forked from KolibriOS/kolibrios
436 lines
14 KiB
C
436 lines
14 KiB
C
/*
|
|
* 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((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;
|
|
bufout[i] = 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->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(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;
|
|
}
|
|
}
|
|
|
|
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
|