Simplify, clean up and rename aout_OutputNextBuffer()

include/vlc_aout.h

@@ -190,8 +190,6 @@ static const uint32_t pi_vlc_chan_order_wg4[] =
  * Prototypes
  *****************************************************************************/
 
-VLC_API aout_buffer_t * aout_OutputNextBuffer( audio_output_t *, mtime_t, bool ) VLC_USED;
-
 /**
  * This function computes the reordering needed to go from pi_chan_order_in to
  * pi_chan_order_out.
@@ -275,6 +273,7 @@ VLC_API void aout_PacketPlay(audio_output_t *, block_t *);
 VLC_API void aout_PacketPause(audio_output_t *, bool, mtime_t);
 VLC_API void aout_PacketFlush(audio_output_t *, bool);
 
-VLC_API block_t *aout_PacketNext(audio_output_t *, mtime_t, bool) VLC_USED;
+VLC_API block_t *aout_PacketNext(audio_output_t *, mtime_t) VLC_USED;
+
 
 #endif /* VLC_AOUT_H */

modules/audio_output/alsa.c

@@ -655,8 +655,7 @@ static void ALSAFill( audio_output_t * p_aout )
         next_date = mdate() + delay_us;
     }
 
-    block_t *p_buffer = aout_OutputNextBuffer( p_aout, next_date,
-           (p_aout->format.i_format ==  VLC_CODEC_SPDIFL) );
+    block_t *p_buffer = aout_PacketNext( p_aout, next_date );
 
     /* Audio output buffer shortage -> stop the fill process and wait */
     if( p_buffer == NULL )

modules/audio_output/auhal.c

@@ -1303,7 +1303,7 @@ static OSStatus RenderCallbackAnalog( vlc_object_t *_p_aout,
     {
         /* We don't have enough data yet */
         aout_buffer_t * p_buffer;
-        p_buffer = aout_OutputNextBuffer( p_aout, current_date , false );
+        p_buffer = aout_PacketNext( p_aout, current_date );
 
         if( p_buffer != NULL )
         {
@@ -1370,7 +1370,7 @@ static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
                    AudioConvertHostTimeToNanos( inOutputTime->mHostTime ) / 1000;
                    //- ((mtime_t) 1000000 / p_aout->format.i_rate * 31 ); // 31 = Latency in Frames. retrieve somewhere
 
-    p_buffer = aout_OutputNextBuffer( p_aout, current_date, true );
+    p_buffer = aout_PacketNext( p_aout, current_date );
 
 #define BUFFER outOutputData->mBuffers[p_sys->i_stream_index]
     if( p_buffer != NULL )

modules/audio_output/directx.c

@@ -1006,9 +1006,6 @@ static void* DirectSoundThread( void *data )
     mtime_t last_time;
     int canc = vlc_savecancel ();
 
-    /* We don't want any resampling when using S/PDIF output */
-    bool b_sleek = (p_aout->format.i_format == VLC_CODEC_SPDIFL);
-
     msg_Dbg( p_aout, "DirectSoundThread ready" );
 
     /* Wait here until Play() is called */
@@ -1077,9 +1074,9 @@ static void* DirectSoundThread( void *data )
 
         for( i = 0; i < l_free_slots; i++ )
         {
-            aout_buffer_t *p_buffer = aout_OutputNextBuffer( p_aout,
+            aout_buffer_t *p_buffer = aout_PacketNext( p_aout,
                 mtime + INT64_C(1000000) * (i * i_frame_siz + l_queued) /
-                p_aout->format.i_rate, b_sleek );
+                p_aout->format.i_rate );
 
             /* If there is no audio data available and we have some buffered
              * already, then just wait for the next time */

modules/audio_output/jack.c

@@ -263,7 +263,7 @@ int Process( jack_nframes_t i_frames, void *p_arg )
     mtime_t play_date = mdate() + (mtime_t) ( dtime );
 
     /* Get the next audio data buffer */
-    aout_buffer_t *p_buffer = aout_OutputNextBuffer( p_aout, play_date, false );
+    aout_buffer_t *p_buffer = aout_PacketNext( p_aout, play_date );
 
     if( p_buffer != NULL )
     {

modules/audio_output/oss.c

@@ -604,8 +604,7 @@ static void* OSSThread( void *obj )
             mtime_t buffered = BufferDuration( p_aout );
 
