Here are patches for SDL12 and SDL_mixer for 4 or 6 channel
authorSam Lantinga <slouken@libsdl.org>
Sat, 21 Aug 2004 12:27:02 +0000
changeset 942 41a59de7f2ed
parent 941 5095c4a264aa
child 943 715c32d8f26c
Here are patches for SDL12 and SDL_mixer for 4 or 6 channel surround sound on Linux using the Alsa driver. To use them, naturally you need a sound card that will do 4 or 6 channels and probably also a recent version of the Alsa drivers and library. Since the only SDL output driver that knows about surround sound is the Alsa driver, you���ll want to choose it, using: export SDL_AUDIODRIVER=alsa There are no syntactic changes to the programming API. No new library calls, no differences in arguments. There are two semantic changes: (1) For library calls with number of channels as an argument, formerly you could use only 1 or 2 for the number of channels. Now you can also use 4 or 6. (2) The two "left" and "right" arguments to Mix_SetPanning, for the case of 4 or 6 channels, no longer simply control the volumes of the left and right channels. Now the "left" argument is converted to an angle and Mix_SetPosition is called, and the "right" argu- ment is ignored. With two exceptions, so far as I know, the modified SDL12 and SDL_mixer work the same way as the original versions, when opened for 1 or 2 channel output. The two exceptions are bugs which I fixed. Well, the first, anyway, is a bug for sure. When rate conversions up or down by a factor of two are applied (in src/audio/SDL_audiocvt.c), streams with different numbers of channels (that is, mono and stereo) are treated the same way: either each sample is copied or every other sample is omitted. This is ok for mono, but for stereo, it is frames that should be copied or omitted, where by "frame" I mean a portion of the stream containing one sample for each channel. (In the SDL source, confusingly, sometimes frames are called "samples".) So for these rate conversions, stereo streams have to be treated differently, and they are, in my modified version. The other problem that might be characterized as a bug arises when SDL_mixer is passed a multichannel chunk which does not have an integral number of frames. Due to the way the effect_position code loops over frames, when the chunk ends with a partial frame, memory outside the chunk buffer will be accessed. In the case of stereo, it���s possible that because malloc may give more memory than requested, this potential problem never actually causes a segment fault. I don���t know. For 6 channel chunks, I do know, and it does cause segment faults. If SDL_mixer is passed defective chunks and this causes a segment fault, arguably, that���s not a bug in SDL_mixer. Still, whether or not it counts as a bug, it���s easy to protect against, so why not? I added code in mixer.c to discard any partial frame at the end of a chunk. Then what about when SDL or SDL_mixer is opened for 4 or 6 chan- nel output? What happens with the parts of the current library designed for stereo? I don���t know whether I���ve covered all the bases, but I���ve tried: (1) For playing 2 channel waves, or other cases where SDL knows it has to match up a 2 channel source with a 4 or 6 channel output, I���ve added code in SDL_audiocvt.c to make the necessary conversions. (2) For playing midis using timidity, I���ve converted timidity to do 4 or 6 channel output, upon request. (3) For playing mods using mikmod, I put ad hoc code in music.c to convert the stereo output that mikmod produces to 4 or 6 chan- nels. Obviously it would be better to change the mikmod code to mix down into 4 or 6 channels, but I have a hard time following the code in mikmod, so I didn���t do that. (4) For playing mp3s, I put ad hoc code in smpeg to