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ffmpeg: move A/V non-streamcopy initialization to a later point

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- For video, this means a single initialization point in do_video_out.
- For audio we unfortunately need to do it in two places just
  before the buffer sink is utilized (if av_buffersink_get_samples
  would still work according to its specification after a call to
  avfilter_graph_request_oldest was made, we could at least remove
  the one in transcode_step).

Other adjustments to make things work:
- As the AVFrame PTS adjustment to encoder time base needs the encoder
  to be initialized, so it is now moved to do_{video,audio}_out,
  right after the encoder has been initialized. Due to this,
  the additional parameter in do_video_out is removed as it is no
  longer necessary.
This commit is contained in:
Jan Ekström 2020-09-11 00:25:21 +03:00
parent 3360c9a567
commit 67be1ce0c6
1 changed files with 77 additions and 35 deletions
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112
fftools/ffmpeg.c
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@ -947,6 +947,28 @@ early_exit:
return float_pts;
}
static int init_output_stream(OutputStream *ost, char *error, int error_len);
static int init_output_stream_wrapper(OutputStream *ost, unsigned int fatal)
{
int ret = AVERROR_BUG;
char error[1024] = {0};
if (ost->initialized)
return 0;
ret = init_output_stream(ost, error, sizeof(error));
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error initializing output stream %d:%d -- %s\n",
ost->file_index, ost->index, error);
if (fatal)
exit_program(1);
}
return ret;
}
static void do_audio_out(OutputFile *of, OutputStream *ost,
AVFrame *frame)
{
@ -958,6 +980,8 @@ static void do_audio_out(OutputFile *of, OutputStream *ost,
pkt.data = NULL;
pkt.size = 0;
adjust_frame_pts_to_encoder_tb(of, ost, frame);
if (!check_recording_time(ost))
return;
@ -1092,8 +1116,7 @@ static void do_subtitle_out(OutputFile *of,
static void do_video_out(OutputFile *of,
OutputStream *ost,
AVFrame *next_picture,
double sync_ipts)
AVFrame *next_picture)
{
int ret, format_video_sync;
AVPacket pkt;
@ -1103,10 +1126,14 @@ static void do_video_out(OutputFile *of,
int nb_frames, nb0_frames, i;
double delta, delta0;
double duration = 0;
double sync_ipts = AV_NOPTS_VALUE;
int frame_size = 0;
InputStream *ist = NULL;
AVFilterContext *filter = ost->filter->filter;
init_output_stream_wrapper(ost, 1);
sync_ipts = adjust_frame_pts_to_encoder_tb(of, ost, next_picture);
if (ost->source_index >= 0)
ist = input_streams[ost->source_index];
@ -1440,28 +1467,6 @@ static void do_video_stats(OutputStream *ost, int frame_size)
}
}
static int init_output_stream(OutputStream *ost, char *error, int error_len);
static int init_output_stream_wrapper(OutputStream *ost, unsigned int fatal)
{
int ret = AVERROR_BUG;
char error[1024] = {0};
if (ost->initialized)
return 0;
ret = init_output_stream(ost, error, sizeof(error));
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error initializing output stream %d:%d -- %s\n",
ost->file_index, ost->index, error);
if (fatal)
exit_program(1);
}
return ret;
}
static void finish_output_stream(OutputStream *ost)
{
OutputFile *of = output_files[ost->file_index];
@ -1498,7 +1503,17 @@ static int reap_filters(int flush)
continue;
filter = ost->filter->filter;
init_output_stream_wrapper(ost, 1);
/*
* Unlike video, with audio the audio frame size matters.
* Currently we are fully reliant on the lavfi filter chain to
* do the buffering deed for us, and thus the frame size parameter
* needs to be set accordingly. Where does one get the required
* frame size? From the initialized AVCodecContext of an audio
* encoder. Thus, if we have gotten to an audio stream, initialize
* the encoder earlier than receiving the first AVFrame.
