/* * 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 */ #include "libavutil/avassert.h" #include "libavutil/dict.h" #include "libavutil/error.h" #include "libavutil/log.h" #include "libavutil/pixdesc.h" #include "libavutil/pixfmt.h" #include "libavutil/timestamp.h" #include "libavcodec/avcodec.h" #include "libavcodec/codec.h" #include "libavfilter/buffersrc.h" #include "ffmpeg.h" static int send_frame_to_filters(InputStream *ist, AVFrame *decoded_frame) { int i, ret; av_assert1(ist->nb_filters > 0); /* ensure ret is initialized */ for (i = 0; i < ist->nb_filters; i++) { ret = ifilter_send_frame(ist->filters[i], decoded_frame, i < ist->nb_filters - 1); if (ret == AVERROR_EOF) ret = 0; /* ignore */ if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "Failed to inject frame into filter network: %s\n", av_err2str(ret)); break; } } return ret; } static AVRational audio_samplerate_update(InputStream *ist, const AVFrame *frame) { const int prev = ist->last_frame_tb.den; const int sr = frame->sample_rate; AVRational tb_new; int64_t gcd; if (frame->sample_rate == ist->last_frame_sample_rate) goto finish; gcd = av_gcd(prev, sr); if (prev / gcd >= INT_MAX / sr) { av_log(ist, AV_LOG_WARNING, "Audio timestamps cannot be represented exactly after " "sample rate change: %d -> %d\n", prev, sr); // LCM of 192000, 44100, allows to represent all common samplerates tb_new = (AVRational){ 1, 28224000 }; } else tb_new = (AVRational){ 1, prev / gcd * sr }; // keep the frame timebase if it is strictly better than // the samplerate-defined one if (frame->time_base.num == 1 && frame->time_base.den > tb_new.den && !(frame->time_base.den % tb_new.den)) tb_new = frame->time_base; if (ist->last_frame_pts != AV_NOPTS_VALUE) ist->last_frame_pts = av_rescale_q(ist->last_frame_pts, ist->last_frame_tb, tb_new); ist->last_frame_duration_est = av_rescale_q(ist->last_frame_duration_est, ist->last_frame_tb, tb_new); ist->last_frame_tb = tb_new; ist->last_frame_sample_rate = frame->sample_rate; finish: return ist->last_frame_tb; } static void audio_ts_process(InputStream *ist, AVFrame *frame) { AVRational tb_filter = (AVRational){1, frame->sample_rate}; AVRational tb; int64_t pts_pred; // on samplerate change, choose a new internal timebase for timestamp // generation that can represent timestamps from all the samplerates // seen so far tb = audio_samplerate_update(ist, frame); pts_pred = ist->last_frame_pts == AV_NOPTS_VALUE ? 0 : ist->last_frame_pts + ist->last_frame_duration_est; if (frame->pts == AV_NOPTS_VALUE) { frame->pts = pts_pred; frame->time_base = tb; } else if (ist->last_frame_pts != AV_NOPTS_VALUE && frame->pts > av_rescale_q_rnd(pts_pred, tb, frame->time_base, AV_ROUND_UP)) { // there was a gap in timestamps, reset conversion state ist->filter_in_rescale_delta_last = AV_NOPTS_VALUE; } frame->pts = av_rescale_delta(frame->time_base, frame->pts, tb, frame->nb_samples, &ist->filter_in_rescale_delta_last, tb); ist->last_frame_pts = frame->pts; ist->last_frame_duration_est = av_rescale_q(frame->nb_samples, tb_filter, tb); // finally convert to filtering timebase frame->pts = av_rescale_q(frame->pts, tb, tb_filter); frame->duration = frame->nb_samples; frame->time_base = tb_filter; } static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *frame) { const InputFile *ifile = input_files[ist->file_index]; int64_t codec_duration = 0; // XXX lavf currently makes up frame durations when they are not provided by // the container. As there is no way to reliably distinguish real container // durations from the fake made-up ones, we use heuristics based on whether // the container has timestamps. Eventually lavf should stop making up // durations, then this should be simplified. // prefer frame duration for containers with timestamps if (frame->duration > 0 && (!ifile->format_nots || ist->framerate.num)) return frame->duration; if (ist->dec_ctx->framerate.den && ist->dec_ctx->framerate.num) { int fields = frame->repeat_pict + 2; AVRational field_rate = av_mul_q(ist->dec_ctx->framerate, (AVRational){ 2, 1 }); codec_duration = av_rescale_q(fields, av_inv_q(field_rate), frame->time_base); } // prefer codec-layer duration for containers without timestamps if (codec_duration > 0 && ifile->format_nots) return codec_duration; // when timestamps are available, repeat last frame's actual duration // (i.e. pts difference between this and last frame) if (frame->pts != AV_NOPTS_VALUE && ist->last_frame_pts != AV_NOPTS_VALUE && frame->pts > ist->last_frame_pts) return frame->pts - ist->last_frame_pts; // try frame/codec duration if (frame->duration > 0) return frame->duration; if (codec_duration > 0) return codec_duration; // try average framerate if (ist->st->avg_frame_rate.num && ist->st->avg_frame_rate.den) { int64_t d = av_rescale_q(1, av_inv_q(ist->st->avg_frame_rate), frame->time_base); if (d > 0) return d; } // last resort is last frame's estimated duration, and 1 return FFMAX(ist->last_frame_duration_est, 1); } static int video_frame_process(InputStream *ist, AVFrame *frame) { // The following line may be required in some cases where there is no parser // or the parser does not has_b_frames correctly if (ist->par->video_delay < ist->dec_ctx->has_b_frames) { if (ist->dec_ctx->codec_id == AV_CODEC_ID_H264) { ist->par->video_delay = ist->dec_ctx->has_b_frames; } else av_log(ist->dec_ctx, AV_LOG_WARNING, "video_delay is larger in decoder than demuxer %d > %d.\n" "If you want to help, upload a sample " "of this file to https://streams.videolan.org/upload/ " "and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)\n", ist->dec_ctx->has_b_frames, ist->par->video_delay); } if (ist->dec_ctx->width != frame->width || ist->dec_ctx->height != frame->height || ist->dec_ctx->pix_fmt != frame->format) { av_log(NULL, AV_LOG_DEBUG, "Frame parameters mismatch context %d,%d,%d != %d,%d,%d\n", frame->width, frame->height, frame->format, ist->dec_ctx->width, ist->dec_ctx->height, ist->dec_ctx->pix_fmt); } if(ist->top_field_first>=0) frame->flags |= AV_FRAME_FLAG_TOP_FIELD_FIRST; if (ist->hwaccel_retrieve_data && frame->format == ist->hwaccel_pix_fmt) { int err = ist->hwaccel_retrieve_data(ist->dec_ctx, frame); if (err < 0) return err; } frame->pts = frame->best_effort_timestamp; // forced fixed framerate if (ist->framerate.num) { frame->pts = AV_NOPTS_VALUE; frame->duration = 1; frame->time_base = av_inv_q(ist->framerate); } // no timestamp available - extrapolate from previous frame duration if (frame->pts == AV_NOPTS_VALUE) frame->pts = ist->last_frame_pts == AV_NOPTS_VALUE ? 0 : ist->last_frame_pts + ist->last_frame_duration_est; // update timestamp history ist->last_frame_duration_est = video_duration_estimate(ist, frame); ist->last_frame_pts = frame->pts; ist->last_frame_tb = frame->time_base; if (debug_ts) { av_log(ist, AV_LOG_INFO, "decoder -> pts:%s pts_time:%s " "pkt_dts:%s pkt_dts_time:%s " "duration:%s duration_time:%s " "keyframe:%d frame_type:%d time_base:%d/%d\n", av_ts2str(frame->pts), av_ts2timestr(frame->pts, &frame->time_base), av_ts2str(frame->pkt_dts), av_ts2timestr(frame->pkt_dts, &frame->time_base), av_ts2str(frame->duration), av_ts2timestr(frame->duration, &frame->time_base), !!(frame->flags & AV_FRAME_FLAG_KEY), frame->pict_type, frame->time_base.num, frame->time_base.den); } if (ist->st->sample_aspect_ratio.num) frame->sample_aspect_ratio = ist->st->sample_aspect_ratio; return 0; } static void sub2video_flush(InputStream *ist) { int i; int ret; if (ist->sub2video.end_pts < INT64_MAX) sub2video_update(ist, INT64_MAX, NULL); for (i = 0; i < ist->nb_filters; i++) { ret = av_buffersrc_add_frame(ist->filters[i]->filter, NULL); if (ret != AVERROR_EOF && ret < 0) av_log(NULL, AV_LOG_WARNING, "Flush the frame error.