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/*
* Copyright (c) 2012 Nicolas George
*
* 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
* concat audio-video filter
*/
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#define FF_BUFQUEUE_SIZE 256
#include "bufferqueue.h"
#include "internal.h"
#include "video.h"
#include "audio.h"
#define TYPE_ALL 2
typedef struct {
const AVClass *class;
unsigned nb_streams[TYPE_ALL]; /**< number of out streams of each type */
unsigned nb_segments;
unsigned cur_idx; /**< index of the first input of current segment */
int64_t delta_ts; /**< timestamp to add to produce output timestamps */
unsigned nb_in_active; /**< number of active inputs in current segment */
unsigned unsafe;
struct concat_in {
int64_t pts;
int64_t nb_frames;
unsigned eof;
struct FFBufQueue queue;
} *in;
} ConcatContext;
#define OFFSET(x) offsetof(ConcatContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
#define F AV_OPT_FLAG_FILTERING_PARAM
#define V AV_OPT_FLAG_VIDEO_PARAM
static const AVOption concat_options[] = {
{ "n", "specify the number of segments", OFFSET(nb_segments),
AV_OPT_TYPE_INT, { .i64 = 2 }, 2, INT_MAX, V|A|F},
{ "v", "specify the number of video streams",
OFFSET(nb_streams[AVMEDIA_TYPE_VIDEO]),
AV_OPT_TYPE_INT, { .i64 = 1 }, 0, INT_MAX, V|F },
{ "a", "specify the number of audio streams",
OFFSET(nb_streams[AVMEDIA_TYPE_AUDIO]),
AV_OPT_TYPE_INT, { .i64 = 0 }, 0, INT_MAX, A|F},
{ "unsafe", "enable unsafe mode",
OFFSET(unsafe),
AV_OPT_TYPE_INT, { .i64 = 0 }, 0, INT_MAX, V|A|F},
{ NULL }
};
AVFILTER_DEFINE_CLASS(concat);
static int query_formats(AVFilterContext *ctx)
{
ConcatContext *cat = ctx->priv;
unsigned type, nb_str, idx0 = 0, idx, str, seg;
AVFilterFormats *formats, *rates = NULL;
AVFilterChannelLayouts *layouts = NULL;
for (type = 0; type < TYPE_ALL; type++) {
nb_str = cat->nb_streams[type];
for (str = 0; str < nb_str; str++) {
idx = idx0;
/* Set the output formats */
formats = ff_all_formats(type);
if (!formats)
return AVERROR(ENOMEM);
ff_formats_ref(formats, &ctx->outputs[idx]->in_formats);
if (type == AVMEDIA_TYPE_AUDIO) {
rates = ff_all_samplerates();
if (!rates)
return AVERROR(ENOMEM);
ff_formats_ref(rates, &ctx->outputs[idx]->in_samplerates);
layouts = ff_all_channel_layouts();
if (!layouts)
return AVERROR(ENOMEM);
ff_channel_layouts_ref(layouts, &ctx->outputs[idx]->in_channel_layouts);
}
/* Set the same formats for each corresponding input */
for (seg = 0; seg < cat->nb_segments; seg++) {
ff_formats_ref(formats, &ctx->inputs[idx]->out_formats);
if (type == AVMEDIA_TYPE_AUDIO) {
ff_formats_ref(rates, &ctx->inputs[idx]->out_samplerates);
ff_channel_layouts_ref(layouts, &ctx->inputs[idx]->out_channel_layouts);
}
idx += ctx->nb_outputs;
}
idx0++;
}
}
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ConcatContext *cat = ctx->priv;
unsigned out_no = FF_OUTLINK_IDX(outlink);
unsigned in_no = out_no, seg;
AVFilterLink *inlink = ctx->inputs[in_no];
/* enhancement: find a common one */
outlink->time_base = AV_TIME_BASE_Q;
outlink->w = inlink->w;
outlink->h = inlink->h;
outlink->sample_aspect_ratio = inlink->sample_aspect_ratio;
outlink->format = inlink->format;
for (seg = 1; seg < cat->nb_segments; seg++) {
inlink = ctx->inputs[in_no += ctx->nb_outputs];
if (!outlink->sample_aspect_ratio.num)
outlink->sample_aspect_ratio = inlink->sample_aspect_ratio;
/* possible enhancement: unsafe mode, do not check */
if (outlink->w != inlink->w ||
outlink->h != inlink->h ||
outlink->sample_aspect_ratio.num != inlink->sample_aspect_ratio.num &&
inlink->sample_aspect_ratio.num ||
outlink->sample_aspect_ratio.den != inlink->sample_aspect_ratio.den) {
av_log(ctx, AV_LOG_ERROR, "Input link %s parameters "
