/*
- * Sample rate convertion for both audio and video
+ * samplerate conversion for both audio and video
* Copyright (c) 2000 Fabrice Bellard.
*
- * This library is free software; you can redistribute it and/or
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * FFmpeg 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
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file resample.c
- * Sample rate convertion for both audio and video.
+ * samplerate conversion 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 AVResampleContext;
struct ReSampleContext {
- ReSampleChannelContext channel_ctx[2];
+ struct AVResampleContext *resample_context;
+ short *temp[2];
+ int temp_len;
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)
{
}
}
-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,
+ReSampleContext *audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate)
{
ReSampleContext *s;
- int i;
-
+
if ( input_channels > 2)
{
- av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
- return NULL;
+ av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
+ return NULL;
}
s = av_mallocz(sizeof(ReSampleContext));
if (!s)
{
- av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
- return NULL;
+ av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
+ 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.
+ * AC-3 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);
- }
+#define TAPS 16
+ s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8);
+
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 *buftmp2[2], *buftmp3[2];
int lenout;
- if (s->input_channels == s->output_channels && s->ratio == 1.0) {
+ if (s->input_channels == s->output_channels && s->ratio == 1.0 && 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) );
-
+ for(i=0; i<s->filter_channels; i++){
+ bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
+ memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
+ buftmp2[i] = bufin[i] + s->temp_len;
+ }
+
/* 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) );
+ lenout= 4*nb_samples * s->ratio + 16;
+ bufout[0]= av_malloc( lenout * sizeof(short) );
+ bufout[1]= 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];
+ memcpy(buftmp2[0], input, nb_samples*sizeof(short));
} 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;
+ memcpy(buftmp2[0], input, nb_samples*sizeof(short));
}
+ nb_samples += s->temp_len;
+
/* 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);
+ int consumed;
+ int is_last= i+1 == s->filter_channels;
+
+ nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
+ s->temp_len= nb_samples - consumed;
+ s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
+ memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
}
if (s->output_channels == 2 && s->input_channels == 1) {
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
}
- av_free(bufin[0]);
- av_free(bufin[1]);
+ for(i=0; i<s->filter_channels; i++)
+ av_free(bufin[i]);
av_free(bufout[0]);
av_free(bufout[1]);
void audio_resample_close(ReSampleContext *s)
{
+ av_resample_close(s->resample_context);
+ av_freep(&s->temp[0]);
+ av_freep(&s->temp[1]);
av_free(s);
}
projects / ffmpeg / blobdiff