mirror of https://git.ffmpeg.org/ffmpeg.git
444 lines
15 KiB
C
444 lines
15 KiB
C
/*
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* Copyright (c) 2012 Nicolas George
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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* See the GNU Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public License
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* along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* concat audio-video filter
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*/
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#include "libavutil/audioconvert.h"
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#include "libavutil/avassert.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#define FF_BUFQUEUE_SIZE 256
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#include "bufferqueue.h"
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#include "internal.h"
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#include "video.h"
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#include "audio.h"
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#define TYPE_ALL 2
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typedef struct {
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const AVClass *class;
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unsigned nb_streams[TYPE_ALL]; /**< number of out streams of each type */
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unsigned nb_segments;
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unsigned cur_idx; /**< index of the first input of current segment */
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int64_t delta_ts; /**< timestamp to add to produce output timestamps */
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unsigned nb_in_active; /**< number of active inputs in current segment */
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struct concat_in {
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int64_t pts;
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int64_t nb_frames;
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unsigned eof;
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struct FFBufQueue queue;
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} *in;
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} ConcatContext;
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#define OFFSET(x) offsetof(ConcatContext, x)
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static const AVOption concat_options[] = {
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{ "n", "specify the number of segments", OFFSET(nb_segments),
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AV_OPT_TYPE_INT, { .dbl = 2 }, 2, INT_MAX },
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{ "v", "specify the number of video streams",
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OFFSET(nb_streams[AVMEDIA_TYPE_VIDEO]),
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AV_OPT_TYPE_INT, { .dbl = 1 }, 0, INT_MAX },
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{ "a", "specify the number of audio streams",
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OFFSET(nb_streams[AVMEDIA_TYPE_AUDIO]),
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AV_OPT_TYPE_INT, { .dbl = 0 }, 0, INT_MAX },
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{ 0 }
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};
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AVFILTER_DEFINE_CLASS(concat);
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static int query_formats(AVFilterContext *ctx)
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{
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ConcatContext *cat = ctx->priv;
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unsigned type, nb_str, idx0 = 0, idx, str, seg;
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AVFilterFormats *formats, *rates;
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AVFilterChannelLayouts *layouts;
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for (type = 0; type < TYPE_ALL; type++) {
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nb_str = cat->nb_streams[type];
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for (str = 0; str < nb_str; str++) {
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idx = idx0;
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/* Set the output formats */
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formats = ff_all_formats(type);
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if (!formats)
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return AVERROR(ENOMEM);
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ff_formats_ref(formats, &ctx->outputs[idx]->in_formats);
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if (type == AVMEDIA_TYPE_AUDIO) {
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rates = ff_all_samplerates();
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if (!rates)
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return AVERROR(ENOMEM);
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ff_formats_ref(rates, &ctx->outputs[idx]->in_samplerates);
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layouts = ff_all_channel_layouts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ff_channel_layouts_ref(layouts, &ctx->outputs[idx]->in_channel_layouts);
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}
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/* Set the same formats for each corresponding input */
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for (seg = 0; seg < cat->nb_segments; seg++) {
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ff_formats_ref(formats, &ctx->inputs[idx]->out_formats);
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if (type == AVMEDIA_TYPE_AUDIO) {
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ff_formats_ref(rates, &ctx->inputs[idx]->out_samplerates);
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ff_channel_layouts_ref(layouts, &ctx->inputs[idx]->out_channel_layouts);
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}
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idx += ctx->nb_outputs;
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}
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idx0++;
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}
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}
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return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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ConcatContext *cat = ctx->priv;
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unsigned out_no = FF_OUTLINK_IDX(outlink);
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unsigned in_no = out_no, seg;
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AVFilterLink *inlink = ctx->inputs[in_no];
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/* enhancement: find a common one */
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outlink->time_base = AV_TIME_BASE_Q;
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outlink->w = inlink->w;
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outlink->h = inlink->h;
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outlink->sample_aspect_ratio = inlink->sample_aspect_ratio;
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outlink->format = inlink->format;
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for (seg = 1; seg < cat->nb_segments; seg++) {