"(size %dx%d, SAR %d:%d) do not match the corresponding "
"output link %s parameters (%dx%d, SAR %d:%d)\n",
ctx->input_pads[in_no].name, inlink->w, inlink->h,
inlink->sample_aspect_ratio.num,
inlink->sample_aspect_ratio.den,
ctx->input_pads[out_no].name, outlink->w, outlink->h,
outlink->sample_aspect_ratio.num,
outlink->sample_aspect_ratio.den);
if (!cat->unsafe)
return AVERROR(EINVAL);
}
}
return 0;
}
static int push_frame(AVFilterContext *ctx, unsigned in_no, AVFrame *buf)
{
ConcatContext *cat = ctx->priv;
unsigned out_no = in_no % ctx->nb_outputs;
AVFilterLink * inlink = ctx-> inputs[ in_no];
AVFilterLink *outlink = ctx->outputs[out_no];
struct concat_in *in = &cat->in[in_no];
buf->pts = av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
in->pts = buf->pts;
in->nb_frames++;
/* add duration to input PTS */
if (inlink->sample_rate)
/* use number of audio samples */
in->pts += av_rescale_q(buf->nb_samples,
av_make_q(1, inlink->sample_rate),
outlink->time_base);
else if (in->nb_frames >= 2)
/* use mean duration */
in->pts = av_rescale(in->pts, in->nb_frames, in->nb_frames - 1);
buf->pts += cat->delta_ts;
return ff_filter_frame(outlink, buf);
}
static int process_frame(AVFilterLink *inlink, AVFrame *buf)
{
AVFilterContext *ctx = inlink->dst;
ConcatContext *cat = ctx->priv;
unsigned in_no = FF_INLINK_IDX(inlink);
if (in_no < cat->cur_idx) {
av_log(ctx, AV_LOG_ERROR, "Frame after EOF on input %s\n",
ctx->input_pads[in_no].name);
av_frame_free(&buf);
} else if (in_no >= cat->cur_idx + ctx->nb_outputs) {
ff_bufqueue_add(ctx, &cat->in[in_no].queue, buf);
} else {
return push_frame(ctx, in_no, buf);
}
return 0;
}
static AVFrame *get_video_buffer(AVFilterLink *inlink, int w, int h)
{
AVFilterContext *ctx = inlink->dst;
unsigned in_no = FF_INLINK_IDX(inlink);
AVFilterLink *outlink = ctx->outputs[in_no % ctx->nb_outputs];
return ff_get_video_buffer(outlink, w, h);
}
static AVFrame *get_audio_buffer(AVFilterLink *inlink, int nb_samples)
{
AVFilterContext *ctx = inlink->dst;
unsigned in_no = FF_INLINK_IDX(inlink);
AVFilterLink *outlink = ctx->outputs[in_no % ctx->nb_outputs];
return ff_get_audio_buffer(outlink, nb_samples);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
return process_frame(inlink, buf);
}
static void close_input(AVFilterContext *ctx, unsigned in_no)
{
ConcatContext *cat = ctx->priv;
cat->in[in_no].eof = 1;
cat->nb_in_active--;
av_log(ctx, AV_LOG_VERBOSE, "EOF on %s, %d streams left in segment.\n",
ctx->input_pads[in_no].name, cat->nb_in_active);
}
static void find_next_delta_ts(AVFilterContext *ctx, int64_t *seg_delta)
{
ConcatContext *cat = ctx->priv;
unsigned i = cat->cur_idx;
unsigned imax = i + ctx->nb_outputs;
int64_t pts;
pts = cat->in[i++].pts;
for (; i < imax; i++)
pts = FFMAX(pts, cat->in[i].pts);
cat->delta_ts += pts;
*seg_delta = pts;
}
static int send_silence(AVFilterContext *ctx, unsigned in_no, unsigned out_no,
int64_t seg_delta)
{
ConcatContext *cat = ctx->priv;
AVFilterLink *outlink = ctx->outputs[out_no];
int64_t base_pts = cat->in[in_no].pts + cat->delta_ts - seg_delta;
int64_t nb_samples, sent = 0;
int frame_nb_samples, ret;
AVRational rate_tb = { 1, ctx->inputs[in_no]->sample_rate };
AVFrame *buf;
int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
if (!rate_tb.den)
return AVERROR_BUG;
nb_samples = av_rescale_q(seg_delta - cat->in[in_no].pts,
outlink->time_base, rate_tb);
frame_nb_samples = FFMAX(9600, rate_tb.den / 5); /* arbitrary */
while (nb_samples) {
frame_nb_samples = FFMIN(frame_nb_samples, nb_samples);
buf = ff_get_audio_buffer(outlink, frame_nb_samples);
if (!buf)
return AVERROR(ENOMEM);
av_samples_set_silence(buf->extended_data, 0, frame_nb_samples,
nb_channels, outlink->format);
