mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
2c0317419b
The additional parameters were never used and are complicating the function interface. Also, they were inconsistent with the way the video API works. So this assumes that a requested samples buffer will have *always* the format specified in the requested link. This breaks audio filtering API and ABI.
349 lines
11 KiB
C
349 lines
11 KiB
C
/*
|
|
* Copyright (c) 2011 Stefano Sabatini
|
|
* Copyright (c) 2011 Mina Nagy Zaki
|
|
*
|
|
* 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
|
|
* resampling audio filter
|
|
*/
|
|
|
|
#include "libavutil/eval.h"
|
|
#include "libavcodec/avcodec.h"
|
|
#include "avfilter.h"
|
|
#include "internal.h"
|
|
|
|
typedef struct {
|
|
struct AVResampleContext *resample;
|
|
int out_rate;
|
|
double ratio;
|
|
AVFilterBufferRef *outsamplesref;
|
|
int unconsumed_nb_samples,
|
|
max_cached_nb_samples;
|
|
int16_t *cached_data[8],
|
|
*resampled_data[8];
|
|
} AResampleContext;
|
|
|
|
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
|
|
{
|
|
AResampleContext *aresample = ctx->priv;
|
|
int ret;
|
|
|
|
if (args) {
|
|
if ((ret = ff_parse_sample_rate(&aresample->out_rate, args, ctx)) < 0)
|
|
return ret;
|
|
} else {
|
|
aresample->out_rate = -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold void uninit(AVFilterContext *ctx)
|
|
{
|
|
AResampleContext *aresample = ctx->priv;
|
|
if (aresample->outsamplesref) {
|
|
int nb_channels =
|
|
av_get_channel_layout_nb_channels(
|
|
aresample->outsamplesref->audio->channel_layout);
|
|
avfilter_unref_buffer(aresample->outsamplesref);
|
|
while (nb_channels--) {
|
|
av_freep(&(aresample->cached_data[nb_channels]));
|
|
av_freep(&(aresample->resampled_data[nb_channels]));
|
|
}
|
|
}
|
|
|
|
if (aresample->resample)
|
|
av_resample_close(aresample->resample);
|
|
}
|
|
|
|
static int config_output(AVFilterLink *outlink)
|
|
{
|
|
AVFilterContext *ctx = outlink->src;
|
|
AVFilterLink *inlink = ctx->inputs[0];
|
|
AResampleContext *aresample = ctx->priv;
|
|
|
|
if (aresample->out_rate == -1)
|
|
aresample->out_rate = outlink->sample_rate;
|
|
else
|
|
outlink->sample_rate = aresample->out_rate;
|
|
|
|
//TODO: make the resampling parameters configurable
|
|
aresample->resample = av_resample_init(aresample->out_rate, inlink->sample_rate,
|
|
16, 10, 0, 0.8);
|
|
|
|
aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
|
|
|
|
av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n",
|
|
inlink->sample_rate, outlink->sample_rate);
|
|
return 0;
|
|
}
|
|
|
|
static int query_formats(AVFilterContext *ctx)
|
|
{
|
|
AVFilterFormats *formats = NULL;
|
|
|
|
avfilter_add_format(&formats, AV_SAMPLE_FMT_S16);
|
|
if (!formats)
|
|
return AVERROR(ENOMEM);
|
|
avfilter_set_common_sample_formats(ctx, formats);
|
|
|
|
formats = avfilter_all_channel_layouts();
|
|
if (!formats)
|
|
return AVERROR(ENOMEM);
|
|
avfilter_set_common_channel_layouts(ctx, formats);
|
|
|
|
formats = avfilter_all_packing_formats();
|
|
if (!formats)
|
|
return AVERROR(ENOMEM);
|
|
