+++ /dev/null
-/*
- * Sample rate convertion for both audio and video
- * Copyright (c) 2000 Fabrice Bellard.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- */
-
-/**
- * @file resample.c
- * Sample rate convertion for both audio and video.
- */
-
-#include "avcodec.h"
-
-typedef struct {
- /* fractional resampling */
- uint32_t incr; /* fractional increment */
- uint32_t frac;
- int last_sample;
- /* integer down sample */
- int iratio; /* integer divison ratio */
- int icount, isum;
- int inv;
-} ReSampleChannelContext;
-
-struct ReSampleContext {
- ReSampleChannelContext channel_ctx[2];
- float ratio;
- /* channel convert */
- int input_channels, output_channels, filter_channels;
-};
-
-
-#define FRAC_BITS 16
-#define FRAC (1 << FRAC_BITS)
-
-static void init_mono_resample(ReSampleChannelContext *s, float ratio)
-{
- ratio = 1.0 / ratio;
- s->iratio = (int)floorf(ratio);
- if (s->iratio == 0)
- s->iratio = 1;
- s->incr = (int)((ratio / s->iratio) * FRAC);
- s->frac = FRAC;
- s->last_sample = 0;
- s->icount = s->iratio;
- s->isum = 0;
- s->inv = (FRAC / s->iratio);
-}
-
-/* fractional audio resampling */
-static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
-{
- unsigned int frac, incr;
- int l0, l1;
- short *q, *p, *pend;
-
- l0 = s->last_sample;
- incr = s->incr;
- frac = s->frac;
-
- p = input;
- pend = input + nb_samples;
- q = output;
-
- l1 = *p++;
- for(;;) {
- /* interpolate */
- *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
- frac = frac + s->incr;
- while (frac >= FRAC) {
- frac -= FRAC;
- if (p >= pend)
- goto the_end;
- l0 = l1;
- l1 = *p++;
- }
- }
- the_end:
- s->last_sample = l1;
- s->frac = frac;
- return q - output;
-}
-
-static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
-{
- short *q, *p, *pend;
- int c, sum;
-
- p = input;
- pend = input + nb_samples;
- q = output;
-
- c = s->icount;
- sum = s->isum;
-
- for(;;) {
- sum += *p++;
- if (--c == 0) {
- *q++ = (sum * s->inv) >> FRAC_BITS;
- c = s->iratio;
- sum = 0;
- }
- if (p >= pend)
- break;
- }
- s->isum = sum;
- s->icount = c;
- return q - output;
-}
-
-/* n1: number of samples */
-static void stereo_to_mono(short *output, short *input, int n1)
-{
- short *p, *q;
- int n = n1;
-
- p = input;
- q = output;
- while (n >= 4) {
- q[0] = (p[0] + p[1]) >> 1;
- q[1] = (p[2] + p[3]) >> 1;
- q[2] = (p[4] + p[5]) >> 1;
- q[3] = (p[6] + p[7]) >> 1;
- q += 4;
- p += 8;
- n -= 4;
- }
- while (n > 0) {
- q[0] = (p[0] + p[1]) >> 1;
- q++;
- p += 2;
- n--;
- }
-}
-
-/* n1: number of samples */
-static void mono_to_stereo(short *output, short *input, int n1)
-{
- short *p, *q;
- int n = n1;
- int v;
-
- p = input;
- q = output;
- while (n >= 4) {
- v = p[0]; q[0] = v; q[1] = v;
- v = p[1]; q[2] = v; q[3] = v;
- v = p[2]; q[4] = v; q[5] = v;
- v = p[3]; q[6] = v; q[7] = v;
- q += 8;
- p += 4;
- n -= 4;
- }
- while (n > 0) {
- v = p[0]; q[0] = v; q[1] = v;
- q += 2;
- p += 1;
- n--;
- }
-}
-
-/* XXX: should use more abstract 'N' channels system */
-static void stereo_split(short *output1, short *output2, short *input, int n)
-{
- int i;
-
- for(i=0;i<n;i++) {
- *output1++ = *input++;
- *output2++ = *input++;
- }
-}
-
-static void stereo_mux(short *output, short *input1, short *input2, int n)
-{
- int i;
-
- for(i=0;i<n;i++) {
- *output++ = *input1++;
- *output++ = *input2++;
- }
-}
-
-static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
-{
- int i;
- short l,r;
-
- for(i=0;i<n;i++) {
- l=*input1++;
- r=*input2++;
- *output++ = l; /* left */
- *output++ = (l/2)+(r/2); /* center */
- *output++ = r; /* right */
- *output++ = 0; /* left surround */
- *output++ = 0; /* right surroud */
- *output++ = 0; /* low freq */
- }
-}
-
-static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
-{
- short *buf1;
- short *buftmp;
-
