src/audio/alsa/SDL_alsa_audio.c
author Sam Lantinga <slouken@libsdl.org>
Sat, 21 Aug 2004 12:27:02 +0000
changeset 942 41a59de7f2ed
parent 939 c7c04f811994
child 1161 05d4b93b911e
permissions -rw-r--r--
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

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2004 Sam Lantinga

    This library is free software; you can redistribute it and/or
    modify it under the terms of the GNU Library General Public
    License as published by the Free Software Foundation; either
    version 2 of the License, or (at your option) any later version.

    This library is distributed in the hope that it will be useful,
    but WITHOUT ANY WARRANTY; without even the implied warranty of
    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
    Library General Public License for more details.

    You should have received a copy of the GNU Library General Public
    License along with this library; if not, write to the Free
    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

    Sam Lantinga
    slouken@libsdl.org
*/



/* Allow access to a raw mixing buffer */

#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/types.h>
#include <sys/time.h>

#include "SDL_audio.h"
#include "SDL_error.h"
#include "SDL_audiomem.h"
#include "SDL_audio_c.h"
#include "SDL_timer.h"
#include "SDL_alsa_audio.h"

#ifdef ALSA_DYNAMIC
#ifdef USE_DLVSYM
#define __USE_GNU
#endif
#include <dlfcn.h>
#include "SDL_name.h"
#include "SDL_loadso.h"
#else
#define SDL_NAME(X)	X
#endif


/* The tag name used by ALSA audio */
#define DRIVER_NAME         "alsa"

/* The default ALSA audio driver */
#define DEFAULT_DEVICE	"default"

/* Audio driver functions */
static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec);
static void ALSA_WaitAudio(_THIS);
static void ALSA_PlayAudio(_THIS);
static Uint8 *ALSA_GetAudioBuf(_THIS);
static void ALSA_CloseAudio(_THIS);

#ifdef ALSA_DYNAMIC

static const char *alsa_library = ALSA_DYNAMIC;
static void *alsa_handle = NULL;
static int alsa_loaded = 0;

static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm);
static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm);
static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm);
static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm);
static const char *(*SDL_NAME(snd_strerror))(int errnum);
static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void);
static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access);
static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);
static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);
static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params);
static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);
static snd_pcm_uframes_t (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int *dir);
static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);
static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock);
#define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof)

static struct {
	const char *name;
	void **func;
} alsa_functions[] = {
	{ "snd_pcm_open",	(void**)&SDL_NAME(snd_pcm_open)		},
	{ "snd_pcm_close",	(void**)&SDL_NAME(snd_pcm_close)	},
	{ "snd_pcm_writei",	(void**)&SDL_NAME(snd_pcm_writei)	},
	{ "snd_pcm_resume",	(void**)&SDL_NAME(snd_pcm_resume)	},
	{ "snd_pcm_prepare",	(void**)&SDL_NAME(snd_pcm_prepare)	},
	{ "snd_pcm_drain",	(void**)&SDL_NAME(snd_pcm_drain)	},
	{ "snd_strerror",	(void**)&SDL_NAME(snd_strerror)		},
	{ "snd_pcm_hw_params_sizeof",		(void**)&SDL_NAME(snd_pcm_hw_params_sizeof)		},
	{ "snd_pcm_hw_params_any",		(void**)&SDL_NAME(snd_pcm_hw_params_any)		},
	{ "snd_pcm_hw_params_set_access",	(void**)&SDL_NAME(snd_pcm_hw_params_set_access)		},
	{ "snd_pcm_hw_params_set_format",	(void**)&SDL_NAME(snd_pcm_hw_params_set_format)		},
	{ "snd_pcm_hw_params_set_channels",	(void**)&SDL_NAME(snd_pcm_hw_params_set_channels)	},
	{ "snd_pcm_hw_params_get_channels",	(void**)&SDL_NAME(snd_pcm_hw_params_get_channels)	},
	{ "snd_pcm_hw_params_set_rate_near",	(void**)&SDL_NAME(snd_pcm_hw_params_set_rate_near)	},
	{ "snd_pcm_hw_params_set_period_size_near",	(void**)&SDL_NAME(snd_pcm_hw_params_set_period_size_near)	},
	{ "snd_pcm_hw_params_set_periods_near",	(void**)&SDL_NAME(snd_pcm_hw_params_set_periods_near)	},
	{ "snd_pcm_hw_params",	(void**)&SDL_NAME(snd_pcm_hw_params)	},
	{ "snd_pcm_nonblock",	(void**)&SDL_NAME(snd_pcm_nonblock)	},
};

