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https://github.com/FFmpeg/FFmpeg.git
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249 lines
6.8 KiB
C
249 lines
6.8 KiB
C
/*
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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 02111-1307 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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struct AVResampleContext;
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struct ReSampleContext {
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struct AVResampleContext *resample_context;
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short *temp[2];
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int temp_len;
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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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/* 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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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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s->resample_context= av_resample_init(output_rate, input_rate);
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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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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 && 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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for(i=0; i<s->filter_channels; i++){
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bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
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memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
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buftmp2[i] = bufin[i] + s->temp_len;
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}
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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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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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buftmp3[0] = bufout[0];
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memcpy(buftmp2[0], input, nb_samples*sizeof(short));
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} else if (s->output_channels >= 2) {
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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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buftmp3[0] = output;
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memcpy(buftmp2[0], input, nb_samples*sizeof(short));
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}
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nb_samples += s->temp_len;
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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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int consumed;
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int is_last= i+1 == s->filter_channels;
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nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
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s->temp_len= nb_samples - consumed;
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s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
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memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
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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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for(i=0; i<s->filter_channels; i++)
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av_free(bufin[i]);
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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_resample_close(s->resample_context);
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av_freep(&s->temp[0]);
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av_freep(&s->temp[1]);
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av_free(s);
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}
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