Spaces:
Runtime error
Runtime error
| /* | |
| * 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 | |
| */ | |
| /* | |
| * How this works: | |
| * -------------- | |
| * time: 0 1 2 3 4 5 6 7 8 9 10 11 12 13 | |
| * ------------------------------------------------------------------- | |
| * | | | | | | | | | | | | | | | |
| * | ┌───┐┌────────┐┌───┐┌─────────────┐ | |
| * stream 0| │d=1││ d=2 ││d=1││ d=3 │ | |
| * | └───┘└────────┘└───┘└─────────────┘ | |
| * ┌───┐ ┌───────────────────────┐ | |
| * stream 1│d=1│ │ d=5 │ | |
| * └───┘ └───────────────────────┘ | |
| * | ┌───┐┌───┐┌───┐┌───┐ | |
| * stream 2| │d=1││d=1││d=1││d=1│ <- stream 2 is the head stream of the queue | |
| * | └───┘└───┘└───┘└───┘ | |
| * ^ ^ | |
| * [stream 2 tail] [stream 2 head] | |
| * | |
| * We have N streams (N=3 in the diagram), each stream is a FIFO. The *tail* of | |
| * each FIFO is the frame with smallest end time, the *head* is the frame with | |
| * the largest end time. Frames submitted to the queue with sq_send() are placed | |
| * after the head, frames returned to the caller with sq_receive() are taken | |
| * from the tail. | |
| * | |
| * The head stream of the whole queue (SyncQueue.head_stream) is the limiting | |
| * stream with the *smallest* head timestamp, i.e. the stream whose source lags | |
| * furthest behind all other streams. It determines which frames can be output | |
| * from the queue. | |
| * | |
| * In the diagram, the head stream is 2, because it head time is t=5, while | |
| * streams 0 and 1 end at t=8 and t=9 respectively. All frames that _end_ at | |
| * or before t=5 can be output, i.e. the first 3 frames from stream 0, first | |
| * frame from stream 1, and all 4 frames from stream 2. | |
| */ | |
| typedef struct SyncQueueStream { | |
| AVFifo *fifo; | |
| AVRational tb; | |
| /* number of audio samples in fifo */ | |
| uint64_t samples_queued; | |
| /* stream head: largest timestamp seen */ | |
| int64_t head_ts; | |
| int limiting; | |
| /* no more frames will be sent for this stream */ | |
| int finished; | |
| uint64_t frames_sent; | |
| uint64_t samples_sent; | |
| uint64_t frames_max; | |
| int frame_samples; | |
| } SyncQueueStream; | |
| struct SyncQueue { | |
| enum SyncQueueType type; | |
| /* no more frames will be sent for any stream */ | |
| int finished; | |
| /* sync head: the stream with the _smallest_ head timestamp | |
| * this stream determines which frames can be output */ | |
| int head_stream; | |
| /* the finished stream with the smallest finish timestamp or -1 */ | |
| int head_finished_stream; | |
| // maximum buffering duration in microseconds | |
| int64_t buf_size_us; | |
| SyncQueueStream *streams; | |
| unsigned int nb_streams; | |
| // pool of preallocated frames to avoid constant allocations | |
| ObjPool *pool; | |
| int have_limiting; | |
| uintptr_t align_mask; | |
| }; | |
| static void frame_move(const SyncQueue *sq, SyncQueueFrame dst, | |
| SyncQueueFrame src) | |
| { | |
| if (sq->type == SYNC_QUEUE_PACKETS) | |
| av_packet_move_ref(dst.p, src.p); | |
| else | |
| av_frame_move_ref(dst.f, src.f); | |
| } | |
| /** | |
| * Compute the end timestamp of a frame. If nb_samples is provided, consider | |
| * the frame to have this number of audio samples, otherwise use frame duration. | |
