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ffmpeg/libavcodec/resample.c

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/*
* 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
* samplerate conversion for both audio and video
*/
#include <string.h>
#include "avcodec.h"
#include "audioconvert.h"
#include "libavutil/opt.h"
#include "libavutil/mem.h"
#include "libavutil/samplefmt.h"
#if FF_API_AVCODEC_RESAMPLE
#define MAX_CHANNELS 8
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, LIBAVUTIL_VERSION_INT
};
struct ReSampleContext {
struct AVResampleContext *resample_context;
short *temp[MAX_CHANNELS];
int temp_len;
float ratio;
/* channel convert */
int input_channels, output_channels, filter_channels;
AVAudioConvert *convert_ctx[2];
enum AVSampleFormat 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--;
}
}
/*
5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
- Left = front_left + rear_gain * rear_left + center_gain * center
- Right = front_right + rear_gain * rear_right + center_gain * center
Where rear_gain is usually around 0.5-1.0 and
center_gain is almost always 0.7 (-3 dB)
*/
static void surround_to_stereo(short **output, short *input, int channels, int samples)
{
int i;
short l, r;
for (i = 0; i < samples; i++) {
int fl,fr,c,rl,rr;
fl = input[0];
fr = input[1];
c = input[2];
// lfe = input[3];
rl = input[4];
rr = input[5];
l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c));
r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c));
/* output l & r. */
*output[0]++ = l;
*output[1]++ = r;
/* increment input. */
input += channels;
}
}
static void deinterleave(short **output, short *input, int channels, int samples)
{
int i, j;
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*output[j]++ = *input++;
}
}
}
static void interleave(short *output, short **input, int channels, int samples)
{
int i, j;
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*output++ = *input[j]++;
}
}
}
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 */
}
}
#define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \
ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0
static const uint8_t supported_resampling[MAX_CHANNELS] = {
// output ch: 1 2 3 4 5 6 7 8
SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
};
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate,
enum AVSampleFormat sample_fmt_out,
enum AVSampleFormat sample_fmt_in,
int filter_length, int log2_phase_count,
int linear, double cutoff)
{
ReSampleContext *s;
if (input_channels > MAX_CHANNELS) {
av_log(NULL, AV_LOG_ERROR,
"Resampling with input channels greater than %d is unsupported.\n",
MAX_CHANNELS);
return NULL;
}
if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) {
int i;
av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed "
"output channels for %d input channel%s", input_channels,
input_channels > 1 ? "s:" : ":");
for (i = 0; i < MAX_CHANNELS; i++)
if (supported_resampling[input_channels-1] & (1<<i))
av_log(NULL, AV_LOG_ERROR, " %d", i + 1);
av_log(NULL, AV_LOG_ERROR, "\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_bytes_per_sample(s->sample_fmt[0]);
s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_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",
av_get_sample_fmt_name(s->sample_fmt[0]));
av_free(s);
return NULL;
}
}
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert s16 sample format to %s sample format\n",
av_get_sample_fmt_name(s->sample_fmt[1]));
av_audio_convert_free(s->convert_ctx[0]);
av_free(s);
return NULL;
}
}
s->resample_context = av_resample_init(output_rate, input_rate,
filter_length, log2_phase_count,
linear, cutoff);
*(const AVClass**)s->resample_context = &audioresample_context_class;
return s;
}
/* 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[MAX_CHANNELS];
short *bufout[MAX_CHANNELS];
short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
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] != AV_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->resample_context, 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->resample_context, AV_LOG_ERROR,
"Audio sample format conversion failed\n");
return 0;
}
input = s->buffer[0];
}
lenout= 2*s->output_channels*nb_samples * s->ratio + 16;
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
s->output_channels;
output_bak = output;
if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
av_free(s->buffer[1]);
s->buffer_size[1] = out_size;
s->buffer[1] = av_malloc(s->buffer_size[1]);
if (!s->buffer[1]) {
av_log(s->resample_context, 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_array((nb_samples + s->temp_len), sizeof(short));
bufout[i] = av_malloc_array(lenout, sizeof(short));
if (!bufin[i] || !bufout[i]) {
av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
nb_samples1 = 0;
goto fail;
}
memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
buftmp2[i] = bufin[i] + s->temp_len;
}
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->input_channels == 6 && s->output_channels ==2) {
buftmp3[0] = bufout[0];
buftmp3[1] = bufout[1];
surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
} else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
for (i = 0; i < s->input_channels; i++) {
buftmp3[i] = bufout[i];
}
deinterleave(buftmp2, input, s->input_channels, 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_array(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 == 6 && s->input_channels == 2) {
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
} else if ((s->output_channels == s->input_channels && s->input_channels >= 2) ||
(s->output_channels == 2 && s->input_channels == 6)) {
interleave(output, buftmp3, s->output_channels, nb_samples1);
}
if (s->sample_fmt[1] != AV_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->resample_context, AV_LOG_ERROR,
"Audio sample format conversion failed\n");
return 0;
}
}
fail:
for (i = 0; i < s->filter_channels; i++) {
av_free(bufin[i]);
av_free(bufout[i]);
}
return nb_samples1;
}
void audio_resample_close(ReSampleContext *s)
{
int i;
av_resample_close(s->resample_context);
for (i = 0; i < s->filter_channels; i++)
av_freep(&s->temp[i]);
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);
}
#endif
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