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inlink = ctx->inputs[in_no += ctx->nb_outputs];
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/* possible enhancement: unsafe mode, do not check */
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if (outlink->w != inlink->w ||
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outlink->h != inlink->h ||
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outlink->sample_aspect_ratio.num != inlink->sample_aspect_ratio.num ||
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outlink->sample_aspect_ratio.den != inlink->sample_aspect_ratio.den) {
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av_log(ctx, AV_LOG_ERROR, "Input link %s parameters "
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"(size %dx%d, SAR %d:%d) do not match the corresponding "
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"output link %s parameters (%dx%d, SAR %d:%d)\n",
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ctx->input_pads[in_no].name, inlink->w, inlink->h,
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inlink->sample_aspect_ratio.num,
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inlink->sample_aspect_ratio.den,
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ctx->input_pads[out_no].name, outlink->w, outlink->h,
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outlink->sample_aspect_ratio.num,
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outlink->sample_aspect_ratio.den);
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return AVERROR(EINVAL);
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}
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}
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return 0;
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}
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static void push_frame(AVFilterContext *ctx, unsigned in_no,
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AVFilterBufferRef *buf)
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{
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ConcatContext *cat = ctx->priv;
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unsigned out_no = in_no % ctx->nb_outputs;
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AVFilterLink * inlink = ctx-> inputs[ in_no];
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AVFilterLink *outlink = ctx->outputs[out_no];
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struct concat_in *in = &cat->in[in_no];
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buf->pts = av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
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in->pts = buf->pts;
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in->nb_frames++;
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/* add duration to input PTS */
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if (inlink->sample_rate)
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/* use number of audio samples */
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in->pts += av_rescale_q(buf->audio->nb_samples,
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(AVRational){ 1, inlink->sample_rate },
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outlink->time_base);
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else if (in->nb_frames >= 2)
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/* use mean duration */
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in->pts = av_rescale(in->pts, in->nb_frames, in->nb_frames - 1);
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buf->pts += cat->delta_ts;
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switch (buf->type) {
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case AVMEDIA_TYPE_VIDEO:
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ff_start_frame(outlink, buf);
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ff_draw_slice(outlink, 0, outlink->h, 1);
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ff_end_frame(outlink);
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break;
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case AVMEDIA_TYPE_AUDIO:
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ff_filter_samples(outlink, buf);
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break;
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}
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}
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static void process_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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AVFilterContext *ctx = inlink->dst;
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ConcatContext *cat = ctx->priv;
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unsigned in_no = FF_INLINK_IDX(inlink);
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if (in_no < cat->cur_idx) {
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av_log(ctx, AV_LOG_ERROR, "Frame after EOF on input %s\n",
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ctx->input_pads[in_no].name);
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avfilter_unref_buffer(buf);
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} if (in_no >= cat->cur_idx + ctx->nb_outputs) {
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ff_bufqueue_add(ctx, &cat->in[in_no].queue, buf);
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} else {
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push_frame(ctx, in_no, buf);
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}
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}
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static AVFilterBufferRef *get_video_buffer(AVFilterLink *inlink, int perms,
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int w, int h)
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{
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AVFilterContext *ctx = inlink->dst;
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unsigned in_no = FF_INLINK_IDX(inlink);
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AVFilterLink *outlink = ctx->outputs[in_no % ctx->nb_outputs];
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return ff_get_video_buffer(outlink, perms, w, h);
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}
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static AVFilterBufferRef *get_audio_buffer(AVFilterLink *inlink, int perms,
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int nb_samples)
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{
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AVFilterContext *ctx = inlink->dst;
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unsigned in_no = FF_INLINK_IDX(inlink);
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AVFilterLink *outlink = ctx->outputs[in_no % ctx->nb_outputs];
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return ff_get_audio_buffer(outlink, perms, nb_samples);
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}
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static int start_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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return 0;
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}
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static int draw_slice(AVFilterLink *inlink, int y, int h, int dir)
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{
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return 0;
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}
