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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 "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavutil/avassert.h"
#include "libswresample/swresample.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"
typedef struct {
double ratio;
struct SwrContext *swr;
int64_t next_pts;
} AResampleContext;
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
{
AResampleContext *aresample = ctx->priv;
int ret = 0;
char *argd = av_strdup(args);
aresample->next_pts = AV_NOPTS_VALUE;
aresample->swr = swr_alloc();
if (!aresample->swr)
return AVERROR(ENOMEM);
if (args) {
char *ptr=argd, *token;
while(token = av_strtok(ptr, ":", &ptr)) {
char *value;
av_strtok(token, "=", &value);
if(value) {
if((ret=av_opt_set(aresample->swr, token, value, 0)) < 0)
goto end;
} else {
int out_rate;
if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)
goto end;
if((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)
goto end;
}
}
}
end:
av_free(argd);
return ret;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AResampleContext *aresample = ctx->priv;
swr_free(&aresample->swr);
}
static int query_formats(AVFilterContext *ctx)
{
AResampleContext *aresample = ctx->priv;
int out_rate = av_get_int(aresample->swr, "osr", NULL);
uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AVFilterFormats *in_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
AVFilterFormats *out_formats;
AVFilterFormats *in_samplerates = ff_all_samplerates();
AVFilterFormats *out_samplerates;
AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
AVFilterChannelLayouts *out_layouts;
avfilter_formats_ref (in_formats, &inlink->out_formats);
avfilter_formats_ref (in_samplerates, &inlink->out_samplerates);
ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
if(out_rate > 0) {
out_samplerates = avfilter_make_format_list((int[]){ out_rate, -1 });
} else {
out_samplerates = ff_all_samplerates();
}
avfilter_formats_ref(out_samplerates, &outlink->in_samplerates);
if(out_format != AV_SAMPLE_FMT_NONE) {
out_formats = avfilter_make_format_list((int[]){ out_format, -1 });
} else
out_formats = avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO);
avfilter_formats_ref(out_formats, &outlink->in_formats);
if(out_layout) {
out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });
} else
out_layouts = ff_all_channel_layouts();
ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
int ret;
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
AResampleContext *aresample = ctx->priv;
int out_rate;
uint64_t out_layout;
enum AVSampleFormat out_format;
aresample->swr = swr_alloc_set_opts(aresample->swr,
outlink->channel_layout, outlink->format, outlink->sample_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;
out_rate = av_get_int(aresample->swr, "osr", NULL);
out_layout = av_get_int(aresample->swr, "ocl", NULL);
out_format = av_get_int(aresample->swr, "osf", NULL);
outlink->time_base = (AVRational) {1, out_rate};
av_assert0(outlink->sample_rate == out_rate);
av_assert0(outlink->channel_layout == out_layout);
av_assert0(outlink->format == out_format);
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 + 1;
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);
if (n_out <= 0) {
avfilter_unref_buffer(outsamplesref);
avfilter_unref_buffer(insamplesref);
return;
}
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
if(insamplesref->pts != AV_NOPTS_VALUE) {
aresample->next_pts =
outsamplesref->pts = av_rescale_q(insamplesref->pts, inlink->time_base, outlink->time_base)
- swr_get_delay(aresample->swr, outlink->time_base.den);
av_assert0(outlink->time_base.num == 1);
} else{
outsamplesref->pts = AV_NOPTS_VALUE; //aresample->next_pts;
}
if(aresample->next_pts != AV_NOPTS_VALUE)
aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
ff_filter_samples(outlink, outsamplesref);
avfilter_unref_buffer(insamplesref);
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AResampleContext *aresample = ctx->priv;
int ret = avfilter_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF) {
AVFilterBufferRef *outsamplesref;
int n_out = 4096;
outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
if (!outsamplesref)
return AVERROR(ENOMEM);
n_out = swr_convert(aresample->swr, outsamplesref->data, n_out, 0, 0);
if (n_out <= 0) {
avfilter_unref_buffer(outsamplesref);
return (n_out == 0) ? AVERROR_EOF : n_out;
}
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
outsamplesref->pts = aresample->next_pts;
if(aresample->next_pts != AV_NOPTS_VALUE)
aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);
ff_filter_samples(outlink, outsamplesref);
return 0;
}
return ret;
}
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,
.request_frame = request_frame,
.type = AVMEDIA_TYPE_AUDIO, },
{ .name = NULL}},
};
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