Merge svn changes up to r30301

9 files changed, 42 insertions, 59 deletions

diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c
index 140b13a62a..92b8b64ea7 100644
--- a/libao2/ao_alsa.c
+++ b/libao2/ao_alsa.c
@@ -128,7 +128,7 @@ static int control(int cmd, void *arg)
       long get_vol, set_vol;
       float f_multi;
 
-      if(ao_data.format == AF_FORMAT_AC3)
+      if(AF_FORMAT_IS_AC3(ao_data.format))
 	return CONTROL_TRUE;
 
       if(mixer_channel) {
@@ -374,15 +374,11 @@ static int init(int rate_hz, int channels, int format, int flags)
       case AF_FORMAT_U16_BE:
 	alsa_format = SND_PCM_FORMAT_U16_BE;
 	break;
-#if !HAVE_BIGENDIAN
-      case AF_FORMAT_AC3:
-#endif
+      case AF_FORMAT_AC3_LE:
       case AF_FORMAT_S16_LE:
 	alsa_format = SND_PCM_FORMAT_S16_LE;
 	break;
-#if HAVE_BIGENDIAN
-      case AF_FORMAT_AC3:
-#endif
+      case AF_FORMAT_AC3_BE:
       case AF_FORMAT_S16_BE:
 	alsa_format = SND_PCM_FORMAT_S16_BE;
 	break;
@@ -437,7 +433,7 @@ static int init(int rate_hz, int channels, int format, int flags)
      * while opening the abstract alias for the spdif subdevice
      * 'iec958'
      */
-    if (format == AF_FORMAT_AC3) {
+    if (AF_FORMAT_IS_AC3(format)) {
 	device.str = "iec958";
 	mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
     }
@@ -496,12 +492,13 @@ static int init(int rate_hz, int channels, int format, int flags)
     }
 
     if (!alsa_handler) {
+      int isac3 =  AF_FORMAT_IS_AC3(format);
       //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
-      if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
+      if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
 	{
 	  if (err != -EBUSY && ao_noblock) {
 	    mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
-	    if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
+	    if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
 	      mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
 	      return 0;
 	    }
@@ -544,6 +541,9 @@ static int init(int rate_hz, int channels, int format, int flags)
          mp_tmsg(MSGT_AO,MSGL_INFO,
 		"[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
          alsa_format = SND_PCM_FORMAT_S16_LE;
+         if (AF_FORMAT_IS_AC3(ao_data.format))
+           ao_data.format = AF_FORMAT_AC3_LE;
+         else
          ao_data.format = AF_FORMAT_S16_LE;
       }
 
@@ -583,7 +583,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	  return 0;
         }
 
-      bytes_per_sample = snd_pcm_format_physical_width(alsa_format) / 8;
+      bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
       bytes_per_sample *= ao_data.channels;
       ao_data.bps = ao_data.samplerate * bytes_per_sample;
 
diff --git a/libao2/ao_alsa5.c b/libao2/ao_alsa5.c
index 8843efb68a..ad17f5895f 100644
--- a/libao2/ao_alsa5.c
+++ b/libao2/ao_alsa5.c
@@ -101,15 +101,11 @@ static int init(int rate_hz, int channels, int format, int flags)
 	case AF_FORMAT_U16_BE:
 	    alsa_format.format = SND_PCM_SFMT_U16_BE;
 	    break;
-#if !HAVE_BIGENDIAN
-	case AF_FORMAT_AC3:
-#endif
+	case AF_FORMAT_AC3_LE:
 	case AF_FORMAT_S16_LE:
 	    alsa_format.format = SND_PCM_SFMT_S16_LE;
 	    break;
-#if HAVE_BIGENDIAN
-	case AF_FORMAT_AC3:
-#endif
+	case AF_FORMAT_AC3_BE:
 	case AF_FORMAT_S16_BE:
 	    alsa_format.format = SND_PCM_SFMT_S16_BE;
 	    break;
diff --git a/libao2/ao_coreaudio.c b/libao2/ao_coreaudio.c
index 130eee8f97..b60fb094e7 100644
--- a/libao2/ao_coreaudio.c
+++ b/libao2/ao_coreaudio.c
@@ -263,7 +263,7 @@ int b_alive;
     ao->b_changed_mixing = 0;
 
