Mercurial > sdl-ios-xcode
annotate test/loopwave.c @ 942:41a59de7f2ed
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
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Sat, 21 Aug 2004 12:27:02 +0000 |
| parents | 74212992fb08 |
| children | be9c9c8f6d53 |
| rev | line source |
|---|---|
| 0 | 1 |
| 2 /* Program to load a wave file and loop playing it using SDL sound */ | |
| 3 | |
| 4 /* loopwaves.c is much more robust in handling WAVE files -- | |
| 5 This is only for simple WAVEs | |
| 6 */ | |
| 7 | |
| 8 #include <stdio.h> | |
| 9 #include <stdlib.h> | |
| 10 #include <signal.h> | |
| 11 | |
| 12 #include "SDL.h" | |
| 13 #include "SDL_audio.h" | |
| 14 | |
| 15 struct { | |
| 16 SDL_AudioSpec spec; | |
| 17 Uint8 *sound; /* Pointer to wave data */ | |
| 18 Uint32 soundlen; /* Length of wave data */ | |
| 19 int soundpos; /* Current play position */ | |
| 20 } wave; | |
| 21 | |
| 22 void fillerup(void *unused, Uint8 *stream, int len) | |
| 23 { | |
| 24 Uint8 *waveptr; | |
| 25 int waveleft; | |
| 26 | |
| 27 /* Set up the pointers */ | |
| 28 waveptr = wave.sound + wave.soundpos; | |
| 29 waveleft = wave.soundlen - wave.soundpos; | |
| 30 | |
| 31 /* Go! */ | |
| 32 while ( waveleft <= len ) { | |
| 33 SDL_MixAudio(stream, waveptr, waveleft, SDL_MIX_MAXVOLUME); | |
| 34 stream += waveleft; | |
| 35 len -= waveleft; | |
| 36 waveptr = wave.sound; | |
| 37 waveleft = wave.soundlen; | |
| 38 wave.soundpos = 0; | |
| 39 } | |
| 40 SDL_MixAudio(stream, waveptr, len, SDL_MIX_MAXVOLUME); | |
| 41 wave.soundpos += len; | |
| 42 } | |
| 43 | |
| 44 static int done = 0; | |
| 45 void poked(int sig) | |
| 46 { | |
| 47 done = 1; | |
| 48 } | |
| 49 | |
| 50 int main(int argc, char *argv[]) | |
| 51 { | |
| 52 char name[32]; | |
| 53 | |
| 54 /* Load the SDL library */ | |
| 55 if ( SDL_Init(SDL_INIT_AUDIO) < 0 ) { | |
| 56 fprintf(stderr, "Couldn't initialize SDL: %s\n",SDL_GetError()); | |
| 57 exit(1); | |
| 58 } | |
| 59 atexit(SDL_Quit); | |
| 60 | |
| 61 if ( argv[1] == NULL ) { | |
| 62 fprintf(stderr, "Usage: %s <wavefile>\n", argv[0]); | |
| 63 exit(1); | |
| 64 } | |
| 65 | |
| 66 /* Load the wave file into memory */ | |
| 67 if ( SDL_LoadWAV(argv[1], | |
| 68 &wave.spec, &wave.sound, &wave.soundlen) == NULL ) { | |
| 69 fprintf(stderr, "Couldn't load %s: %s\n", | |
| 70 argv[1], SDL_GetError()); | |
| 71 exit(1); | |
| 72 } | |
| 73 wave.spec.callback = fillerup; | |
| 74 | |
| 75 /* Set the signals */ | |
| 76 #ifdef SIGHUP | |
| 77 signal(SIGHUP, poked); | |
| 78 #endif | |
| 79 signal(SIGINT, poked); | |
| 80 #ifdef SIGQUIT | |
| 81 signal(SIGQUIT, poked); | |
| 82 #endif | |
| 83 signal(SIGTERM, poked); | |
| 84 | |
| 85 /* Initialize fillerup() variables */ | |
| 86 if ( SDL_OpenAudio(&wave.spec, NULL) < 0 ) { | |
| 87 fprintf(stderr, "Couldn't open audio: %s\n", SDL_GetError()); | |
| 88 SDL_FreeWAV(wave.sound); | |
| 89 exit(2); | |
| 90 } | |
| 91 SDL_PauseAudio(0); | |
| 92 | |
| 93 /* Let the audio run */ | |
| 94 printf("Using audio driver: %s\n", SDL_AudioDriverName(name, 32)); | |
| 95 while ( ! done && (SDL_GetAudioStatus() == SDL_AUDIO_PLAYING) ) | |
| 96 SDL_Delay(1000); | |
| 97 | |
| 98 /* Clean up on signal */ | |
| 99 SDL_CloseAudio(); | |
| 100 SDL_FreeWAV(wave.sound); | |
| 101 return(0); | |
| 102 } |