mirror of https://github.com/FFmpeg/FFmpeg.git
You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
672 lines
21 KiB
672 lines
21 KiB
/* |
|
* 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 <stdint.h> |
|
#include <string.h> |
|
|
|
#include "libavutil/avassert.h" |
|
#include "libavutil/channel_layout.h" |
|
#include "libavutil/cpu.h" |
|
#include "libavutil/error.h" |
|
#include "libavutil/fifo.h" |
|
#include "libavutil/mathematics.h" |
|
#include "libavutil/mem.h" |
|
#include "libavutil/samplefmt.h" |
|
|
|
#include "objpool.h" |
|
#include "sync_queue.h" |
|
|
|
/* |
|
* 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; |
|
|
|
return 0; |
|
} |
|
|
|
// 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; |
|
} |
|
|
|
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); |
|
}
|
|
|