------------------------------------------------
     29 
     30 void AudioResamplerCubic::init() {
     31     memset(&left, 0, sizeof(state));
     32     memset(&right, 0, sizeof(state));
     33 }
     34 
     35 size_t AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
     36         AudioBufferProvider* provider) {
     37 
     38     // should never happen, but we overflow if it does
     39     // ALOG_ASSERT(outFrameCount < 32767);
     40 
     41     // select the appropriate resampler
     42     switch (mChannelCount) {
     43     case 1:
     44         return resampleMono16(out, outFrameCount, provider);
     45     case 2:
     46         return resampleStereo16(out, outFrameCount, provider);
     47     default:
     48         LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
     49         return 0;
     50     }
     51 }
     52 
     53 size_t AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
     54         AudioBufferProvider* provider) {
     55 
     56     int32_t vl = mVolume[0];
     57     int32_t vr = mVolume[1];
     58 
     59     size_t inputIndex = mInputIndex;
     60     uint32_t phaseFraction = mPhaseFraction;
     61     uint32_t phaseIncrement = mPhaseIncrement;
     62     size_t outputIndex = 0;
     63     size_t outputSampleCount = outFrameCount * 2;
     64     size_t inFrameCount = getInFrameCountRequired(outFrameCount);
     65 
     66     // fetch first buffer
     67     if (mBuffer.frameCount == 0) {
     68         mBuffer.frameCount = inFrameCount;
     69         provider->getNextBuffer(&mBuffer);
     70         if (mBuffer.raw == NULL) {
     71             return 0;
     72         }
     73         // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
     74     }
     75     int16_t *in = mBuffer.i16;
     76 
     77     while (outputIndex < outputSampleCount) {
     78         int32_t x;
     79 
     80         // calculate output sample
     81         x = phaseFraction >> kPreInterpShift;
     82         out[outputIndex++] += vl * interp(&left, x);
     83         out[outputIndex++] += vr * interp(&right, x);
     84         // out[outputIndex++] += vr * in[inputIndex*2];
     85 
     86         // increment phase
     87         phaseFraction += phaseIncrement;
     88         uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
     89         phaseFraction &= kPhaseMask;
     90 
     91         // time to fetch another sample
     92         while (indexIncrement--) {
     93 
     94             inputIndex++;
     95             if (inputIndex == mBuffer.frameCount) {
     96                 inputIndex = 0;
     97                 provider->releaseBuffer(&mBuffer);
     98                 mBuffer.frameCount = inFrameCount;
     99                 provider->getNextBuffer(&mBuffer);
    100                 if (mBuffer.raw == NULL) {
    101                     goto save_state;  // ugly, but efficient
    102                 }
    103                 in = mBuffer.i16;
    104                 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
    105             }
    106 
    107             // advance sample state
    108             advance(&left, in[inputIndex*2]);
    109             advance(&right, in[inputIndex*2+1]);
    110         }
    111     }
    112 
    113 save_state:
    114     // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
    115     mInputIndex = inputIndex;
    116     mPhaseFraction = phaseFraction;
    117     return outputIndex / 2 /* channels for stereo */;
    118 }
    119 
    120 size_t AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
    121         AudioBufferProvider* provider) {
    122 
    123     int32_t vl = mVolume[0];
    124     int32_t vr = mVolume[1];
    125 
    126     size_t inputIndex = mInputIndex;
    127     uint32_t phaseFraction = mPhaseFraction;
    128     uint32_t phaseIncrement = mPhaseIncrement;
    129     size_t outputIndex = 0;
    130     size_t outputSampleCount = outFrameCount * 2;
    131     size_t inFrameCount = getInFrameCountRequired(outFrameCount);
    132 
    133     // fetch first buffer
    134     if (mBuffer.frameCount == 0) {
    135         mBuffer.frameCount = inFrameCount;
    136         provider->getNextBuffer(&mBuffer);
    137         if (mBuffer.raw == NULL) {
    138             return 0;
    139         }
    140         // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
    141     }
    142     int16_t *in = mBuffer.i16;
    143 
    144     while (outputIndex < outputSampleCount) {
    145         int32_t sample;
    146         int32_t x;
    147 
    148         // calculate output sample
    149         x = phaseFraction >> kPreInterpShift;
    150         sample = interp(&left, x);
    151         out[outputIndex++] += vl * sample;
    152         out[outputIndex++] += vr * sample;
    153 
    154         // increment phase
    155         phaseFraction += phaseIncrement;
    156         uint32_t indexIncrement = (phaseFraction >> kNumPhaseBits);
    157         phaseFraction &= kPhaseMask;
    158 
    159         // time to fetch another sample
    160         while (indexIncrement--) {
    161 
    162             inputIndex++;
    163             if (inputIndex == mBuffer.frameCount) {
    164                 inputIndex = 0;
    165                 provider->releaseBuffer(&mBuffer);
    166                 mBuffer.frameCount = inFrameCount;
    167                 provider->getNextBuffer(&mBuffer);
    168                 if (mBuffer.raw == NULL) {
    169                     goto save_state;  // ugly, but efficient
    170                 }
    171                 // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
    172                 in = mBuffer.i16;
    173             }
    174 
    175             // advance sample state
    176             advance(&left, in[inputIndex]);
    177         }
    178     }
    179 
    180 save_state:
    181     // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
    182     mInputIndex = inputIndex;
    183     mPhaseFraction = phaseFraction;
    184     return outputIndex;
    185 }
    186 
    187 // ----------------------------------------------------------------------------
    188 } // namespace android
    189