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fftools/ffmpeg_dec: move timestamp estimation state to Decoder 

It is purely internal to decoding.
This commit is contained in:
Anton Khirnov 2023-05-18 05:52:23 +02:00
parent 5b05e9e32a
commit cad59cccaf
3 changed files with 52 additions and 50 deletions
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10
fftools/ffmpeg.h
View File

@ -352,16 +352,6 @@ typedef struct InputStream {
AVRational framerate_guessed;
// pts/estimated duration of the last decoded frame
// * in decoder timebase for video,
// * in last_frame_tb (may change during decoding) for audio
int64_t last_frame_pts;
int64_t last_frame_duration_est;
AVRational last_frame_tb;
int last_frame_sample_rate;
int64_t filter_in_rescale_delta_last;
int64_t nb_samples; /* number of samples in the last decoded audio frame before looping */
AVDictionary *decoder_opts;

87
fftools/ffmpeg_dec.c
View File

@ -34,6 +34,15 @@
struct Decoder {
AVFrame *frame;
AVPacket *pkt;
// pts/estimated duration of the last decoded frame
// * in decoder timebase for video,
// * in last_frame_tb (may change during decoding) for audio
int64_t last_frame_pts;
int64_t last_frame_duration_est;
AVRational last_frame_tb;
int64_t last_filter_in_rescale_delta;
int last_frame_sample_rate;
};
void dec_free(Decoder **pdec)
@ -67,6 +76,9 @@ static int dec_alloc(Decoder **pdec)
if (!dec->pkt)
goto fail;
dec->last_filter_in_rescale_delta = AV_NOPTS_VALUE;
dec->last_frame_pts = AV_NOPTS_VALUE;
dec->last_frame_tb = (AVRational){ 1, 1 };
*pdec = dec;
@ -94,21 +106,22 @@ static int send_frame_to_filters(InputStream *ist, AVFrame *decoded_frame)
return ret;
}
static AVRational audio_samplerate_update(InputStream *ist, const AVFrame *frame)
static AVRational audio_samplerate_update(void *logctx, Decoder *d,
const AVFrame *frame)
{
const int prev = ist->last_frame_tb.den;
const int prev = d->last_frame_tb.den;
const int sr = frame->sample_rate;
AVRational tb_new;
int64_t gcd;
if (frame->sample_rate == ist->last_frame_sample_rate)
if (frame->sample_rate == d->last_frame_sample_rate)
goto finish;
gcd = av_gcd(prev, sr);
if (prev / gcd >= INT_MAX / sr) {
av_log(ist, AV_LOG_WARNING,
av_log(logctx, AV_LOG_WARNING,
"Audio timestamps cannot be represented exactly after "
"sample rate change: %d -> %d\n", prev, sr);
@ -123,20 +136,20 @@ static AVRational audio_samplerate_update(InputStream *ist, const AVFrame *frame
!(frame->time_base.den % tb_new.den))
tb_new = frame->time_base;
if (ist->last_frame_pts != AV_NOPTS_VALUE)
ist->last_frame_pts = av_rescale_q(ist->last_frame_pts,
ist->last_frame_tb, tb_new);
ist->last_frame_duration_est = av_rescale_q(ist->last_frame_duration_est,
ist->last_frame_tb, tb_new);
if (d->last_frame_pts != AV_NOPTS_VALUE)
d->last_frame_pts = av_rescale_q(d->last_frame_pts,
d->last_frame_tb, tb_new);
d->last_frame_duration_est = av_rescale_q(d->last_frame_duration_est,
d->last_frame_tb, tb_new);
ist->last_frame_tb = tb_new;
ist->last_frame_sample_rate = frame->sample_rate;
d->last_frame_tb = tb_new;
d->last_frame_sample_rate = frame->sample_rate;
finish:
return ist->last_frame_tb;
return d->last_frame_tb;
}
static void audio_ts_process(InputStream *ist, AVFrame *frame)
static void audio_ts_process(void *logctx, Decoder *d, AVFrame *frame)
{
AVRational tb_filter = (AVRational){1, frame->sample_rate};
AVRational tb;
@ -145,27 +158,27 @@ static void audio_ts_process(InputStream *ist, AVFrame *frame)
// on samplerate change, choose a new internal timebase for timestamp
// generation that can represent timestamps from all the samplerates
// seen so far
tb = audio_samplerate_update(ist, frame);
pts_pred = ist->last_frame_pts == AV_NOPTS_VALUE ? 0 :
ist->last_frame_pts + ist->last_frame_duration_est;
tb = audio_samplerate_update(logctx, d, frame);
pts_pred = d->last_frame_pts == AV_NOPTS_VALUE ? 0 :
d->last_frame_pts + d->last_frame_duration_est;
if (frame->pts == AV_NOPTS_VALUE) {
frame->pts = pts_pred;
frame->time_base = tb;
} else if (ist->last_frame_pts != AV_NOPTS_VALUE &&
} else if (d->last_frame_pts != AV_NOPTS_VALUE &&
frame->pts > av_rescale_q_rnd(pts_pred, tb, frame->time_base,
