[vlc-commits] auhal: Fix SPDIF support

[Ron Frederick]

vlc | branch: master | Ron Frederick <ronf at timeheart.net> | Sat Jul 23 19:34:48 2011 -0700| [8af2eab6adb552fdd45652b7a0839571981ddef6] | committer: Felix Paul Kühne

auhal: Fix SPDIF support

The recent changes to auhal had it using an AudioStreamBasicDescription
array when looking at the stream properties. However, the proper type is
AudioStreamRangedDescription. This patch corrects that, and then accesses
the AudioStreamBasicDescription inside it as necessary.

Signed-off-by: Felix Paul Kühne <fkuehne at videolan.org>

> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=8af2eab6adb552fdd45652b7a0839571981ddef6
---

 modules/audio_output/auhal.c |   26 +++++++++++++-------------
 1 files changed, 13 insertions(+), 13 deletions(-)

diff --git a/modules/audio_output/auhal.c b/modules/audio_output/auhal.c
index de3944a..9383148 100644
--- a/modules/audio_output/auhal.c
+++ b/modules/audio_output/auhal.c
@@ -677,7 +677,7 @@ static int OpenSPDIF( audio_output_t * p_aout )
     for( int i = 0; i < i_streams && p_sys->i_stream_index < 0 ; i++ )
     {
         /* Find a stream with a cac3 stream */
-        AudioStreamBasicDescription *p_format_list = NULL;
+        AudioStreamRangedDescription *p_format_list = NULL;
         int                         i_formats = 0;
         bool                  b_digital = false;
 
@@ -689,8 +689,8 @@ static int OpenSPDIF( audio_output_t * p_aout )
             continue;
         }
 
-        i_formats = i_param_size / sizeof( AudioStreamBasicDescription );
-        p_format_list = (AudioStreamBasicDescription *)malloc( i_param_size );
+        i_formats = i_param_size / sizeof( AudioStreamRangedDescription );
+        p_format_list = (AudioStreamRangedDescription *)malloc( i_param_size );
         if( p_format_list == NULL )
             continue;
 
@@ -705,8 +705,8 @@ static int OpenSPDIF( audio_output_t * p_aout )
         /* Check if one of the supported formats is a digital format */
         for( int j = 0; j < i_formats; j++ )
         {
-            if( p_format_list[j].mFormatID == 'IAC3' ||
-                  p_format_list[j].mFormatID == kAudioFormat60958AC3 )
+            if( p_format_list[j].mFormat.mFormatID == 'IAC3' ||
+                  p_format_list[j].mFormat.mFormatID == kAudioFormat60958AC3 )
             {
                 b_digital = true;
                 break;
@@ -738,21 +738,21 @@ static int OpenSPDIF( audio_output_t * p_aout )
 
             for( int j = 0; j < i_formats; j++ )
             {
-                if( p_format_list[j].mFormatID == 'IAC3' ||
-                      p_format_list[j].mFormatID == kAudioFormat60958AC3 )
+                if( p_format_list[j].mFormat.mFormatID == 'IAC3' ||
+                      p_format_list[j].mFormat.mFormatID == kAudioFormat60958AC3 )
                 {
-                    if( p_format_list[j].mSampleRate == p_aout->format.i_rate )
+                    if( p_format_list[j].mFormat.mSampleRate == p_aout->format.i_rate )
                     {
                         i_requested_rate_format = j;
                         break;
                     }
-                    else if( p_format_list[j].mSampleRate == p_sys->sfmt_revert.mSampleRate )
+                    else if( p_format_list[j].mFormat.mSampleRate == p_sys->sfmt_revert.mSampleRate )
                     {
                         i_current_rate_format = j;
                     }
                     else
                     {
-                        if( i_backup_rate_format < 0 || p_format_list[j].mSampleRate > p_format_list[i_backup_rate_format].mSampleRate )
+                        if( i_backup_rate_format < 0 || p_format_list[j].mFormat.mSampleRate > p_format_list[i_backup_rate_format].mFormat.mSampleRate )
                             i_backup_rate_format = j;
                     }
                 }
@@ -760,10 +760,10 @@ static int OpenSPDIF( audio_output_t * p_aout )
             }
 
             if( i_requested_rate_format >= 0 ) /* We prefer to output at the samplerate of the original audio */
-                p_sys->stream_format = p_format_list[i_requested_rate_format];
+                p_sys->stream_format = p_format_list[i_requested_rate_format].mFormat;
             else if( i_current_rate_format >= 0 ) /* If not possible, we will try to use the current samplerate of the device */
-                p_sys->stream_format = p_format_list[i_current_rate_format];
-            else p_sys->stream_format = p_format_list[i_backup_rate_format]; /* And if we have to, any digital format will be just fine (highest rate possible) */
+                p_sys->stream_format = p_format_list[i_current_rate_format].mFormat;
+            else p_sys->stream_format = p_format_list[i_backup_rate_format].mFormat; /* And if we have to, any digital format will be just fine (highest rate possible) */
         }
         free( p_format_list );
     }