\n"); } } int process_subtitle(InputStream *ist, AVSubtitle *subtitle, int *got_output) { int ret = 0; int free_sub = 1; if (ist->fix_sub_duration) { int end = 1; if (ist->prev_sub.got_output) { end = av_rescale(subtitle->pts - ist->prev_sub.subtitle.pts, 1000, AV_TIME_BASE); if (end < ist->prev_sub.subtitle.end_display_time) { av_log(NULL, AV_LOG_DEBUG, "Subtitle duration reduced from %"PRId32" to %d%s\n", ist->prev_sub.subtitle.end_display_time, end, end <= 0 ? ", dropping it" : ""); ist->prev_sub.subtitle.end_display_time = end; } } FFSWAP(int, *got_output, ist->prev_sub.got_output); FFSWAP(int, ret, ist->prev_sub.ret); FFSWAP(AVSubtitle, *subtitle, ist->prev_sub.subtitle); if (end <= 0) goto out; } if (!*got_output) return ret; if (ist->sub2video.frame) { sub2video_update(ist, INT64_MIN, subtitle); } else if (ist->nb_filters) { if (!ist->sub2video.sub_queue) ist->sub2video.sub_queue = av_fifo_alloc2(8, sizeof(AVSubtitle), AV_FIFO_FLAG_AUTO_GROW); if (!ist->sub2video.sub_queue) report_and_exit(AVERROR(ENOMEM)); ret = av_fifo_write(ist->sub2video.sub_queue, subtitle, 1); if (ret < 0) exit_program(1); free_sub = 0; } if (!subtitle->num_rects) goto out; for (int oidx = 0; oidx < ist->nb_outputs; oidx++) { OutputStream *ost = ist->outputs[oidx]; if (!ost->enc || ost->type != AVMEDIA_TYPE_SUBTITLE) continue; enc_subtitle(output_files[ost->file_index], ost, subtitle); } out: if (free_sub) avsubtitle_free(subtitle); return ret; } static int transcode_subtitles(InputStream *ist, const AVPacket *pkt) { AVSubtitle subtitle; int got_output; int ret = avcodec_decode_subtitle2(ist->dec_ctx, &subtitle, &got_output, pkt); if (ret < 0) { av_log(ist, AV_LOG_ERROR, "Error decoding subtitles: %s\n", av_err2str(ret)); if (exit_on_error) exit_program(1); ist->decode_errors++; } if (ret < 0 || !got_output) { if (!pkt->size) sub2video_flush(ist); return ret < 0 ? ret : AVERROR_EOF; } ist->frames_decoded++; return process_subtitle(ist, &subtitle, &got_output); } static int send_filter_eof(InputStream *ist) { int i, ret; for (i = 0; i < ist->nb_filters; i++) { int64_t end_pts = ist->last_frame_pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE : ist->last_frame_pts + ist->last_frame_duration_est; ret = ifilter_send_eof(ist->filters[i], end_pts, ist->last_frame_tb); if (ret < 0) return ret; } return 0; } int dec_packet(InputStream *ist, const AVPacket *pkt, int no_eof) { AVCodecContext *dec = ist->dec_ctx; const char *type_desc = av_get_media_type_string(dec->codec_type); int ret; if (dec->codec_type == AVMEDIA_TYPE_SUBTITLE) return transcode_subtitles(ist, pkt ? pkt : ist->pkt); // With fate-indeo3-2, we're getting 0-sized packets before EOF for some // reason. This seems like a semi-critical bug. Don't trigger EOF, and // skip the packet. if (pkt && pkt->size == 0) return 0; ret = avcodec_send_packet(dec, pkt); if (ret < 0 && !(ret == AVERROR_EOF && !pkt)) { // In particular, we don't expect AVERROR(EAGAIN), because we read all // decoded frames with avcodec_receive_frame() until done. av_log(ist, AV_LOG_ERROR, "Error submitting %s to decoder: %s\n", pkt ? "packet" : "EOF", av_err2str(ret)); if (exit_on_error) exit_program(1); if (ret != AVERROR_EOF) ist->decode_errors++; return ret; } while (1) { AVFrame *frame = ist->decoded_frame; update_benchmark(NULL); ret = avcodec_receive_frame(dec, frame); update_benchmark("decode_%s %d.