static void UnloadALSALibrary(void) {
	if (alsa_loaded) {
/*		SDL_UnloadObject(alsa_handle);*/
		dlclose(alsa_handle);
		alsa_handle = NULL;
		alsa_loaded = 0;
	}
}

static int LoadALSALibrary(void) {
	int i, retval = -1;

/*	alsa_handle = SDL_LoadObject(alsa_library);*/
	alsa_handle = dlopen(alsa_library,RTLD_NOW);
	if (alsa_handle) {
		alsa_loaded = 1;
		retval = 0;
		for (i = 0; i < SDL_TABLESIZE(alsa_functions); i++) {
/*			*alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/
#ifdef USE_DLVSYM
			*alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9");
			if (!*alsa_functions[i].func)
#endif
				*alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name);
			if (!*alsa_functions[i].func) {
				retval = -1;
				UnloadALSALibrary();
				break;
			}
		}
	}
	return retval;
}

#else

static void UnloadALSALibrary(void) {
	return;
}

static int LoadALSALibrary(void) {
	return 0;
}

#endif /* ALSA_DYNAMIC */

static const char *get_audio_device(int channels)
{
	const char *device;
	
	device = getenv("AUDIODEV");	/* Is there a standard variable name? */
	if ( device == NULL ) {
		if (channels == 6) device = "surround51";
		else if (channels == 4) device = "surround40";
		else device = DEFAULT_DEVICE;
	}
	return device;
}

/* Audio driver bootstrap functions */

static int Audio_Available(void)
{
	int available;
	int status;
	snd_pcm_t *handle;

	available = 0;
	if (LoadALSALibrary() < 0) {
		return available;
	}
	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);
	}
	UnloadALSALibrary();
	return(available);
}

static void Audio_DeleteDevice(SDL_AudioDevice *device)
{
	free(device->hidden);
	free(device);
	UnloadALSALibrary();
}

static SDL_AudioDevice *Audio_CreateDevice(int devindex)
{
	SDL_AudioDevice *this;

	/* Initialize all variables that we clean on shutdown */
	LoadALSALibrary();
	this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
	if ( this ) {
		memset(this, 0, (sizeof *this));
		this->hidden = (struct SDL_PrivateAudioData *)
				malloc((sizeof *this->hidden));
	}
	if ( (this == NULL) || (this->hidden == NULL) ) {
		SDL_OutOfMemory();
		if ( this ) {
			free(this);
		}
		return(0);
	}
	memset(this->hidden, 0, (sizeof *this->hidden));

	/* Set the function pointers */
	this->OpenAudio = ALSA_OpenAudio;
	this->WaitAudio = ALSA_WaitAudio;
	this->PlayAudio = ALSA_PlayAudio;
	this->GetAudioBuf = ALSA_GetAudioBuf;
	this->CloseAudio = ALSA_CloseAudio;

	this->free = Audio_DeleteDevice;

	return this;
}

AudioBootStrap ALSA_bootstrap = {
	DRIVER_NAME, "ALSA 0.9 PCM audio",
	Audio_Available, Audio_CreateDevice
};

