* Correct socklen_t detection.

* Fix for S/PDIF encapsulation on big endian systems.
* S/PDIF output for Mac OS X, courtesy of Heiko Panther <heiko_panthe@mac.com>.

configure.ac.in

@@ -283,7 +283,21 @@ if ${have_nanosleep}; then
   AC_DEFINE(HAVE_NANOSLEEP, 1,
             Define if nanosleep is available.)
 fi
-# HP/UX port
+
+dnl Check for socklen_t
+AC_CACHE_CHECK([for socklen_t], ac_cv_type_socklen_t,
+  [AC_TRY_COMPILE(
+     [#include <sys/types.h>
+      #include <sys/socket.h>],
+     [socklen_t len = 42; return len;],
+     ac_cv_type_socklen_t=yes,
+     ac_cv_type_socklen_t=no)])
+if test x$ac_cv_type_socklen_t != xno; then
+  AC_DEFINE(HAVE_SOCKLEN_T, 1,
+            Define if <sys/socket.h> defines socklen_t.)
+fi
+
+dnl HP/UX port
 AC_CHECK_LIB(rt,sem_init, [LDFLAGS_vlc="${LDFLAGS_vlc} -lrt"])
 
 AC_CHECK_FUNC(inet_aton,,[

include/audio_output.h

@@ -2,7 +2,7 @@
  * audio_output.h : audio output interface
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: audio_output.h,v 1.71 2002/11/14 22:38:46 massiot Exp $
+ * $Id: audio_output.h,v 1.72 2002/11/28 23:24:14 massiot Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *
@@ -36,13 +36,16 @@ struct audio_sample_format_t
     /* Describes from which original channels, before downmixing, the
      * buffer is derived. */
     u32                 i_original_channels;
-    /* Optional - for A52, SPDIF and DTS types */
+    /* Optional - for A/52, SPDIF and DTS types : */
+    /* Bytes used by one compressed frame, depends on bitrate. */
     unsigned int        i_bytes_per_frame;
+    /* Number of sampleframes contained in one compressed frame. */
     unsigned int        i_frame_length;
     /* Please note that it may be completely arbitrary - buffers are not
      * obliged to contain a integral number of so-called "frames". It's
      * just here for the division :
-     * i_nb_samples * i_bytes_per_frame / i_frame_length */
+     * buffer_size = i_nb_samples * i_bytes_per_frame / i_frame_length
+     */
 };
 
 #define AOUT_FMTS_IDENTICAL( p_first, p_second ) (                          \

include/vlc_common.h

@@ -3,7 +3,7 @@
  * Collection of useful common types and macros definitions
  *****************************************************************************
  * Copyright (C) 1998, 1999, 2000 VideoLAN
- * $Id: vlc_common.h,v 1.41 2002/11/25 03:12:42 ipkiss Exp $
+ * $Id: vlc_common.h,v 1.42 2002/11/28 23:24:14 massiot Exp $
  *
  * Authors: Samuel Hocevar <sam@via.ecp.fr>
  *          Vincent Seguin <seguin@via.ecp.fr>
@@ -116,6 +116,10 @@ typedef uint8_t             yuv_data_t;
 /* Audio volume */
 typedef uint16_t            audio_volume_t;
 
+#ifndef HAVE_SOCKLEN_T
+typedef int                 socklen_t;
+#endif
+
 /*****************************************************************************
  * Old types definitions
  *****************************************************************************

