[FFmpeg-devel] [PATCH 2/2] lavfi: add concat filter.
Nicolas George
nicolas.george at normalesup.org
Fri Jul 20 10:40:55 CEST 2012
Signed-off-by: Nicolas George <nicolas.george at normalesup.org>
---
Changelog | 1 +
doc/filters.texi | 75 ++++++++
libavfilter/Makefile | 1 +
libavfilter/allfilters.c | 1 +
libavfilter/f_concat.c | 428 ++++++++++++++++++++++++++++++++++++++++++++++
5 files changed, 506 insertions(+)
create mode 100644 libavfilter/f_concat.c
diff --git a/Changelog b/Changelog
index 4242cea..15ac8d6 100644
--- a/Changelog
+++ b/Changelog
@@ -32,6 +32,7 @@ version next:
- 3GPP Timed Text decoder
- GeoTIFF decoder support
- ffmpeg -(no)stdin option
+- concat filter
version 0.11:
diff --git a/doc/filters.texi b/doc/filters.texi
index 4a6c092..306d82d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -3988,6 +3988,81 @@ tools.
Below is a description of the currently available transmedia filters.
+ at 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:
+ at table @option
+
+ at item n
+Set the number of segments. Default is 2.
+
+ at item v
+Set the number of output video streams, that is also the number of video
+streams in each segment. Default is 1.
+
+ at item a
+Set the number of output audio streams, that is also the number of video
+streams in each segment. Default is 0.
+
+ at end table
+
+The filter has @var{v}+ at var{a} outputs: first @var{v} video outputs, then
+ at var{a} audio outputs.
+
+There are @var{s}×(@var{v}+ at 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:
+ at itemize
+ at item
+Concatenate an opening, an episode and an ending, all in bilingual version
+(video in stream 0, audio in streams 1 and 2):
+ at 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=s=3:v=1:a=2 [v] [a1] [a2]' \
+ -map '[v]' -map '[a1]' -map '[a2]' output.mkv
+ at end example
+
+ at item
+Concatenate two parts, handling audio and video separately, using the
+(a)movie sources, and adjusting the resolution:
+ at 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]
+ at 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.
+
+ at end itemize
+
@section showwaves
Convert input audio to a video output, representing the samples waves.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index b094f59..642a105 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -197,6 +197,7 @@ OBJS-$(CONFIG_MP_FILTER) += libmpcodecs/vf_yvu9.o
OBJS-$(CONFIG_MP_FILTER) += libmpcodecs/pullup.o
# transmedia filters
+OBJS-$(CONFIG_CONCAT_FILTER) += f_concat.o
OBJS-$(CONFIG_SHOWWAVES_FILTER) += avf_showwaves.o
TOOLS = graph2dot
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 706405e..cae4c99 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -134,6 +134,7 @@ 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
diff --git a/libavfilter/f_concat.c b/libavfilter/f_concat.c
new file mode 100644
index 0000000..fb8fb15
--- /dev/null
+++ b/libavfilter/f_concat.c
@@ -0,0 +1,428 @@
+/*
+ * 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/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 "
+ "(%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 void start_frame(AVFilterLink *inlink, AVFilterBufferRef *buf) { }
+
+static void draw_slice(AVFilterLink *inlink, int y, int h, int dir) { }
+
+static void end_frame(AVFilterLink *inlink)
+{
+ process_frame(inlink, inlink->cur_buf);
+}
+
+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_size;
+ AVRational rate_tb = { 1, ctx->inputs[in_no]->sample_rate };
+ AVFilterBufferRef *buf;
+
+ if (!rate_tb.den)
+ return;
+ nb_samples = av_rescale_q(cat->delta_ts - base_pts,
+ outlink->time_base, rate_tb);
+ frame_size = FFMAX(9600, rate_tb.den / 5); /* arbitrary */
+ while (nb_samples) {
+ frame_size = FFMIN(frame_size, nb_samples);
+ buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, frame_size);
+ if (!buf)
+ return;
+ buf->pts = base_pts + av_rescale_q(sent, rate_tb, outlink->time_base);
+ ff_filter_samples(outlink, buf);
+ sent += frame_size;
+ nb_samples -= frame_size;
+ }
+}
+
+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 } },
+};
--
1.7.10.4
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