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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/channel_layout.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 {
const AVClass *class;
int sample_rate_arg;
double ratio;
struct SwrContext *swr;
int64_t next_pts;
int req_fullfilled;
int more_data;
} AResampleContext;
static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
{
AResampleContext *aresample = ctx->priv;
int ret = 0;
aresample->next_pts = AV_NOPTS_VALUE;
aresample->swr = swr_alloc();
if (!aresample->swr) {
ret = AVERROR(ENOMEM);
goto end;
}
if (opts) {
AVDictionaryEntry *e = NULL;
while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
goto end;
}
av_dict_free(opts);
}
if (aresample->sample_rate_arg > 0)
av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
end:
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 = ff_all_formats(AVMEDIA_TYPE_AUDIO);
AVFilterFormats *out_formats;
AVFilterFormats *in_samplerates = ff_all_samplerates();
AVFilterFormats *out_samplerates;
AVFilterChannelLayouts *in_layouts = ff_all_channel_counts();
AVFilterChannelLayouts *out_layouts;
ff_formats_ref (in_formats, &inlink->out_formats);
ff_formats_ref (in_samplerates, &inlink->out_samplerates);
ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
if(out_rate > 0) {
int ratelist[] = { out_rate, -1 };
out_samplerates = ff_make_format_list(ratelist);
} else {
out_samplerates = ff_all_samplerates();
}
if (!out_samplerates) {
av_log(ctx, AV_LOG_ERROR, "Cannot allocate output samplerates.\n");
return AVERROR(ENOMEM);
}
ff_formats_ref(out_samplerates, &outlink->in_samplerates);
if(out_format != AV_SAMPLE_FMT_NONE) {
int formatlist[] = { out_format, -1 };
out_formats = ff_make_format_list(formatlist);
} else
out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
ff_formats_ref(out_formats, &outlink->in_formats);
if(out_layout) {
int64_t layout_list[] = { out_layout, -1 };
out_layouts = avfilter_make_format64_list(layout_list);
} else
out_layouts = ff_all_channel_counts();
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;
char inchl_buf[128], outchl_buf[128];
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);
if (!inlink->channel_layout)
av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
if (!outlink->channel_layout)
av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
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 || !outlink->channel_layout);
av_assert0(outlink->format == out_format);
aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->nb_samples;
int64_t delay;
int n_out = n_in * aresample->ratio + 32;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFrame *outsamplesref;
int ret;
delay = swr_get_delay(aresample->swr, outlink->sample_rate);
if (delay > 0)
n_out += FFMIN(delay, FFMAX(4096, n_out));
outsamplesref = ff_get_audio_buffer(outlink, n_out);
if(!outsamplesref)
return AVERROR(ENOMEM);
av_frame_copy_props(outsamplesref, insamplesref);
outsamplesref->format = outlink->format;
av_frame_set_channels(outsamplesref, outlink->channels);
outsamplesref->channel_layout = outlink->channel_layout;
outsamplesref->sample_rate = outlink->sample_rate;
if(insamplesref->pts != AV_NOPTS_VALUE) {
int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
int64_t outpts= swr_next_pts(aresample->swr, inpts);
aresample->next_pts =
outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
} else {
outsamplesref->pts = AV_NOPTS_VALUE;
}
n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
(void *)insamplesref->extended_data, n_in);
if (n_out <= 0) {
av_frame_free(&outsamplesref);
av_frame_free(&insamplesref);
return 0;
}
aresample->more_data = outsamplesref->nb_samples == n_out; // Indicate that there is probably more data in our buffers
outsamplesref->nb_samples = n_out;
ret = ff_filter_frame(outlink, outsamplesref);
aresample->req_fullfilled= 1;
av_frame_free(&insamplesref);
return ret;
}
static int flush_frame(AVFilterLink *outlink, int final, AVFrame **outsamplesref_ret)
{
AVFilterContext *ctx = outlink->src;
AResampleContext *aresample = ctx->priv;
AVFilterLink *const inlink = outlink->src->inputs[0];
AVFrame *outsamplesref;
int n_out = 4096;
int64_t pts;
outsamplesref = ff_get_audio_buffer(outlink, n_out);
*outsamplesref_ret = outsamplesref;
if (!outsamplesref)
return AVERROR(ENOMEM);
pts = swr_next_pts(aresample->swr, INT64_MIN);
pts = ROUNDED_DIV(pts, inlink->sample_rate);
n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, final ? NULL : (void*)outsamplesref->extended_data, 0);
if (n_out <= 0) {
av_frame_free(&outsamplesref);
return (n_out == 0) ? AVERROR_EOF : n_out;
}
outsamplesref->sample_rate = outlink->sample_rate;
outsamplesref->nb_samples = n_out;
outsamplesref->pts = pts;
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AResampleContext *aresample = ctx->priv;
int ret;
// First try to get data from the internal buffers
if (aresample->more_data) {
AVFrame *outsamplesref;
if (flush_frame(outlink, 0, &outsamplesref) >= 0) {
return ff_filter_frame(outlink, outsamplesref);
}
}
aresample->more_data = 0;
// Second request more data from the input
aresample->req_fullfilled = 0;
do{
ret = ff_request_frame(ctx->inputs[0]);
}while(!aresample->req_fullfilled && ret>=0);
// Third if we hit the end flush
if (ret == AVERROR_EOF) {
AVFrame *outsamplesref;
if ((ret = flush_frame(outlink, 1, &outsamplesref)) < 0)
return ret;
return ff_filter_frame(outlink, outsamplesref);
}
return ret;
}
static const AVClass *resample_child_class_next(const AVClass *prev)
{
return prev ? NULL : swr_get_class();
}
static void *resample_child_next(void *obj, void *prev)
{
AResampleContext *s = obj;
return prev ? NULL : s->swr;
}
#define OFFSET(x) offsetof(AResampleContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption options[] = {
{"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
{NULL}
};
static const AVClass aresample_class = {
.class_name = "aresample",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
.child_class_next = resample_child_class_next,
.child_next = resample_child_next,
};
static const AVFilterPad aresample_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad aresample_outputs[] = {
{
.name = "default",
.config_props = config_output,
.request_frame = request_frame,
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_aresample = {
.name = "aresample",
.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
.init_dict = init_dict,
.uninit = uninit,
.query_formats = query_formats,
.priv_size = sizeof(AResampleContext),
.priv_class = &aresample_class,
.inputs = aresample_inputs,
.outputs = aresample_outputs,
};