modules/audio_filter/converter/a52tofloat32.c

@@ -4,7 +4,7 @@
  *   (http://liba52.sf.net/).
  *****************************************************************************
  * Copyright (C) 2001, 2002 VideoLAN
- * $Id: a52tofloat32.c,v 1.8 2002/11/21 23:06:08 massiot Exp $
+ * $Id: a52tofloat32.c,v 1.9 2002/11/28 23:24:14 massiot Exp $
  *
  * Authors: Gildas Bazin <gbazin@netcourrier.com>
  *          Christophe Massiot <massiot@via.ecp.fr>
@@ -222,37 +222,37 @@ static void Interleave( float * p_out, const float * p_in, int i_nb_channels )
      * The WG-4 order is appropriate for stereo, quadrophonia, and 5.1 surround.
      *
      * 6 channel mode
-     * channel	liba52 order	WG-4 order
-     * 0	LFE		// L
-     * 1	L		// R
-     * 2	C		// LS
-     * 3	R		// RS
-     * 4	LS		// C
-     * 5	RS		// LFE
+     * channel  liba52 order    WG-4 order
+     * 0        LFE             // L
+     * 1        L               // R
+     * 2        C               // LS
+     * 3        R               // RS
+     * 4        LS              // C
+     * 5        RS              // LFE
      *
      * The liba52 moves channels to the front if there are unused spaces, so
      * there is no gap between channels. The translation table says which
-     * channel of the new stream [use the new channel # as index] is taken
-     * from which original channel [use the number from the array to address
-     * the original channel].
+     * channel of the new stream is taken from which original channel [use
+     * the new channel as the array index, use the number you get from the
+     * array to address the original channel].
      */
-    
+
     static const int translation[7][6] =
-    {{ 0, 0, 0, 0, 0, 0 },	/* 0 channels (rarely used) */
-    { 0, 0, 0, 0, 0, 0 },	/* 1 ch */
-    { 0, 1, 0, 0, 0, 0 },	/* 2 */
-    { 1, 2, 0, 0, 0, 0 },	/* 3 */
-    { 1, 3, 2, 0, 0, 0 },	/* 4 */
-    { 1, 3, 4, 2, 0, 0 },	/* 5 */
-    { 1, 3, 4, 5, 2, 0 }};	/* 6 */
-	
+    {{ 0, 0, 0, 0, 0, 0 },      /* 0 channels (rarely used) */
+    { 0, 0, 0, 0, 0, 0 },       /* 1 ch */
+    { 0, 1, 0, 0, 0, 0 },       /* 2 */
+    { 1, 2, 0, 0, 0, 0 },       /* 3 */
+    { 1, 3, 2, 0, 0, 0 },       /* 4 */
+    { 1, 3, 4, 2, 0, 0 },       /* 5 */
+    { 1, 3, 4, 5, 2, 0 }};      /* 6 */
+
     int i, j;
     for ( j = 0; j < i_nb_channels; j++ )
     {
         for ( i = 0; i < 256; i++ )
         {
-            p_out[i * i_nb_channels + translation[i_nb_channels][j]]
-                    = p_in[j * 256 + i];
+            p_out[i * i_nb_channels + j] = p_in[translation[i_nb_channels][j]
+                                                 * 256 + i];
         }
     }
 }

modules/audio_filter/converter/a52tospdif.c

@@ -2,7 +2,7 @@
  * a52tospdif.c : encapsulates A/52 frames into S/PDIF packets
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: a52tospdif.c,v 1.14 2002/11/20 13:37:35 sam Exp $
+ * $Id: a52tospdif.c,v 1.15 2002/11/28 23:24:14 massiot Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *          Stphane Borel <stef@via.ecp.fr>
@@ -78,42 +78,39 @@ static int Create( vlc_object_t *p_this )
 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
                     aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
 {
-#ifdef WORDS_BIGENDIAN
-    static const u8 p_sync[6] = { 0xF8, 0x72, 0x4E, 0x1F, 0x00, 0x01 };
-#else
+    /* It is not entirely clear which endianness the AC3 stream should have.
+     * I have been told endianness does not matter, AC3 can be both endian.
+     * But then, I could not get it to work on Mac OS X and a JVC RX-6000R
+     * decoder without using little endian. So right now, I convert to little
+     * endian.
+     */
+
     static const u8 p_sync[6] = { 0x72, 0xF8, 0x1F, 0x4E, 0x01, 0x00 };
-#   ifndef HAVE_SWAB
+#ifndef HAVE_SWAB
     u16 i;
-#   endif
 #endif
     u16 i_length = p_in_buf->i_nb_bytes;
-    u16 * pi_length;
+    u8 * pi_length;
     byte_t * p_in = p_in_buf->p_buffer;
     byte_t * p_out = p_out_buf->p_buffer;
+    byte_t * p_tmp;
 
     /* Copy the S/PDIF headers. */
     memcpy( p_out, p_sync, 6 );
-    pi_length = (u16 *)(p_out + 6);
-    *pi_length = i_length * 8;
-
-    /* FIXME : if i_length is odd, the following code sucks. What should
-     * we do ? --Meuuh */
+    pi_length = (p_out + 6);
+    *pi_length = (i_length * 8) & 0xff;
+    *(pi_length + 1) = (i_length * 8) >> 8;
 
