diff options
| -rw-r--r-- | filters/f_decoder_wrapper.c | 14 | ||||
| -rw-r--r-- | filters/f_decoder_wrapper.h | 3 | ||||
| -rw-r--r-- | player/audio.c | 23 | ||||
| -rw-r--r-- | player/core.h | 1 |
4 files changed, 20 insertions, 21 deletions
diff --git a/filters/f_decoder_wrapper.c b/filters/f_decoder_wrapper.c index 0e078c0575..061f056a1e 100644 --- a/filters/f_decoder_wrapper.c +++ b/filters/f_decoder_wrapper.c @@ -105,6 +105,7 @@ static void reset_decoder(struct priv *p) p->last_format = p->fixed_format = (struct mp_image_params){0}; p->public.dropped_frames = 0; p->public.attempt_framedrops = 0; + p->public.pts_reset = false; p->packets_without_output = 0; mp_frame_unref(&p->packet); talloc_free(p->new_segment); @@ -402,9 +403,20 @@ static void process_audio_frame(struct priv *p, struct mp_aframe *aframe) if (p->pts != MP_NOPTS_VALUE) MP_STATS(p, "value %f audio-pts-err", p->pts - frame_pts); + double diff = fabs(p->pts - frame_pts); + + // Attempt to detect jumps in PTS. Even for the lowest sample rates and + // with worst container rounded timestamp, this should be a margin more + // than enough. + if (p->pts != MP_NOPTS_VALUE && diff > 0.1) { + MP_WARN(p, "Invalid audio PTS: %f -> %f\n", p->pts, frame_pts); + if (diff >= 5) + p->public.pts_reset = true; + } + // Keep the interpolated timestamp if it doesn't deviate more // than 1 ms from the real one. (MKV rounded timestamps.) - if (p->pts == MP_NOPTS_VALUE || fabs(p->pts - frame_pts) > 0.001) + if (p->pts == MP_NOPTS_VALUE || diff > 0.001) p->pts = frame_pts; } diff --git a/filters/f_decoder_wrapper.h b/filters/f_decoder_wrapper.h index e6601052a2..119e0f9eb6 100644 --- a/filters/f_decoder_wrapper.h +++ b/filters/f_decoder_wrapper.h @@ -51,6 +51,9 @@ struct mp_decoder_wrapper { // Prefer spdif wrapper over real decoders. bool try_spdif; + + // A pts reset was observed (audio only, heuristic). + bool pts_reset; }; // Create the decoder wrapper for the given stream, plus underlying decoder. diff --git a/player/audio.c b/player/audio.c index 5b061efca1..37c5446bd9 100644 --- a/player/audio.c +++ b/player/audio.c @@ -179,7 +179,6 @@ void update_playback_speed(struct MPContext *mpctx) static void ao_chain_reset_state(struct ao_chain *ao_c) { ao_c->last_out_pts = MP_NOPTS_VALUE; - ao_c->pts_reset = false; TA_FREEP(&ao_c->output_frame); ao_c->out_eof = false; @@ -658,21 +657,6 @@ static bool copy_output(struct MPContext *mpctx, struct ao_chain *ao_c, if (frame.type == MP_FRAME_AUDIO) { ao_c->output_frame = frame.data; ao_c->out_eof = false; - - double pts = mp_aframe_get_pts(ao_c->output_frame); - if (pts != MP_NOPTS_VALUE) { - // Attempt to detect jumps in PTS. Even for the lowest - // sample rates and with worst container rounded timestamp, - // this should be a margin more than enough. - double desync = pts - ao_c->last_out_pts; - if (ao_c->last_out_pts != MP_NOPTS_VALUE && fabs(desync) > 0.1) - { - MP_WARN(ao_c, "Invalid audio PTS: %f -> %f\n", - ao_c->last_out_pts, pts); - if (desync >= 5) - ao_c->pts_reset = true; - } - } ao_c->last_out_pts = mp_aframe_end_pts(ao_c->output_frame); } else if (frame.type == MP_FRAME_EOF) { ao_c->out_eof = true; @@ -786,8 +770,7 @@ void fill_audio_out_buffers(struct MPContext *mpctx) // (if AO is set due to gapless from previous file, then we can try to // filter normally until the filter tells us to change the AO) if (!mpctx->ao) { - // Probe the initial audio format. Returns AD_OK (and does nothing) if - // the format is already known. + // Probe the initial audio format. mp_pin_out_request_data(ao_c->filter->f->pins[1]); reinit_audio_filters_and_output(mpctx); return; // try again next iteration @@ -799,7 +782,9 @@ void fill_audio_out_buffers(struct MPContext *mpctx) return; } - if (mpctx->vo_chain && ao_c->pts_reset) { + if (mpctx->vo_chain && ao_c->track && ao_c->track->dec && + ao_c->track->dec->pts_reset) + { MP_VERBOSE(mpctx, "Reset playback due to audio timestamp reset.\n"); reset_playback_state(mpctx); mp_wakeup_core(mpctx); diff --git a/player/core.h b/player/core.h index 104d26cd48..27696e8f06 100644 --- a/player/core.h +++ b/player/core.h @@ -187,7 +187,6 @@ struct ao_chain { struct mp_log *log; bool spdif_passthrough, spdif_failed; - bool pts_reset; struct mp_output_chain *filter; |