[FFmpeg-cvslog] ffmpeg: move A/V non-streamcopy initialization to a later point
Jan Ekström
git at videolan.org
Thu Oct 29 17:32:47 EET 2020
ffmpeg | branch: master | Jan Ekström <jeebjp at gmail.com> | Fri Sep 11 00:25:21 2020 +0300| [67be1ce0c6de330b1c10d1d121819d8a74a7b1f5] | committer: Jan Ekström
ffmpeg: move A/V non-streamcopy initialization to a later point
- 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.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=67be1ce0c6de330b1c10d1d121819d8a74a7b1f5
---
fftools/ffmpeg.c | 112 ++++++++++++++++++++++++++++++++++++++-----------------
1 file changed, 77 insertions(+), 35 deletions(-)
diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c
index 498e5f08a6..52d1b09c78 100644
--- a/fftools/ffmpeg.c
+++ b/fftools/ffmpeg.c
@@ -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;
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