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/*
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* sample format and channel layout conversion audio filter
*/
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "libavresample/avresample.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
typedef struct ResampleContext {
AVAudioResampleContext *avr;
int64_t next_pts;
/* set by filter_samples() to signal an output frame to request_frame() */
int got_output;
} ResampleContext;
static av_cold void uninit(AVFilterContext *ctx)
{
ResampleContext *s = ctx->priv;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
AVFilterFormats *out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
AVFilterFormats *in_samplerates = ff_all_samplerates();
AVFilterFormats *out_samplerates = ff_all_samplerates();
AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts();
ff_formats_ref(in_formats, &inlink->out_formats);
ff_formats_ref(out_formats, &outlink->in_formats);
ff_formats_ref(in_samplerates, &inlink->out_samplerates);
ff_formats_ref(out_samplerates, &outlink->in_samplerates);
ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
ResampleContext *s = ctx->priv;
char buf1[64], buf2[64];
int ret;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
if (inlink->channel_layout == outlink->channel_layout &&
inlink->sample_rate == outlink->sample_rate &&
(inlink->format == outlink->format ||
(av_get_channel_layout_nb_channels(inlink->channel_layout) == 1 &&
av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
av_get_planar_sample_fmt(inlink->format) ==
av_get_planar_sample_fmt(outlink->format))))
return 0;
if (!(s->avr = avresample_alloc_context()))
return AVERROR(ENOMEM);
av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
if ((ret = avresample_open(s->avr)) < 0)
return ret;
outlink->time_base = (AVRational){ 1, outlink->sample_rate };
s->next_pts = AV_NOPTS_VALUE;
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, outlink->channel_layout);
av_log(ctx, AV_LOG_VERBOSE,
"fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ResampleContext *s = ctx->priv;
int ret = 0;
s->got_output = 0;
while (ret >= 0 && !s->got_output)
ret = ff_request_frame(ctx->inputs[0]);
/* flush the lavr delay buffer */
if (ret == AVERROR_EOF && s->avr) {
AVFilterBufferRef *buf;
int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
outlink->sample_rate,
ctx->inputs[0]->sample_rate,
AV_ROUND_UP);
if (!nb_samples)
return ret;
buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
if (!buf)
return AVERROR(ENOMEM);
ret = avresample_convert(s->avr, buf->extended_data,
buf->linesize[0], nb_samples,
NULL, 0, 0);
if (ret <= 0) {
avfilter_unref_buffer(buf);
return (ret == 0) ? AVERROR_EOF : ret;
}
buf->pts = s->next_pts;
return ff_filter_samples(outlink, buf);
}
return ret;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret;
if (s->avr) {
AVFilterBufferRef *buf_out;
int delay, nb_samples;
/* maximum possible samples lavr can output */
delay = avresample_get_delay(s->avr);
nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
outlink->sample_rate, inlink->sample_rate,
AV_ROUND_UP);
buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
if (!buf_out) {
ret = AVERROR(ENOMEM);
goto fail;
}
ret = avresample_convert(s->avr, buf_out->extended_data,
buf_out->linesize[0], nb_samples,
buf->extended_data, buf->linesize[0],
buf->audio->nb_samples);
if (ret <= 0) {
avfilter_unref_buffer(buf_out);
if (ret < 0)
goto fail;
}
av_assert0(!avresample_available(s->avr));
if (s->next_pts == AV_NOPTS_VALUE) {
if (buf->pts == AV_NOPTS_VALUE) {
av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
"assuming 0.\n");
s->next_pts = 0;
} else
s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base);
}
if (ret > 0) {
buf_out->audio->nb_samples = ret;
if (buf->pts != AV_NOPTS_VALUE) {
buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base) -
av_rescale(delay, outlink->sample_rate,
inlink->sample_rate);
} else
buf_out->pts = s->next_pts;
s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
ret = ff_filter_samples(outlink, buf_out);
s->got_output = 1;
}
fail:
avfilter_unref_buffer(buf);
} else {
buf->format = outlink->format;
ret = ff_filter_samples(outlink, buf);
s->got_output = 1;
}
return ret;
}
static const AVFilterPad avfilter_af_resample_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ
},
{ NULL }
};
static const AVFilterPad avfilter_af_resample_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame
},
{ NULL }
};
AVFilter avfilter_af_resample = {
.name = "resample",
.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
.priv_size = sizeof(ResampleContext),
.uninit = uninit,
.query_formats = query_formats,
.inputs = avfilter_af_resample_inputs,
.outputs = avfilter_af_resample_outputs,
};
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