-#ifndef WORDS_BIGENDIAN
-#   ifdef HAVE_SWAB
+#ifdef HAVE_SWAB
     swab( p_in, p_out + 8, i_length );
-#   else
-    p_out += 8;
+#else
+    p_tmp = p_out + 8;
     for ( i = i_length / 2 ; i-- ; )
     {
-        p_out[0] = p_in[1];
-        p_out[1] = p_in[0];
-        p_out += 2; p_in += 2;
+        p_tmp[0] = p_in[1];
+        p_tmp[1] = p_in[0];
+        p_tmp += 2; p_in += 2;
     }
-#   endif
-
-#else
-    p_filter->p_vlc->pf_memcpy( p_out + 8, p_in, i_length );
 #endif
 
     p_filter->p_vlc->pf_memset( p_out + 8 + i_length, 0,

modules/gui/macosx/aout.m

@@ -2,7 +2,7 @@
  * aout.m: CoreAudio output plugin
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: aout.m,v 1.15 2002/11/18 23:00:41 massiot Exp $
+ * $Id: aout.m,v 1.16 2002/11/28 23:24:15 massiot Exp $
  *
  * Authors: Colin Delacroix <colin@zoy.org>
  *          Jon Lech Johansen <jon-vl@nanocrew.net>
@@ -107,13 +107,17 @@ int E_(OpenAudio)( vlc_object_t * p_this )
 
     /* Get a description of the data format used by the device */
     i_param_size = sizeof(AudioStreamBasicDescription); 
-    err = AudioDeviceGetProperty(p_sys->device, 0, false, kAudioDevicePropertyStreamFormat, &i_param_size, &p_sys->stream_format );
+    err = AudioDeviceGetProperty(p_sys->device, 0, false, kAudioDevicePropertyStreamFormat,
+				 &i_param_size, &p_sys->stream_format );
     if( err != noErr )
     {
         msg_Err( p_aout, "failed to get stream format: %4.4s", &err );
         return -1 ;
     }
 
+    /* Now we know the sample rate of the device */
+    p_aout->output.output.i_rate = p_sys->stream_format.mSampleRate;
+
     msg_Dbg( p_aout, "mSampleRate %ld, mFormatID %4.4s, mFormatFlags %ld, mBytesPerPacket %ld, mFramesPerPacket %ld, mBytesPerFrame %ld, mChannelsPerFrame %ld, mBitsPerChannel %ld",
            (UInt32)p_sys->stream_format.mSampleRate, &p_sys->stream_format.mFormatID,
            p_sys->stream_format.mFormatFlags, p_sys->stream_format.mBytesPerPacket,
@@ -122,43 +126,14 @@ int E_(OpenAudio)( vlc_object_t * p_this )
 