copy channels in the case when 4 or 6 channel output is needed. (5) There seems to be no problem with .ogg files - stereo .oggs can be up converted as .wavs are. (6) The effect_position code in SDL_mixer is now generalized to in- clude the cases of 4 and 6 channel streams. I���ve done a very limited amount of compatibility testing for some of the games using SDL I happen to have. For details, see the file TESTS. I���ve put into a separate archive, Surround-SDL-testfiles.tgz, a couple of 6 channel wave files for testing and a 6 channel ogg file. If you have the right hardware and version of Alsa, you should be able to play the wave files with the Alsa utility aplay (and hear all channels, except maybe lfe, for chan-id.wav, since it���s rather faint). Don���t expect aplay to give good sound, though. There���s something wrong with the current version of aplay. The canyon.ogg file is to test loading of 6 channel oggs. After patching and compiling, you can play it with playmus. (My version of ogg123 will not play it, and I had to patch mplayer to get it to play 6 channel oggs.) Greg Lee <greg@ling.lll.hawaii.edu> Thus, July 1, 2004
src/audio/SDL_audio.c
src/audio/SDL_audiocvt.c
src/audio/alsa/SDL_alsa_audio.c
--- a/src/audio/SDL_audio.c	Sat Aug 21 11:34:24 2004 +0000
+++ b/src/audio/SDL_audio.c	Sat Aug 21 12:27:02 2004 +0000
@@ -391,6 +391,8 @@
 	switch ( desired->channels ) {
 	    case 1:	/* Mono */
 	    case 2:	/* Stereo */
+	    case 4:	/* surround */
+	    case 6:	/* surround with center and lfe */
 		break;
 	    default:
 		SDL_SetError("1 (mono) and 2 (stereo) channels supported");
--- a/src/audio/SDL_audiocvt.c	Sat Aug 21 11:34:24 2004 +0000
+++ b/src/audio/SDL_audiocvt.c	Sat Aug 21 12:27:02 2004 +0000
@@ -175,6 +175,244 @@
 	}
 }
 
+/* Discard top 4 channels */
+void SDL_ConvertStrip(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Sint32 lsample, rsample;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting down to stereo\n");
+#endif
+	switch (format&0x8018) {
+
+		case AUDIO_U8: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			for ( i=cvt->len_cvt/6; i; --i ) {
+				lsample = src[0];
+				rsample = src[1];
+				dst[0] = lsample;
+				dst[1] = rsample;
+				src += 6;
+				dst += 2;
+			}
+		}
+		break;
+
+		case AUDIO_S8: {
+			Sint8 *src, *dst;
+
+			src = (Sint8 *)cvt->buf;
+			dst = (Sint8 *)cvt->buf;
+			for ( i=cvt->len_cvt/6; i; --i ) {
+				lsample = src[0];
+				rsample = src[1];
+				dst[0] = lsample;
+				dst[1] = rsample;
+				src += 6;
+				dst += 2;
+			}
+		}
+		break;
+
+		case AUDIO_U16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/12; i; --i ) {
+					lsample = (Uint16)((src[0]<<8)|src[1]);
+					rsample = (Uint16)((src[2]<<8)|src[3]);
+						dst[1] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[0] = (lsample&0xFF);
+						dst[3] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[2] = (rsample&0xFF);
+					src += 12;
+					dst += 4;
+				}
+			} else {
+				for ( i=cvt->len_cvt/12; i; --i ) {
+					lsample = (Uint16)((src[1]<<8)|src[0]);
+					rsample = (Uint16)((src[3]<<8)|src[2]);
+						dst[0] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[1] = (lsample&0xFF);
+						dst[2] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[3] = (rsample&0xFF);
+					src += 12;
+					dst += 4;
+				}
+			}
+		}
+		break;
+
+		case AUDIO_S16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/12; i; --i ) {
+					lsample = (Sint16)((src[0]<<8)|src[1]);
+					rsample = (Sint16)((src[2]<<8)|src[3]);
+						dst[1] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[0] = (lsample&0xFF);
+						dst[3] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[2] = (rsample&0xFF);
+					src += 12;
+					dst += 4;
+				}
+			} else {