     /* Probe whether device support S/PDIF stream output if input is AC3 stream. */
-    if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3)
+    if (AF_FORMAT_IS_AC3(format))
     {
         /* Find the ID of the default Device. */
         i_param_size = sizeof(AudioDeviceID);
@@ -314,23 +314,7 @@ int b_alive;
 	inDesc.mSampleRate=rate;
 	inDesc.mFormatID=ao->b_supports_digital ? kAudioFormat60958AC3 : kAudioFormatLinearPCM;
 	inDesc.mChannelsPerFrame=channels;
-	switch(format&AF_FORMAT_BITS_MASK){
-	case AF_FORMAT_8BIT:
-		inDesc.mBitsPerChannel=8;
-		break;
-	case AF_FORMAT_16BIT:
-		inDesc.mBitsPerChannel=16;
-		break;
-	case AF_FORMAT_24BIT:
-		inDesc.mBitsPerChannel=24;
-		break;
-	case AF_FORMAT_32BIT:
-		inDesc.mBitsPerChannel=32;
-		break;
-	default:
-		ao_msg(MSGT_AO, MSGL_WARN, "Unsupported format (0x%08x)\n", format);
-		goto err_out;
-	}
+	inDesc.mBitsPerChannel=af_fmt2bits(format);
 
     if((format&AF_FORMAT_POINT_MASK)==AF_FORMAT_F) {
 	// float
@@ -344,13 +328,7 @@ int b_alive;
 	// unsigned int
 		inDesc.mFormatFlags = kAudioFormatFlagIsPacked;
     }
-    if ((format & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_AC3) {
-        // Currently ac3 input (comes from hwac3) is always in native byte-order.
-#if HAVE_BIGENDIAN
-        inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
-#endif
-    }
-    else if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
+    if ((format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
         inDesc.mFormatFlags |= kAudioFormatFlagIsBigEndian;
 
     inDesc.mFramesPerPacket = 1;
diff --git a/libao2/ao_dsound.c b/libao2/ao_dsound.c
index f66042a793..c53820638d 100644
--- a/libao2/ao_dsound.c
+++ b/libao2/ao_dsound.c
@@ -327,7 +327,7 @@ static int write_buffer(unsigned char *data, int len)
 
   if (SUCCEEDED(res))
   {
-  	if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) {
+  	if( (ao_data.channels == 6) && !AF_FORMAT_IS_AC3(ao_data.format) ) {
   	    // reorder channels while writing to pointers.
   	    // it's this easy because buffer size and len are always
   	    // aligned to multiples of channels*bytespersample
@@ -432,8 +432,11 @@ static int init(int rate, int channels, int format, int flags)
 		mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: 8 channel audio not yet supported\n");
 		return 0;
 	}
+
+	if (AF_FORMAT_IS_AC3(format))
+		format = AF_FORMAT_AC3_NE;
 	switch(format){
-		case AF_FORMAT_AC3:
+		case AF_FORMAT_AC3_NE:
 		case AF_FORMAT_S24_LE:
 		case AF_FORMAT_S16_LE:
 		case AF_FORMAT_U8:
@@ -456,7 +459,7 @@ static int init(int rate, int channels, int format, int flags)
 	wformat.Format.cbSize          = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
 	wformat.Format.nChannels       = channels;
 	wformat.Format.nSamplesPerSec  = rate;
-	if (format == AF_FORMAT_AC3) {
+	if (AF_FORMAT_IS_AC3(format)) {
 		wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
 		wformat.Format.wBitsPerSample  = 16;
 		wformat.Format.nBlockAlign     = 4;
diff --git a/libao2/ao_dxr2.c b/libao2/ao_dxr2.c
index 4712f3341f..f9e31836d5 100644
--- a/libao2/ao_dxr2.c
+++ b/libao2/ao_dxr2.c
@@ -201,7 +201,7 @@ static int play(void* data,int len,int flags){
   // MPEG and AC3 don't work :-(
     if(ao_data.format==AF_FORMAT_MPEG2)
       send_mpeg_ps_packet (data, len, 0xC0, ao_data.pts, 2, write_dxr2);
-    else if(ao_data.format==AF_FORMAT_AC3)
+    else if(AF_FORMAT_IS_AC3(ao_data.format))
       send_mpeg_ps_packet (data, len, 0x80, ao_data.pts, 2, write_dxr2);
     else {
 	int i;
diff --git a/libao2/ao_mpegpes.c b/libao2/ao_mpegpes.c
index 47a75d7545..f6ab1c6aca 100644
--- a/libao2/ao_mpegpes.c
+++ b/libao2/ao_mpegpes.c
@@ -250,9 +250,12 @@ static int init(int rate,int channels,int format,int flags){
     switch(format){
 	case AF_FORMAT_S16_BE:
 	case AF_FORMAT_MPEG2:
-	case AF_FORMAT_AC3:
+	case AF_FORMAT_AC3_BE:
 	    ao_data.format=format;
 	    break;
+	case AF_FORMAT_AC3_LE:
+	    ao_data.format=AF_FORMAT_AC3_BE;
+	    break;
 	default:
 	    ao_data.format=AF_FORMAT_S16_BE;
     }
@@ -331,8 +334,6 @@ static int play(void* data,int len,int flags){
 	unsigned short *s=data;
 //	if(len>2000) len=2000;
 //	printf("ao_mpegpes: len=%d  \n",len);
-	if(ao_data.format==AF_FORMAT_AC3)
-	    for(i=0;i<len/2;i++) s[i]=(s[i]>>8)|(s[i]<<8); // le<->be
 	send_mpeg_lpcm_packet(data, len, 0xA0, ao_data.pts, freq_id, my_ao_write);
     }
     return len;
diff --git a/libao2/ao_oss.c b/libao2/ao_oss.c
index 6872b15946..58d3b233cc 100644
--- a/libao2/ao_oss.c
+++ b/libao2/ao_oss.c
@@ -96,7 +96,7 @@ static int format2oss(int format)
     case AF_FORMAT_MPEG2: return AFMT_MPEG;
 #endif
 #ifdef AFMT_AC3
-    case AF_FORMAT_AC3: return AFMT_AC3;
+    case AF_FORMAT_AC3_NE: return AFMT_AC3;
 #endif
     }
     mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format));
@@ -139,7 +139,7 @@ static int oss2format(int format)
     case AFMT_MPEG: return AF_FORMAT_MPEG2;
 #endif
 #ifdef AFMT_AC3
-    case AFMT_AC3: return AF_FORMAT_AC3;
+    case AFMT_AC3: return AF_FORMAT_AC3_NE;
 #endif
     }
     mp_tmsg(MSGT_GLOBAL,MSGL_ERR,"[AO OSS] Unknown/Unsupported OSS format: %x.\n", format);
@@ -208,7 +208,7 @@ static int control(int cmd,void *arg){
             return CONTROL_OK;
 #endif
 