     msg_Dbg( p_aout, "vlc format %4.4s, mac output format '%4.4s'",
              (char *)&p_aout->output.output.i_format, &p_sys->stream_format.mFormatID );
-
-    switch(p_sys->stream_format.mFormatID)
-    {
-    case 0:
-    case kAudioFormatLinearPCM:
-        p_aout->output.output.i_format = VLC_FOURCC('f','l','3','2');
-        if ( p_sys->stream_format.mChannelsPerFrame < 6 )
-            p_aout->output.output.i_physical_channels
-                 = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
-        else
-            p_aout->output.output.i_physical_channels
-                  = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT
-                  | AOUT_CHAN_CENTER | AOUT_CHAN_REARRIGHT
-                  | AOUT_CHAN_REARLEFT | AOUT_CHAN_LFE;
-        break;
-
-    case kAudioFormat60958AC3:
-    case 'IAC3':
-        p_aout->output.output.i_format = VLC_FOURCC('s','p','d','i');
-        //not necessary, use the input's format by default --Meuuh
-        //p_aout->output.output.i_channels = AOUT_CHAN_DOLBY | AOUT_CHAN_LFE;
-        p_aout->output.output.i_bytes_per_frame = AOUT_SPDIF_SIZE; //p_sys->stream_format.mBytesPerFrame;
-        p_aout->output.output.i_frame_length = A52_FRAME_NB; //p_sys->stream_format.mFramesPerPacket;
-        break;
-
-    default:
-        msg_Err( p_aout, "Unknown hardware format '%4.4s'. Go ask Heiko.", &p_sys->stream_format.mFormatID );
-        return -1;
-    }
-
-    /* Set sample rate and channels per frame */
-    p_aout->output.output.i_rate = p_sys->stream_format.mSampleRate;
-
+    
     /* Get the buffer size that the device uses for IO */
+    // If we do PCM, use the device's given buffer size
+    // If we do raw AC3, we could use the devices given size too
+    // If we do AC3 over SPDIF, force the size of one AC3 frame
+    // (I think we need to do that because of the packetizer)
     i_param_size = sizeof( p_sys->i_buffer_size );
-#if 1	// i have a feeling we should use the buffer size imposed by the AC3 device (usually about 6144)
-    err = AudioDeviceGetProperty( p_sys->device, 1, false, 
+    err = AudioDeviceGetProperty( p_sys->device, 0, false, 
                                   kAudioDevicePropertyBufferSize, 
                                   &i_param_size, &p_sys->i_buffer_size );
     if(err) {
@@ -166,26 +141,130 @@ int E_(OpenAudio)( vlc_object_t * p_this )
 	return -1;
     }
     else msg_Dbg( p_aout, "native buffer Size: %d", p_sys->i_buffer_size );
+    
+    if((p_sys->stream_format.mFormatID==kAudioFormat60958AC3
+       || p_sys->stream_format.mFormatID=='IAC3')
+       && p_sys->i_buffer_size != AOUT_SPDIF_SIZE)
+    {
+	p_sys->i_buffer_size = AOUT_SPDIF_SIZE;
+	i_param_size = sizeof( p_sys->i_buffer_size );
+	err = AudioDeviceSetProperty( p_sys->device, 0, 0, false,
+			       kAudioDevicePropertyBufferSize,
+			       i_param_size, &p_sys->i_buffer_size );
+	if( err != noErr )
+	{
+	    msg_Err( p_aout, "failed to set device buffer size: %4.4s", &err );
+	    return -1;
+	}
+	else msg_Dbg(p_aout, "bufferSize set to %d", p_sys->i_buffer_size);
+    };
+
+
+    // We now know the buffer size in bytes. Set the values for the vlc converters.
+    switch(p_sys->stream_format.mFormatID)
+    {
+	case 0:
+	case kAudioFormatLinearPCM:
+	    p_aout->output.output.i_format = VLC_FOURCC('f','l','3','2');
+	    if ( p_sys->stream_format.mChannelsPerFrame < 6 )
+		p_aout->output.output.i_physical_channels
+		    = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
+	    else
+		p_aout->output.output.i_physical_channels
+		    = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT
+		    | AOUT_CHAN_CENTER | AOUT_CHAN_REARRIGHT
+		    | AOUT_CHAN_REARLEFT | AOUT_CHAN_LFE;
+
+	    p_aout->output.i_nb_samples = p_sys->i_buffer_size / p_sys->stream_format.mBytesPerFrame;
+	    break;
+
+	case kAudioFormat60958AC3:
+	case 'IAC3':
+	    p_aout->output.output.i_format = VLC_FOURCC('s','p','d','i');
+	    msg_Dbg(p_aout, "phychan %d, ochan %d, bytes/fr %d, frlen %d",
+	     p_aout->output.output.i_physical_channels,
+	     p_aout->output.output.i_original_channels,
+	     p_aout->output.output.i_bytes_per_frame,
+	     p_aout->output.output.i_frame_length);
+
+	    p_aout->output.output.i_bytes_per_frame = AOUT_SPDIF_SIZE; //p_sys->stream_format.mBytesPerFrame;
+	    p_aout->output.output.i_frame_length = A52_FRAME_NB; //p_sys->stream_format.mFramesPerPacket;
+	    p_aout->output.i_nb_samples = p_aout->output.output.i_frame_length;
+	    
+	    
+	     //Probably not needed after all
+	    
+	    // Some more settings to make the SPDIF device work... Other SPDIF Devices might need additional
+	    // values here. But don't change these, in order to not break existing devices. Either add values
+	    // which are not being set here, or check if the SetProperty was successful, and try another version
+	    // if not.
+//	    p_sys->stream_format.mBytesPerFrame=4;	// usually set to 0 for AC3 by the system
+//	    p_sys->stream_format.mFormatFlags|=kAudioFormatFlagIsBigEndian;
+//		+kAudioFormatFlagIsPacked
+//		+kAudioFormatFlagIsNonInterleaved;
+//	    p_sys->stream_format.mBytesPerPacket=6144;
+//	    p_sys->stream_format.mFramesPerPacket=1536;
+	    
+	    break;
+
+	default:
+	    msg_Err( p_aout, "Unknown hardware format '%4.4s'. Go ask Heiko.", &p_sys->stream_format.mFormatID );
+	    return -1;
+    }
+
+
+    // Now tell the device how many sample frames to expect in each buffer
+    i_param_size=sizeof(p_aout->output.i_nb_samples);
+#if 0
+    err = AudioDeviceGetProperty( p_sys->device, 0, false,
+                                  kAudioDevicePropertyBufferFrameSize,
+                                  &i_param_size, &p_aout->output.i_nb_samples);
+    if(err) {
+	msg_Err(p_aout, "failed to get BufferFrameSize - err %4.4s, device %ld", &err, p_sys->device);
+	return -1;
+    }
+    else msg_Dbg( p_aout, "native BufferFrameSize: %d", p_aout->output.i_nb_samples);
 #else
-    p_sys->i_buffer_size = p_aout->output.output.i_bytes_per_frame;
-    err = AudioDeviceSetProperty( p_sys->device, 0, 1, false,
-                                  kAudioDevicePropertyBufferSize,
-                                  i_param_size, &p_sys->i_buffer_size );
+    err = AudioDeviceSetProperty( p_sys->device, 0, 0, false,
+                                  kAudioDevicePropertyBufferFrameSize,
+                                  i_param_size, &p_aout->output.i_nb_samples);
     if( err != noErr )
     {
-        msg_Err( p_aout, "failed to set device buffer size: %4.4s", err );
-        return( -1 );
+        msg_Err( p_aout, "failed to set BufferFrameSize: %4.4s", &err );
+        return -1;
     }
-    else msg_Dbg(p_aout, "bufferSize set to %d", p_sys->i_buffer_size);
+    else msg_Dbg(p_aout, "bufferFrameSize set to %d", p_aout->output.i_nb_samples);
 #endif
 