+				for ( i=cvt->len_cvt/12; i; --i ) {
+					lsample = (Sint16)((src[1]<<8)|src[0]);
+					rsample = (Sint16)((src[3]<<8)|src[2]);
+						dst[0] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[1] = (lsample&0xFF);
+						dst[2] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[3] = (rsample&0xFF);
+					src += 12;
+					dst += 4;
+				}
+			}
+		}
+		break;
+	}
+	cvt->len_cvt /= 3;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Discard top 2 channels of 6 */
+void SDL_ConvertStrip_2(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Sint32 lsample, rsample;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting 6 down to quad\n");
+#endif
+	switch (format&0x8018) {
+
+		case AUDIO_U8: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				lsample = src[0];
+				rsample = src[1];
+				dst[0] = lsample;
+				dst[1] = rsample;
+				src += 4;
+				dst += 2;
+			}
+		}
+		break;
+
+		case AUDIO_S8: {
+			Sint8 *src, *dst;
+
+			src = (Sint8 *)cvt->buf;
+			dst = (Sint8 *)cvt->buf;
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				lsample = src[0];
+				rsample = src[1];
+				dst[0] = lsample;
+				dst[1] = rsample;
+				src += 4;
+				dst += 2;
+			}
+		}
+		break;
+
+		case AUDIO_U16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/8; i; --i ) {
+					lsample = (Uint16)((src[0]<<8)|src[1]);
+					rsample = (Uint16)((src[2]<<8)|src[3]);
+						dst[1] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[0] = (lsample&0xFF);
+						dst[3] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[2] = (rsample&0xFF);
+					src += 8;
+					dst += 4;
+				}
+			} else {
+				for ( i=cvt->len_cvt/8; i; --i ) {
+					lsample = (Uint16)((src[1]<<8)|src[0]);
+					rsample = (Uint16)((src[3]<<8)|src[2]);
+						dst[0] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[1] = (lsample&0xFF);
+						dst[2] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[3] = (rsample&0xFF);
+					src += 8;
+					dst += 4;
+				}
+			}
+		}
+		break;
+
+		case AUDIO_S16: {
+			Uint8 *src, *dst;
+
+			src = cvt->buf;
+			dst = cvt->buf;
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/8; i; --i ) {
+					lsample = (Sint16)((src[0]<<8)|src[1]);
+					rsample = (Sint16)((src[2]<<8)|src[3]);
+						dst[1] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[0] = (lsample&0xFF);
+						dst[3] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[2] = (rsample&0xFF);
+					src += 8;
+					dst += 4;
+				}
+			} else {
+				for ( i=cvt->len_cvt/8; i; --i ) {
+					lsample = (Sint16)((src[1]<<8)|src[0]);
+					rsample = (Sint16)((src[3]<<8)|src[2]);
+						dst[0] = (lsample&0xFF);
+						lsample >>= 8;
+						dst[1] = (lsample&0xFF);
+						dst[2] = (rsample&0xFF);
+						rsample >>= 8;
+						dst[3] = (rsample&0xFF);
+					src += 8;
+					dst += 4;
+				}
+			}
+		}
+		break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
 
 /* Duplicate a mono channel to both stereo channels */
 void SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format)
@@ -213,6 +451,337 @@
 	}
 }
 
+
+/* Duplicate a stereo channel to a pseudo-5.1 stream */
+void SDL_ConvertSurround(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting stereo to surround\n");
+#endif
+	switch (format&0x8018) {
+
+		case AUDIO_U8: {
+			Uint8 *src, *dst, lf, rf, ce;
+
+			src = (Uint8 *)(cvt->buf+cvt->len_cvt);
+			dst = (Uint8 *)(cvt->buf+cvt->len_cvt*3);
+			for ( i=cvt->len_cvt; i; --i ) {
+				dst -= 6;
+				src -= 2;
+				lf = src[0];
+				rf = src[1];
+				ce = (lf/2) + (rf/2);
+				dst[0] = lf;
+				dst[1] = rf;
+				dst[2] = lf - ce;
+				dst[3] = rf - ce;
+				dst[4] = ce;
+				dst[5] = ce;
+			}
+		}
+		break;
+
+		case AUDIO_S8: {
+			Sint8 *src, *dst, lf, rf, ce;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*3;