*/
if (av_buffersink_get_type(filter) == AVMEDIA_TYPE_AUDIO)
init_output_stream_wrapper(ost, 1);
if (!ost->filtered_frame && !(ost->filtered_frame = av_frame_alloc())) {
return AVERROR(ENOMEM);
@ -1506,7 +1521,6 @@ static int reap_filters(int flush)
filtered_frame = ost->filtered_frame;
while (1) {
double float_pts = AV_NOPTS_VALUE; // this is identical to filtered_frame.pts but with higher precision
ret = av_buffersink_get_frame_flags(filter, filtered_frame,
AV_BUFFERSINK_FLAG_NO_REQUEST);
if (ret < 0) {
@ -1515,7 +1529,7 @@ static int reap_filters(int flush)
"Error in av_buffersink_get_frame_flags(): %s\n", av_err2str(ret));
} else if (flush && ret == AVERROR_EOF) {
if (av_buffersink_get_type(filter) == AVMEDIA_TYPE_VIDEO)
do_video_out(of, ost, NULL, AV_NOPTS_VALUE);
do_video_out(of, ost, NULL);
}
break;
}
@ -1524,15 +1538,12 @@ static int reap_filters(int flush)
continue;
}
float_pts = adjust_frame_pts_to_encoder_tb(of, ost,
filtered_frame);
switch (av_buffersink_get_type(filter)) {
case AVMEDIA_TYPE_VIDEO:
if (!ost->frame_aspect_ratio.num)
enc->sample_aspect_ratio = filtered_frame->sample_aspect_ratio;
do_video_out(of, ost, filtered_frame, float_pts);
do_video_out(of, ost, filtered_frame);
break;
case AVMEDIA_TYPE_AUDIO:
if (!(enc->codec->capabilities & AV_CODEC_CAP_PARAM_CHANGE) &&
@ -3698,10 +3709,19 @@ static int transcode_init(void)
goto dump_format;
}
/* open each encoder */
/*
* initialize stream copy and subtitle/data streams.
* Encoded AVFrame based streams will get initialized as follows:
* - when the first AVFrame is received in do_video_out
* - just before the first AVFrame is received in either transcode_step
* or reap_filters due to us requiring the filter chain buffer sink
* to be configured with the correct audio frame size, which is only
* known after the encoder is initialized.
*/
for (i = 0; i < nb_output_streams; i++) {
// skip streams fed from filtergraphs until we have a frame for them
if (output_streams[i]->filter)
if (!output_streams[i]->stream_copy &&
(output_streams[i]->enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO ||
output_streams[i]->enc_ctx->codec_type == AVMEDIA_TYPE_AUDIO))
continue;
ret = init_output_stream_wrapper(output_streams[i], 0);
@ -4615,7 +4635,29 @@ static int transcode_step(void)
}
if (ost->filter && ost->filter->graph->graph) {
init_output_stream_wrapper(ost, 1);
/*
* Similar case to the early audio initialization in reap_filters.
* Audio is special in ffmpeg.c currently as we depend on lavfi's
* audio frame buffering/creation to get the output audio frame size
* in samples correct. The audio frame size for the filter chain is
* configured during the output stream initialization.
*
* Apparently avfilter_graph_request_oldest (called in
* transcode_from_filter just down the line) peeks. Peeking already
* puts one frame "ready to be given out", which means that any
* update in filter buffer sink configuration afterwards will not
* help us. And yes, even if it would be utilized,
* av_buffersink_get_samples is affected, as it internally utilizes
* the same early exit for peeked frames.
*
* In other words, if avfilter_graph_request_oldest would not make
* further filter chain configuration or usage of
* av_buffersink_get_samples useless (by just causing the return
* of the peeked AVFrame as-is), we could get rid of this additional
* early encoder initialization.
*/
if (av_buffersink_get_type(ost->filter->filter) == AVMEDIA_TYPE_AUDIO)
init_output_stream_wrapper(ost, 1);
if ((ret = transcode_from_filter(ost->filter->graph, &ist)) < 0)
return ret;