| */ | |
| static int64_t frame_end(const SyncQueue *sq, SyncQueueFrame frame, int nb_samples) | |
| { | |
| if (nb_samples) { | |
| int64_t d = av_rescale_q(nb_samples, (AVRational){ 1, frame.f->sample_rate}, | |
| frame.f->time_base); | |
| return frame.f->pts + d; | |
| } | |
| return (sq->type == SYNC_QUEUE_PACKETS) ? | |
| frame.p->pts + frame.p->duration : | |
| frame.f->pts + frame.f->duration; | |
| } | |
| static int frame_samples(const SyncQueue *sq, SyncQueueFrame frame) | |
| { | |
| return (sq->type == SYNC_QUEUE_PACKETS) ? 0 : frame.f->nb_samples; | |
| } | |
| static int frame_null(const SyncQueue *sq, SyncQueueFrame frame) | |
| { | |
| return (sq->type == SYNC_QUEUE_PACKETS) ? (frame.p == NULL) : (frame.f == NULL); | |
| } | |
| static void tb_update(const SyncQueue *sq, SyncQueueStream *st, | |
| const SyncQueueFrame frame) | |
| { | |
| AVRational tb = (sq->type == SYNC_QUEUE_PACKETS) ? | |
| frame.p->time_base : frame.f->time_base; | |
| av_assert0(tb.num > 0 && tb.den > 0); | |
| if (tb.num == st->tb.num && tb.den == st->tb.den) | |
| return; | |
| // timebase should not change after the first frame | |
| av_assert0(!av_fifo_can_read(st->fifo)); | |
| if (st->head_ts != AV_NOPTS_VALUE) | |
| st->head_ts = av_rescale_q(st->head_ts, st->tb, tb); | |
| st->tb = tb; | |
| } | |
| static void finish_stream(SyncQueue *sq, unsigned int stream_idx) | |
| { | |
| SyncQueueStream *st = &sq->streams[stream_idx]; | |
| st->finished = 1; | |
| if (st->limiting && st->head_ts != AV_NOPTS_VALUE) { | |
| /* check if this stream is the new finished head */ | |
| if (sq->head_finished_stream < 0 || | |
| av_compare_ts(st->head_ts, st->tb, | |
| sq->streams[sq->head_finished_stream].head_ts, | |
| sq->streams[sq->head_finished_stream].tb) < 0) { | |
| sq->head_finished_stream = stream_idx; | |
| } | |
| /* mark as finished all streams that should no longer receive new frames, | |
| * due to them being ahead of some finished stream */ | |
| st = &sq->streams[sq->head_finished_stream]; | |
| for (unsigned int i = 0; i < sq->nb_streams; i++) { | |
| SyncQueueStream *st1 = &sq->streams[i]; | |
| if (st != st1 && st1->head_ts != AV_NOPTS_VALUE && | |
| av_compare_ts(st->head_ts, st->tb, st1->head_ts, st1->tb) <= 0) | |
| st1->finished = 1; | |
| } | |
| } | |
| /* mark the whole queue as finished if all streams are finished */ | |
| for (unsigned int i = 0; i < sq->nb_streams; i++) { | |
| if (!sq->streams[i].finished) | |
| return; | |
| } | |
| sq->finished = 1; | |
| } | |
| static void queue_head_update(SyncQueue *sq) | |
| { | |
| if (sq->head_stream < 0) { | |
| /* wait for one timestamp in each stream before determining | |
| * the queue head */ | |
| for (unsigned int i = 0; i < sq->nb_streams; i++) { | |
| SyncQueueStream *st = &sq->streams[i]; | |
| if (st->limiting && st->head_ts == AV_NOPTS_VALUE) | |
| return; | |
| } | |
| // placeholder value, correct one will be found below | |
| sq->head_stream = 0; | |
| } | |
| for (unsigned int i = 0; i < sq->nb_streams; i++) { | |
| SyncQueueStream *st_head = &sq->streams[sq->head_stream]; | |
| SyncQueueStream *st_other = &sq->streams[i]; | |
| if (st_other->limiting && st_other->head_ts != AV_NOPTS_VALUE && | |
| av_compare_ts(st_other->head_ts, st_other->tb, | |