+			for ( i=cvt->len_cvt; i; --i ) {
+				dst -= 6;
+				src -= 2;
+				lf = src[0];
+				rf = src[1];
+				ce = (lf/2) + (rf/2);
+				dst[0] = lf;
+				dst[1] = rf;
+				dst[2] = lf - ce;
+				dst[3] = rf - ce;
+				dst[4] = ce;
+				dst[5] = ce;
+			}
+		}
+		break;
+
+		case AUDIO_U16: {
+			Uint8 *src, *dst;
+			Uint16 lf, rf, ce, lr, rr;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*3;
+
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 12;
+					src -= 4;
+					lf = (Uint16)((src[0]<<8)|src[1]);
+					rf = (Uint16)((src[2]<<8)|src[3]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[1] = (lf&0xFF);
+						dst[0] = ((lf>>8)&0xFF);
+						dst[3] = (rf&0xFF);
+						dst[2] = ((rf>>8)&0xFF);
+
+						dst[1+4] = (lr&0xFF);
+						dst[0+4] = ((lr>>8)&0xFF);
+						dst[3+4] = (rr&0xFF);
+						dst[2+4] = ((rr>>8)&0xFF);
+
+						dst[1+8] = (ce&0xFF);
+						dst[0+8] = ((ce>>8)&0xFF);
+						dst[3+8] = (ce&0xFF);
+						dst[2+8] = ((ce>>8)&0xFF);
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 12;
+					src -= 4;
+					lf = (Uint16)((src[1]<<8)|src[0]);
+					rf = (Uint16)((src[3]<<8)|src[2]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[0] = (lf&0xFF);
+						dst[1] = ((lf>>8)&0xFF);
+						dst[2] = (rf&0xFF);
+						dst[3] = ((rf>>8)&0xFF);
+
+						dst[0+4] = (lr&0xFF);
+						dst[1+4] = ((lr>>8)&0xFF);
+						dst[2+4] = (rr&0xFF);
+						dst[3+4] = ((rr>>8)&0xFF);
+
+						dst[0+8] = (ce&0xFF);
+						dst[1+8] = ((ce>>8)&0xFF);
+						dst[2+8] = (ce&0xFF);
+						dst[3+8] = ((ce>>8)&0xFF);
+				}
+			}
+		}
+		break;
+
+		case AUDIO_S16: {
+			Uint8 *src, *dst;
+			Sint16 lf, rf, ce, lr, rr;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*3;
+
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 12;
+					src -= 4;
+					lf = (Sint16)((src[0]<<8)|src[1]);
+					rf = (Sint16)((src[2]<<8)|src[3]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[1] = (lf&0xFF);
+						dst[0] = ((lf>>8)&0xFF);
+						dst[3] = (rf&0xFF);
+						dst[2] = ((rf>>8)&0xFF);
+
+						dst[1+4] = (lr&0xFF);
+						dst[0+4] = ((lr>>8)&0xFF);
+						dst[3+4] = (rr&0xFF);
+						dst[2+4] = ((rr>>8)&0xFF);
+
+						dst[1+8] = (ce&0xFF);
+						dst[0+8] = ((ce>>8)&0xFF);
+						dst[3+8] = (ce&0xFF);
+						dst[2+8] = ((ce>>8)&0xFF);
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 12;
+					src -= 4;
+					lf = (Sint16)((src[1]<<8)|src[0]);
+					rf = (Sint16)((src[3]<<8)|src[2]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[0] = (lf&0xFF);
+						dst[1] = ((lf>>8)&0xFF);
+						dst[2] = (rf&0xFF);
+						dst[3] = ((rf>>8)&0xFF);
+
+						dst[0+4] = (lr&0xFF);
+						dst[1+4] = ((lr>>8)&0xFF);
+						dst[2+4] = (rr&0xFF);
+						dst[3+4] = ((rr>>8)&0xFF);
+
+						dst[0+8] = (ce&0xFF);
+						dst[1+8] = ((ce>>8)&0xFF);
+						dst[2+8] = (ce&0xFF);
+						dst[3+8] = ((ce>>8)&0xFF);
+				}
+			}
+		}
+		break;
+	}
+	cvt->len_cvt *= 3;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Duplicate a stereo channel to a pseudo-4.0 stream */
+void SDL_ConvertSurround_4(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting stereo to quad\n");
+#endif
+	switch (format&0x8018) {
+
+		case AUDIO_U8: {
+			Uint8 *src, *dst, lf, rf, ce;
+
+			src = (Uint8 *)(cvt->buf+cvt->len_cvt);
+			dst = (Uint8 *)(cvt->buf+cvt->len_cvt*2);
+			for ( i=cvt->len_cvt; i; --i ) {
+				dst -= 4;
+				src -= 2;
+				lf = src[0];
+				rf = src[1];
+				ce = (lf/2) + (rf/2);
+				dst[0] = lf;
+				dst[1] = rf;
+				dst[2] = lf - ce;
+				dst[3] = rf - ce;
+			}
+		}
+		break;
+
+		case AUDIO_S8: {
+			Sint8 *src, *dst, lf, rf, ce;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*2;
+			for ( i=cvt->len_cvt; i; --i ) {
+				dst -= 4;
+				src -= 2;
+				lf = src[0];
+				rf = src[1];
