/*
* 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
* sample format and channel layout conversion audio filter
*/
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/dict.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 {
const AVClass *class;
AVAudioResampleContext *avr;
AVDictionary *options;
int resampling;
int64_t next_pts;
int64_t next_in_pts;
/* set by filter_frame() to signal an output frame to request_frame() */
int got_output;
} ResampleContext;
static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
{
ResampleContext *s = ctx->priv;
const AVClass *avr_class = avresample_get_class();
AVDictionaryEntry *e = NULL;
while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
if (av_opt_find(&avr_class, e->key, NULL, 0,
AV_OPT_SEARCH_FAKE_OBJ | AV_OPT_SEARCH_CHILDREN))
av_dict_set(&s->options, e->key, e->value, 0);
}
e = NULL;
while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
av_dict_set(opts, e->key, NULL, 0);
/* do not allow the user to override basic format options */
av_dict_set(&s->options, "in_channel_layout", NULL, 0);
av_dict_set(&s->options, "out_channel_layout", NULL, 0);
av_dict_set(&s->options, "in_sample_fmt", NULL, 0);
av_dict_set(&s->options, "out_sample_fmt", NULL, 0);
av_dict_set(&s->options, "in_sample_rate", NULL, 0);
av_dict_set(&s->options, "out_sample_rate", NULL, 0);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
ResampleContext *s = ctx->priv;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
av_dict_free(&s->options);
}
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;
int64_t resampling_forced;
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);
if (s->options) {
int ret;
AVDictionaryEntry *e = NULL;
while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
ret = av_opt_set_dict(s->avr, &s->options);
if (ret < 0)
return ret;
}
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;
av_opt_get_int(s->avr, "force_resampling", 0, &resampling_forced);
s->resampling = resampling_forced || (inlink->sample_rate != outlink->sample_rate);
if (s->resampling) {
outlink->time_base = (AVRational){ 1, outlink->sample_rate };
s->next_pts = AV_NOPTS_VALUE;
s->next_in_pts = AV_NOPTS_VALUE;
} else
outlink->time_base = inlink->time_base;
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) {
AVFrame *frame;
int nb_samples = avresample_get_out_samples(s->avr, 0);
if (!nb_samples)
return ret;
frame = ff_get_audio_buffer(outlink, nb_samples);
if (!frame)
return AVERROR(ENOMEM);
ret = avresample_convert(s->avr, frame->extended_data,
frame->linesize[0], nb_samples,
NULL, 0, 0);
if (ret <= 0) {
av_frame_free(&frame);
return (ret == 0) ? AVERROR_EOF : ret;
}
frame->nb_samples = ret;
frame->pts = s->next_pts;
return ff_filter_frame(outlink, frame);
}
return ret;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret;
if (s->avr) {
AVFrame *out;
int delay, nb_samples;
/* maximum possible samples lavr can output */
delay = avresample_get_delay(s->avr);
nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
out = ff_get_audio_buffer(outlink, nb_samples);
if (!out) {
ret = AVERROR(ENOMEM);
goto fail;
}
ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
nb_samples, in->extended_data, in->linesize[0],
in->nb_samples);
if (ret <= 0) {
av_frame_free(&out);
if (ret < 0)
goto fail;
}
av_assert0(!avresample_available(s->avr));
if (s->resampling && s->next_pts == AV_NOPTS_VALUE) {
if (in->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(in->pts, inlink->time_base,
outlink->time_base);
}
if (ret > 0) {
out->nb_samples = ret;
ret = av_frame_copy_props(out, in);
if (ret < 0) {
av_frame_free(&out);
goto fail;
}
if (s->resampling) {
out->sample_rate = outlink->sample_rate;
/* Only convert in->pts if there is a discontinuous jump.
This ensures that out->pts tracks the number of samples actually
output by the resampler in the absence of such a jump.
Otherwise, the rounding in av_rescale_q() and av_rescale()
causes off-by-1 errors. */
if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
out->pts = av_rescale_q(in->pts, inlink->time_base,
outlink->time_base) -
av_rescale(delay, outlink->sample_rate,
inlink->sample_rate);
} else
out->pts = s->next_pts;
s->next_pts = out->pts + out->nb_samples;
s->next_in_pts = in->pts + in->nb_samples;
} else
out->pts = in->pts;
ret = ff_filter_frame(outlink, out);
s->got_output = 1;
}
fail:
av_frame_free(&in);
} else {
in->format = outlink->format;
ret = ff_filter_frame(outlink, in);
s->got_output = 1;
}
return ret;
}
static const AVClass *resample_child_class_next(const AVClass *prev)
{
return prev ? NULL : avresample_get_class();
}
static void *resample_child_next(void *obj, void *prev)
{
ResampleContext *s = obj;
return prev ? NULL : s->avr;
}
static const AVClass resample_class = {
.class_name = "resample",
.item_name = av_default_item_name,
.version = LIBAVUTIL_VERSION_INT,
.child_class_next = resample_child_class_next,
.child_next = resample_child_next,
};
static const AVFilterPad avfilter_af_resample_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad avfilter_af_resample_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame
},
{ NULL }
};
AVFilter ff_af_resample = {
.name = "resample",
.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
.priv_size = sizeof(ResampleContext),
.priv_class = &resample_class,
.init_dict = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = avfilter_af_resample_inputs,
.outputs = avfilter_af_resample_outputs,
};