| st_head->head_ts, st_head->tb) < 0) | |
| sq->head_stream = i; | |
| } | |
| } | |
| /* update this stream's head timestamp */ | |
| static void stream_update_ts(SyncQueue *sq, unsigned int stream_idx, int64_t ts) | |
| { | |
| SyncQueueStream *st = &sq->streams[stream_idx]; | |
| if (ts == AV_NOPTS_VALUE || | |
| (st->head_ts != AV_NOPTS_VALUE && st->head_ts >= ts)) | |
| return; | |
| st->head_ts = ts; | |
| /* if this stream is now ahead of some finished stream, then | |
| * this stream is also finished */ | |
| if (sq->head_finished_stream >= 0 && | |
| av_compare_ts(sq->streams[sq->head_finished_stream].head_ts, | |
| sq->streams[sq->head_finished_stream].tb, | |
| ts, st->tb) <= 0) | |
| finish_stream(sq, stream_idx); | |
| /* update the overall head timestamp if it could have changed */ | |
| if (st->limiting && | |
| (sq->head_stream < 0 || sq->head_stream == stream_idx)) | |
| queue_head_update(sq); | |
| } | |
| /* If the queue for the given stream (or all streams when stream_idx=-1) | |
| * is overflowing, trigger a fake heartbeat on lagging streams. | |
| * | |
| * @return 1 if heartbeat triggered, 0 otherwise | |
| */ | |
| static int overflow_heartbeat(SyncQueue *sq, int stream_idx) | |
| { | |
| SyncQueueStream *st; | |
| SyncQueueFrame frame; | |
| int64_t tail_ts = AV_NOPTS_VALUE; | |
| /* if no stream specified, pick the one that is most ahead */ | |
| if (stream_idx < 0) { | |
| int64_t ts = AV_NOPTS_VALUE; | |
| for (int i = 0; i < sq->nb_streams; i++) { | |
| st = &sq->streams[i]; | |
| if (st->head_ts != AV_NOPTS_VALUE && | |
| (ts == AV_NOPTS_VALUE || | |
| av_compare_ts(ts, sq->streams[stream_idx].tb, | |
| st->head_ts, st->tb) < 0)) { | |
| ts = st->head_ts; | |
| stream_idx = i; | |
| } | |
| } | |
| /* no stream has a timestamp yet -> nothing to do */ | |
| if (stream_idx < 0) | |
| return 0; | |
| } | |
| st = &sq->streams[stream_idx]; | |
| /* get the chosen stream's tail timestamp */ | |
| for (size_t i = 0; tail_ts == AV_NOPTS_VALUE && | |
| av_fifo_peek(st->fifo, &frame, 1, i) >= 0; i++) | |
| tail_ts = frame_end(sq, frame, 0); | |
| /* overflow triggers when the tail is over specified duration behind the head */ | |
| if (tail_ts == AV_NOPTS_VALUE || tail_ts >= st->head_ts || | |
| av_rescale_q(st->head_ts - tail_ts, st->tb, AV_TIME_BASE_Q) < sq->buf_size_us) | |
| return 0; | |
| /* signal a fake timestamp for all streams that prevent tail_ts from being output */ | |
| tail_ts++; | |
| for (unsigned int i = 0; i < sq->nb_streams; i++) { | |
| SyncQueueStream *st1 = &sq->streams[i]; | |
| int64_t ts; | |
| if (st == st1 || st1->finished || | |
| (st1->head_ts != AV_NOPTS_VALUE && | |
| av_compare_ts(tail_ts, st->tb, st1->head_ts, st1->tb) <= 0)) | |
| continue; | |
| ts = av_rescale_q(tail_ts, st->tb, st1->tb); | |
| if (st1->head_ts != AV_NOPTS_VALUE) | |
| ts = FFMAX(st1->head_ts + 1, ts); | |
| stream_update_ts(sq, i, ts); | |
| } | |
| return 1; | |
| } | |
| int sq_send(SyncQueue *sq, unsigned int stream_idx, SyncQueueFrame frame) | |
| { | |
| SyncQueueStream *st; | |
| SyncQueueFrame dst; | |
| int64_t ts; | |
| int ret, nb_samples; | |
| av_assert0(stream_idx < sq->nb_streams); | |
| st = &sq->streams[stream_idx]; | |
| if (frame_null(sq, frame)) { | |
| finish_stream(sq, stream_idx); | |
| return 0; | |
