Cog Audio: Resampler now extrapolates into latency padding for better gapless playback
Christopher Snowhill
parent
7d4841b1c6
commit
ed882e25cb
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@ -13,6 +13,8 @@
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#import <audio/conversion/s16_to_float.h>
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#import <audio/conversion/s32_to_float.h>
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#import "lpc.h"
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void PrintStreamDesc (AudioStreamBasicDescription *inDesc)
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{
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if (!inDesc) {
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@ -368,6 +370,32 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
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}
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}
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static const int extrapolate_order = 16;
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static void extrapolate(float *buffer, size_t channels, size_t frameSize, size_t size, BOOL backward)
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{
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const size_t delta = (backward ? -1 : 1) * channels;
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float lpc[extrapolate_order];
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float work[frameSize];
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for (size_t ch = 0; ch < channels; ch++)
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{
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if (frameSize - size > extrapolate_order * 2)
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{
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float *chPcmBuf = buffer + ch + (backward ? frameSize : -1) * channels;
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for (size_t i = 0; i < frameSize; i++) work[i] = *(chPcmBuf += delta);
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vorbis_lpc_from_data(work, lpc, (int)(frameSize - size), extrapolate_order);
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vorbis_lpc_predict(lpc, work + frameSize - size - extrapolate_order, extrapolate_order, work + frameSize - size, size);
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chPcmBuf = buffer + ch + (backward ? frameSize : -1) * channels;
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for (size_t i = 0; i < frameSize; i++) *(chPcmBuf += delta) = work[i];
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}
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}
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}
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-(void)process
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{
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char writeBuf[CHUNK_SIZE];
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@ -384,7 +412,11 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
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UInt32 ioNumberPackets;
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int amountReadFromFC;
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int amountRead = 0;
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int extrapolateStart = 0;
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int extrapolateEnd = 0;
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size_t amountToSkip = 0;
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BOOL inputRetry = NO;
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if (emittingSilence)
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{
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memset(dest, 0, amount);
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@ -416,6 +448,9 @@ tryagain:
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if (!inputBuffer || inputBufferSize < newSize)
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inputBuffer = realloc( inputBuffer, inputBufferSize = newSize * 3 );
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// Pad end of track with floats. For simplicity, pad start in track
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// native format.
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if (stopping || [self shouldContinue] == NO || [self endOfStream] == YES)
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{
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if (!skipResampler && !latencyPostfill)
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@ -425,6 +460,12 @@ tryagain:
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if (!inputBuffer || inputBufferSize < newSize)
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inputBuffer = realloc( inputBuffer, inputBufferSize = newSize * 64);
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extrapolateEnd = ioNumberPackets;
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// Extrapolate end samples
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if (inpSize)
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extrapolate( inputBuffer, floatFormat.mChannelsPerFrame, inpSize / floatFormat.mBytesPerPacket, extrapolateEnd, NO);
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inpSize = newSize;
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inpOffset = 0;
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latencyPostfill = YES;
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@ -438,13 +479,17 @@ tryagain:
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}
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size_t amountToWrite = ioNumberPackets * inputFormat.mBytesPerPacket;
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size_t amountToSkip = 0;
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if (!inputRetry) amountToSkip = 0;
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BOOL isFloat = !!(inputFormat.mFormatFlags & kAudioFormatFlagIsFloat);
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BOOL isUnsigned = !isFloat && !(inputFormat.mFormatFlags & kAudioFormatFlagIsSignedInteger);
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if (!skipResampler)
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{
