Do not read avg_buffering from the player thread

On buffer underflow, the average buffering must be updated, but it is
intended to be accessed only from the receiver thread.

Make the player and the receiver thread communicate the underflow via a
new field (ap->underflow).

app/src/audio_player.c

@@ -67,7 +67,7 @@ sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
 
         if (ap->received) {
             // Inserting additional samples immediately increases buffering
-            ap->avg_buffering.avg += silence;
+            ap->underflow += silence;
         }
     }
 
@@ -194,9 +194,12 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
         // Number of samples added (or removed, if negative) for compensation
         int32_t instant_compensation =
             (int32_t) samples_written - frame->nb_samples;
+        int32_t inserted_silence = (int32_t) ap->underflow;
 
         // The compensation must apply instantly, it must not be smoothed
-        ap->avg_buffering.avg += instant_compensation;
+        ap->avg_buffering.avg += instant_compensation + inserted_silence;
+
+        ap->underflow = 0; // reset
 
         // However, the buffering level must be smoothed
         sc_average_push(&ap->avg_buffering, buffered_samples);
@@ -358,6 +361,7 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
 
     ap->received = false;
     ap->played = false;
+    ap->underflow = 0;
 
     // The thread calling open() is the thread calling push(), which fills the
     // audio buffer consumed by the SDL audio thread.

app/src/audio_player.h

@@ -56,6 +56,10 @@ struct sc_audio_player {
     // (only used by the receiver thread)
     uint32_t samples_since_resync;
 
+    // Number of silence samples inserted since the last received packet
+    // (protected by SDL_AudioDeviceLock())
+    uint32_t underflow;
+
     // Set to true the first time a sample is received (protected by
     // SDL_AudioDeviceLock())
     bool received;