| } | |
| if (st->finished) | |
| return AVERROR_EOF; | |
| tb_update(sq, st, frame); | |
| ret = objpool_get(sq->pool, (void**)&dst); | |
| if (ret < 0) | |
| return ret; | |
| frame_move(sq, dst, frame); | |
| nb_samples = frame_samples(sq, dst); | |
| // make sure frame duration is consistent with sample count | |
| if (nb_samples) { | |
| av_assert0(dst.f->sample_rate > 0); | |
| dst.f->duration = av_rescale_q(nb_samples, (AVRational){ 1, dst.f->sample_rate }, | |
| dst.f->time_base); | |
| } | |
| ts = frame_end(sq, dst, 0); | |
| ret = av_fifo_write(st->fifo, &dst, 1); | |
| if (ret < 0) { | |
| frame_move(sq, frame, dst); | |
| objpool_release(sq->pool, (void**)&dst); | |
| return ret; | |
| } | |
| stream_update_ts(sq, stream_idx, ts); | |
| st->samples_queued += nb_samples; | |
| st->samples_sent += nb_samples; | |
| if (st->frame_samples) | |
| st->frames_sent = st->samples_sent / st->frame_samples; | |
| else | |
| st->frames_sent++; | |
| if (st->frames_sent >= st->frames_max) | |
| finish_stream(sq, stream_idx); | |
| return 0; | |
| } | |
| static void offset_audio(AVFrame *f, int nb_samples) | |
| { | |
| const int planar = av_sample_fmt_is_planar(f->format); | |
| const int planes = planar ? f->ch_layout.nb_channels : 1; | |
| const int bps = av_get_bytes_per_sample(f->format); | |
| const int offset = nb_samples * bps * (planar ? 1 : f->ch_layout.nb_channels); | |
| av_assert0(bps > 0); | |
| av_assert0(nb_samples < f->nb_samples); | |
| for (int i = 0; i < planes; i++) { | |
| f->extended_data[i] += offset; | |
| if (i < FF_ARRAY_ELEMS(f->data)) | |
| f->data[i] = f->extended_data[i]; | |
| } | |
| f->linesize[0] -= offset; | |
| f->nb_samples -= nb_samples; | |
| f->duration = av_rescale_q(f->nb_samples, (AVRational){ 1, f->sample_rate }, | |
| f->time_base); | |
| f->pts += av_rescale_q(nb_samples, (AVRational){ 1, f->sample_rate }, | |
| f->time_base); | |
| } | |
| static int frame_is_aligned(const SyncQueue *sq, const AVFrame *frame) | |
| { | |
| // only checks linesize[0], so only works for audio | |
| av_assert0(frame->nb_samples > 0); | |
| av_assert0(sq->align_mask); | |
| // only check data[0], because we always offset all data pointers | |
| // by the same offset, so if one is aligned, all are | |
| if (!((uintptr_t)frame->data[0] & sq->align_mask) && | |
| !(frame->linesize[0] & sq->align_mask) && | |
| frame->linesize[0] > sq->align_mask) | |
| return 1; | |
| return 0; | |
| } | |
| static int receive_samples(SyncQueue *sq, SyncQueueStream *st, | |
| AVFrame *dst, int nb_samples) | |
| { | |
| SyncQueueFrame src; | |
| int ret; | |
| av_assert0(st->samples_queued >= nb_samples); | |
| ret = av_fifo_peek(st->fifo, &src, 1, 0); | |
| av_assert0(ret >= 0); | |
| // peeked frame has enough samples and its data is aligned | |
| // -> we can just make a reference and limit its sample count | |
| if (src.f->nb_samples > nb_samples && frame_is_aligned(sq, src.f)) { | |
| ret = av_frame_ref(dst, src.f); | |
| if (ret < 0) | |
| return ret; | |
| dst->nb_samples = nb_samples; | |
| offset_audio(src.f, nb_samples); | |
| st->samples_queued -= nb_samples; | |
| goto finish; | |
| } | |
| // otherwise allocate a new frame and copy the data | |
| ret = av_channel_layout_copy(&dst->ch_layout, &src.f->ch_layout); | |
