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fftools/sync_queue: allow requesting a specific number of audio samples

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This will be made useful in following commits.
This commit is contained in:
Anton Khirnov 2023-03-23 08:38:19 +01:00
parent 81cca3dae3
commit f9d3c06533
2 changed files with 185 additions and 9 deletions
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184
fftools/sync_queue.c
View File

@ -20,10 +20,13 @@
#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"
@ -67,6 +70,8 @@ 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;
@ -74,7 +79,9 @@ typedef struct SyncQueueStream {
int finished;
uint64_t frames_sent;
uint64_t samples_sent;
uint64_t frames_max;
int frame_samples;
} SyncQueueStream;
struct SyncQueue {
@ -98,6 +105,8 @@ struct SyncQueue {
ObjPool *pool;
int have_limiting;
uintptr_t align_mask;
};
static void frame_move(const SyncQueue *sq, SyncQueueFrame dst,
@ -109,8 +118,18 @@ static void frame_move(const SyncQueue *sq, SyncQueueFrame dst,
av_frame_move_ref(dst.f, src.f);
}
static int64_t frame_ts(const SyncQueue *sq, SyncQueueFrame frame)
/**
* 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;
@ -265,7 +284,7 @@ static int overflow_heartbeat(SyncQueue *sq, int 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_ts(sq, frame);
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 ||
@ -326,7 +345,7 @@ int sq_send(SyncQueue *sq, unsigned int stream_idx, SyncQueueFrame frame)
dst.f->time_base);
}
ts = frame_ts(sq, dst);
ts = frame_end(sq, dst, 0);
ret = av_fifo_write(st->fifo, &dst, 1);
if (ret < 0) {
@ -337,13 +356,131 @@ int sq_send(SyncQueue *sq, unsigned int stream_idx, SyncQueueFrame frame)
stream_update_ts(sq, stream_idx, ts);
st->frames_sent++;
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)
{
@ -354,13 +491,18 @@ static int receive_for_stream(SyncQueue *sq, unsigned int stream_idx,
av_assert0(stream_idx < sq->nb_streams);
st = &sq->streams[stream_idx];
if (av_fifo_can_read(st->fifo)) {
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_ts(sq, peek);
ts = frame_end(sq, peek, nb_samples);
/* check if this stream's tail timestamp does not overtake
* the overall queue head */
@ -372,9 +514,19 @@ static int receive_for_stream(SyncQueue *sq, unsigned int stream_idx,
* Frames are also passed through when there are no limiting streams.
*/
if (cmp <= 0 || ts == AV_NOPTS_VALUE || !sq->have_limiting) {
frame_move(sq, frame, peek);
objpool_release(sq->pool, (void**)&peek);
av_fifo_drain2(st->fifo, 1);
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;
}
}
@ -460,6 +612,20 @@ void sq_limit_frames(SyncQueue *sq, unsigned int stream_idx, uint64_t frames)
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));

10
fftools/sync_queue.h
View File

@ -71,6 +71,16 @@ int sq_add_stream(SyncQueue *sq, int limiting);
void sq_limit_frames(SyncQueue *sq, unsigned int stream_idx,
uint64_t max_frames);
/**
* Set a constant output audio frame size, in samples. Can only be used with
* SYNC_QUEUE_FRAMES queues and audio streams.
*
* All output frames will have exactly frame_samples audio samples, except
* possibly for the last one, which may have fewer.
*/
void sq_frame_samples(SyncQueue *sq, unsigned int stream_idx,
int frame_samples);
/**
* Submit a frame for the stream with index stream_idx.
*