+				ce = (lf/2) + (rf/2);
+				dst[0] = lf;
+				dst[1] = rf;
+				dst[2] = lf - ce;
+				dst[3] = rf - ce;
+			}
+		}
+		break;
+
+		case AUDIO_U16: {
+			Uint8 *src, *dst;
+			Uint16 lf, rf, ce, lr, rr;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*2;
+
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 8;
+					src -= 4;
+					lf = (Uint16)((src[0]<<8)|src[1]);
+					rf = (Uint16)((src[2]<<8)|src[3]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[1] = (lf&0xFF);
+						dst[0] = ((lf>>8)&0xFF);
+						dst[3] = (rf&0xFF);
+						dst[2] = ((rf>>8)&0xFF);
+
+						dst[1+4] = (lr&0xFF);
+						dst[0+4] = ((lr>>8)&0xFF);
+						dst[3+4] = (rr&0xFF);
+						dst[2+4] = ((rr>>8)&0xFF);
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 8;
+					src -= 4;
+					lf = (Uint16)((src[1]<<8)|src[0]);
+					rf = (Uint16)((src[3]<<8)|src[2]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[0] = (lf&0xFF);
+						dst[1] = ((lf>>8)&0xFF);
+						dst[2] = (rf&0xFF);
+						dst[3] = ((rf>>8)&0xFF);
+
+						dst[0+4] = (lr&0xFF);
+						dst[1+4] = ((lr>>8)&0xFF);
+						dst[2+4] = (rr&0xFF);
+						dst[3+4] = ((rr>>8)&0xFF);
+				}
+			}
+		}
+		break;
+
+		case AUDIO_S16: {
+			Uint8 *src, *dst;
+			Sint16 lf, rf, ce, lr, rr;
+
+			src = cvt->buf+cvt->len_cvt;
+			dst = cvt->buf+cvt->len_cvt*2;
+
+			if ( (format & 0x1000) == 0x1000 ) {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 8;
+					src -= 4;
+					lf = (Sint16)((src[0]<<8)|src[1]);
+					rf = (Sint16)((src[2]<<8)|src[3]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[1] = (lf&0xFF);
+						dst[0] = ((lf>>8)&0xFF);
+						dst[3] = (rf&0xFF);
+						dst[2] = ((rf>>8)&0xFF);
+
+						dst[1+4] = (lr&0xFF);
+						dst[0+4] = ((lr>>8)&0xFF);
+						dst[3+4] = (rr&0xFF);
+						dst[2+4] = ((rr>>8)&0xFF);
+				}
+			} else {
+				for ( i=cvt->len_cvt/4; i; --i ) {
+					dst -= 8;
+					src -= 4;
+					lf = (Sint16)((src[1]<<8)|src[0]);
+					rf = (Sint16)((src[3]<<8)|src[2]);
+					ce = (lf/2) + (rf/2);
+					rr = lf - ce;
+					lr = rf - ce;
+						dst[0] = (lf&0xFF);
+						dst[1] = ((lf>>8)&0xFF);
+						dst[2] = (rf&0xFF);
+						dst[3] = ((rf>>8)&0xFF);
+
+						dst[0+4] = (lr&0xFF);
+						dst[1+4] = ((lr>>8)&0xFF);
+						dst[2+4] = (rr&0xFF);
+						dst[3+4] = ((rr>>8)&0xFF);
+				}
+			}
+		}
+		break;
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
 /* Convert 8-bit to 16-bit - LSB */
 void SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format)
 {
@@ -374,6 +943,173 @@
 	}
 }
 
+
+/* Convert rate up by multiple of 2, for stereo */
+void SDL_RateMUL2_c2(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate * 2\n");
+#endif
+	src = cvt->buf+cvt->len_cvt;
+	dst = cvt->buf+cvt->len_cvt*2;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/2; i; --i ) {
+				src -= 2;
+				dst -= 4;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[0];
+				dst[3] = src[1];
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				src -= 4;
+				dst -= 8;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[0];
+				dst[5] = src[1];
+				dst[6] = src[2];
+				dst[7] = src[3];
+			}
+			break;
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Convert rate up by multiple of 2, for quad */
+void SDL_RateMUL2_c4(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate * 2\n");
+#endif
+	src = cvt->buf+cvt->len_cvt;
+	dst = cvt->buf+cvt->len_cvt*2;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				src -= 4;
+				dst -= 8;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[0];
+				dst[5] = src[1];
+				dst[6] = src[2];
+				dst[7] = src[3];
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/8; i; --i ) {
+				src -= 8;
+				dst -= 16;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[6];