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static int end_frame(AVFilterLink *inlink)
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{
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process_frame(inlink, inlink->cur_buf);
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inlink->cur_buf = NULL;
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return 0;
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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process_frame(inlink, buf);
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return 0; /* enhancement: handle error return */
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}
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static void close_input(AVFilterContext *ctx, unsigned in_no)
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{
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ConcatContext *cat = ctx->priv;
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cat->in[in_no].eof = 1;
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cat->nb_in_active--;
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av_log(ctx, AV_LOG_VERBOSE, "EOF on %s, %d streams left in segment.\n",
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ctx->input_pads[in_no].name, cat->nb_in_active);
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}
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static void find_next_delta_ts(AVFilterContext *ctx)
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{
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ConcatContext *cat = ctx->priv;
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unsigned i = cat->cur_idx;
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unsigned imax = i + ctx->nb_outputs;
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int64_t pts;
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pts = cat->in[i++].pts;
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for (; i < imax; i++)
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pts = FFMAX(pts, cat->in[i].pts);
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cat->delta_ts += pts;
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}
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static void send_silence(AVFilterContext *ctx, unsigned in_no, unsigned out_no)
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{
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ConcatContext *cat = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[out_no];
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int64_t base_pts = cat->in[in_no].pts + cat->delta_ts;
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int64_t nb_samples, sent = 0;
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int frame_nb_samples;
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AVRational rate_tb = { 1, ctx->inputs[in_no]->sample_rate };
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AVFilterBufferRef *buf;
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int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
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if (!rate_tb.den)
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return;
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nb_samples = av_rescale_q(cat->delta_ts - base_pts,
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outlink->time_base, rate_tb);
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frame_nb_samples = FFMAX(9600, rate_tb.den / 5); /* arbitrary */
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while (nb_samples) {
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frame_nb_samples = FFMIN(frame_nb_samples, nb_samples);
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buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, frame_nb_samples);
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if (!buf)
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return;
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av_samples_set_silence(buf->extended_data, 0, frame_nb_samples,
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nb_channels, outlink->format);
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buf->pts = base_pts + av_rescale_q(sent, rate_tb, outlink->time_base);
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ff_filter_samples(outlink, buf);
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sent += frame_nb_samples;
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nb_samples -= frame_nb_samples;
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}
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}
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static void flush_segment(AVFilterContext *ctx)
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{
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ConcatContext *cat = ctx->priv;
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unsigned str, str_max;
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find_next_delta_ts(ctx);
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cat->cur_idx += ctx->nb_outputs;
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cat->nb_in_active = ctx->nb_outputs;
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av_log(ctx, AV_LOG_VERBOSE, "Segment finished at pts=%"PRId64"\n",
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cat->delta_ts);
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if (cat->cur_idx < ctx->nb_inputs) {
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/* pad audio streams with silence */
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str = cat->nb_streams[AVMEDIA_TYPE_VIDEO];
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str_max = str + cat->nb_streams[AVMEDIA_TYPE_AUDIO];
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for (; str < str_max; str++)
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send_silence(ctx, cat->cur_idx - ctx->nb_outputs + str, str);
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/* flush queued buffers */
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/* possible enhancement: flush in PTS order */
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str_max = cat->cur_idx + ctx->nb_outputs;
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for (str = cat->cur_idx; str < str_max; str++)
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while (cat->in[str].queue.available)
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push_frame(ctx, str, ff_bufqueue_get(&cat->in[str].queue));
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}
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}
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static int request_frame(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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ConcatContext *cat = ctx->priv;
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unsigned out_no = FF_OUTLINK_IDX(outlink);
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unsigned in_no = out_no + cat->cur_idx;
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unsigned str, str_max;
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int ret;
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while (1) {
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if (in_no >= ctx->nb_inputs)
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return AVERROR_EOF;
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if (!cat->in[in_no].eof) {
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ret = ff_request_frame(ctx->inputs[in_no]);