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if (latencyStarted < 0)
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{
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latencyStarted = resampler->latency(resampler_data);
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extrapolateStart = (int)latencyStarted;
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}
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if (latencyStarted)
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@ -452,8 +497,8 @@ tryagain:
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size_t latencyToWrite = latencyStarted * inputFormat.mBytesPerPacket;
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if (latencyToWrite > amountToWrite)
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latencyToWrite = amountToWrite;
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if (inputFormat.mBitsPerChannel <= 8)
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if (isUnsigned)
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memset(inputBuffer, 0x80, latencyToWrite);
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else
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memset(inputBuffer, 0, latencyToWrite);
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@ -467,9 +512,15 @@ tryagain:
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}
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}
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size_t bytesReadFromInput = [self readData:inputBuffer + amountToSkip amount:(int)amountToWrite] + amountToSkip;
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size_t bytesReadFromInput = [self readData:inputBuffer + amountToSkip amount:(int)amountToWrite];
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BOOL isFloat = !!(inputFormat.mFormatFlags & kAudioFormatFlagIsFloat);
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if (!bytesReadFromInput)
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{
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inputRetry = YES;
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continue;
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}
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bytesReadFromInput += amountToSkip;
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if (bytesReadFromInput &&
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(inputFormat.mFormatFlags & kAudioFormatFlagIsBigEndian))
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@ -490,7 +541,6 @@ tryagain:
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if (bytesReadFromInput && !isFloat)
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{
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size_t bitsPerSample = inputFormat.mBitsPerChannel;
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BOOL isUnsigned = !(inputFormat.mFormatFlags & kAudioFormatFlagIsSignedInteger);
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if (bitsPerSample <= 8) {
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samplesRead = bytesReadFromInput;
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if (isUnsigned)
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@ -537,6 +587,13 @@ tryagain:
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}
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}
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// Extrapolate start
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if (extrapolateStart)
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{
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extrapolate( inputBuffer, floatFormat.mChannelsPerFrame, bytesReadFromInput / floatFormat.mBytesPerPacket, extrapolateStart, YES);
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extrapolateStart = 0;
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}
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// Input now contains bytesReadFromInput worth of floats, in the input sample rate
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inpSize = bytesReadFromInput;
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inpOffset = 0;
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@ -43,56 +43,58 @@
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runOnlyForDeploymentPostprocessing = 0;
|
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};
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||||
|
|
|
|||
|
|
@ -0,0 +1,156 @@
|
|||
/********************************************************************
|
||||
* *
|
||||
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
|
||||
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
|
||||
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
|
||||
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
|
||||
* *
|
||||
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
|
||||
* by the Xiph.Org Foundation http://www.xiph.org/ *
|
||||
* *
|
||||
********************************************************************
|
||||
|
||||
function: LPC low level routines
|
||||
last mod: $Id: lpc.c 16227 2009-07-08 06:58:46Z xiphmont $
|
||||
|
||||
********************************************************************/
|
||||
|
||||
/* Some of these routines (autocorrelator, LPC coefficient estimator)
|
||||
are derived from code written by Jutta Degener and Carsten Bormann;
|
||||
thus we include their copyright below. The entirety of this file
|
||||
is freely redistributable on the condition that both of these
|
||||
copyright notices are preserved without modification. */
|
||||
|
||||
/* Preserved Copyright: *********************************************/
|
||||
|
||||
/* Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
|
||||
Technische Universita"t Berlin
|
||||
|
||||
Any use of this software is permitted provided that this notice is not
|
||||
removed and that neither the authors nor the Technische Universita"t
|
||||
Berlin are deemed to have made any representations as to the
|
||||
suitability of this software for any purpose nor are held responsible
|
||||
for any defects of this software. THERE IS ABSOLUTELY NO WARRANTY FOR
|
||||
THIS SOFTWARE.
|
||||
|
||||
As a matter of courtesy, the authors request to be informed about uses
|
||||
this software has found, about bugs in this software, and about any
|
||||
improvements that may be of general interest.
