ao_jack: fix termination on the end of file

The player didn't quit when the end of a file was reached. The reason
for this is that jack reported a constant audio delay even when all
audio was done playing. Whether that was recognized as EOF by the player
depended whether the exact value was higher or lower than the player's
threshhold for what it considers no more audio.

get_delay() should return amount of time it takes until the last sample
written to the audio buffer reaches the speaker. Therefore, we have to
track the estimated time when the last sample is done, and subtract it
from the calculated latency. Basically, the latency is the only amount
of time left in the delay, and it should go towards 0 as audio reaches
ths speakers.

I'm not sure if this is correct, but at least it solves the problem. One
suspicious thing is that we use system time to estimate the end of the
audio time. Maybe using jack_frame_time() would be more correct. But
apart from this, there doesn't seem to be a better way to handle this.

audio/out/ao_jack.c

@@ -60,10 +60,12 @@ struct priv {
     int connect;
     int autostart;
     int stdlayout;
+    int last_chunk;
     volatile int paused;
     volatile int underrun;
     volatile float callback_interval;
     volatile float callback_time;
+    volatile float last_ok_time;      // last time real audio was played
 
     int num_ports;
     struct port_ring ports[MP_NUM_CHANNELS];
@@ -128,11 +130,15 @@ process(jack_nframes_t nframes, void *arg)
             underrun = 1;
     }
 
+    float now = mp_time_us() / 1000000.0;
+
     if (underrun)
         p->underrun = 1;
 
+    if (!p->underrun)
+        p->last_ok_time = now;
+
     if (p->estimate) {
-        float now = mp_time_us() / 1000000.0;
         float diff = p->callback_time + p->callback_interval - now;
         if ((diff > -0.002) && (diff < 0.002))
             p->callback_time += p->callback_interval;
@@ -289,14 +295,23 @@ static float get_delay(struct ao *ao)
     struct priv *p = ao->priv;
     int buffered = mp_ring_buffered(p->ports[0].ring); // could be less
     float in_jack = p->jack_latency;
+    float now = mp_time_us() / 1000000.0;
 
     if (p->estimate && p->callback_interval > 0) {
         float elapsed = mp_time_us() / 1000000.0 - p->callback_time;
         in_jack += p->callback_interval - elapsed;
-        if (in_jack < 0)
-            in_jack = 0;
     }
 
+    if (p->underrun && !buffered && p->last_chunk) {
+        // Report correct buffer drainage, when our buffer is empty, but jack
+        // and/or the audio device still have some audio to play.
+        // Assumes audio clock goes at about the same speed as the system time.
+        in_jack += p->last_ok_time - now;
+    }
+
+    if (in_jack < 0)
+        in_jack = 0;
+
     return (float)buffered / (float)ao->bps + in_jack;
 }
 
@@ -369,6 +384,7 @@ static int play(struct ao *ao, void **data, int samples, int flags)
     }
 
     p->underrun = 0;
+    p->last_chunk = flags & AOPLAY_FINAL_CHUNK;
     return ret / ao->sstride;
 }