aout: fix assertion when S/PDIF is used

[vlc] / src / audio_output / input.c

diff --git a/src/audio_output/input.c b/src/audio_output/input.c
index 3056809bf612e5f4bd83c6f295df31cfd5bf150f..9820854bbcd79487503cb08c420dea45e44fedc3 100644 (file)
--- a/src/audio_output/input.c
+++ b/src/audio_output/input.c
@@ -43,23 +43,12 @@ static void inputResamplingStop( audio_output_t *, aout_input_t * );
 /*****************************************************************************
  * aout_InputNew : allocate a new input and rework the filter pipeline
  *****************************************************************************/
-aout_input_t *aout_InputNew (audio_output_t * p_aout,
-                             const audio_sample_format_t *restrict infmt,
-                             const audio_sample_format_t *restrict outfmt,
-                             const aout_request_vout_t *p_request_vout)
+aout_input_t *aout_InputNew (const audio_sample_format_t *restrict infmt)
 {
-    aout_input_t *p_input = malloc (sizeof (*p_input));
-    if (unlikely(p_input == NULL))
-        return NULL;
+    aout_input_t *p_input = xmalloc (sizeof (*p_input));
 
     p_input->samplerate = infmt->i_rate;
 
-    if (aout_FiltersNew (p_aout, infmt, outfmt, p_request_vout))
-    {
-        free(p_input);
-        return NULL;
-    }
-
     p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
     p_input->i_last_input_rate = INPUT_RATE_DEFAULT;
     p_input->i_buffer_lost = 0;
@@ -71,11 +60,9 @@ aout_input_t *aout_InputNew (audio_output_t * p_aout,
  *****************************************************************************
  * This function must be entered with the mixer lock.
  *****************************************************************************/
-int aout_InputDelete( audio_output_t * p_aout, aout_input_t * p_input )
+void aout_InputDelete (aout_input_t * p_input )
 {
-    aout_FiltersDestroy (p_aout);
-    (void) p_input;
-    return 0;
+    free (p_input);
 }
 
 /*****************************************************************************
@@ -191,7 +178,6 @@ block_t *aout_InputPlay(audio_output_t *p_aout, aout_input_t *p_input,
          *    synchronization
          * Solution : resample the buffer to avoid a scratch.
          */
-        p_input->i_resamp_start_date = now;
         p_input->i_resamp_start_drift = (int)-drift;
         p_input->i_resampling_type = (drift < 0) ? AOUT_RESAMPLING_DOWN
                                                  : AOUT_RESAMPLING_UP;
@@ -220,9 +206,7 @@ block_t *aout_InputPlay(audio_output_t *p_aout, aout_input_t *p_input,
         if( owner->resampler->fmt_in.audio.i_rate == i_nominal_rate )
         {
             p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
-            msg_Warn( p_aout, "resampling stopped after %"PRIi64" usec "
-                      "(drift: %"PRIi64")",
-                      now - p_input->i_resamp_start_date,
+            msg_Warn( p_aout, "resampling stopped (drift: %"PRIi64")",
                       p_buffer->i_pts - start_date);
         }
         else if( abs( (int)(p_buffer->i_pts - start_date) ) <