+				dst[7] = src[7];
+				dst[8] = src[0];
+				dst[9] = src[1];
+				dst[10] = src[2];
+				dst[11] = src[3];
+				dst[12] = src[4];
+				dst[13] = src[5];
+				dst[14] = src[6];
+				dst[15] = src[7];
+			}
+			break;
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Convert rate up by multiple of 2, for 5.1 */
+void SDL_RateMUL2_c6(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate * 2\n");
+#endif
+	src = cvt->buf+cvt->len_cvt;
+	dst = cvt->buf+cvt->len_cvt*2;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/6; i; --i ) {
+				src -= 6;
+				dst -= 12;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[0];
+				dst[7] = src[1];
+				dst[8] = src[2];
+				dst[9] = src[3];
+				dst[10] = src[4];
+				dst[11] = src[5];
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/12; i; --i ) {
+				src -= 12;
+				dst -= 24;
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[6];
+				dst[7] = src[7];
+				dst[8] = src[8];
+				dst[9] = src[9];
+				dst[10] = src[10];
+				dst[11] = src[11];
+				dst[12] = src[0];
+				dst[13] = src[1];
+				dst[14] = src[2];
+				dst[15] = src[3];
+				dst[16] = src[4];
+				dst[17] = src[5];
+				dst[18] = src[6];
+				dst[19] = src[7];
+				dst[20] = src[8];
+				dst[21] = src[9];
+				dst[22] = src[10];
+				dst[23] = src[11];
+			}
+			break;
+	}
+	cvt->len_cvt *= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
 /* Convert rate down by multiple of 2 */
 void SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format)
 {
@@ -408,6 +1144,137 @@
 	}
 }
 
+
+/* Convert rate down by multiple of 2, for stereo */
+void SDL_RateDIV2_c2(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate / 2\n");
+#endif
+	src = cvt->buf;
+	dst = cvt->buf;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/4; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				src += 4;
+				dst += 2;
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/8; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				src += 8;
+				dst += 4;
+			}
+			break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+
+/* Convert rate down by multiple of 2, for quad */
+void SDL_RateDIV2_c4(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate / 2\n");
+#endif
+	src = cvt->buf;
+	dst = cvt->buf;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/8; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				src += 8;
+				dst += 4;
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/16; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[6];
+				dst[7] = src[7];
+				src += 16;
+				dst += 8;
+			}
+			break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
+/* Convert rate down by multiple of 2, for 5.1 */
+void SDL_RateDIV2_c6(SDL_AudioCVT *cvt, Uint16 format)
+{
+	int i;
+	Uint8 *src, *dst;
+
+#ifdef DEBUG_CONVERT
+	fprintf(stderr, "Converting audio rate / 2\n");
+#endif
+	src = cvt->buf;
+	dst = cvt->buf;
+	switch (format & 0xFF) {
+		case 8:
+			for ( i=cvt->len_cvt/12; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				src += 12;
+				dst += 6;
+			}
+			break;
+		case 16:
+			for ( i=cvt->len_cvt/24; i; --i ) {
+				dst[0] = src[0];
+				dst[1] = src[1];
+				dst[2] = src[2];
+				dst[3] = src[3];
+				dst[4] = src[4];
+				dst[5] = src[5];
+				dst[6] = src[6];
+				dst[7] = src[7];
+				dst[8] = src[8];
+				dst[9] = src[9];
+				dst[10] = src[10];
+				dst[11] = src[11];
+				src += 24;
+				dst += 12;
+			}
+			break;
+	}
+	cvt->len_cvt /= 2;
+	if ( cvt->filters[++cvt->filter_index] ) {
+		cvt->filters[cvt->filter_index](cvt, format);
+	}
+}
+
 /* Very slow rate conversion routine */
 void SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format)
 {
@@ -511,6 +1378,8 @@
 	Uint16 src_format, Uint8 src_channels, int src_rate,
