ffmpeg / libavcodec / resample.c @ 8dbcc9f2
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/*


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* Sample rate convertion for both audio and video

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* Copyright (c) 2000 Fabrice Bellard.

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*

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* This library is free software; you can redistribute it and/or

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* modify it under the terms of the GNU Lesser General Public

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* License as published by the Free Software Foundation; either

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* version 2 of the License, or (at your option) any later version.

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*

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* This library is distributed in the hope that it will be useful,

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* but WITHOUT ANY WARRANTY; without even the implied warranty of

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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

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* Lesser General Public License for more details.

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*

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* You should have received a copy of the GNU Lesser General Public

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* License along with this library; if not, write to the Free Software

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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 021111307 USA

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*/

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/**

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* @file resample.c

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* Sample rate convertion for both audio and video.

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*/

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#include "avcodec.h" 
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typedef struct { 
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/* fractional resampling */

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uint32_t incr; /* fractional increment */

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uint32_t frac; 
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int last_sample;

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/* integer down sample */

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int iratio; /* integer divison ratio */ 
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int icount, isum;

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int inv;

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} ReSampleChannelContext; 
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struct ReSampleContext {

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ReSampleChannelContext channel_ctx[2];

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float ratio;

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/* channel convert */

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int input_channels, output_channels, filter_channels;

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}; 
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#define FRAC_BITS 16 
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#define FRAC (1 << FRAC_BITS) 
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static void init_mono_resample(ReSampleChannelContext *s, float ratio) 
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{ 
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ratio = 1.0 / ratio; 
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s>iratio = (int)floorf(ratio);

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if (s>iratio == 0) 
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s>iratio = 1;

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s>incr = (int)((ratio / s>iratio) * FRAC);

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s>frac = FRAC; 
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s>last_sample = 0;

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s>icount = s>iratio; 
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s>isum = 0;

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s>inv = (FRAC / s>iratio); 
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} 
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/* fractional audio resampling */

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static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) 
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{ 
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unsigned int frac, incr; 
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int l0, l1;

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short *q, *p, *pend;

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l0 = s>last_sample; 
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incr = s>incr; 
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frac = s>frac; 
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p = input; 
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pend = input + nb_samples; 
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q = output; 
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l1 = *p++; 
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for(;;) {

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/* interpolate */

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*q++ = (l0 * (FRAC  frac) + l1 * frac) >> FRAC_BITS; 
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frac = frac + s>incr; 
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while (frac >= FRAC) {

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frac = FRAC; 
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if (p >= pend)

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goto the_end;

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l0 = l1; 
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l1 = *p++; 
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} 
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} 
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the_end:

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s>last_sample = l1; 
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s>frac = frac; 
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return q  output;

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} 
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static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) 
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{ 
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short *q, *p, *pend;

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int c, sum;

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p = input; 
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pend = input + nb_samples; 
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q = output; 
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c = s>icount; 
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sum = s>isum; 
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for(;;) {

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sum += *p++; 
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if (c == 0) { 
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*q++ = (sum * s>inv) >> FRAC_BITS; 
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c = s>iratio; 
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sum = 0;

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} 
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if (p >= pend)

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break;

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} 
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s>isum = sum; 
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s>icount = c; 
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return q  output;

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} 
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/* n1: number of samples */

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static void stereo_to_mono(short *output, short *input, int n1) 
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{ 
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short *p, *q;

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int n = n1;

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p = input; 
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q = output; 
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while (n >= 4) { 
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q[0] = (p[0] + p[1]) >> 1; 
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q[1] = (p[2] + p[3]) >> 1; 
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q[2] = (p[4] + p[5]) >> 1; 
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q[3] = (p[6] + p[7]) >> 1; 
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q += 4;

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p += 8;

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n = 4;

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} 
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while (n > 0) { 
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q[0] = (p[0] + p[1]) >> 1; 
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q++; 
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p += 2;

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n; 
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} 
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} 
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/* n1: number of samples */

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static void mono_to_stereo(short *output, short *input, int n1) 
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{ 
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short *p, *q;

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int n = n1;

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int v;

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p = input; 
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q = output; 
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while (n >= 4) { 
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v = p[0]; q[0] = v; q[1] = v; 
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v = p[1]; q[2] = v; q[3] = v; 
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v = p[2]; q[4] = v; q[5] = v; 
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v = p[3]; q[6] = v; q[7] = v; 
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q += 8;

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p += 4;

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n = 4;

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} 
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while (n > 0) { 
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v = p[0]; q[0] = v; q[1] = v; 
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q += 2;

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p += 1;

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n; 
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} 
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} 
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/* XXX: should use more abstract 'N' channels system */

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static void stereo_split(short *output1, short *output2, short *input, int n) 
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{ 
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int i;