| if (ret < 0) | |
| return ret; | |
| dst->format = src.f->format; | |
| dst->nb_samples = nb_samples; | |
| ret = av_frame_get_buffer(dst, 0); | |
| if (ret < 0) | |
| goto fail; | |
| ret = av_frame_copy_props(dst, src.f); | |
| if (ret < 0) | |
| goto fail; | |
| dst->nb_samples = 0; | |
| while (dst->nb_samples < nb_samples) { | |
| int to_copy; | |
| ret = av_fifo_peek(st->fifo, &src, 1, 0); | |
| av_assert0(ret >= 0); | |
| to_copy = FFMIN(nb_samples - dst->nb_samples, src.f->nb_samples); | |
| av_samples_copy(dst->extended_data, src.f->extended_data, dst->nb_samples, | |
| 0, to_copy, dst->ch_layout.nb_channels, dst->format); | |
| if (to_copy < src.f->nb_samples) | |
| offset_audio(src.f, to_copy); | |
| else { | |
| av_frame_unref(src.f); | |
| objpool_release(sq->pool, (void**)&src); | |
| av_fifo_drain2(st->fifo, 1); | |
| } | |
| st->samples_queued -= to_copy; | |
| dst->nb_samples += to_copy; | |
| } | |
| finish: | |
| dst->duration = av_rescale_q(nb_samples, (AVRational){ 1, dst->sample_rate }, | |
| dst->time_base); | |
| return 0; | |
| fail: | |
| av_frame_unref(dst); | |
| return ret; | |
| } | |
| static int receive_for_stream(SyncQueue *sq, unsigned int stream_idx, | |
| SyncQueueFrame frame) | |
| { | |
| SyncQueueStream *st_head = sq->head_stream >= 0 ? | |
| &sq->streams[sq->head_stream] : NULL; | |
| SyncQueueStream *st; | |
| av_assert0(stream_idx < sq->nb_streams); | |
| st = &sq->streams[stream_idx]; | |
| if (av_fifo_can_read(st->fifo) && | |
| (st->frame_samples <= st->samples_queued || st->finished)) { | |
| int nb_samples = st->frame_samples; | |
| SyncQueueFrame peek; | |
| int64_t ts; | |
| int cmp = 1; | |
| if (st->finished) | |
| nb_samples = FFMIN(nb_samples, st->samples_queued); | |
| av_fifo_peek(st->fifo, &peek, 1, 0); | |
| ts = frame_end(sq, peek, nb_samples); | |
| /* check if this stream's tail timestamp does not overtake | |
| * the overall queue head */ | |
| if (ts != AV_NOPTS_VALUE && st_head) | |
| cmp = av_compare_ts(ts, st->tb, st_head->head_ts, st_head->tb); | |
| /* We can release frames that do not end after the queue head. | |
| * Frames with no timestamps are just passed through with no conditions. | |
| * Frames are also passed through when there are no limiting streams. | |
| */ | |
| if (cmp <= 0 || ts == AV_NOPTS_VALUE || !sq->have_limiting) { | |
| if (nb_samples && | |
| (nb_samples != peek.f->nb_samples || !frame_is_aligned(sq, peek.f))) { | |
| int ret = receive_samples(sq, st, frame.f, nb_samples); | |
| if (ret < 0) | |
| return ret; | |
| } else { | |
| frame_move(sq, frame, peek); | |
| objpool_release(sq->pool, (void**)&peek); | |
| av_fifo_drain2(st->fifo, 1); | |
| av_assert0(st->samples_queued >= frame_samples(sq, frame)); | |
| st->samples_queued -= frame_samples(sq, frame); | |
| } | |
| return 0; | |
| } | |
| } | |
| return (sq->finished || (st->finished && !av_fifo_can_read(st->fifo))) ? | |
| AVERROR_EOF : AVERROR(EAGAIN); | |
| } | |
| static int receive_internal(SyncQueue *sq, int stream_idx, SyncQueueFrame frame) | |
| { | |
| int nb_eof = 0; | |
| int ret; | |
| /* read a frame for a specific stream */ | |
| if (stream_idx >= 0) { | |
| ret = receive_for_stream(sq, stream_idx, frame); | |
| return (ret < 0) ? ret : stream_idx; | |
| } | |