/* This function waits until it is possible to write a full sound buffer */
static void ALSA_WaitAudio(_THIS)
{
	/* Check to see if the thread-parent process is still alive */
	{ static int cnt = 0;
		/* Note that this only works with thread implementations 
		   that use a different process id for each thread.
		*/
		if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
			if ( kill(parent, 0) < 0 ) {
				this->enabled = 0;
			}
		}
	}
}

static void ALSA_PlayAudio(_THIS)
{
	int           status;
	int           sample_len;
	signed short *sample_buf;

	sample_len = this->spec.samples;
	sample_buf = (signed short *)mixbuf;
	while ( sample_len > 0 ) {
		status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len);
		if ( status < 0 ) {
			if ( status == -EAGAIN ) {
				SDL_Delay(1);
				continue;
			}
			if ( status == -ESTRPIPE ) {
				do {
					SDL_Delay(1);
					status = SDL_NAME(snd_pcm_resume)(pcm_handle);
				} while ( status == -EAGAIN );
			}
			if ( status < 0 ) {
				status = SDL_NAME(snd_pcm_prepare)(pcm_handle);
			}
			if ( status < 0 ) {
				/* Hmm, not much we can do - abort */
				this->enabled = 0;
				return;
			}
			continue;
		}
		sample_buf += status * this->spec.channels;
		sample_len -= status;
	}
}

static Uint8 *ALSA_GetAudioBuf(_THIS)
{
	return(mixbuf);
}

static void ALSA_CloseAudio(_THIS)
{
	if ( mixbuf != NULL ) {
		SDL_FreeAudioMem(mixbuf);
		mixbuf = NULL;
	}
	if ( pcm_handle ) {
		SDL_NAME(snd_pcm_drain)(pcm_handle);
		SDL_NAME(snd_pcm_close)(pcm_handle);
		pcm_handle = NULL;
	}
}

static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
	int                  status;
	snd_pcm_hw_params_t *params;
	snd_pcm_format_t     format;
	snd_pcm_uframes_t    frames;
	Uint16               test_format;

	/* Open the audio device */
	/* 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);
	}

	/* Figure out what the hardware is capable of */
	snd_pcm_hw_params_alloca(&params);
	status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params);
	if ( status < 0 ) {
		SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	/* SDL only uses interleaved sample output */
	status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	/* Try for a closest match on audio format */
	status = -1;
	for ( test_format = SDL_FirstAudioFormat(spec->format);
	      test_format && (status < 0); ) {
		switch ( test_format ) {
			case AUDIO_U8:
				format = SND_PCM_FORMAT_U8;
				break;
			case AUDIO_S8:
				format = SND_PCM_FORMAT_S8;
				break;
			case AUDIO_S16LSB:
				format = SND_PCM_FORMAT_S16_LE;
				break;
			case AUDIO_S16MSB:
				format = SND_PCM_FORMAT_S16_BE;
				break;
			case AUDIO_U16LSB:
				format = SND_PCM_FORMAT_U16_LE;
				break;
			case AUDIO_U16MSB:
				format = SND_PCM_FORMAT_U16_BE;
				break;
			default:
				format = 0;
				break;
		}
		if ( format != 0 ) {
			status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format);
		}
		if ( status < 0 ) {
			test_format = SDL_NextAudioFormat();
		}
	}
	if ( status < 0 ) {
		SDL_SetError("Couldn't find any hardware audio formats");
		ALSA_CloseAudio(this);
		return(-1);
	}
	spec->format = test_format;

	/* Set the number of channels */
	status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels);
	if ( status < 0 ) {
		status = SDL_NAME(snd_pcm_hw_params_get_channels)(params);
		if ( (status <= 0) || (status > 2) ) {
			SDL_SetError("Couldn't set audio channels");
			ALSA_CloseAudio(this);
			return(-1);
		}
		spec->channels = status;
	}

	/* Set the audio rate */
	status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}
	spec->freq = status;

	/* Set the buffer size, in samples */
	frames = spec->samples;
	frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL);
	spec->samples = frames;
	SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL);

	/* "set" the hardware with the desired parameters */
	status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	/* Calculate the final parameters for this audio specification */
	SDL_CalculateAudioSpec(spec);

	/* Allocate mixing buffer */
	mixlen = spec->size;
	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
	if ( mixbuf == NULL ) {
		ALSA_CloseAudio(this);
		return(-1);
	}
	memset(mixbuf, spec->silence, spec->size);

	/* Get the parent process id (we're the parent of the audio thread) */
	parent = getpid();

	/* Switch to blocking mode for playback */
	SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);

	/* We're ready to rock and roll. :-) */
	return(0);
}