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for(i=0;i<n;i++) { 
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*output1++ = *input++; 
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*output2++ = *input++; 
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} 
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} 
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static void stereo_mux(short *output, short *input1, short *input2, int n) 
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{ 
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int i;

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for(i=0;i<n;i++) { 
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*output++ = *input1++; 
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*output++ = *input2++; 
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} 
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} 
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static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) 
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{ 
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int i;

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short l,r;

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for(i=0;i<n;i++) { 
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l=*input1++; 
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r=*input2++; 
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*output++ = l; /* left */

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*output++ = (l/2)+(r/2); /* center */ 
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*output++ = r; /* right */

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*output++ = 0; /* left surround */ 
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*output++ = 0; /* right surroud */ 
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*output++ = 0; /* low freq */ 
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} 
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} 
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static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) 
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{ 
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short *buf1;

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short *buftmp;

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buf1= (short*)av_malloc( nb_samples * sizeof(short) ); 
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/* first downsample by an integer factor with averaging filter */

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if (s>iratio > 1) { 
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buftmp = buf1; 
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nb_samples = integer_downsample(s, buftmp, input, nb_samples); 
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} else {

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buftmp = input; 
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} 
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/* then do a fractional resampling with linear interpolation */

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if (s>incr != FRAC) {

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nb_samples = fractional_resample(s, output, buftmp, nb_samples); 
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} else {

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memcpy(output, buftmp, nb_samples * sizeof(short)); 
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} 
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av_free(buf1); 
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return nb_samples;

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} 
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ReSampleContext *audio_resample_init(int output_channels, int input_channels, 
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int output_rate, int input_rate) 
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{ 
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ReSampleContext *s; 
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int i;

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if ( input_channels > 2) 
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{ 
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av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported."); 
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return NULL; 
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} 
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s = av_mallocz(sizeof(ReSampleContext));

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if (!s)

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{ 
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av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context."); 
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return NULL; 
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} 
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s>ratio = (float)output_rate / (float)input_rate; 
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s>input_channels = input_channels; 
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s>output_channels = output_channels; 
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s>filter_channels = s>input_channels; 
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if (s>output_channels < s>filter_channels)

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s>filter_channels = s>output_channels; 
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/*

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* ac3 output is the only case where filter_channels could be greater than 2.

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* input channels can't be greater than 2, so resample the 2 channels and then

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* expand to 6 channels after the resampling.

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*/

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if(s>filter_channels>2) 
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s>filter_channels = 2;

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for(i=0;i<s>filter_channels;i++) { 
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init_mono_resample(&s>channel_ctx[i], s>ratio); 
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} 
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return s;

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} 
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/* resample audio. 'nb_samples' is the number of input samples */

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/* XXX: optimize it ! */

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/* XXX: do it with polyphase filters, since the quality here is

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HORRIBLE. Return the number of samples available in output */

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int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) 
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{ 
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int i, nb_samples1;

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short *bufin[2]; 
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short *bufout[2]; 
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short *buftmp2[2], *buftmp3[2]; 
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int lenout;

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if (s>input_channels == s>output_channels && s>ratio == 1.0) { 
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/* nothing to do */

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memcpy(output, input, nb_samples * s>input_channels * sizeof(short)); 
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return nb_samples;

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} 
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/* XXX: move those malloc to resample init code */

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bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) ); 
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bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) ); 
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/* make some zoom to avoid round pb */

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lenout= (int)(nb_samples * s>ratio) + 16; 
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bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); 
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bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); 
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if (s>input_channels == 2 && 
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s>output_channels == 1) {

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buftmp2[0] = bufin[0]; 
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buftmp3[0] = output;

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stereo_to_mono(buftmp2[0], input, nb_samples);

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} else if (s>output_channels >= 2 && s>input_channels == 1) { 
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buftmp2[0] = input;

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buftmp3[0] = bufout[0]; 
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} else if (s>output_channels >= 2) { 
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buftmp2[0] = bufin[0]; 
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buftmp2[1] = bufin[1]; 
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buftmp3[0] = bufout[0]; 
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buftmp3[1] = bufout[1]; 
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stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); 
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} else {

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buftmp2[0] = input;

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buftmp3[0] = output;

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} 
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/* resample each channel */

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nb_samples1 = 0; /* avoid warning */ 
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for(i=0;i<s>filter_channels;i++) { 
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nb_samples1 = mono_resample(&s>channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); 
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} 
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if (s>output_channels == 2 && s>input_channels == 1) { 
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mono_to_stereo(output, buftmp3[0], nb_samples1);

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} else if (s>output_channels == 2) { 
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stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); 
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} else if (s>output_channels == 6) { 
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ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); 
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} 
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av_free(bufin[0]);

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av_free(bufin[1]);

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av_free(bufout[0]);

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av_free(bufout[1]);

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return nb_samples1;

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} 
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void audio_resample_close(ReSampleContext *s)

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{ 
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av_free(s); 
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} 