lavfi: add concat filter.

Changelog

@@ -35,6 +35,7 @@ version next:
 - Opus decoder using libopus
 - caca output device using libcaca
 - alphaextract and alphamerge filters
+- concat filter
 
 
 version 0.11:

doc/filters.texi

@@ -4013,6 +4013,81 @@ tools.
 
 Below is a description of the currently available transmedia filters.
 
+@section concat
+
+Concatenate audio and video streams, joining them together one after the
+other.
+
+The filter works on segments of synchronized video and audio streams. All
+segments must have the same number of streams of each type, and that will
+also be the number of streams at output.
+
+The filter accepts the following named parameters:
+@table @option
+
+@item n
+Set the number of segments. Default is 2.
+
+@item v
+Set the number of output video streams, that is also the number of video
+streams in each segment. Default is 1.
+
+@item a
+Set the number of output audio streams, that is also the number of video
+streams in each segment. Default is 0.
+
+@end table
+
+The filter has @var{v}+@var{a} outputs: first @var{v} video outputs, then
+@var{a} audio outputs.
+
+There are @var{n}×(@var{v}+@var{a}) inputs: first the inputs for the first
+segment, in the same order as the outputs, then the inputs for the second
+segment, etc.
+
+Related streams do not always have exactly the same duration, for various
+reasons including codec frame size or sloppy authoring. For that reason,
+related synchronized streams (e.g. a video and its audio track) should be
+concatenated at once. The concat filter will use the duration of the longest
+stream in each segment (except the last one), and if necessary pad shorter
+audio streams with silence.
+
+For this filter to work correctly, all segments must start at timestamp 0.
+
+All corresponding streams must have the same parameters in all segments; the
+filtering system will automatically select a common pixel format for video
+streams, and a common sample format, sample rate and channel layout for
+audio streams, but other settings, such as resolution, must be converted
+explicitly by the user.
+
+Different frame rates are acceptable but will result in variable frame rate
+at output; be sure to configure the output file to handle it.
+
+Examples:
+@itemize
+@item
+Concatenate an opening, an episode and an ending, all in bilingual version
+(video in stream 0, audio in streams 1 and 2):
+@example
+ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \
+  '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2]
+   concat=n=3:v=1:a=2 [v] [a1] [a2]' \
+  -map '[v]' -map '[a1]' -map '[a2]' output.mkv
+@end example
+
+@item
+Concatenate two parts, handling audio and video separately, using the
+(a)movie sources, and adjusting the resolution:
+@example
+movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ;
+movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ;
+[v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa]
+@end example
+Note that a desync will happen at the stitch if the audio and video streams
+do not have exactly the same duration in the first file.
+
+@end itemize
+
 @section showwaves
 
 Convert input audio to a video output, representing the samples waves.

libavfilter/Makefile

@@ -199,6 +199,7 @@ OBJS-$(CONFIG_MP_FILTER) += libmpcodecs/vf_yvu9.o
 OBJS-$(CONFIG_MP_FILTER) += libmpcodecs/pullup.o
 
 # transmedia filters
+OBJS-$(CONFIG_CONCAT_FILTER)                 += avf_concat.o
 OBJS-$(CONFIG_SHOWWAVES_FILTER)              += avf_showwaves.o
 
 TOOLS     = graph2dot

libavfilter/allfilters.c

@@ -136,6 +136,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER (NULLSINK,    nullsink,    vsink);
 
     /* transmedia filters */
+    REGISTER_FILTER (CONCAT,      concat,      avf);
     REGISTER_FILTER (SHOWWAVES,   showwaves,   avf);
 