-	    if(ao_data.format == AF_FORMAT_AC3)
+	    if(AF_FORMAT_IS_AC3(ao_data.format))
 		return CONTROL_TRUE;
 
 	    if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
@@ -326,12 +326,14 @@ static int init(int rate,int channels,int format,int flags){
   fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
 #endif
 
-  if(format == AF_FORMAT_AC3) {
+  if(AF_FORMAT_IS_AC3(format)) {
     ao_data.samplerate=rate;
     ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
   }
 
 ac3_retry:
+  if (AF_FORMAT_IS_AC3(format))
+    format = AF_FORMAT_AC3_NE;
   ao_data.format=format;
   oss_format=format2oss(format);
   if (oss_format == -1) {
@@ -361,7 +363,7 @@ ac3_retry:
     af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
 
   ao_data.channels = channels;
-  if(format != AF_FORMAT_AC3) {
+  if(!AF_FORMAT_IS_AC3(format)) {
     // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
     if (ao_data.channels > 2) {
       if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
@@ -476,10 +478,10 @@ static void reset(void){
 #endif
 
   oss_format = format2oss(ao_data.format);
-  if(ao_data.format == AF_FORMAT_AC3)
+  if(AF_FORMAT_IS_AC3(ao_data.format))
     ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
   ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
-  if(ao_data.format != AF_FORMAT_AC3) {
+  if(!AF_FORMAT_IS_AC3(ao_data.format)) {
     if (ao_data.channels > 2)
       ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
     else {
diff --git a/libao2/ao_pcm.c b/libao2/ao_pcm.c
index a03f41193e..5c418d0f5a 100644
--- a/libao2/ao_pcm.c
+++ b/libao2/ao_pcm.c
@@ -127,7 +127,8 @@ static int init(int rate,int channels,int format,int flags){
         format=AF_FORMAT_U8;
     case AF_FORMAT_U8:
         break;
-    case AF_FORMAT_AC3:
+    case AF_FORMAT_AC3_BE:
+    case AF_FORMAT_AC3_LE:
         bits=16;
         break;
     default:
diff --git a/libao2/ao_win32.c b/libao2/ao_win32.c
index a0475455ec..55ed17b457 100644
--- a/libao2/ao_win32.c
+++ b/libao2/ao_win32.c
@@ -143,8 +143,10 @@ static int init(int rate,int channels,int format,int flags)
 	unsigned char* buffer;
 	int i;
 
+	if (AF_FORMAT_IS_AC3(format))
+		format = AF_FORMAT_AC3_NE;
 	switch(format){
-		case AF_FORMAT_AC3:
+		case AF_FORMAT_AC3_NE:
 		case AF_FORMAT_S24_LE:
 		case AF_FORMAT_S16_LE:
 		case AF_FORMAT_U8:
@@ -180,7 +182,7 @@ static int init(int rate,int channels,int format,int flags)
     wformat.Format.cbSize          = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
     wformat.Format.nChannels       = channels;
     wformat.Format.nSamplesPerSec  = rate;
-    if(format == AF_FORMAT_AC3)
+    if(AF_FORMAT_IS_AC3(format))
     {
         wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
         wformat.Format.wBitsPerSample  = 16;