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/*
* 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 "libswresample/swresample.h"
#include "avfilter.h"
#include "internal.h"
typedef struct {
int out_rate;
double ratio;
struct SwrContext *swr;
} 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;
swr_free(&aresample->swr);
}
static int config_output(AVFilterLink *outlink)
{
int ret;
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 (filter size, phrase shift, linear, cutoff) configurable
aresample->swr = swr_alloc_set_opts(aresample->swr,
inlink->channel_layout, inlink->format, aresample->out_rate,
inlink->channel_layout, inlink->format, inlink->sample_rate,
0, ctx);
if (!aresample->swr)
return AVERROR(ENOMEM);
ret = swr_init(aresample->swr);
if (ret < 0)
return ret;
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 void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->audio->nb_samples;
int n_out = n_in * aresample->ratio;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
n_out = swr_convert(aresample->swr, outsamplesref->data, n_out,
(void *)insamplesref->data, n_in);
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
outsamplesref->pts = insamplesref->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
av_rescale(outlink->sample_rate, insamplesref->pts, inlink ->sample_rate);
ff_filter_samples(outlink, outsamplesref);
avfilter_unref_buffer(insamplesref);
}
AVFilter avfilter_af_aresample = {
.name = "aresample",
.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
.init = init,
.uninit = uninit,
.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}},
};
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