 	Uint16 dst_format, Uint8 dst_channels, int dst_rate)
 {
+/*printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
+		src_format, dst_format, src_channels, dst_channels, src_rate, dst_rate);*/
 	/* Start off with no conversion necessary */
 	cvt->needed = 0;
 	cvt->filter_index = 0;
@@ -554,6 +1423,29 @@
 
 	/* Last filter:  Mono/Stereo conversion */
 	if ( src_channels != dst_channels ) {
+		if ( (src_channels == 1) && (dst_channels > 1) ) {
+			cvt->filters[cvt->filter_index++] = 
+						SDL_ConvertStereo;
+			cvt->len_mult *= 2;
+			src_channels = 2;
+			cvt->len_ratio *= 2;
+		}
+		if ( (src_channels == 2) &&
+				(dst_channels == 6) ) {
+			cvt->filters[cvt->filter_index++] =
+						 SDL_ConvertSurround;
+			src_channels = 6;
+			cvt->len_mult *= 3;
+			cvt->len_ratio *= 3;
+		}
+		if ( (src_channels == 2) &&
+				(dst_channels == 4) ) {
+			cvt->filters[cvt->filter_index++] =
+						 SDL_ConvertSurround_4;
+			src_channels = 4;
+			cvt->len_mult *= 2;
+			cvt->len_ratio *= 2;
+		}
 		while ( (src_channels*2) <= dst_channels ) {
 			cvt->filters[cvt->filter_index++] = 
 						SDL_ConvertStereo;
@@ -561,6 +1453,20 @@
 			src_channels *= 2;
 			cvt->len_ratio *= 2;
 		}
+		if ( (src_channels == 6) &&
+				(dst_channels <= 2) ) {
+			cvt->filters[cvt->filter_index++] =
+						 SDL_ConvertStrip;
+			src_channels = 2;
+			cvt->len_ratio /= 3;
+		}
+		if ( (src_channels == 6) &&
+				(dst_channels == 4) ) {
+			cvt->filters[cvt->filter_index++] =
+						 SDL_ConvertStrip_2;
+			src_channels = 4;
+			cvt->len_ratio /= 2;
+		}
 		/* This assumes that 4 channel audio is in the format:
 		     Left {front/back} + Right {front/back}
 		   so converting to L/R stereo works properly.
@@ -588,13 +1494,25 @@
 		if ( src_rate > dst_rate ) {
 			hi_rate = src_rate;
 			lo_rate = dst_rate;
-			rate_cvt = SDL_RateDIV2;
+			switch (src_channels) {
+				case 1: rate_cvt = SDL_RateDIV2; break;
+				case 2: rate_cvt = SDL_RateDIV2_c2; break;
+				case 4: rate_cvt = SDL_RateDIV2_c4; break;
+				case 6: rate_cvt = SDL_RateDIV2_c6; break;
+				default: return -1;
+			}
 			len_mult = 1;
 			len_ratio = 0.5;
 		} else {
 			hi_rate = dst_rate;
 			lo_rate = src_rate;
-			rate_cvt = SDL_RateMUL2;
+			switch (src_channels) {
+				case 1: rate_cvt = SDL_RateMUL2; break;
+				case 2: rate_cvt = SDL_RateMUL2_c2; break;
+				case 4: rate_cvt = SDL_RateMUL2_c4; break;
+				case 6: rate_cvt = SDL_RateMUL2_c6; break;
+				default: return -1;
+			}
 			len_mult = 2;
 			len_ratio = 2.0;
 		}
--- a/src/audio/alsa/SDL_alsa_audio.c	Sat Aug 21 11:34:24 2004 +0000
+++ b/src/audio/alsa/SDL_alsa_audio.c	Sat Aug 21 12:27:02 2004 +0000
@@ -163,13 +163,15 @@
 
 #endif /* ALSA_DYNAMIC */
 
-static const char *get_audio_device()
+static const char *get_audio_device(int channels)
 {
 	const char *device;
 	
 	device = getenv("AUDIODEV");	/* Is there a standard variable name? */
 	if ( device == NULL ) {
-		device = DEFAULT_DEVICE;
+		if (channels == 6) device = "surround51";
+		else if (channels == 4) device = "surround40";
+		else device = DEFAULT_DEVICE;
 	}
 	return device;
 }
@@ -186,7 +188,7 @@
 	if (LoadALSALibrary() < 0) {
 		return available;
 	}
-	status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+	status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
 	if ( status >= 0 ) {
 		available = 1;
         	SDL_NAME(snd_pcm_close)(handle);
@@ -319,7 +321,9 @@
 	Uint16               test_format;
 
 	/* Open the audio device */
-	status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+	/* Name of device should depend on # channels in spec */
+	status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+
 	if ( status < 0 ) {
 		SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
 		return(-1);