view include/SDL_audio.h @ 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 02759105b989
children c9b51268668f
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/*
    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

/* Access to the raw audio mixing buffer for the SDL library */

#ifndef _SDL_audio_h
#define _SDL_audio_h

#include <stdio.h>

#include "SDL_main.h"
#include "SDL_types.h"
#include "SDL_error.h"
#include "SDL_rwops.h"
#include "SDL_byteorder.h"

#include "begin_code.h"
/* Set up for C function definitions, even when using C++ */
#ifdef __cplusplus
extern "C" {
#endif

/* The calculated values in this structure are calculated by SDL_OpenAudio() */
typedef struct SDL_AudioSpec {
	int freq;		/* DSP frequency -- samples per second */
	Uint16 format;		/* Audio data format */
	Uint8  channels;	/* Number of channels: 1 mono, 2 stereo */
	Uint8  silence;		/* Audio buffer silence value (calculated) */
	Uint16 samples;		/* Audio buffer size in samples (power of 2) */
	Uint16 padding;		/* Necessary for some compile environments */
	Uint32 size;		/* Audio buffer size in bytes (calculated) */
	/* This function is called when the audio device needs more data.
	   'stream' is a pointer to the audio data buffer
	   'len' is the length of that buffer in bytes.
	   Once the callback returns, the buffer will no longer be valid.
	   Stereo samples are stored in a LRLRLR ordering.
	*/
	void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
	void  *userdata;
} SDL_AudioSpec;

/* Audio format flags (defaults to LSB byte order) */
#define AUDIO_U8	0x0008	/* Unsigned 8-bit samples */
#define AUDIO_S8	0x8008	/* Signed 8-bit samples */
#define AUDIO_U16LSB	0x0010	/* Unsigned 16-bit samples */
#define AUDIO_S16LSB	0x8010	/* Signed 16-bit samples */
#define AUDIO_U16MSB	0x1010	/* As above, but big-endian byte order */
#define AUDIO_S16MSB	0x9010	/* As above, but big-endian byte order */
#define AUDIO_U16	AUDIO_U16LSB
#define AUDIO_S16	AUDIO_S16LSB

/* Native audio byte ordering */
#if SDL_BYTEORDER == SDL_LIL_ENDIAN
#define AUDIO_U16SYS	AUDIO_U16LSB
#define AUDIO_S16SYS	AUDIO_S16LSB
#else
#define AUDIO_U16SYS	AUDIO_U16MSB
#define AUDIO_S16SYS	AUDIO_S16MSB
#endif


/* A structure to hold a set of audio conversion filters and buffers */
typedef struct SDL_AudioCVT {
	int needed;			/* Set to 1 if conversion possible */
	Uint16 src_format;		/* Source audio format */
	Uint16 dst_format;		/* Target audio format */
	double rate_incr;		/* Rate conversion increment */
	Uint8 *buf;			/* Buffer to hold entire audio data */
	int    len;			/* Length of original audio buffer */
	int    len_cvt;			/* Length of converted audio buffer */
	int    len_mult;		/* buffer must be len*len_mult big */
	double len_ratio; 	/* Given len, final size is len*len_ratio */
	void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
	int filter_index;		/* Current audio conversion function */
} SDL_AudioCVT;


/* Function prototypes */

/* These functions are used internally, and should not be used unless you
 * have a specific need to specify the audio driver you want to use.
 * You should normally use SDL_Init() or SDL_InitSubSystem().
 */
extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
extern DECLSPEC void SDLCALL SDL_AudioQuit(void);

/* This function fills the given character buffer with the name of the
 * current audio driver, and returns a pointer to it if the audio driver has
 * been initialized.  It returns NULL if no driver has been initialized.
 */
extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);