| /* read a frame for any stream with available output */ | |
| for (unsigned int i = 0; i < sq->nb_streams; i++) { | |
| ret = receive_for_stream(sq, i, frame); | |
| if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN)) { | |
| nb_eof += (ret == AVERROR_EOF); | |
| continue; | |
| } | |
| return (ret < 0) ? ret : i; | |
| } | |
| return (nb_eof == sq->nb_streams) ? AVERROR_EOF : AVERROR(EAGAIN); | |
| } | |
| int sq_receive(SyncQueue *sq, int stream_idx, SyncQueueFrame frame) | |
| { | |
| int ret = receive_internal(sq, stream_idx, frame); | |
| /* try again if the queue overflowed and triggered a fake heartbeat | |
| * for lagging streams */ | |
| if (ret == AVERROR(EAGAIN) && overflow_heartbeat(sq, stream_idx)) | |
| ret = receive_internal(sq, stream_idx, frame); | |
| return ret; | |
| } | |
| int sq_add_stream(SyncQueue *sq, int limiting) | |
| { | |
| SyncQueueStream *tmp, *st; | |
| tmp = av_realloc_array(sq->streams, sq->nb_streams + 1, sizeof(*sq->streams)); | |
| if (!tmp) | |
| return AVERROR(ENOMEM); | |
| sq->streams = tmp; | |
| st = &sq->streams[sq->nb_streams]; | |
| memset(st, 0, sizeof(*st)); | |
| st->fifo = av_fifo_alloc2(1, sizeof(SyncQueueFrame), AV_FIFO_FLAG_AUTO_GROW); | |
| if (!st->fifo) | |
| return AVERROR(ENOMEM); | |
| /* we set a valid default, so that a pathological stream that never | |
| * receives even a real timebase (and no frames) won't stall all other | |
| * streams forever; cf. overflow_heartbeat() */ | |
| st->tb = (AVRational){ 1, 1 }; | |
| st->head_ts = AV_NOPTS_VALUE; | |
| st->frames_max = UINT64_MAX; | |
| st->limiting = limiting; | |
| sq->have_limiting |= limiting; | |
| return sq->nb_streams++; | |
| } | |
| void sq_limit_frames(SyncQueue *sq, unsigned int stream_idx, uint64_t frames) | |
| { | |
| SyncQueueStream *st; | |
| av_assert0(stream_idx < sq->nb_streams); | |
| st = &sq->streams[stream_idx]; | |
| st->frames_max = frames; | |
| if (st->frames_sent >= st->frames_max) | |
| finish_stream(sq, stream_idx); | |
| } | |
| void sq_frame_samples(SyncQueue *sq, unsigned int stream_idx, | |
| int frame_samples) | |
| { | |
| SyncQueueStream *st; | |
| av_assert0(sq->type == SYNC_QUEUE_FRAMES); | |
| av_assert0(stream_idx < sq->nb_streams); | |
| st = &sq->streams[stream_idx]; | |
| st->frame_samples = frame_samples; | |
| sq->align_mask = av_cpu_max_align() - 1; | |
| } | |
| SyncQueue *sq_alloc(enum SyncQueueType type, int64_t buf_size_us) | |
| { | |
| SyncQueue *sq = av_mallocz(sizeof(*sq)); | |
| if (!sq) | |
| return NULL; | |
| sq->type = type; | |
| sq->buf_size_us = buf_size_us; | |
| sq->head_stream = -1; | |
| sq->head_finished_stream = -1; | |
| sq->pool = (type == SYNC_QUEUE_PACKETS) ? objpool_alloc_packets() : | |
| objpool_alloc_frames(); | |
| if (!sq->pool) { | |
| av_freep(&sq); | |
| return NULL; | |
| } | |
| return sq; | |
| } | |
| void sq_free(SyncQueue **psq) | |
| { | |
| SyncQueue *sq = *psq; | |
| if (!sq) | |
| return; | |
| for (unsigned int i = 0; i < sq->nb_streams; i++) { | |
| SyncQueueFrame frame; | |
| while (av_fifo_read(sq->streams[i].fifo, &frame, 1) >= 0) | |
| objpool_release(sq->pool, (void**)&frame); | |
| av_fifo_freep2(&sq->streams[i].fifo); | |
| } | |
| av_freep(&sq->streams); | |
| objpool_free(&sq->pool); | |
| av_freep(psq); | |
| } | |