             /* Next buffer will be played at mdate() + buffered */
-            p_buffer = aout_OutputNextBuffer( p_aout, mdate() + buffered,
-                                              false );
+            p_buffer = aout_PacketNext( p_aout, mdate() + buffered );
 
             if( p_buffer == NULL &&
                 buffered > ( p_aout->sys->max_buffer_duration
@@ -642,7 +641,7 @@ static void* OSSThread( void *obj )
             for( ;; )
             {
                 canc = vlc_savecancel ();
-                p_buffer = aout_OutputNextBuffer( p_aout, next_date, true );
+                p_buffer = aout_PacketNext( p_aout );
                 if ( p_buffer )
                     break;
                 vlc_restorecancel (canc);

modules/audio_output/portaudio.c

@@ -137,7 +137,7 @@ static int paCallback( const void *inputBuffer, void *outputBuffer,
 
     out_date = mdate() + (mtime_t) ( 1000000 *
         ( paDate->outputBufferDacTime - paDate->currentTime ) );
-    p_buffer = aout_OutputNextBuffer( p_aout, out_date, true );
+    p_buffer = aout_PacketNext( p_aout, out_date );
 
     if ( p_buffer != NULL )
     {
@@ -150,12 +150,6 @@ static int paCallback( const void *inputBuffer, void *outputBuffer,
         }
         vlc_memcpy( outputBuffer, p_buffer->p_buffer,
                     framesPerBuffer * p_sys->i_sample_size );
-        /* aout_BufferFree may be dangereous here, but then so is
-         * aout_OutputNextBuffer (calls aout_BufferFree internally).
-         * one solution would be to link the no longer useful buffers
-         * in a second fifo (in aout_OutputNextBuffer too) and to
-         * wait until we are in Play to do the actual free.
-         */
         aout_BufferFree( p_buffer );
     }
     else

modules/audio_output/waveout.c

@@ -898,7 +898,7 @@ static void* WaveOutThread( void *data )
     // than wait a short time... before grabbing first frames
     mwait( p_sys->start_date - AOUT_MAX_PTS_ADVANCE/4 );
 
-#define waveout_warn(msg) msg_Warn( p_aout, "aout_OutputNextBuffer no buffer "\
+#define waveout_warn(msg) msg_Warn( p_aout, "aout_PacketNext no buffer "\
                            "got next_date=%d ms, "\
                            "%d frames to play, %s",\
                            (int)(next_date/(mtime_t)1000), \
@@ -927,25 +927,19 @@ static void* WaveOutThread( void *data )
 
 
                 /* Take into account the latency */
-                p_buffer = aout_OutputNextBuffer( p_aout,
-                    next_date,
-                    b_sleek );
-
+                p_buffer = aout_PacketNext( p_aout, next_date );
                 if(!p_buffer)
                 {
 #if 0
-                    msg_Dbg( p_aout, "aout_OutputNextBuffer no buffer "
-                                      "got next_date=%d ms, "
-                                      "%d frames to play",
-                                      (int)(next_date/(mtime_t)1000),
-                                      i_queued_frames);
+                    msg_Dbg( p_aout, "aout_PacketNext no buffer got "
+                             "next_date=%"PRId64" ms, %d frames to play",
+                             next_date/1000, i_queued_frames);
 #endif
                     // means we are too early to request a new buffer?
                     waveout_warn("waiting...")
                     mwait( next_date - AOUT_MAX_PTS_ADVANCE/4 );
                     next_date = mdate();
-                    p_buffer = aout_OutputNextBuffer( p_aout, next_date,
-                                                      b_sleek );
+                    p_buffer = aout_PacketNext( p_aout, next_date );
                 }
 
                 if( !p_buffer && i_queued_frames )

src/audio_output/output.c

@@ -599,74 +599,67 @@ static block_t *aout_OutputSlice (audio_output_t *p_aout)
     return p_buffer;
 }
 
-/*****************************************************************************
- * aout_OutputNextBuffer : give the audio output plug-in the right buffer
- *****************************************************************************
- * If b_can_sleek is 1, the aout core functions won't try to resample
- * new buffers to catch up - that is we suppose that the output plug-in can
- * compensate it by itself. S/PDIF outputs should always set b_can_sleek = 1.
- * This function is entered with no lock at all :-).
- *****************************************************************************/
-aout_buffer_t * aout_OutputNextBuffer( audio_output_t * p_aout,
-                                       mtime_t start_date,
-                                       bool b_can_sleek )
+/**
+ * Dequeues the next audio packet (a.k.a. audio fragment).
+ * The audio output plugin must first call aout_PacketPlay() to queue the
+ * decoded audio samples. Typically, audio_output_t.pf_play is set to, or calls
+ * aout_PacketPlay().
+ * @note This function is considered legacy. Please do not use this function in
+ * new audio output plugins.
+ * @param p_aout audio output instance
+ * @param start_date expected PTS of the audio packet
+ */
+block_t *aout_PacketNext (audio_output_t *p_aout, mtime_t start_date)
 {
     aout_packet_t *p = aout_packet (p_aout);
     aout_fifo_t *p_fifo = &p->fifo;
     aout_buffer_t *p_buffer = NULL;
-    mtime_t now = mdate();
+    const bool b_can_sleek = AOUT_FMT_NON_LINEAR (&p_aout->format);
+    const mtime_t threshold =
+        b_can_sleek ? start_date : mdate () - AOUT_MAX_PTS_DELAY;
 
