opensles: take in account all buffered audio in latency

We only considered the audio already buffered through opensles engine,
so the calculation was incorrect.

Use samples rather than bytes for calculation, and a function rather than
hardcoding 2*2 all over the place when a size in bytes is needed.

modules/audio_output/opensles_android.c

@@ -103,16 +103,18 @@ struct aout_sys_t
 
     /* audio buffered through opensles */
     uint8_t                        *buf;
-    size_t                          buf_unit_size;
+    size_t                          samples_per_buf;
     int                             next_buf;
 
+    int                             rate;
+
     /* if we can measure latency already */
     bool                            started;
 
     /* audio not yet buffered through opensles */
     block_t                        *p_buffer_chain;
     block_t                       **pp_buffer_last;
-
+    size_t                          samples;
 };
 
 /*****************************************************************************
@@ -140,6 +142,11 @@ vlc_module_end ()
  *
  *****************************************************************************/
 
+static inline int bytesPerSample(void)
+{
+    return 2 /* S16 */ * 2 /* stereo */;
+}
+
 static int TimeGet(audio_output_t* aout, mtime_t* restrict drift)
 {
     aout_sys_t *sys = aout->sys;
@@ -158,8 +165,12 @@ static int TimeGet(audio_output_t* aout, mtime_t* restrict drift)
     if (!started)
         return -1;
 
-    *drift = CLOCK_FREQ * OPENSLES_BUFLEN * st.count / 1000;
-    //msg_Dbg(aout, "latency %"PRId64"", *drift);
+    *drift = (CLOCK_FREQ * OPENSLES_BUFLEN * st.count / 1000)
+        + sys->samples * CLOCK_FREQ / sys->rate;
+
+    /* msg_Dbg(aout, "latency %"PRId64" ms, %d/%d buffers", *drift / 1000,
+        (int)st.count, OPENSLES_BUFFERS); */
+
     return 0;
 }
 
@@ -221,7 +232,7 @@ static void Pause(audio_output_t *aout, bool pause, mtime_t date)
 static int WriteBuffer(audio_output_t *aout)
 {
     aout_sys_t *sys = aout->sys;
-    const size_t unit_size = sys->buf_unit_size;
+    const size_t unit_size = sys->samples_per_buf * bytesPerSample();
 
     block_t *b = sys->p_buffer_chain;
     if (!b)
@@ -282,6 +293,8 @@ static int WriteBuffer(audio_output_t *aout)
     SLresult r = Enqueue(sys->playerBufferQueue,
         &sys->buf[unit_size * sys->next_buf], unit_size);
 
+    sys->samples -= sys->samples_per_buf;
+
     if (r == SL_RESULT_SUCCESS) {
         if (++sys->next_buf == OPENSLES_BUFFERS)
             sys->next_buf = 0;
@@ -305,6 +318,8 @@ static void Play(audio_output_t *aout, block_t *p_buffer)
     p_buffer->p_next = NULL; /* Make sur our linked list doesn't use old references */
     vlc_mutex_lock(&sys->lock);
 
+    sys->samples += p_buffer->i_buffer / bytesPerSample();
+
     /* Hold this block until we can write it into the OpenSL buffer */
     block_ChainLastAppend(&sys->pp_buffer_last, p_buffer);
 
@@ -400,9 +415,9 @@ static int Start(audio_output_t *aout, audio_sample_format_t *restrict fmt)
     CHECK_OPENSL_ERROR("Failed to switch to playing state");
 
     /* XXX: rounding shouldn't affect us at normal sampling rate */
-    sys->buf_unit_size = OPENSLES_BUFLEN * fmt->i_rate * 2 /* channels */ * 2 /* bps */ / 1000;
-    sys->buf_unit_size = (sys->buf_unit_size + 3) & ~3; // align on sample boundary
-    sys->buf = malloc(OPENSLES_BUFFERS * sys->buf_unit_size);
+    sys->rate = fmt->i_rate;
+    sys->samples_per_buf = OPENSLES_BUFLEN * fmt->i_rate / 1000;
+    sys->buf = malloc(OPENSLES_BUFFERS * sys->samples_per_buf * bytesPerSample());
     if (!sys->buf)
         goto error;