avfilter_set_common_packing_formats(ctx, formats);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void deinterleave(int16_t **outp, int16_t *in,
|
|
int nb_channels, int nb_samples)
|
|
{
|
|
int16_t *out[8];
|
|
memcpy(out, outp, nb_channels * sizeof(int16_t*));
|
|
|
|
switch (nb_channels) {
|
|
case 2:
|
|
while (nb_samples--) {
|
|
*out[0]++ = *in++;
|
|
*out[1]++ = *in++;
|
|
}
|
|
break;
|
|
case 3:
|
|
while (nb_samples--) {
|
|
*out[0]++ = *in++;
|
|
*out[1]++ = *in++;
|
|
*out[2]++ = *in++;
|
|
}
|
|
break;
|
|
case 4:
|
|
while (nb_samples--) {
|
|
*out[0]++ = *in++;
|
|
*out[1]++ = *in++;
|
|
*out[2]++ = *in++;
|
|
*out[3]++ = *in++;
|
|
}
|
|
break;
|
|
case 5:
|
|
while (nb_samples--) {
|
|
*out[0]++ = *in++;
|
|
*out[1]++ = *in++;
|
|
*out[2]++ = *in++;
|
|
*out[3]++ = *in++;
|
|
*out[4]++ = *in++;
|
|
}
|
|
break;
|
|
case 6:
|
|
while (nb_samples--) {
|
|
*out[0]++ = *in++;
|
|
*out[1]++ = *in++;
|
|
*out[2]++ = *in++;
|
|
*out[3]++ = *in++;
|
|
*out[4]++ = *in++;
|
|
*out[5]++ = *in++;
|
|
}
|
|
break;
|
|
case 8:
|
|
while (nb_samples--) {
|
|
*out[0]++ = *in++;
|
|
*out[1]++ = *in++;
|
|
*out[2]++ = *in++;
|
|
*out[3]++ = *in++;
|
|
*out[4]++ = *in++;
|
|
*out[5]++ = *in++;
|
|
*out[6]++ = *in++;
|
|
*out[7]++ = *in++;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void interleave(int16_t *out, int16_t **inp,
|
|
int nb_channels, int nb_samples)
|
|
{
|
|
int16_t *in[8];
|
|
memcpy(in, inp, nb_channels * sizeof(int16_t*));
|
|
|
|
switch (nb_channels) {
|
|
case 2:
|
|
while (nb_samples--) {
|
|
*out++ = *in[0]++;
|
|
*out++ = *in[1]++;
|
|
}
|
|
break;
|
|
case 3:
|
|
while (nb_samples--) {
|
|
*out++ = *in[0]++;
|
|
*out++ = *in[1]++;
|
|
*out++ = *in[2]++;
|
|
}
|
|
break;
|
|
case 4:
|
|
while (nb_samples--) {
|
|
*out++ = *in[0]++;
|
|
*out++ = *in[1]++;
|
|
*out++ = *in[2]++;
|
|
*out++ = *in[3]++;
|
|
}
|
|
break;
|
|
case 5:
|
|
while (nb_samples--) {
|
|
*out++ = *in[0]++;
|
|
*out++ = *in[1]++;
|
|
*out++ = *in[2]++;
|
|
*out++ = *in[3]++;
|
|
*out++ = *in[4]++;
|
|
}
|
|
break;
|
|
case 6:
|
|
while (nb_samples--) {
|
|
*out++ = *in[0]++;
|
|
*out++ = *in[1]++;
|
|
*out++ = *in[2]++;
|
|
*out++ = *in[3]++;
|
|
*out++ = *in[4]++;
|
|
*out++ = *in[5]++;
|
|
}
|
|
break;
|
|
case 8:
|
|
while (nb_samples--) {
|
|
*out++ = *in[0]++;
|
|
*out++ = *in[1]++;
|
|
*out++ = *in[2]++;
|
|
*out++ = *in[3]++;
|
|
*out++ = *in[4]++;
|
|
*out++ = *in[5]++;
|
|
*out++ = *in[6]++;
|
|
*out++ = *in[7]++;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
|
|
{
|
|
AResampleContext *aresample = inlink->dst->priv;
|
|
AVFilterLink * const outlink = inlink->dst->outputs[0];
|
|
int i,
|
|
in_nb_samples = insamplesref->audio->nb_samples,
|
|
cached_nb_samples = in_nb_samples + aresample->unconsumed_nb_samples,
|
|
requested_out_nb_samples = aresample->ratio * cached_nb_samples,
|
|
nb_channels =
|
|
av_get_channel_layout_nb_channels(inlink->channel_layout);
|
|
|
|
if (cached_nb_samples > aresample->max_cached_nb_samples) {
|
|
for (i = 0; i < nb_channels; i++) {
|
|
aresample->cached_data[i] =
|
|
av_realloc(aresample->cached_data[i], cached_nb_samples * sizeof(int16_t));
|
|
aresample->resampled_data[i] =