- buf1= (short*)av_malloc( nb_samples * sizeof(short) );
-
- /* first downsample by an integer factor with averaging filter */
- if (s->iratio > 1) {
- buftmp = buf1;
- nb_samples = integer_downsample(s, buftmp, input, nb_samples);
- } else {
- buftmp = input;
- }
-
- /* then do a fractional resampling with linear interpolation */
- if (s->incr != FRAC) {
- nb_samples = fractional_resample(s, output, buftmp, nb_samples);
- } else {
- memcpy(output, buftmp, nb_samples * sizeof(short));
- }
- av_free(buf1);
- return nb_samples;
-}
-
-ReSampleContext *audio_resample_init(int output_channels, int input_channels,
- int output_rate, int input_rate)
-{
- ReSampleContext *s;
- int i;
-
- if ( input_channels > 2)
- {
- printf("Resampling with input channels greater than 2 unsupported.");
- return NULL;
- }
-
- s = av_mallocz(sizeof(ReSampleContext));
- if (!s)
- {
- printf("Can't allocate memory for resample context.");
- return NULL;
- }
-
- s->ratio = (float)output_rate / (float)input_rate;
-
- s->input_channels = input_channels;
- s->output_channels = output_channels;
-
- s->filter_channels = s->input_channels;
- if (s->output_channels < s->filter_channels)
- s->filter_channels = s->output_channels;
-
-/*
- * ac3 output is the only case where filter_channels could be greater than 2.
- * input channels can't be greater than 2, so resample the 2 channels and then
- * expand to 6 channels after the resampling.
- */
- if(s->filter_channels>2)
- s->filter_channels = 2;
-
- for(i=0;i<s->filter_channels;i++) {
- init_mono_resample(&s->channel_ctx[i], s->ratio);
- }
- return s;
-}
-
-/* resample audio. 'nb_samples' is the number of input samples */
-/* XXX: optimize it ! */
-/* XXX: do it with polyphase filters, since the quality here is
- HORRIBLE. Return the number of samples available in output */
-int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
-{
- int i, nb_samples1;
- short *bufin[2];
- short *bufout[2];
- short *buftmp2[2], *buftmp3[2];
- int lenout;
-
- if (s->input_channels == s->output_channels && s->ratio == 1.0) {
- /* nothing to do */
- memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
- return nb_samples;
- }
-
- /* XXX: move those malloc to resample init code */
- bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
- bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
-
- /* make some zoom to avoid round pb */
- lenout= (int)(nb_samples * s->ratio) + 16;
- bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
- bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
-
- if (s->input_channels == 2 &&
- s->output_channels == 1) {
- buftmp2[0] = bufin[0];
- buftmp3[0] = output;
- stereo_to_mono(buftmp2[0], input, nb_samples);
- } else if (s->output_channels >= 2 && s->input_channels == 1) {
- buftmp2[0] = input;
- buftmp3[0] = bufout[0];
- } else if (s->output_channels >= 2) {
- buftmp2[0] = bufin[0];
- buftmp2[1] = bufin[1];
- buftmp3[0] = bufout[0];
- buftmp3[1] = bufout[1];
- stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
- } else {
- buftmp2[0] = input;
- buftmp3[0] = output;
- }
-
- /* resample each channel */
- nb_samples1 = 0; /* avoid warning */
- for(i=0;i<s->filter_channels;i++) {
- nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
- }
-
- if (s->output_channels == 2 && s->input_channels == 1) {
- mono_to_stereo(output, buftmp3[0], nb_samples1);
- } else if (s->output_channels == 2) {
- stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
- } else if (s->output_channels == 6) {
- ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
- }
-
- av_free(bufin[0]);
- av_free(bufin[1]);
-
- av_free(bufout[0]);
- av_free(bufout[1]);
- return nb_samples1;
-}
-
-void audio_resample_close(ReSampleContext *s)
-{
- av_free(s);
-}
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