src/audio/SDL_audio.c
author Sam Lantinga <slouken@libsdl.org>
Sat, 21 Aug 2004 12:27:02 +0000
changeset 942 41a59de7f2ed
parent 936 84f930aebaeb
child 955 d74fbf56f2f6
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
*/

#ifdef SAVE_RCSID
static char rcsid =
 "@(#) $Id$";
#endif

/* Allow access to a raw mixing buffer */
#include <stdlib.h>
#include <stdio.h>
#include <string.h>

#include "SDL.h"
#include "SDL_audio.h"
#include "SDL_timer.h"
#include "SDL_error.h"
#include "SDL_audio_c.h"
#include "SDL_audiomem.h"
#include "SDL_sysaudio.h"

/* Available audio drivers */
static AudioBootStrap *bootstrap[] = {
#ifdef OPENBSD_AUDIO_SUPPORT
	&OPENBSD_AUDIO_bootstrap,
#endif
#ifdef OSS_SUPPORT
	&DSP_bootstrap,
	&DMA_bootstrap,
#endif
#ifdef ALSA_SUPPORT
	&ALSA_bootstrap,
#endif
#ifdef QNXNTOAUDIO_SUPPORT
	&QNXNTOAUDIO_bootstrap,
#endif
#ifdef SUNAUDIO_SUPPORT
	&SUNAUDIO_bootstrap,
#endif
#ifdef DMEDIA_SUPPORT
	&DMEDIA_bootstrap,
#endif
#ifdef ARTSC_SUPPORT
	&ARTSC_bootstrap,
#endif
#ifdef ESD_SUPPORT
	&ESD_bootstrap,
#endif
#ifdef NAS_SUPPORT
	&NAS_bootstrap,
#endif
#ifdef ENABLE_DIRECTX
	&DSOUND_bootstrap,
#endif
#ifdef ENABLE_WINDIB
	&WAVEOUT_bootstrap,
#endif
#ifdef __BEOS__
	&BAUDIO_bootstrap,
#endif
#ifdef MACOSX
	&COREAUDIO_bootstrap,
#endif
#if defined(macintosh) || TARGET_API_MAC_CARBON
	&SNDMGR_bootstrap,
#endif
#ifdef _AIX
	&Paud_bootstrap,
#endif
#ifdef ENABLE_AHI
	&AHI_bootstrap,
#endif
#ifdef MMEAUDIO_SUPPORT
	&MMEAUDIO_bootstrap,
#endif
#ifdef MINTAUDIO_SUPPORT
	&MINTAUDIO_GSXB_bootstrap,
	&MINTAUDIO_MCSN_bootstrap,
	&MINTAUDIO_STFA_bootstrap,
	&MINTAUDIO_XBIOS_bootstrap,
	&MINTAUDIO_DMA8_bootstrap,
#endif
#ifdef DISKAUD_SUPPORT
	&DISKAUD_bootstrap,
#endif
#ifdef ENABLE_DC
	&DCAUD_bootstrap,
#endif
#ifdef DRENDERER_SUPPORT
	&DRENDERER_bootstrap,
#endif
	NULL
};
SDL_AudioDevice *current_audio = NULL;

/* Various local functions */
int SDL_AudioInit(const char *driver_name);
void SDL_AudioQuit(void);

#ifdef ENABLE_AHI
static int audio_configured = 0;
#endif

/* The general mixing thread function */
int SDL_RunAudio(void *audiop)
{
	SDL_AudioDevice *audio = (SDL_AudioDevice *)audiop;
	Uint8 *stream;
	int    stream_len;
	void  *udata;
	void (*fill)(void *userdata,Uint8 *stream, int len);
	int    silence;
#ifdef ENABLE_AHI
	int started = 0;

/* AmigaOS NEEDS that the audio driver is opened in the thread that uses it! */

	D(bug("Task audio started audio struct:<%lx>...\n",audiop));

	D(bug("Before Openaudio..."));
	if(audio->OpenAudio(audio, &audio->spec)==-1)
	{
		D(bug("Open audio failed...\n"));
		return(-1);
	}
	D(bug("OpenAudio...OK\n"));
#endif

	/* Perform any thread setup */
	if ( audio->ThreadInit ) {
		audio->ThreadInit(audio);
	}
	audio->threadid = SDL_ThreadID();

	/* Set up the mixing function */
	fill  = audio->spec.callback;
	udata = audio->spec.userdata;

#ifdef ENABLE_AHI
	audio_configured = 1;