|
|
av_realloc(aresample->resampled_data[i],
|
|
FFALIGN(sizeof(int16_t) * requested_out_nb_samples, 16));
|
|
|
|
if (aresample->cached_data[i] == NULL || aresample->resampled_data[i] == NULL)
|
|
return;
|
|
}
|
|
aresample->max_cached_nb_samples = cached_nb_samples;
|
|
|
|
if (aresample->outsamplesref)
|
|
avfilter_unref_buffer(aresample->outsamplesref);
|
|
|
|
aresample->outsamplesref = avfilter_get_audio_buffer(outlink,
|
|
AV_PERM_WRITE | AV_PERM_REUSE2,
|
|
requested_out_nb_samples);
|
|
|
|
avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
|
|
aresample->outsamplesref->pts =
|
|
insamplesref->pts / inlink->sample_rate * outlink->sample_rate;
|
|
aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
|
|
outlink->out_buf = aresample->outsamplesref;
|
|
}
|
|
|
|
/* av_resample() works with planar audio buffers */
|
|
if (!inlink->planar && nb_channels > 1) {
|
|
int16_t *out[8];
|
|
for (i = 0; i < nb_channels; i++)
|
|
out[i] = aresample->cached_data[i] + aresample->unconsumed_nb_samples;
|
|
|
|
deinterleave(out, (int16_t *)insamplesref->data[0],
|
|
nb_channels, in_nb_samples);
|
|
} else {
|
|
for (i = 0; i < nb_channels; i++)
|
|
memcpy(aresample->cached_data[i] + aresample->unconsumed_nb_samples,
|
|
insamplesref->data[i],
|
|
in_nb_samples * sizeof(int16_t));
|
|
}
|
|
|
|
for (i = 0; i < nb_channels; i++) {
|
|
int consumed_nb_samples;
|
|
const int is_last = i+1 == nb_channels;
|
|
|
|
aresample->outsamplesref->audio->nb_samples =
|
|
av_resample(aresample->resample,
|
|
aresample->resampled_data[i], aresample->cached_data[i],
|
|
&consumed_nb_samples,
|
|
cached_nb_samples,
|
|
requested_out_nb_samples, is_last);
|
|
|
|
/* move unconsumed data back to the beginning of the cache */
|
|
aresample->unconsumed_nb_samples = cached_nb_samples - consumed_nb_samples;
|
|
memmove(aresample->cached_data[i],
|
|
aresample->cached_data[i] + consumed_nb_samples,
|
|
aresample->unconsumed_nb_samples * sizeof(int16_t));
|
|
}
|
|
|
|
|
|
/* copy resampled data to the output samplesref */
|
|
if (!inlink->planar && nb_channels > 1) {
|
|
interleave((int16_t *)aresample->outsamplesref->data[0],
|
|
aresample->resampled_data,
|
|
nb_channels, aresample->outsamplesref->audio->nb_samples);
|
|
} else {
|
|
for (i = 0; i < nb_channels; i++)
|
|
memcpy(aresample->outsamplesref->data[i], aresample->resampled_data[i],
|
|
aresample->outsamplesref->audio->nb_samples * sizeof(int16_t));
|
|
}
|
|
|
|
avfilter_filter_samples(outlink, avfilter_ref_buffer(aresample->outsamplesref, ~0));
|
|
avfilter_unref_buffer(insamplesref);
|
|
}
|
|
|
|
AVFilter avfilter_af_aresample = {
|
|
.name = "aresample",
|
|
.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.query_formats = query_formats,
|
|
.priv_size = sizeof(AResampleContext),
|
|
|
|
.inputs = (AVFilterPad[]) {{ .name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_samples = filter_samples,
|
|
.min_perms = AV_PERM_READ, },
|
|
{ .name = NULL}},
|
|
.outputs = (AVFilterPad[]) {{ .name = "default",
|
|
.config_props = config_output,
|
|
.type = AVMEDIA_TYPE_AUDIO, },
|
|
{ .name = NULL}},
|
|
};
|