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DEFINITIONS
This source file includes following definitions.
- context_to_name
- stereo_to_mono
- mono_to_stereo
- stereo_split
- stereo_mux
- ac3_5p1_mux
- av_audio_resample_init
- audio_resample_init
- audio_resample
- audio_resample_close
/*
* samplerate conversion for both audio and video
* Copyright (c) 2000 Fabrice Bellard
*
* 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 2.1 of the License, or (at your option) any later version.
*
* 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file libavcodec/resample.c
* samplerate conversion for both audio and video
*/
#include "avcodec.h"
#include "audioconvert.h"
#include "opt.h"
struct AVResampleContext;
static const char *context_to_name(void *ptr)
{
return "audioresample";
}
static const AVOption options[] = {{NULL}};
static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options };
struct ReSampleContext {
const AVClass *av_class;
struct AVResampleContext *resample_context;
short *temp[2];
int temp_len;
float ratio;
/* channel convert */
int input_channels, output_channels, filter_channels;
AVAudioConvert *convert_ctx[2];
enum SampleFormat sample_fmt[2]; ///< input and output sample format
unsigned sample_size[2]; ///< size of one sample in sample_fmt
short *buffer[2]; ///< buffers used for conversion to S16
unsigned buffer_size[2]; ///< sizes of allocated buffers
};
/* 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 */
}
}
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate,
enum SampleFormat sample_fmt_out,
enum SampleFormat sample_fmt_in,
int filter_length, int log2_phase_count,
int linear, double cutoff)
{
ReSampleContext *s;
if ( input_channels > 2)
{
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.\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;
s->sample_fmt [0] = sample_fmt_in;
s->sample_fmt [1] = sample_fmt_out;
s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
s->sample_fmt[0], 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert %s sample format to s16 sample format\n",
avcodec_get_sample_fmt_name(s->sample_fmt[0]));
av_free(s);
return NULL;
}
}
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
SAMPLE_FMT_S16, 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert s16 sample format to %s sample format\n",
avcodec_get_sample_fmt_name(s->sample_fmt[1]));
av_audio_convert_free(s->convert_ctx[0]);
av_free(s);
return NULL;
}
}
/*
* 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;
#define TAPS 16
s->resample_context= av_resample_init(output_rate, input_rate,
filter_length, log2_phase_count, linear, cutoff);
s->av_class= &audioresample_context_class;
return s;
}
#if LIBAVCODEC_VERSION_MAJOR < 53
ReSampleContext *audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate)
{
return av_audio_resample_init(output_channels, input_channels,
output_rate, input_rate,
SAMPLE_FMT_S16, SAMPLE_FMT_S16,
TAPS, 10, 0, 0.8);
}
#endif
/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
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];
short *output_bak = NULL;
int lenout;
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;
}
if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
int istride[1] = { s->sample_size[0] };
int ostride[1] = { 2 };
const void *ibuf[1] = { input };
void *obuf[1];
unsigned input_size = nb_samples*s->input_channels*2;
if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
av_free(s->buffer[0]);
s->buffer_size[0] = input_size;
s->buffer[0] = av_malloc(s->buffer_size[0]);
if (!s->buffer[0]) {
av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
return 0;
}
}
obuf[0] = s->buffer[0];
if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
ibuf, istride, nb_samples*s->input_channels) < 0) {
av_log(s, AV_LOG_ERROR, "Audio sample format conversion failed\n");
return 0;
}
input = s->buffer[0];
}
lenout= 4*nb_samples * s->ratio + 16;
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
output_bak = output;
if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
av_free(s->buffer[1]);
s->buffer_size[1] = lenout;
s->buffer[1] = av_malloc(s->buffer_size[1]);
if (!s->buffer[1]) {
av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
return 0;
}
}
output = s->buffer[1];
}
/* XXX: move those malloc to resample init code */
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 */
bufout[0]= av_malloc( lenout * sizeof(short) );
bufout[1]= av_malloc( lenout * sizeof(short) );
if (s->input_channels == 2 &&
s->output_channels == 1) {
buftmp3[0] = output;
stereo_to_mono(buftmp2[0], input, nb_samples);
} else if (s->output_channels >= 2 && s->input_channels == 1) {
buftmp3[0] = bufout[0];
memcpy(buftmp2[0], input, nb_samples*sizeof(short));
} else if (s->output_channels >= 2) {
buftmp3[0] = bufout[0];
buftmp3[1] = bufout[1];
stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
} else {
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++) {
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) {
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);
}
if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
int istride[1] = { 2 };
int ostride[1] = { s->sample_size[1] };
const void *ibuf[1] = { output };
void *obuf[1] = { output_bak };
if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
ibuf, istride, nb_samples1*s->output_channels) < 0) {
av_log(s, AV_LOG_ERROR, "Audio sample format convertion failed\n");
return 0;
}
}
for(i=0; i<s->filter_channels; i++)
av_free(bufin[i]);
av_free(bufout[0]);
av_free(bufout[1]);
return nb_samples1;
}
void audio_resample_close(ReSampleContext *s)
{
av_resample_close(s->resample_context);
av_freep(&s->temp[0]);
av_freep(&s->temp[1]);
av_freep(&s->buffer[0]);
av_freep(&s->buffer[1]);
av_audio_convert_free(s->convert_ctx[0]);
av_audio_convert_free(s->convert_ctx[1]);
av_free(s);
}