-    p_aout->output.i_nb_samples = p_sys->i_buffer_size / p_sys->stream_format.mBytesPerFrame;
-
+/*    
+    // And set the device format, since we might have changed some of it above
+    i_param_size = sizeof(AudioStreamBasicDescription);
+    err = AudioDeviceSetProperty(p_sys->device, 0, 0, false, kAudioDevicePropertyStreamFormat,
+				 i_param_size, &p_sys->stream_format );
+    if( err != noErr )
+    {
+        msg_Err( p_aout, "failed to set stream format: %4.4s", &err );
+        return -1 ;
+    }
+    else
+	msg_Dbg( p_aout, "set: mSampleRate %ld, mFormatID %4.4s, mFormatFlags %ld, mBytesPerPacket %ld, mFramesPerPacket %ld, mBytesPerFrame %ld, mChannelsPerFrame %ld, mBitsPerChannel %ld",
+	  (UInt32)p_sys->stream_format.mSampleRate, &p_sys->stream_format.mFormatID,
+	  p_sys->stream_format.mFormatFlags, p_sys->stream_format.mBytesPerPacket,
+	  p_sys->stream_format.mFramesPerPacket, p_sys->stream_format.mBytesPerFrame,
+	  p_sys->stream_format.mChannelsPerFrame, p_sys->stream_format.mBitsPerChannel );
+  */  
+    
+    
     /* Add callback */
     err = AudioDeviceAddIOProc( p_sys->device,
                                 (AudioDeviceIOProc)IOCallback,
                                 (void *)p_aout );
-
+    if( err != noErr )
+    {
+        msg_Err( p_aout, "AudioDeviceAddIOProc failed: %4.4s", &err );
+	return -1;
+    }
+    
     /* Open the output with callback IOCallback */
     err = AudioDeviceStart( p_sys->device,
                             (AudioDeviceIOProc)IOCallback );
@@ -216,7 +295,14 @@ void E_(CloseAudio)( aout_instance_t * p_aout )
                            (AudioDeviceIOProc)IOCallback ); 
     if( err != noErr )
     {
-        msg_Err( p_aout, "AudioDeviceStop failed: %d", err );
+        msg_Err( p_aout, "AudioDeviceStop failed: %4.4s", &err );
+    }
+
+    err = AudioDeviceRemoveIOProc( p_sys->device,
+                                (AudioDeviceIOProc)IOCallback );
+    if( err != noErr )
+    {
+        msg_Err( p_aout, "AudioDeviceRemoveIOProc failed: %4.4s", &err );
     }
 
     free( p_sys );
@@ -229,6 +315,7 @@ static void Play( aout_instance_t * p_aout )
 {
 }
 
