From 02f9087de9103784a236f50d4b0990069ae9cf1e Mon Sep 17 00:00:00 2001
From: wm4 <wm4@nowhere>
Date: Mon, 12 Feb 2018 18:00:23 +0100
Subject: 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.
---
 player/audio.c | 23 ++++-------------------
 player/core.h  |  1 -
 2 files changed, 4 insertions(+), 20 deletions(-)

(limited to 'player')

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;
 
-- 
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