Commit d371e7b9 authored by Anton Khirnov's avatar Anton Khirnov
Browse files

lavfi: add lavr-based audio resampling filter.

parent ea60dfe2
......@@ -1531,6 +1531,7 @@ frei0r_filter_extralibs='$ldl'
frei0r_src_filter_deps="frei0r dlopen strtok_r"
frei0r_src_filter_extralibs='$ldl'
hqdn3d_filter_deps="gpl"
resample_filter_deps="avresample"
ocv_filter_deps="libopencv"
yadif_filter_deps="gpl"
......
......@@ -111,6 +111,12 @@ Below is a description of the currently available audio filters.
Pass the audio source unchanged to the output.
@section resample
Convert the audio sample format, sample rate and channel layout. This filter is
not meant to be used directly, it is inserted automatically by libavfilter
whenever conversion is needed. Use the @var{aformat} filter to force a specific
conversion.
@c man end AUDIO FILTERS
@chapter Audio Sources
......
NAME = avfilter
FFLIBS = avutil swscale
FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
FFLIBS-$(CONFIG_RESAMPLE_FILTER) += avresample
HEADERS = avfilter.h \
avfiltergraph.h \
......@@ -22,6 +23,7 @@ OBJS = allfilters.o \
vsrc_buffer.o \
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
......
/*
*
* 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/mathematics.h"
#include "libavutil/opt.h"
#include "libavresample/avresample.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct ResampleContext {
AVAudioResampleContext *avr;
int64_t next_pts;
} 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 = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
AVFilterFormats *out_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
avfilter_formats_ref(in_formats, &inlink->out_formats);
avfilter_formats_ref(out_formats, &outlink->in_formats);
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)
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 both the input and output formats are s16 or u8, use s16 as
the internal sample format */
if (av_get_bytes_per_sample(inlink->format) <= 2 &&
av_get_bytes_per_sample(outlink->format) <= 2)
av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 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 = avfilter_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, (void**)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;
ff_filter_samples(outlink, buf);
return 0;
}
return ret;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
if (s->avr) {
AVFilterBufferRef *buf_out;
int delay, nb_samples, ret;
/* 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);
ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
buf_out->linesize[0], nb_samples,
(void**)buf->extended_data, buf->linesize[0],
buf->audio->nb_samples);
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;
ff_filter_samples(outlink, buf_out);
}
avfilter_unref_buffer(buf);
} else
ff_filter_samples(outlink, buf);
}
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 = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ },
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame },
{ .name = NULL}},
};
......@@ -35,6 +35,7 @@ void avfilter_register_all(void)
initialized = 1;
REGISTER_FILTER (ANULL, anull, af);
REGISTER_FILTER (RESAMPLE, resample, af);
REGISTER_FILTER (ANULLSRC, anullsrc, asrc);
......
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