+#include <syslog.h>
 /*****************************************************************************
  * IOCallback : callback for audio output
  *****************************************************************************/
@@ -245,13 +332,12 @@ static OSStatus IOCallback( AudioDeviceID inDevice,
     mtime_t         current_date;
     AudioTimeStamp  host_time;
     aout_buffer_t * p_buffer;
-
+    
     host_time.mFlags = kAudioTimeStampHostTimeValid;
     AudioDeviceTranslateTime( inDevice, inOutputTime, &host_time );
     current_date = p_sys->clock_diff
                  + AudioConvertHostTimeToNanos(host_time.mHostTime) / 1000;
 
-//    msg_Dbg(p_aout, "Now fetching audio data");
     p_buffer = aout_OutputNextBuffer( p_aout, current_date, (p_aout->output.output.i_format == VLC_FOURCC('s','p','d','i')) );
 
     /* move data into output data buffer */
@@ -260,14 +346,7 @@ static OSStatus IOCallback( AudioDeviceID inDevice,
 	BlockMoveData( p_buffer->p_buffer,
                        outOutputData->mBuffers[ 0 ].mData, 
                        p_sys->i_buffer_size );
-
-/*	msg_Dbg(p_aout, "This buffer has %d bytes, i take %d: %f %f %f %f",
-        p_buffer->i_nb_bytes, p_sys->i_buffer_size,
-        ((float*)p_buffer->p_buffer)[0], ((float*)p_buffer->p_buffer)[1],
-        ((float*)p_buffer->p_buffer)[2], ((float*)p_buffer->p_buffer)[3]);
-*/
-
-    
+//	syslog(LOG_INFO, "convert: %08lX %08lX %08lX", ((long*)p_buffer->p_buffer)[0], ((long*)p_buffer->p_buffer)[1], ((long*)p_buffer->p_buffer)[2]);
 	aout_BufferFree( p_buffer );
     }
     else
@@ -275,6 +354,9 @@ static OSStatus IOCallback( AudioDeviceID inDevice,
         memset(outOutputData->mBuffers[ 0 ].mData, 0, p_sys->i_buffer_size);
     }
 
+
+//    outOutputData->mBuffers[0].mDataByteSize=p_sys->i_buffer_size;
+    
     return noErr;     
 }
 

modules/gui/macosx/asystm.m

@@ -3,7 +3,6 @@
 //  
 //
 //  Created by Heiko Panther on Tue Sep 10 2002.
-//  Copyright (c) 2002 __MyCompanyName__. All rights reserved.
 //
 
 #import "asystm.h"

modules/misc/network/ipv4.c

@@ -2,7 +2,7 @@
  * ipv4.c: IPv4 network abstraction layer
  *****************************************************************************
  * Copyright (C) 2001, 2002 VideoLAN
- * $Id: ipv4.c,v 1.6 2002/11/23 04:40:53 sam Exp $
+ * $Id: ipv4.c,v 1.7 2002/11/28 23:24:15 massiot Exp $
  *
  * Authors: Christophe Massiot <massiot@via.ecp.fr>
  *          Mathias Kretschmer <mathias@research.att.com>
@@ -145,7 +145,7 @@ static int OpenUDP( vlc_object_t * p_this, network_socket_t * p_socket )
 #endif
 
     int i_handle, i_opt;
-    unsigned int i_opt_size;
+    socklen_t i_opt_size;
     struct sockaddr_in sock;
 
     if( i_bind_port == 0 )

modules/misc/network/ipv6.c

@@ -2,7 +2,7 @@
  * ipv6.c: IPv6 network abstraction layer
  *****************************************************************************
  * Copyright (C) 2002 VideoLAN
- * $Id: ipv6.c,v 1.4 2002/11/19 20:56:45 gbazin Exp $
+ * $Id: ipv6.c,v 1.5 2002/11/28 23:24:15 massiot Exp $
  *
  * Authors: Alexis Guillard <alexis.guillard@bt.com>
  *          Christophe Massiot <massiot@via.ecp.fr>
@@ -230,7 +230,7 @@ static int OpenUDP( vlc_object_t * p_this, network_socket_t * p_socket )
     int i_server_port = p_socket->i_server_port;
 
     int i_handle, i_opt;
-    unsigned int i_opt_size;
+    socklen_t i_opt_size;
     struct sockaddr_in6 sock;
 
     /* Open a SOCK_DGRAM (UDP) socket, in the AF_INET6 domain, automatic (0)