     vlc_mutex_lock( &p->lock );
     if( p->pause_date != VLC_TS_INVALID )
-        goto out;
+        goto out; /* paused: do not dequeue buffers */
 
-    /* Drop the audio sample if the audio output is really late.
-     * In the case of b_can_sleek, we don't use a resampler so we need to be
-     * a lot more severe. */
-    while( ((p_buffer = p_fifo->p_first) != NULL)
-     && p_buffer->i_pts < (b_can_sleek ? start_date : now) - AOUT_MAX_PTS_DELAY )
+    for (;;)
     {
-        msg_Dbg( p_aout, "audio output is too slow (%"PRId64"), "
-                 "trashing %"PRId64"us", now - p_buffer->i_pts,
-                 p_buffer->i_length );
-        aout_BufferFree( aout_FifoPop( p_fifo ) );
-    }
+        p_buffer = p_fifo->p_first;
+        if (p_buffer == NULL)
+            goto out; /* nothing to play */
 
-    if( p_buffer == NULL )
-    {
-#if 0 /* This is bad because the audio output might just be trying to fill
-       * in its internal buffers. And anyway, it's up to the audio output
-       * to deal with this kind of starvation. */
-
-        /* Set date to 0, to allow the mixer to send a new buffer ASAP */
-        aout_FifoReset( &p->fifo );
-        if ( !p->starving )
-            msg_Dbg( p_aout,
-                 "audio output is starving (no input), playing silence" );
-        p_aout->starving = true;
-#endif
-        goto out;
+        if (p_buffer->i_pts >= threshold)
+            break;
+
+        /* Drop the audio sample if the audio output is really late.
+         * In the case of b_can_sleek, we don't use a resampler so we need to
+         * be a lot more severe. */
+        msg_Dbg (p_aout, "audio output is too slow (%"PRId64" us): "
+                 " trashing %"PRId64" us", threshold - p_buffer->i_pts,
+                 p_buffer->i_length);
+        block_Release (aout_FifoPop (p_fifo));
     }
 
-    mtime_t delta = start_date - p_buffer->i_pts;
-    /* Here we suppose that all buffers have the same duration - this is
-     * generally true, and anyway if it's wrong it won't be a disaster.
-     */
-    if ( 0 > delta + p_buffer->i_length )
+    mtime_t delta = p_buffer->i_pts - start_date;
+    /* This assumes that all buffers have the same duration. This is true
+     * since aout_PacketPlay() (aout_OutputSlice()) is used. */
+    if (delta >= p_buffer->i_length)
     {
         if (!p->starving)
-            msg_Dbg( p_aout, "audio output is starving (%"PRId64"), "
-                     "playing silence", -delta );
+        {
+            msg_Dbg (p_aout, "audio output is starving (%"PRId64"), "
+                     "playing silence", delta);
             p->starving = true;
+        }
         p_buffer = NULL;
-        goto out;
+        goto out; /* nothing to play _yet_ */
     }
 
     p->starving = false;
     p_buffer = aout_FifoPop( p_fifo );
 
-    if( !b_can_sleek
-     && ( delta > AOUT_MAX_PTS_DELAY || delta < -AOUT_MAX_PTS_ADVANCE ) )
+    if (!b_can_sleek
+     && (delta > AOUT_MAX_PTS_ADVANCE || delta < -AOUT_MAX_PTS_DELAY))
     {
         /* Try to compensate the drift by doing some resampling. */
         msg_Warn( p_aout, "output date isn't PTS date, requesting "

src/libvlccore.sym

@@ -17,12 +17,12 @@ aout_filter_RequestVout
 aout_FormatPrepare
 aout_FormatPrint
 aout_FormatPrintChannels
-aout_OutputNextBuffer
 aout_PacketInit
 aout_PacketDestroy
 aout_PacketPlay
 aout_PacketPause
 aout_PacketFlush
+aout_PacketNext
 aout_VolumeGet
 aout_VolumeSet
 aout_VolumeUp