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authorwm4 <wm4@nowhere>2018-02-12 18:00:23 +0100
committerKevin Mitchell <kevmitch@gmail.com>2018-02-13 17:45:29 -0800
commit02f9087de9103784a236f50d4b0990069ae9cf1e (patch)
treeffedf5a29e66ab95494a67e69e761383441a44cf
parent562f563ff1835daa0d749d1e9bcbe389412ffd98 (diff)
downloadmpv-02f9087de9103784a236f50d4b0990069ae9cf1e.tar.bz2
mpv-02f9087de9103784a236f50d4b0990069ae9cf1e.tar.xz
audio: move back PTS jump detection to before filter chain
The recent changes to player/audio.c moved PTS jump detection to after audio filtering. This was mostly done for convenience, because dataflow between decoder and filters was made "automatic", and jump detection would have to be done as filter. Now move it back to after decoders, again out of convenience. The future direction is to make the dataflow between filters and AO automatic, so this is a bit in the way. Another reason is that speed changes tend to cause jumps - these are legitimate, but get annoying quickly.
-rw-r--r--filters/f_decoder_wrapper.c14
-rw-r--r--filters/f_decoder_wrapper.h3
-rw-r--r--player/audio.c23
-rw-r--r--player/core.h1
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;