/*
 * This function opens the audio device with the desired parameters, and
 * returns 0 if successful, placing the actual hardware parameters in the
 * structure pointed to by 'obtained'.  If 'obtained' is NULL, the audio
 * data passed to the callback function will be guaranteed to be in the
 * requested format, and will be automatically converted to the hardware
 * audio format if necessary.  This function returns -1 if it failed 
 * to open the audio device, or couldn't set up the audio thread.
 *
 * When filling in the desired audio spec structure,
 *  'desired->freq' should be the desired audio frequency in samples-per-second.
 *  'desired->format' should be the desired audio format.
 *  'desired->samples' is the desired size of the audio buffer, in samples.
 *     This number should be a power of two, and may be adjusted by the audio
 *     driver to a value more suitable for the hardware.  Good values seem to
 *     range between 512 and 8096 inclusive, depending on the application and
 *     CPU speed.  Smaller values yield faster response time, but can lead
 *     to underflow if the application is doing heavy processing and cannot
 *     fill the audio buffer in time.  A stereo sample consists of both right
 *     and left channels in LR ordering.
 *     Note that the number of samples is directly related to time by the
 *     following formula:  ms = (samples*1000)/freq
 *  'desired->size' is the size in bytes of the audio buffer, and is
 *     calculated by SDL_OpenAudio().
 *  'desired->silence' is the value used to set the buffer to silence,
 *     and is calculated by SDL_OpenAudio().
 *  'desired->callback' should be set to a function that will be called
 *     when the audio device is ready for more data.  It is passed a pointer
 *     to the audio buffer, and the length in bytes of the audio buffer.
 *     This function usually runs in a separate thread, and so you should
 *     protect data structures that it accesses by calling SDL_LockAudio()
 *     and SDL_UnlockAudio() in your code.
 *  'desired->userdata' is passed as the first parameter to your callback
 *     function.
 *
 * The audio device starts out playing silence when it's opened, and should
 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready
 * for your audio callback function to be called.  Since the audio driver
 * may modify the requested size of the audio buffer, you should allocate
 * any local mixing buffers after you open the audio device.
 */
extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);

/*
 * Get the current audio state:
 */
typedef enum {
	SDL_AUDIO_STOPPED = 0,
	SDL_AUDIO_PLAYING,
	SDL_AUDIO_PAUSED
} SDL_audiostatus;
extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);

/*
 * This function pauses and unpauses the audio callback processing.
 * It should be called with a parameter of 0 after opening the audio
 * device to start playing sound.  This is so you can safely initialize
 * data for your callback function after opening the audio device.
 * Silence will be written to the audio device during the pause.
 */
extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);

/*
 * This function loads a WAVE from the data source, automatically freeing
 * that source if 'freesrc' is non-zero.  For example, to load a WAVE file,
 * you could do:
 *	SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
 *
 * If this function succeeds, it returns the given SDL_AudioSpec,
 * filled with the audio data format of the wave data, and sets
 * 'audio_buf' to a malloc()'d buffer containing the audio data,
 * and sets 'audio_len' to the length of that audio buffer, in bytes.
 * You need to free the audio buffer with SDL_FreeWAV() when you are 
 * done with it.
 *
 * This function returns NULL and sets the SDL error message if the 
 * wave file cannot be opened, uses an unknown data format, or is 
 * corrupt.  Currently raw and MS-ADPCM WAVE files are supported.
 */
extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);

/* Compatibility convenience function -- loads a WAV from a file */
#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
	SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)

/*
 * This function frees data previously allocated with SDL_LoadWAV_RW()
 */
extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);

/*
 * This function takes a source format and rate and a destination format
 * and rate, and initializes the 'cvt' structure with information needed
 * by SDL_ConvertAudio() to convert a buffer of audio data from one format
 * to the other.
 * This function returns 0, or -1 if there was an error.
 */
extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
		Uint16 src_format, Uint8 src_channels, int src_rate,
		Uint16 dst_format, Uint8 dst_channels, int dst_rate);

/* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),
 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of
 * audio data in the source format, this function will convert it in-place
 * to the desired format.
 * The data conversion may expand the size of the audio data, so the buffer
 * cvt->buf should be allocated after the cvt structure is initialized by
 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.
 */
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);

/*
 * This takes two audio buffers of the playing audio format and mixes
 * them, performing addition, volume adjustment, and overflow clipping.
 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
 * for full audio volume.  Note this does not change hardware volume.
 * This is provided for convenience -- you can mix your own audio data.
 */
#define SDL_MIX_MAXVOLUME 128
extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);

/*
 * The lock manipulated by these functions protects the callback function.
 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the
 * callback function is not running.  Do not call these from the callback
 * function or you will cause deadlock.
 */
extern DECLSPEC void SDLCALL SDL_LockAudio(void);
extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);

/*
 * This function shuts down audio processing and closes the audio device.
 */
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);


/* Ends C function definitions when using C++ */
#ifdef __cplusplus
}
#endif
#include "close_code.h"

#endif /* _SDL_audio_h */