|
||||
|
||||
Berlin, 28.11.1994
|
||||
Jutta Degener
|
||||
Carsten Bormann
|
||||
|
||||
*********************************************************************/
|
||||
|
||||
#include <stdlib.h>
|
||||
#include <string.h>
|
||||
#include <math.h>
|
||||
#include "lpc.h"
|
||||
|
||||
/* Autocorrelation LPC coeff generation algorithm invented by
|
||||
N. Levinson in 1947, modified by J. Durbin in 1959. */
|
||||
|
||||
/* Input : n elements of time doamin data
|
||||
Output: m lpc coefficients, excitation energy */
|
||||
|
||||
float vorbis_lpc_from_data(float *data,float *lpci,int n,int m){
|
||||
double aut[m+1];
|
||||
double lpc[m];
|
||||
double error;
|
||||
double epsilon;
|
||||
int i,j;
|
||||
|
||||
/* autocorrelation, p+1 lag coefficients */
|
||||
j=m+1;
|
||||
while(j--){
|
||||
double d=0; /* double needed for accumulator depth */
|
||||
for(i=j;i<n;i++)d+=(double)data[i]*data[i-j];
|
||||
aut[j]=d;
|
||||
}
|
||||
|
||||
/* Generate lpc coefficients from autocorr values */
|
||||
|
||||
/* set our noise floor to about -100dB */
|
||||
error=aut[0] * (1. + 1e-10);
|
||||
epsilon=1e-9*aut[0]+1e-10;
|
||||
|
||||
for(i=0;i<m;i++){
|
||||
double r= -aut[i+1];
|
||||
|
||||
if(error<epsilon){
|
||||
memset(lpc+i,0,(m-i)*sizeof(*lpc));
|
||||
goto done;
|
||||
}
|
||||
|
||||
/* Sum up this iteration's reflection coefficient; note that in
|
||||
Vorbis we don't save it. If anyone wants to recycle this code
|
||||
and needs reflection coefficients, save the results of 'r' from
|
||||
each iteration. */
|
||||
|
||||
for(j=0;j<i;j++)r-=lpc[j]*aut[i-j];
|
||||
r/=error;
|
||||
|
||||
/* Update LPC coefficients and total error */
|
||||
|
||||
lpc[i]=r;
|
||||
for(j=0;j<i/2;j++){
|
||||
double tmp=lpc[j];
|
||||
|
||||
lpc[j]+=r*lpc[i-1-j];
|
||||
lpc[i-1-j]+=r*tmp;
|
||||
}
|
||||
if(i&1)lpc[j]+=lpc[j]*r;
|
||||
|
||||
error*=1.-r*r;
|
||||
|
||||
}
|
||||
|
||||
done:
|
||||
|
||||
/* slightly damp the filter */
|
||||
{
|
||||
double g = .99;
|
||||
double damp = g;
|
||||
for(j=0;j<m;j++){
|
||||
lpc[j]*=damp;
|
||||
damp*=g;
|
||||
}
|
||||
}
|
||||
|
||||
for(j=0;j<m;j++)lpci[j]=(float)lpc[j];
|
||||
|
||||
/* we need the error value to know how big an impulse to hit the
|
||||
filter with later */
|
||||
|
||||
return error;
|
||||
}
|
||||
|
||||
void vorbis_lpc_predict(float *coeff,float *prime,int m,
|
||||
float *data,long n){
|
||||
|
||||
/* in: coeff[0...m-1] LPC coefficients
|
||||
prime[0...m-1] initial values (allocated size of n+m-1)
|
||||
out: data[0...n-1] data samples */
|
||||
|
||||
long i,j,o,p;
|
||||
float y;
|
||||
float work[m+n];
|
||||
|
||||
if(!prime)
|
||||
for(i=0;i<m;i++)
|
||||
work[i]=0.f;
|
||||
else
|
||||
for(i=0;i<m;i++)
|
||||
work[i]=prime[i];
|
||||
|
||||
for(i=0;i<n;i++){
|
||||
y=0;
|
||||
o=i;
|
||||
p=m;
|
||||
for(j=0;j<m;j++)
|
||||
y-=work[o++]*coeff[--p];
|
||||
|
||||
data[i]=work[o]=y;
|
||||
}
|
||||
}
|
||||
|
|
@ -0,0 +1,27 @@
|
|||
/********************************************************************
|
||||
* *
|
||||
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
|
||||
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
|
||||
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
|
||||
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
|
||||
* *
|
||||
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
|
||||
* by the Xiph.Org Foundation http://www.xiph.org/ *
|
||||
* *
|
||||
********************************************************************
|
||||
|
||||
function: LPC low level routines
|
||||
last mod: $Id: lpc.h 16037 2009-05-26 21:10:58Z xiphmont $
|
||||
|
||||
********************************************************************/
|
||||
|
||||
#ifndef _V_LPC_H_
|
||||
#define _V_LPC_H_
|
||||
|
||||
/* simple linear scale LPC code */
|
||||
extern float vorbis_lpc_from_data(float *data,float *lpc,int n,int m);
|
||||
|
||||
extern void vorbis_lpc_predict(float *coeff,float *prime,int m,
|
||||
float *data,long n);
|
||||
|
||||
#endif
|
||||
Loading…
Reference in New Issue