[Output] [Downmixer] Optimize a bit
Rewrite some of the output and a lot of the downmixer to use Accelerate framework instead of dumb for loops. Signed-off-by: Christopher Snowhill <kode54@gmail.com>swiftingly
Christopher Snowhill
parent
0ba766023e
commit
18af8a06df
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@ -12,6 +12,8 @@
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#import "AudioChunk.h"
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#import <Accelerate/Accelerate.h>
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static void downmix_to_stereo(const float *inBuffer, int channels, uint32_t config, float *outBuffer, size_t count) {
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float FrontRatios[2] = { 0.0F, 0.0F };
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float FrontCenterRatio = 0.0F;
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@ -71,73 +73,77 @@ static void downmix_to_stereo(const float *inBuffer, int channels, uint32_t conf
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channelIndexes[i] = [AudioChunk findChannelIndex:[AudioChunk extractChannelFlag:i fromConfig:config]];
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}
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for(size_t i = 0; i < count; ++i) {
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float left = 0.0F, right = 0.0F;
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for(uint32_t j = 0; j < channels; ++j) {
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float inSample = inBuffer[i * channels + j];
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switch(channelIndexes[j]) {
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case 0:
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left += inSample * FrontRatios[0];
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right += inSample * FrontRatios[1];
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break;
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vDSP_vclr(outBuffer, 1, count * 2);
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case 1:
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left += inSample * FrontRatios[1];
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right += inSample * FrontRatios[0];
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break;
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float tempBuffer[count * 2];
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case 2:
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left += inSample * FrontCenterRatio;
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right += inSample * FrontCenterRatio;
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break;
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for(uint32_t i = 0; i < channels; ++i) {
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float leftRatio = 0.0F;
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float rightRatio = 0.0F;
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switch(channelIndexes[i]) {
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case 0:
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leftRatio = FrontRatios[0];
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rightRatio = FrontRatios[1];
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break;
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case 3:
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left += inSample * LFERatio;
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right += inSample * LFERatio;
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break;
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case 1:
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leftRatio = FrontRatios[1];
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rightRatio = FrontRatios[0];
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break;
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case 4:
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left += inSample * BackRatios[0];
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right += inSample * BackRatios[1];
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break;
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case 2:
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leftRatio = FrontCenterRatio;
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rightRatio = FrontCenterRatio;
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break;
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case 5:
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left += inSample * BackRatios[1];
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right += inSample * BackRatios[0];
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break;
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case 3:
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leftRatio = LFERatio;
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rightRatio = LFERatio;
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break;
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case 6:
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case 7:
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break;
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case 4:
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leftRatio = BackRatios[0];
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rightRatio = BackRatios[1];
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break;
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case 8:
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left += inSample * BackCenterRatio;
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right += inSample * BackCenterRatio;
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break;
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case 5:
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leftRatio = BackRatios[1];
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rightRatio = BackRatios[0];
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break;
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case 9:
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left += inSample * SideRatios[0];
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right += inSample * SideRatios[1];
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break;
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case 6:
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case 7:
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break;
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case 10:
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left += inSample * SideRatios[1];
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right += inSample * SideRatios[0];
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break;
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case 8:
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leftRatio = BackCenterRatio;
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rightRatio = BackCenterRatio;
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break;
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case 11:
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case 12:
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case 13:
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case 14:
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case 15:
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case 16:
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case 17:
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default:
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break;
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}
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case 9:
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leftRatio = SideRatios[0];
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rightRatio = SideRatios[1];
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break;
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case 10:
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leftRatio = SideRatios[1];
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rightRatio = SideRatios[0];
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break;
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case 11:
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case 12:
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case 13:
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case 14:
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case 15:
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case 16:
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case 17:
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default:
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break;
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}
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outBuffer[i * 2 + 0] = left;
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outBuffer[i * 2 + 1] = right;
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vDSP_vsmul(inBuffer + i, channels, &leftRatio, tempBuffer, 1, count);
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vDSP_vsmul(inBuffer + i, channels, &rightRatio, tempBuffer + count, 1, count);
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vDSP_vadd(outBuffer, 2, tempBuffer, 1, outBuffer, 2, count);
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vDSP_vadd(outBuffer + 1, 2, tempBuffer + count, 1, outBuffer + 1, 2, count);
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}
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}
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@ -147,13 +153,8 @@ static void downmix_to_mono(const float *inBuffer, int channels, uint32_t config
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inBuffer = tempBuffer;
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channels = 2;
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config = AudioConfigStereo;
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for(size_t i = 0; i < count; ++i) {
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float sample = 0;
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for(int j = 0; j < channels; ++j) {
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sample += inBuffer[i * channels + j];
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}
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outBuffer[i] = sample;
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}
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cblas_scopy((int)count, inBuffer, 2, outBuffer, 1);
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vDSP_vadd(outBuffer, 1, inBuffer + 1, 2, outBuffer, 1, count);
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}
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static void upmix(const float *inBuffer, int inchannels, uint32_t inconfig, float *outBuffer, int outchannels, uint32_t outconfig, size_t count) {
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@ -15,6 +15,8 @@
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#import "Logging.h"
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#import <Accelerate/Accelerate.h>
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extern void scale_by_volume(float *buffer, size_t count, float volume);
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static NSString *CogPlaybackDidBeginNotficiation = @"CogPlaybackDidBeginNotficiation";
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@ -27,11 +29,7 @@ static void fillBuffers(AudioBufferList *ioData, float *inbuffer, size_t count,
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const size_t maxCount = (ioData->mBuffers[i].mDataByteSize / sizeof(float)) - offset;
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float *output = ((float *)ioData->mBuffers[i].mData) + offset;
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const float *input = inbuffer + i;
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for(size_t j = 0, k = (count > maxCount) ? maxCount : count; j < k; ++j) {
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*output = *input;
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output++;
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input += channels;
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}
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cblas_scopy((int)((count > maxCount) ? maxCount : count), input, (int)channels, output, 1);
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ioData->mBuffers[i].mNumberChannels = 1;
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}
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}
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@ -724,11 +722,7 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
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float *outBuffer = ((float *)inputData->mBuffers[0].mData) + i;
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const float *inBuffer = ((float *)ioData->mBuffers[i].mData);
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const int frameCount = ioData->mBuffers[i].mDataByteSize / sizeof(float);
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for(int j = 0; j < frameCount; ++j) {
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*outBuffer = *inBuffer;
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inBuffer++;
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outBuffer += channels;
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}
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cblas_scopy(frameCount, inBuffer, 1, outBuffer, channels);
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}
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#ifdef OUTPUT_LOG
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