	D(bug("Audio configured... Checking for conversion\n"));
	SDL_mutexP(audio->mixer_lock);
	D(bug("Semaphore obtained...\n"));
#endif

	if ( audio->convert.needed ) {
		if ( audio->convert.src_format == AUDIO_U8 ) {
			silence = 0x80;
		} else {
			silence = 0;
		}
		stream_len = audio->convert.len;
	} else {
		silence = audio->spec.silence;
		stream_len = audio->spec.size;
	}
	stream = audio->fake_stream;

#ifdef ENABLE_AHI
	SDL_mutexV(audio->mixer_lock);
	D(bug("Entering audio loop...\n"));
#endif


	/* Loop, filling the audio buffers */
	while ( audio->enabled ) {

		/* Wait for new current buffer to finish playing */
		if ( stream == audio->fake_stream ) {
			SDL_Delay((audio->spec.samples*1000)/audio->spec.freq);
		} else {
#ifdef ENABLE_AHI
			if ( started > 1 )
#endif
			audio->WaitAudio(audio);
		}

		/* Fill the current buffer with sound */
		if ( audio->convert.needed ) {
			if ( audio->convert.buf ) {
				stream = audio->convert.buf;
			} else {
				continue;
			}
		} else {
			stream = audio->GetAudioBuf(audio);
			if ( stream == NULL ) {
				stream = audio->fake_stream;
			}
		}
		memset(stream, silence, stream_len);

		if ( ! audio->paused ) {
			SDL_mutexP(audio->mixer_lock);
			(*fill)(udata, stream, stream_len);
			SDL_mutexV(audio->mixer_lock);
		}

		/* Convert the audio if necessary */
		if ( audio->convert.needed ) {
			SDL_ConvertAudio(&audio->convert);
			stream = audio->GetAudioBuf(audio);
			if ( stream == NULL ) {
				stream = audio->fake_stream;
			}
			memcpy(stream, audio->convert.buf,
			               audio->convert.len_cvt);
		}

		/* Ready current buffer for play and change current buffer */
		if ( stream != audio->fake_stream ) {
			audio->PlayAudio(audio);
#ifdef ENABLE_AHI
/* AmigaOS don't have to wait the first time audio is played! */
			started++;
#endif
		}
	}
	/* Wait for the audio to drain.. */
	if ( audio->WaitDone ) {
		audio->WaitDone(audio);
	}

#ifdef ENABLE_AHI
	D(bug("WaitAudio...Done\n"));

	audio->CloseAudio(audio);

	D(bug("CloseAudio..Done, subtask exiting...\n"));
	audio_configured = 0;
#endif
	return(0);
}

static void SDL_LockAudio_Default(SDL_AudioDevice *audio)
{
	if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) {
		return;
	}
	SDL_mutexP(audio->mixer_lock);
}

static void SDL_UnlockAudio_Default(SDL_AudioDevice *audio)
{
	if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) {
		return;
	}
	SDL_mutexV(audio->mixer_lock);
}

int SDL_AudioInit(const char *driver_name)
{
	SDL_AudioDevice *audio;
	int i = 0, idx;

	/* Check to make sure we don't overwrite 'current_audio' */
	if ( current_audio != NULL ) {
		SDL_AudioQuit();
	}

	/* Select the proper audio driver */
	audio = NULL;
	idx = 0;
#ifdef unix
	if ( (driver_name == NULL) && (getenv("ESPEAKER") != NULL) ) {
		/* Ahem, we know that if ESPEAKER is set, user probably wants
		   to use ESD, but don't start it if it's not already running.
		   This probably isn't the place to do this, but... Shh! :)
		 */
		for ( i=0; bootstrap[i]; ++i ) {
			if ( strcmp(bootstrap[i]->name, "esd") == 0 ) {
				const char *esd_no_spawn;