%d", type_desc, ist->file_index, ist->st->index); if (ret == AVERROR(EAGAIN)) { av_assert0(pkt); // should never happen during flushing return 0; } else if (ret == AVERROR_EOF) { /* after flushing, send an EOF on all the filter inputs attached to the stream */ /* except when looping we need to flush but not to send an EOF */ if (!no_eof) { ret = send_filter_eof(ist); if (ret < 0) { av_log(NULL, AV_LOG_FATAL, "Error marking filters as finished\n"); exit_program(1); } } return AVERROR_EOF; } else if (ret < 0) { av_log(ist, AV_LOG_ERROR, "Decoding error: %s\n", av_err2str(ret)); if (exit_on_error) exit_program(1); ist->decode_errors++; return ret; } if (frame->decode_error_flags || (frame->flags & AV_FRAME_FLAG_CORRUPT)) { av_log(ist, exit_on_error ? AV_LOG_FATAL : AV_LOG_WARNING, "corrupt decoded frame\n"); if (exit_on_error) exit_program(1); } if (ist->want_frame_data) { FrameData *fd; av_assert0(!frame->opaque_ref); frame->opaque_ref = av_buffer_allocz(sizeof(*fd)); if (!frame->opaque_ref) { av_frame_unref(frame); report_and_exit(AVERROR(ENOMEM)); } fd = (FrameData*)frame->opaque_ref->data; fd->pts = frame->pts; fd->tb = dec->pkt_timebase; fd->idx = dec->frame_num - 1; } frame->time_base = dec->pkt_timebase; if (dec->codec_type == AVMEDIA_TYPE_AUDIO) { ist->samples_decoded += frame->nb_samples; ist->nb_samples = frame->nb_samples; audio_ts_process(ist, frame); } else { ret = video_frame_process(ist, frame); if (ret < 0) { av_log(NULL, AV_LOG_FATAL, "Error while processing the decoded " "data for stream #%d:%d\n", ist->file_index, ist->st->index); exit_program(1); } } ist->frames_decoded++; ret = send_frame_to_filters(ist, frame); av_frame_unref(frame); if (ret < 0) exit_program(1); } } static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat *pix_fmts) { InputStream *ist = s->opaque; const enum AVPixelFormat *p; int ret; for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) { const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(*p); const AVCodecHWConfig *config = NULL; int i; if (!(desc->flags & AV_PIX_FMT_FLAG_HWACCEL)) break; if (ist->hwaccel_id == HWACCEL_GENERIC || ist->hwaccel_id == HWACCEL_AUTO) { for (i = 0;; i++) { config = avcodec_get_hw_config(s->codec, i); if (!config) break; if (!(config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX)) continue; if (config->pix_fmt == *p) break; } } if (config && config->device_type == ist->hwaccel_device_type) { ret = hwaccel_decode_init(s); if (ret < 0) { if (ist->hwaccel_id == HWACCEL_GENERIC) { av_log(NULL, AV_LOG_FATAL, "%s hwaccel requested for input stream #%d:%d, " "but cannot be initialized.\n", av_hwdevice_get_type_name(config->device_type), ist->file_index, ist->st->index); return AV_PIX_FMT_NONE; } continue; } ist->hwaccel_pix_fmt = *p; break; } } return *p; } int dec_open(InputStream *ist) { const AVCodec *codec = ist->dec; int ret; if (!codec) { av_log(ist, AV_LOG_ERROR, "Decoding requested, but no decoder found for: %s\n", avcodec_get_name(ist->dec_ctx->codec_id)); return AVERROR(EINVAL); } ist->dec_ctx->opaque = ist; ist->dec_ctx->get_format = get_format; if (ist->dec_ctx->codec_id == AV_CODEC_ID_DVB_SUBTITLE && (ist->decoding_needed & DECODING_FOR_OST)) { av_dict_set(&ist->decoder_opts, "compute_edt", "1", AV_DICT_DONT_OVERWRITE); if (ist->decoding_needed & DECODING_FOR_FILTER) av_log(NULL, AV_LOG_WARNING, "Warning using DVB subtitles for filtering and output at the same time is not fully supported, also see -compute_edt [0|1]\n"); } /* Useful for subtitles retiming by lavf (FIXME), skipping samples in * audio, and video decoders such as cuvid or mediacodec */ ist->dec_ctx->pkt_timebase = ist->st->time_base; if (!av_dict_get(ist->decoder_opts, "threads", NULL, 0)) av_dict_set(&ist->decoder_opts, "threads", "auto", 0); /* Attached pics are sparse, therefore we would not want to delay their decoding till EOF. */ if (ist->st->disposition & AV_DISPOSITION_ATTACHED_PIC) av_dict_set(&ist->decoder_opts, "threads", "1", 0); ret = hw_device_setup_for_decode(ist); if (ret < 0) { av_log(ist, AV_LOG_ERROR, "Hardware device setup failed for decoder: %s\n", av_err2str(ret)); return ret; } if ((ret = avcodec_open2(ist->dec_ctx, codec, &ist->decoder_opts)) < 0) { if (ret == AVERROR_EXPERIMENTAL) exit_program(1); av_log(ist, AV_LOG_ERROR, "Error while opening decoder: %s\n", av_err2str(ret)); return ret; } assert_avoptions(ist->decoder_opts); return 0; }