/* * 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 }, 1, 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_freep(&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, };