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if (ret != AVERROR_EOF)
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return ret;
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close_input(ctx, in_no);
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}
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/* cycle on all inputs to finish the segment */
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/* possible enhancement: request in PTS order */
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str_max = cat->cur_idx + ctx->nb_outputs - 1;
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for (str = cat->cur_idx; cat->nb_in_active;
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str = str == str_max ? cat->cur_idx : str + 1) {
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if (cat->in[str].eof)
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continue;
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ret = ff_request_frame(ctx->inputs[str]);
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if (ret == AVERROR_EOF)
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close_input(ctx, str);
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else if (ret < 0)
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return ret;
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}
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flush_segment(ctx);
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in_no += ctx->nb_outputs;
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}
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}
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static av_cold int init(AVFilterContext *ctx, const char *args)
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{
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ConcatContext *cat = ctx->priv;
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int ret;
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unsigned seg, type, str;
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char name[32];
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cat->class = &concat_class;
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av_opt_set_defaults(cat);
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ret = av_set_options_string(cat, args, "=", ":");
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if (ret < 0) {
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av_log(ctx, AV_LOG_ERROR, "Error parsing options: '%s'\n", args);
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return ret;
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}
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/* create input pads */
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for (seg = 0; seg < cat->nb_segments; seg++) {
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for (type = 0; type < TYPE_ALL; type++) {
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for (str = 0; str < cat->nb_streams[type]; str++) {
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AVFilterPad pad = {
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.type = type,
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.min_perms = AV_PERM_READ,
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.rej_perms = AV_PERM_REUSE2,
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.get_video_buffer = get_video_buffer,
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.get_audio_buffer = get_audio_buffer,
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};
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snprintf(name, sizeof(name), "in%d:%c%d", seg, "va"[type], str);
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pad.name = av_strdup(name);
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if (type == AVMEDIA_TYPE_VIDEO) {
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pad.start_frame = start_frame;
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pad.draw_slice = draw_slice;
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pad.end_frame = end_frame;
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} else {
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pad.filter_samples = filter_samples;
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}
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ff_insert_inpad(ctx, ctx->nb_inputs, &pad);
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}
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}
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}
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/* create output pads */
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for (type = 0; type < TYPE_ALL; type++) {
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for (str = 0; str < cat->nb_streams[type]; str++) {
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AVFilterPad pad = {
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.type = type,
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.config_props = config_output,
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.request_frame = request_frame,
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};
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snprintf(name, sizeof(name), "out:%c%d", "va"[type], str);
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pad.name = av_strdup(name);
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ff_insert_outpad(ctx, ctx->nb_outputs, &pad);
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}
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}
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cat->in = av_calloc(ctx->nb_inputs, sizeof(*cat->in));
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if (!cat->in)
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return AVERROR(ENOMEM);
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cat->nb_in_active = ctx->nb_outputs;
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ConcatContext *cat = ctx->priv;
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unsigned i;
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for (i = 0; i < ctx->nb_inputs; i++) {
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av_freep(&ctx->input_pads[i].name);
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ff_bufqueue_discard_all(&cat->in[i].queue);
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}
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for (i = 0; i < ctx->nb_outputs; i++)
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av_freep(&ctx->output_pads[i].name);
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av_free(cat->in);
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}
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AVFilter avfilter_avf_concat = {
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.name = "concat",
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.description = NULL_IF_CONFIG_SMALL("Concatenate audio and video streams."),
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.init = init,
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.uninit = uninit,
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.query_formats = query_formats,
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.priv_size = sizeof(ConcatContext),
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.inputs = (const AVFilterPad[]) { { .name = NULL } },
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.outputs = (const AVFilterPad[]) { { .name = NULL } },
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};
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