Convert audio output core to audio filter2

[vlc] / src / audio_output / output.c

diff --git a/src/audio_output/output.c b/src/audio_output/output.c
index 8271868079a6d1cbaff60fb606384b35b512c9c0..d8aca7f37733d9756a4517a0dbec45dc7071a1ef 100644 (file)
--- a/src/audio_output/output.c
+++ b/src/audio_output/output.c
@@ -30,6 +30,8 @@
 
 #include <vlc_common.h>
 #include <vlc_aout.h>
+#include <vlc_cpu.h>
+
 #include "aout_internal.h"
 
 /*****************************************************************************
@@ -170,28 +172,27 @@ int aout_OutputNew( aout_instance_t * p_aout,
     aout_FormatPrint( p_aout, "output", &p_aout->output.output );
 
     /* Calculate the resulting mixer output format. */
-    memcpy( &p_aout->mixer.mixer, &p_aout->output.output,
-            sizeof(audio_sample_format_t) );
+    p_aout->mixer_format = p_aout->output.output;
     if ( !AOUT_FMT_NON_LINEAR(&p_aout->output.output) )
     {
         /* Non-S/PDIF mixer only deals with float32 or fixed32. */
-        p_aout->mixer.mixer.i_format
+        p_aout->mixer_format.i_format
                      = (vlc_CPU() & CPU_CAPABILITY_FPU) ?
                         VLC_CODEC_FL32 : VLC_CODEC_FI32;
-        aout_FormatPrepare( &p_aout->mixer.mixer );
+        aout_FormatPrepare( &p_aout->mixer_format );
     }
     else
     {
-        p_aout->mixer.mixer.i_format = p_format->i_format;
+        p_aout->mixer_format.i_format = p_format->i_format;
     }
 
-    aout_FormatPrint( p_aout, "mixer", &p_aout->mixer.mixer );
+    aout_FormatPrint( p_aout, "mixer", &p_aout->mixer_format );
 
     /* Create filters. */
     p_aout->output.i_nb_filters = 0;
     if ( aout_FiltersCreatePipeline( p_aout, p_aout->output.pp_filters,
                                      &p_aout->output.i_nb_filters,
-                                     &p_aout->mixer.mixer,
+                                     &p_aout->mixer_format,
                                      &p_aout->output.output ) < 0 )
     {
         msg_Err( p_aout, "couldn't create audio output pipeline" );
@@ -200,15 +201,15 @@ int aout_OutputNew( aout_instance_t * p_aout,
     }
 
     /* Prepare hints for the buffer allocator. */
-    p_aout->mixer.output_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
-    p_aout->mixer.output_alloc.i_bytes_per_sec
-                        = p_aout->mixer.mixer.i_bytes_per_frame
-                           * p_aout->mixer.mixer.i_rate
-                           / p_aout->mixer.mixer.i_frame_length;
+    p_aout->mixer_allocation.b_alloc = true;
+    p_aout->mixer_allocation.i_bytes_per_sec
+                        = p_aout->mixer_format.i_bytes_per_frame
+                           * p_aout->mixer_format.i_rate
+                           / p_aout->mixer_format.i_frame_length;
 
     aout_FiltersHintBuffers( p_aout, p_aout->output.pp_filters,
                              p_aout->output.i_nb_filters,
-                             &p_aout->mixer.output_alloc );
+                             &p_aout->mixer_allocation );
 
     p_aout->output.b_error = 0;
     return 0;
@@ -245,11 +246,10 @@ void aout_OutputDelete( aout_instance_t * p_aout )
  *****************************************************************************/
 void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer )
 {
-    aout_FiltersPlay( p_aout, p_aout->output.pp_filters,
-                      p_aout->output.i_nb_filters,
+    aout_FiltersPlay( p_aout->output.pp_filters, p_aout->output.i_nb_filters,
                       &p_buffer );
 
-    if( p_buffer->i_nb_bytes == 0 )
+    if( p_buffer->i_buffer == 0 )
     {
         aout_BufferFree( p_buffer );
         return;
@@ -282,12 +282,12 @@ aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
     /* 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_buffer->start_date <
+    while ( p_buffer && p_buffer->i_pts <
             (b_can_sleek ? start_date : mdate()) - AOUT_PTS_TOLERANCE )
     {
         msg_Dbg( p_aout, "audio output is too slow (%"PRId64"), "
-                 "trashing %"PRId64"us", mdate() - p_buffer->start_date,
-                 p_buffer->end_date - p_buffer->start_date );
+                 "trashing %"PRId64"us", mdate() - p_buffer->i_pts,
+                 p_buffer->i_length );
         p_buffer = p_buffer->p_next;
         aout_BufferFree( p_aout->output.fifo.p_first );
         p_aout->output.fifo.p_first = p_buffer;
@@ -316,8 +316,7 @@ aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
     /* 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 ( p_buffer->start_date > start_date
-                         + (p_buffer->end_date - p_buffer->start_date) )
+    if ( p_buffer->i_pts > start_date + p_buffer->i_length )
     /*
      *                   + AOUT_PTS_TOLERANCE )
      * There is no reason to want that, it just worsen the scheduling of
@@ -325,7 +324,7 @@ aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
      * --Gibalou
      */
     {
-        const mtime_t i_delta = p_buffer->start_date - start_date;
+        const mtime_t i_delta = p_buffer->i_pts - start_date;
         aout_unlock_output_fifo( p_aout );
 
         if ( !p_aout->output.b_starving )
@@ -338,19 +337,19 @@ aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
     p_aout->output.b_starving = 0;
 
     if ( !b_can_sleek &&
-          ( (p_buffer->start_date - start_date > AOUT_PTS_TOLERANCE)
-             || (start_date - p_buffer->start_date > AOUT_PTS_TOLERANCE) ) )
+          ( (p_buffer->i_pts - start_date > AOUT_PTS_TOLERANCE)
+             || (start_date - p_buffer->i_pts > AOUT_PTS_TOLERANCE) ) )
     {
         /* Try to compensate the drift by doing some resampling. */
         int i;
-        mtime_t difference = start_date - p_buffer->start_date;
+        mtime_t difference = start_date - p_buffer->i_pts;
         msg_Warn( p_aout, "output date isn't PTS date, requesting "
                   "resampling (%"PRId64")", difference );
 
         aout_lock_input_fifos( p_aout );
         for ( i = 0; i < p_aout->i_nb_inputs; i++ )
         {
-            aout_fifo_t * p_fifo = &p_aout->pp_inputs[i]->fifo;
+            aout_fifo_t * p_fifo = &p_aout->pp_inputs[i]->mixer.fifo;
 
             aout_FifoMoveDates( p_aout, p_fifo, difference );
         }