				/* Don't start ESD if it's not running */
				esd_no_spawn = getenv("ESD_NO_SPAWN");
				if ( esd_no_spawn == NULL ) {
					putenv("ESD_NO_SPAWN=1");
				}
				if ( bootstrap[i]->available() ) {
					audio = bootstrap[i]->create(0);
					break;
				}
#ifdef linux	/* No unsetenv() on most platforms */
				if ( esd_no_spawn == NULL ) {
					unsetenv("ESD_NO_SPAWN");
				}
#endif
			}
		}
	}
#endif /* unix */
	if ( audio == NULL ) {
		if ( driver_name != NULL ) {
#if 0	/* This will be replaced with a better driver selection API */
			if ( strrchr(driver_name, ':') != NULL ) {
				idx = atoi(strrchr(driver_name, ':')+1);
			}
#endif
			for ( i=0; bootstrap[i]; ++i ) {
				if (strncmp(bootstrap[i]->name, driver_name,
				            strlen(bootstrap[i]->name)) == 0) {
					if ( bootstrap[i]->available() ) {
						audio=bootstrap[i]->create(idx);
						break;
					}
				}
			}
		} else {
			for ( i=0; bootstrap[i]; ++i ) {
				if ( bootstrap[i]->available() ) {
					audio = bootstrap[i]->create(idx);
					if ( audio != NULL ) {
						break;
					}
				}
			}
		}
		if ( audio == NULL ) {
			SDL_SetError("No available audio device");
#if 0 /* Don't fail SDL_Init() if audio isn't available.
         SDL_OpenAudio() will handle it at that point.  *sigh*
       */
			return(-1);
#endif
		}
	}
	current_audio = audio;
	if ( current_audio ) {
		current_audio->name = bootstrap[i]->name;
		if ( !current_audio->LockAudio && !current_audio->UnlockAudio ) {
			current_audio->LockAudio = SDL_LockAudio_Default;
			current_audio->UnlockAudio = SDL_UnlockAudio_Default;
		}
	}
	return(0);
}

char *SDL_AudioDriverName(char *namebuf, int maxlen)
{
	if ( current_audio != NULL ) {
		strncpy(namebuf, current_audio->name, maxlen-1);
		namebuf[maxlen-1] = '\0';
		return(namebuf);
	}
	return(NULL);
}

int SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained)
{
	SDL_AudioDevice *audio;

	/* Start up the audio driver, if necessary */
	if ( ! current_audio ) {
		if ( (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) ||
		     (current_audio == NULL) ) {
			return(-1);
		}
	}
	audio = current_audio;

	if (audio->opened) {
		SDL_SetError("Audio device is already opened");
		return(-1);
	}

	/* Verify some parameters */
	if ( desired->callback == NULL ) {
		SDL_SetError("SDL_OpenAudio() passed a NULL callback");
		return(-1);
	}
	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");
		return(-1);
	}

#if defined(macintosh) || defined(__riscos__)
	/* FIXME: Need to implement PPC interrupt asm for SDL_LockAudio() */
#else
#if defined(__MINT__) && !defined(ENABLE_THREADS)
	/* Uses interrupt driven audio, without thread */
#else
	/* Create a semaphore for locking the sound buffers */
	audio->mixer_lock = SDL_CreateMutex();
	if ( audio->mixer_lock == NULL ) {
		SDL_SetError("Couldn't create mixer lock");
		SDL_CloseAudio();
		return(-1);
	}
#endif /* __MINT__ */
#endif /* macintosh */

	/* Calculate the silence and size of the audio specification */
	SDL_CalculateAudioSpec(desired);

	/* Open the audio subsystem */
	memcpy(&audio->spec, desired, sizeof(audio->spec));
	audio->convert.needed = 0;
	audio->enabled = 1;
	audio->paused  = 1;

#ifndef ENABLE_AHI

/* AmigaOS opens audio inside the main loop */
	audio->opened = audio->OpenAudio(audio, &audio->spec)+1;

	if ( ! audio->opened ) {
		SDL_CloseAudio();
		return(-1);
	}
#else
	D(bug("Locking semaphore..."));
	SDL_mutexP(audio->mixer_lock);

	audio->thread = SDL_CreateThread(SDL_RunAudio, audio);
	D(bug("Created thread...\n"));

	if ( audio->thread == NULL ) {
		SDL_mutexV(audio->mixer_lock);
		SDL_CloseAudio();
		SDL_SetError("Couldn't create audio thread");
		return(-1);
	}

	while(!audio_configured)
		SDL_Delay(100);
#endif

	/* If the audio driver changes the buffer size, accept it */
	if ( audio->spec.samples != desired->samples ) {
		desired->samples = audio->spec.samples;
		SDL_CalculateAudioSpec(desired);
	}

	/* Allocate a fake audio memory buffer */
	audio->fake_stream = SDL_AllocAudioMem(audio->spec.size);
	if ( audio->fake_stream == NULL ) {
		SDL_CloseAudio();
		SDL_OutOfMemory();
		return(-1);
	}

