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fftools/ffmpeg: use sync queues for enforcing audio frame size

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The code currently uses lavfi for this, which creates a sort of
configuration dependency loop - the encoder should be ideally
initialized with information from the first audio frame, but to get this
frame one needs to first open the encoder to know the frame size. This
necessitates an awkward workaround, which causes audio handling to be
different from video.

With this change, audio encoder initialization is congruent with video.
This commit is contained in:
Anton Khirnov 2023-03-24 08:59:05 +01:00
parent f9d3c06533
commit 923c6ab170
3 changed files with 25 additions and 60 deletions
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58
fftools/ffmpeg.c
View File

@ -1028,6 +1028,8 @@ static void do_audio_out(OutputFile *of, OutputStream *ost,
AVCodecContext *enc = ost->enc_ctx;
int ret;
init_output_stream_wrapper(ost, frame, 1);
if (frame->pts == AV_NOPTS_VALUE)
frame->pts = ost->next_pts;
else {
@ -1378,18 +1380,6 @@ static int reap_filters(int flush)
continue;
filter = ost->filter->filter;
/*
* 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, NULL, 1);
filtered_frame = ost->filtered_frame;
while (1) {
@ -1432,6 +1422,7 @@ static int reap_filters(int flush)
break;
case AVMEDIA_TYPE_AUDIO:
if (!(enc->codec->capabilities & AV_CODEC_CAP_PARAM_CHANGE) &&
avcodec_is_open(enc) &&
enc->ch_layout.nb_channels != filtered_frame->ch_layout.nb_channels) {
av_log(NULL, AV_LOG_ERROR,
"Audio filter graph output is not normalized and encoder does not support parameter changes\n");
@ -3238,10 +3229,13 @@ static int init_output_stream(OutputStream *ost, AVFrame *frame,
ost->file_index, ost->index);
return ret;
}
if (codec->type == AVMEDIA_TYPE_AUDIO &&
!(codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE))
av_buffersink_set_frame_size(ost->filter->filter,
ost->enc_ctx->frame_size);
if (ost->enc_ctx->frame_size) {
av_assert0(ost->sq_idx_encode >= 0);
sq_frame_samples(output_files[ost->file_index]->sq_encode,
ost->sq_idx_encode, ost->enc_ctx->frame_size);
}
assert_avoptions(ost->encoder_opts);
if (ost->enc_ctx->bit_rate && ost->enc_ctx->bit_rate < 1000 &&
ost->enc_ctx->codec_id != AV_CODEC_ID_CODEC2 /* don't complain about 700 bit/s modes */)
@ -3331,12 +3325,8 @@ static int transcode_init(void)
/*
* 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.
* Encoded AVFrame based streams will get initialized when the first AVFrame
* is received in do_video_out
*/
for (OutputStream *ost = ost_iter(NULL); ost; ost = ost_iter(ost)) {
if (ost->enc_ctx &&
@ -3943,30 +3933,6 @@ static int transcode_step(void)
}
if (ost->filter && ost->filter->graph->graph) {
/*
* 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, NULL, 1);
if ((ret = transcode_from_filter(ost->filter->graph, &ist)) < 0)
return ret;
if (!ist)

8
fftools/ffmpeg_filter.c
View File

@ -1242,14 +1242,6 @@ int configure_filtergraph(FilterGraph *fg)
fg->reconfiguration = 1;
for (i = 0; i < fg->nb_outputs; i++) {
OutputStream *ost = fg->outputs[i]->ost;
if (ost->enc_ctx->codec_type == AVMEDIA_TYPE_AUDIO &&
!(ost->enc_ctx->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE))
av_buffersink_set_frame_size(ost->filter->filter,
ost->enc_ctx->frame_size);
}
for (i = 0; i < fg->nb_inputs; i++) {
AVFrame *tmp;
while (av_fifo_read(fg->inputs[i]->frame_queue, &tmp, 1) >= 0) {

19
fftools/ffmpeg_mux_init.c
View File

@ -1451,7 +1451,7 @@ static void create_streams(Muxer *mux, const OptionsContext *o)
static int setup_sync_queues(Muxer *mux, AVFormatContext *oc, int64_t buf_size_us)
{
OutputFile *of = &mux->of;
int nb_av_enc = 0, nb_interleaved = 0;
int nb_av_enc = 0, nb_audio_fs = 0, nb_interleaved = 0;
int limit_frames = 0, limit_frames_av_enc = 0;
#define IS_AV_ENC(ost, type) \
@ -1468,19 +1468,26 @@ static int setup_sync_queues(Muxer *mux, AVFormatContext *oc, int64_t buf_size_u
nb_interleaved += IS_INTERLEAVED(type);
nb_av_enc += IS_AV_ENC(ost, type);
nb_audio_fs += (ost->enc_ctx && type == AVMEDIA_TYPE_AUDIO &&
!(ost->enc_ctx->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE));
limit_frames |= ms->max_frames < INT64_MAX;
limit_frames_av_enc |= (ms->max_frames < INT64_MAX) && IS_AV_ENC(ost, type);
}
if (!((nb_interleaved > 1 && of->shortest) ||
(nb_interleaved > 0 && limit_frames)))
(nb_interleaved > 0 && limit_frames) ||
nb_audio_fs))
return 0;
/* if we have more than one encoded audio/video streams, or at least
* one encoded audio/video stream is frame-limited, then we
* synchronize them before encoding */
if ((of->shortest && nb_av_enc > 1) || limit_frames_av_enc) {
/* we use a sync queue before encoding when:
* - 'shortest' is in effect and we have two or more encoded audio/video
* streams
* - at least one encoded audio/video stream is frame-limited, since
* that has similar semantics to 'shortest'
* - at least one audio encoder requires constant frame sizes
*/
if ((of->shortest && nb_av_enc > 1) || limit_frames_av_enc || nb_audio_fs) {
of->sq_encode = sq_alloc(SYNC_QUEUE_FRAMES, buf_size_us);
if (!of->sq_encode)
return AVERROR(ENOMEM);