/*
* 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;
outlink->time_base = (AVRational) {1, 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_make_all_channel_layouts();
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_channel_layouts(ctx, formats);
formats = avfilter_make_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, requested_out_nb_samples);
outlink->out_buf = aresample->outsamplesref;
}
avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
aresample->outsamplesref->pts =
av_rescale(outlink->sample_rate, insamplesref->pts, inlink->sample_rate);
/* 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 = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.config_props = config_output,
.type = AVMEDIA_TYPE_AUDIO, },
{ .name = NULL}},
};