	/* See if we need to do any conversion */
	if ( obtained != NULL ) {
		memcpy(obtained, &audio->spec, sizeof(audio->spec));
	} else if ( desired->freq != audio->spec.freq ||
                    desired->format != audio->spec.format ||
	            desired->channels != audio->spec.channels ) {
		/* Build an audio conversion block */
		if ( SDL_BuildAudioCVT(&audio->convert,
			desired->format, desired->channels,
					desired->freq,
			audio->spec.format, audio->spec.channels,
					audio->spec.freq) < 0 ) {
			SDL_CloseAudio();
			return(-1);
		}
		if ( audio->convert.needed ) {
			audio->convert.len = desired->size;
			audio->convert.buf =(Uint8 *)SDL_AllocAudioMem(
			   audio->convert.len*audio->convert.len_mult);
			if ( audio->convert.buf == NULL ) {
				SDL_CloseAudio();
				SDL_OutOfMemory();
				return(-1);
			}
		}
	}

#ifndef ENABLE_AHI
	/* Start the audio thread if necessary */
	switch (audio->opened) {
		case  1:
			/* Start the audio thread */
			audio->thread = SDL_CreateThread(SDL_RunAudio, audio);
			if ( audio->thread == NULL ) {
				SDL_CloseAudio();
				SDL_SetError("Couldn't create audio thread");
				return(-1);
			}
			break;

		default:
			/* The audio is now playing */
			break;
	}
#else
	SDL_mutexV(audio->mixer_lock);
	D(bug("SDL_OpenAudio USCITA...\n"));

#endif

	return(0);
}

SDL_audiostatus SDL_GetAudioStatus(void)
{
	SDL_AudioDevice *audio = current_audio;
	SDL_audiostatus status;

	status = SDL_AUDIO_STOPPED;
	if ( audio && audio->enabled ) {
		if ( audio->paused ) {
			status = SDL_AUDIO_PAUSED;
		} else {
			status = SDL_AUDIO_PLAYING;
		}
	}
	return(status);
}

void SDL_PauseAudio (int pause_on)
{
	SDL_AudioDevice *audio = current_audio;

	if ( audio ) {
		audio->paused = pause_on;
	}
}

void SDL_LockAudio (void)
{
	SDL_AudioDevice *audio = current_audio;

	/* Obtain a lock on the mixing buffers */
	if ( audio && audio->LockAudio ) {
		audio->LockAudio(audio);
	}
}

void SDL_UnlockAudio (void)
{
	SDL_AudioDevice *audio = current_audio;

	/* Release lock on the mixing buffers */
	if ( audio && audio->UnlockAudio ) {
		audio->UnlockAudio(audio);
	}
}

void SDL_CloseAudio (void)
{
	SDL_QuitSubSystem(SDL_INIT_AUDIO);
}

void SDL_AudioQuit(void)
{
	SDL_AudioDevice *audio = current_audio;

	if ( audio ) {
		audio->enabled = 0;
		if ( audio->thread != NULL ) {
			SDL_WaitThread(audio->thread, NULL);
		}
		if ( audio->mixer_lock != NULL ) {
			SDL_DestroyMutex(audio->mixer_lock);
		}
		if ( audio->fake_stream != NULL ) {
			SDL_FreeAudioMem(audio->fake_stream);
		}
		if ( audio->convert.needed ) {
			SDL_FreeAudioMem(audio->convert.buf);

		}
#ifndef ENABLE_AHI
		if ( audio->opened ) {
			audio->CloseAudio(audio);
			audio->opened = 0;
		}
#endif
		/* Free the driver data */
		audio->free(audio);
		current_audio = NULL;
	}
}

#define NUM_FORMATS	6
static int format_idx;
static int format_idx_sub;
static Uint16 format_list[NUM_FORMATS][NUM_FORMATS] = {
 { AUDIO_U8, AUDIO_S8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB },
 { AUDIO_S8, AUDIO_U8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB },
 { AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8 },
 { AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8 },
 { AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U8, AUDIO_S8 },
 { AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U8, AUDIO_S8 },
};

Uint16 SDL_FirstAudioFormat(Uint16 format)
{
	for ( format_idx=0; format_idx < NUM_FORMATS; ++format_idx ) {
		if ( format_list[format_idx][0] == format ) {
			break;
		}
	}
	format_idx_sub = 0;
	return(SDL_NextAudioFormat());
}

Uint16 SDL_NextAudioFormat(void)
{
	if ( (format_idx == NUM_FORMATS) || (format_idx_sub == NUM_FORMATS) ) {
		return(0);
	}
	return(format_list[format_idx][format_idx_sub++]);
}

void SDL_CalculateAudioSpec(SDL_AudioSpec *spec)
{
	switch (spec->format) {
		case AUDIO_U8:
			spec->silence = 0x80;
			break;
		default:
			spec->silence = 0x00;
			break;
	}
	spec->size = (spec->format&0xFF)/8;
	spec->size *= spec->channels;
	spec->size *= spec->samples;
}