AV_ROUND_UP)) {
// there was a gap in timestamps, reset conversion state
ist->filter_in_rescale_delta_last = AV_NOPTS_VALUE;
d->last_filter_in_rescale_delta = AV_NOPTS_VALUE;
}
frame->pts = av_rescale_delta(frame->time_base, frame->pts,
tb, frame->nb_samples,
&ist->filter_in_rescale_delta_last, tb);
&d->last_filter_in_rescale_delta, tb);
ist->last_frame_pts = frame->pts;
ist->last_frame_duration_est = av_rescale_q(frame->nb_samples,
tb_filter, tb);
d->last_frame_pts = frame->pts;
d->last_frame_duration_est = av_rescale_q(frame->nb_samples,
tb_filter, tb);
// finally convert to filtering timebase
frame->pts = av_rescale_q(frame->pts, tb, tb_filter);
@ -175,6 +188,7 @@ static void audio_ts_process(InputStream *ist, AVFrame *frame)
static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *frame)
{
const Decoder *d = ist->decoder;
const InputFile *ifile = input_files[ist->file_index];
int64_t codec_duration = 0;
@ -202,9 +216,9 @@ static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *fr
// when timestamps are available, repeat last frame's actual duration
// (i.e. pts difference between this and last frame)
if (frame->pts != AV_NOPTS_VALUE && ist->last_frame_pts != AV_NOPTS_VALUE &&
frame->pts > ist->last_frame_pts)
return frame->pts - ist->last_frame_pts;
if (frame->pts != AV_NOPTS_VALUE && d->last_frame_pts != AV_NOPTS_VALUE &&
frame->pts > d->last_frame_pts)
return frame->pts - d->last_frame_pts;
// try frame/codec duration
if (frame->duration > 0)
@ -221,11 +235,13 @@ static int64_t video_duration_estimate(const InputStream *ist, const AVFrame *fr
}
// last resort is last frame's estimated duration, and 1
return FFMAX(ist->last_frame_duration_est, 1);
return FFMAX(d->last_frame_duration_est, 1);
}
static int video_frame_process(InputStream *ist, AVFrame *frame)
{
Decoder *d = ist->decoder;
// The following line may be required in some cases where there is no parser
// or the parser does not has_b_frames correctly
if (ist->par->video_delay < ist->dec_ctx->has_b_frames) {
@ -273,13 +289,13 @@ static int video_frame_process(InputStream *ist, AVFrame *frame)
// no timestamp available - extrapolate from previous frame duration
if (frame->pts == AV_NOPTS_VALUE)
frame->pts = ist->last_frame_pts == AV_NOPTS_VALUE ? 0 :
ist->last_frame_pts + ist->last_frame_duration_est;
frame->pts = d->last_frame_pts == AV_NOPTS_VALUE ? 0 :
d->last_frame_pts + d->last_frame_duration_est;
// update timestamp history
ist->last_frame_duration_est = video_duration_estimate(ist, frame);
ist->last_frame_pts = frame->pts;
ist->last_frame_tb = frame->time_base;
d->last_frame_duration_est = video_duration_estimate(ist, frame);
d->last_frame_pts = frame->pts;
d->last_frame_tb = frame->time_base;
if (debug_ts) {
av_log(ist, AV_LOG_INFO,
@ -404,12 +420,13 @@ static int transcode_subtitles(InputStream *ist, const AVPacket *pkt)
static int send_filter_eof(InputStream *ist)
{
Decoder *d = ist->decoder;
int i, ret;
for (i = 0; i < ist->nb_filters; i++) {
int64_t end_pts = ist->last_frame_pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
ist->last_frame_pts + ist->last_frame_duration_est;
ret = ifilter_send_eof(ist->filters[i], end_pts, ist->last_frame_tb);
int64_t end_pts = d->last_frame_pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
d->last_frame_pts + d->last_frame_duration_est;
ret = ifilter_send_eof(ist->filters[i], end_pts, d->last_frame_tb);
if (ret < 0)
return ret;
}
@ -511,7 +528,7 @@ int dec_packet(InputStream *ist, const AVPacket *pkt, int no_eof)
ist->samples_decoded += frame->nb_samples;
ist->nb_samples = frame->nb_samples;
audio_ts_process(ist, frame);
audio_ts_process(ist, ist->decoder, frame);
} else {
ret = video_frame_process(ist, frame);
if (ret < 0) {

5
fftools/ffmpeg_demux.c
View File

@ -1181,11 +1181,6 @@ static void add_input_streams(const OptionsContext *o, Demuxer *d)
exit_program(1);
}
ist->filter_in_rescale_delta_last = AV_NOPTS_VALUE;
ist->last_frame_pts = AV_NOPTS_VALUE;
ist->last_frame_tb = (AVRational){ 1, 1 };
ist->dec_ctx = avcodec_alloc_context3(ist->dec);
if (!ist->dec_ctx)
report_and_exit(AVERROR(ENOMEM));