     /* those filters are part of public or internal API => registered

libavfilter/avf_concat.c

@@ -0,0 +1,443 @@
+/*
+ * 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/audioconvert.h"
+#include "libavutil/avassert.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 */
+    struct concat_in {
+        int64_t pts;
+        int64_t nb_frames;
+        unsigned eof;
+        struct FFBufQueue queue;
+    } *in;
+} ConcatContext;
+
+#define OFFSET(x) offsetof(ConcatContext, x)
+
+static const AVOption concat_options[] = {
+    { "n", "specify the number of segments", OFFSET(nb_segments),
+      AV_OPT_TYPE_INT, { .dbl = 2 }, 2, INT_MAX },
+    { "v", "specify the number of video streams",
+      OFFSET(nb_streams[AVMEDIA_TYPE_VIDEO]),
+      AV_OPT_TYPE_INT, { .dbl = 1 }, 1, INT_MAX },
+    { "a", "specify the number of audio streams",
+      OFFSET(nb_streams[AVMEDIA_TYPE_AUDIO]),
+      AV_OPT_TYPE_INT, { .dbl = 0 }, 0, INT_MAX },
+    { 0 }
+};
+
+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;
+    AVFilterChannelLayouts *layouts;
+
+    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];
+        /* 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 ||
+            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);
+            return AVERROR(EINVAL);
+        }
+    }
+
+    return 0;
+}
+
+static void push_frame(AVFilterContext *ctx, unsigned in_no,
+                       AVFilterBufferRef *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->audio->nb_samples,
+                                (AVRational){ 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;
+    switch (buf->type) {
+    case AVMEDIA_TYPE_VIDEO:
+        ff_start_frame(outlink, buf);
+        ff_draw_slice(outlink, 0, outlink->h, 1);
+        ff_end_frame(outlink);
+        break;
+    case AVMEDIA_TYPE_AUDIO:
+        ff_filter_samples(outlink, buf);
+        break;
+    }
+}
+
+static void process_frame(AVFilterLink *inlink, AVFilterBufferRef *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);
+        avfilter_unref_buffer(buf);
+    } if (in_no >= cat->cur_idx + ctx->nb_outputs) {
+        ff_bufqueue_add(ctx, &cat->in[in_no].queue, buf);
+    } else {
+        push_frame(ctx, in_no, buf);
+    }
+}
+
+static AVFilterBufferRef *get_video_buffer(AVFilterLink *inlink, int perms,
+                                           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, perms, w, h);
+}
+
+static AVFilterBufferRef *get_audio_buffer(AVFilterLink *inlink, int perms,
+                                           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, perms, nb_samples);
+}
+
+static int start_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+    return 0;
+}
+
+static int draw_slice(AVFilterLink *inlink, int y, int h, int dir)
+{
+    return 0;
+}
+
+static int end_frame(AVFilterLink *inlink)
+{
+    process_frame(inlink, inlink->cur_buf);
+    inlink->cur_buf = NULL;
+    return 0;
+}
+
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+    process_frame(inlink, buf);
+    return 0; /* enhancement: handle error return */
+}
+
+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)
+{
+    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;
+}
+
+static void send_silence(AVFilterContext *ctx, unsigned in_no, unsigned out_no)
+{
+    ConcatContext *cat = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[out_no];
+    int64_t base_pts = cat->in[in_no].pts;
+    int64_t nb_samples, sent = 0;
+    int frame_nb_samples;
+    AVRational rate_tb = { 1, ctx->inputs[in_no]->sample_rate };
+    AVFilterBufferRef *buf;
+    int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
+
+    if (!rate_tb.den)
+        return;
+    nb_samples = av_rescale_q(cat->delta_ts - base_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, AV_PERM_WRITE, frame_nb_samples);
+        if (!buf)
+            return;
+        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);
+        ff_filter_samples(outlink, buf);
+        sent       += frame_nb_samples;
+        nb_samples -= frame_nb_samples;
+    }
+}
+
+static void flush_segment(AVFilterContext *ctx)
+{
+    ConcatContext *cat = ctx->priv;
+    unsigned str, str_max;
+
+    find_next_delta_ts(ctx);
+    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++)
+            send_silence(ctx, cat->cur_idx - ctx->nb_outputs + str, str);
+        /* 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)
+                push_frame(ctx, str, ff_bufqueue_get(&cat->in[str].queue));
+    }
+}
+
+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;
+        }
+        flush_segment(ctx);
+        in_no += ctx->nb_outputs;
+    }
+}
+
+static av_cold int init(AVFilterContext *ctx, const char *args)
+{
+    ConcatContext *cat = ctx->priv;
+    int ret;
+    unsigned seg, type, str;
+    char name[32];
+
+    cat->class = &concat_class;
+    av_opt_set_defaults(cat);
+    ret = av_set_options_string(cat, args, "=", ":");
+    if (ret < 0) {
+        av_log(ctx, AV_LOG_ERROR, "Error parsing options: '%s'\n", args);
+        return ret;
+    }
+
+    /* 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,
+                    .min_perms        = AV_PERM_READ,
+                    .rej_perms        = AV_PERM_REUSE2,
+                    .get_video_buffer = get_video_buffer,
+                    .get_audio_buffer = get_audio_buffer,
+                };
+                snprintf(name, sizeof(name), "in%d:%c%d", seg, "va"[type], str);
+                pad.name = av_strdup(name);
+                if (type == AVMEDIA_TYPE_VIDEO) {
+                    pad.start_frame = start_frame;
+                    pad.draw_slice  = draw_slice;
+                    pad.end_frame   = end_frame;
+                } else {
+                    pad.filter_samples = filter_samples;
+                }
+                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,
+            };
+            snprintf(name, sizeof(name), "out:%c%d", "va"[type], str);
+            pad.name = av_strdup(name);
+            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 avfilter_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        = (const AVFilterPad[]) { { .name = NULL } },
+    .outputs       = (const AVFilterPad[]) { { .name = NULL } },
+};