buf->pts = base_pts + av_rescale_q(sent, rate_tb, outlink->time_base);
ret = ff_filter_frame(outlink, buf);
if (ret < 0)
return ret;
sent += frame_nb_samples;
nb_samples -= frame_nb_samples;
}
return 0;
}
static int flush_segment(AVFilterContext *ctx)
{
int ret;
ConcatContext *cat = ctx->priv;
unsigned str, str_max;
int64_t seg_delta;
find_next_delta_ts(ctx, &seg_delta);
cat->cur_idx += ctx->nb_outputs;
cat->nb_in_active = ctx->nb_outputs;
av_log(ctx, AV_LOG_VERBOSE, "Segment finished at pts=%"PRId64"\n",
cat->delta_ts);
if (cat->cur_idx < ctx->nb_inputs) {
/* pad audio streams with silence */
str = cat->nb_streams[AVMEDIA_TYPE_VIDEO];
str_max = str + cat->nb_streams[AVMEDIA_TYPE_AUDIO];
for (; str < str_max; str++) {
ret = send_silence(ctx, cat->cur_idx - ctx->nb_outputs + str, str,
seg_delta);
if (ret < 0)
return ret;
}
/* flush queued buffers */
/* possible enhancement: flush in PTS order */
str_max = cat->cur_idx + ctx->nb_outputs;
for (str = cat->cur_idx; str < str_max; str++) {
while (cat->in[str].queue.available) {
ret = push_frame(ctx, str, ff_bufqueue_get(&cat->in[str].queue));
if (ret < 0)
return ret;
}
}
}
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ConcatContext *cat = ctx->priv;
unsigned out_no = FF_OUTLINK_IDX(outlink);
unsigned in_no = out_no + cat->cur_idx;
unsigned str, str_max;
int ret;
while (1) {
if (in_no >= ctx->nb_inputs)
return AVERROR_EOF;
if (!cat->in[in_no].eof) {
ret = ff_request_frame(ctx->inputs[in_no]);
if (ret != AVERROR_EOF)
return ret;
close_input(ctx, in_no);
}
/* cycle on all inputs to finish the segment */
/* possible enhancement: request in PTS order */
str_max = cat->cur_idx + ctx->nb_outputs - 1;
for (str = cat->cur_idx; cat->nb_in_active;
str = str == str_max ? cat->cur_idx : str + 1) {
if (cat->in[str].eof)
continue;
ret = ff_request_frame(ctx->inputs[str]);
if (ret == AVERROR_EOF)
close_input(ctx, str);
else if (ret < 0)
return ret;
}
ret = flush_segment(ctx);
if (ret < 0)
return ret;
in_no += ctx->nb_outputs;
}
}
static av_cold int init(AVFilterContext *ctx)
{
ConcatContext *cat = ctx->priv;
unsigned seg, type, str;
/* create input pads */
for (seg = 0; seg < cat->nb_segments; seg++) {
for (type = 0; type < TYPE_ALL; type++) {
for (str = 0; str < cat->nb_streams[type]; str++) {
AVFilterPad pad = {
.type = type,
.get_video_buffer = get_video_buffer,
.get_audio_buffer = get_audio_buffer,
.filter_frame = filter_frame,
};
pad.name = av_asprintf("in%d:%c%d", seg, "va"[type], str);
ff_insert_inpad(ctx, ctx->nb_inputs, &pad);
}
}
}
/* create output pads */
for (type = 0; type < TYPE_ALL; type++) {
for (str = 0; str < cat->nb_streams[type]; str++) {
AVFilterPad pad = {
.type = type,
.config_props = config_output,
.request_frame = request_frame,
};
pad.name = av_asprintf("out:%c%d", "va"[type], str);
ff_insert_outpad(ctx, ctx->nb_outputs, &pad);
}
}
cat->in = av_calloc(ctx->nb_inputs, sizeof(*cat->in));
if (!cat->in)
return AVERROR(ENOMEM);
cat->nb_in_active = ctx->nb_outputs;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
ConcatContext *cat = ctx->priv;
unsigned i;
for (i = 0; i < ctx->nb_inputs; i++) {
av_freep(&ctx->input_pads[i].name);
ff_bufqueue_discard_all(&cat->in[i].queue);
}
for (i = 0; i < ctx->nb_outputs; i++)
av_freep(&ctx->output_pads[i].name);
av_free(cat->in);
}
AVFilter ff_avf_concat = {
.name = "concat",
.description = NULL_IF_CONFIG_SMALL("Concatenate audio and video streams."),
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.priv_size = sizeof(ConcatContext),
.inputs = NULL,
.outputs = NULL,
.priv_class = &concat_class,
.flags = AVFILTER_FLAG_DYNAMIC_INPUTS | AVFILTER_FLAG_DYNAMIC_OUTPUTS,
};
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