view src/audio/alsa/SDL_alsa_audio.c @ 3818:49eadd6e8962 SDL-ryan-multiple-audio-device

Mangled ALSA dynamic loading...static loading will still work if you have the right libraries, but this is expected to break in general cases as we start dealing with different symbol versions, etc (but may work with, say, distro-specific packages, etc).
author Ryan C. Gordon <icculus@icculus.org>
date Fri, 06 Oct 2006 19:45:11 +0000
parents c8b3d3d13ed1
children b225d9820ee3
line wrap: on
line source

/*
    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
*/
#include "SDL_config.h"

/* Allow access to a raw mixing buffer */

#include <sys/types.h>
#include <signal.h>             /* For kill() */
#include <dlfcn.h>
#include <errno.h>
#include <string.h>

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


/* 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);

static int (*ALSA_snd_pcm_open)
                (snd_pcm_t **, const char *, snd_pcm_stream_t, int);
static int (*ALSA_snd_pcm_close)(snd_pcm_t * pcm);
static snd_pcm_sframes_t(*ALSA_snd_pcm_writei)
                (snd_pcm_t *,const void *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_resume)(snd_pcm_t *);
static int (*ALSA_snd_pcm_prepare)(snd_pcm_t *);
static int (*ALSA_snd_pcm_drain)(snd_pcm_t *);
static const char *(*ALSA_snd_strerror)(int);
static size_t(*ALSA_snd_pcm_hw_params_sizeof)(void);
static size_t(*ALSA_snd_pcm_sw_params_sizeof)(void);
static int (*ALSA_snd_pcm_hw_params_any)(snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_set_access)
                (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t);
static int (*ALSA_snd_pcm_hw_params_set_format)
                (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t);
static int (*ALSA_snd_pcm_hw_params_set_channels)
                (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int);
static int (*ALSA_snd_pcm_hw_params_get_channels)(const snd_pcm_hw_params_t *);
static unsigned int (*ALSA_snd_pcm_hw_params_set_rate_near)
                (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int, int *);
static snd_pcm_uframes_t (*ALSA_snd_pcm_hw_params_set_period_size_near)
                (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t, int *);
static snd_pcm_sframes_t (*ALSA_snd_pcm_hw_params_get_period_size)
                (const snd_pcm_hw_params_t *);
static unsigned int (*ALSA_snd_pcm_hw_params_set_periods_near)
                (snd_pcm_t *,snd_pcm_hw_params_t *, unsigned int, int *);
static int (*ALSA_snd_pcm_hw_params_get_periods)(snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params)(snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_sw_params_current)(snd_pcm_t*, snd_pcm_sw_params_t*);
static int (*ALSA_snd_pcm_sw_params_set_start_threshold)
                (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_sw_params_set_avail_min)
                (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_sw_params)(snd_pcm_t *, snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_nonblock)(snd_pcm_t *, int);
#define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof
#define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof


#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC

static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
static void *alsa_handle = NULL;

static int
load_alsa_sym(const char *fn, void **addr)
{
#if HAVE_DLVSYM
    *addr = dlvsym(alsa_handle, fn, "ALSA_0.9");
    if (*addr == NULL)
#endif
    {
        *addr = dlsym(alsa_handle, fn);
        if (*addr == NULL) {
            return 0;
        }
    }

    return 1;
}

/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_ALSA_SYM(x) \
    if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1
#else
#define SDL_ALSA_SYM(x) ALSA_##x = x
#endif

static int load_alsa_syms(void)
{
    SDL_ALSA_SYM(snd_pcm_open);
    SDL_ALSA_SYM(snd_pcm_close);
    SDL_ALSA_SYM(snd_pcm_writei);
    SDL_ALSA_SYM(snd_pcm_resume);
    SDL_ALSA_SYM(snd_pcm_prepare);
    SDL_ALSA_SYM(snd_pcm_drain);
    SDL_ALSA_SYM(snd_strerror);
    SDL_ALSA_SYM(snd_pcm_hw_params_sizeof);
    SDL_ALSA_SYM(snd_pcm_sw_params_sizeof);
    SDL_ALSA_SYM(snd_pcm_hw_params_any);
    SDL_ALSA_SYM(snd_pcm_hw_params_set_access);
    SDL_ALSA_SYM(snd_pcm_hw_params_set_format);
    SDL_ALSA_SYM(snd_pcm_hw_params_set_channels);
    SDL_ALSA_SYM(snd_pcm_hw_params_get_channels);
    SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near);
    SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near);
    SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size);
    SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_near);
    SDL_ALSA_SYM(snd_pcm_hw_params_get_periods);
    SDL_ALSA_SYM(snd_pcm_hw_params);
    SDL_ALSA_SYM(snd_pcm_sw_params_current);
    SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold);
    SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min);
    SDL_ALSA_SYM(snd_pcm_sw_params);
    SDL_ALSA_SYM(snd_pcm_nonblock);
    return 0;
}

static void
UnloadALSALibrary(void)
{
    if (alsa_handle != NULL) {
        dlclose(alsa_handle);
        alsa_handle = NULL;
    }
}

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

    alsa_handle = dlopen(alsa_library, RTLD_NOW);
    if (alsa_handle == NULL) {
        SDL_SetError("ALSA: dlopen('%s') failed: %s\n",
                      alsa_library, strerror(errno));
    } else {
        retval = load_alsa_syms();
        if (retval < 0) {
            UnloadALSALibrary();
        }
    }
    return retval;
}

#else

static void
UnloadALSALibrary(void)
{
    return;
}

static int
LoadALSALibrary(void)
{
    load_alsa_syms();
    return 0;
}

#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */

static const char *
get_audio_device(int channels)
{
    const char *device;

    device = SDL_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 = ALSA_snd_pcm_open(&handle, get_audio_device(2),
                               SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
    if (status >= 0) {
        available = 1;
        ALSA_snd_pcm_close(handle);
    }
    UnloadALSALibrary();
    return (available);
}

static void
Audio_DeleteDevice(SDL_AudioDevice * device)
{
    SDL_free(device->hidden);
    SDL_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 *) SDL_malloc(sizeof(SDL_AudioDevice));
    if (this) {
        SDL_memset(this, 0, (sizeof *this));
        this->hidden = (struct SDL_PrivateAudioData *)
            SDL_malloc((sizeof *this->hidden));
    }
    if ((this == NULL) || (this->hidden == NULL)) {
        SDL_OutOfMemory();
        if (this) {
            SDL_free(this);
        }
        return (0);
    }
    SDL_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, 0
};

/* 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;
            }
        }
    }
}


/*
 * http://bugzilla.libsdl.org/show_bug.cgi?id=110
 * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
 *  and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
 */
#define SWIZ6(T) \
    T *ptr = (T *) mixbuf; \
    const Uint32 count = (this->spec.samples / 6); \
    Uint32 i; \
    for (i = 0; i < count; i++, ptr += 6) { \
        T tmp; \
        tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
        tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
    }

static __inline__ void
swizzle_alsa_channels_6_64bit(_THIS)
{
    SWIZ6(Uint64);
}
static __inline__ void
swizzle_alsa_channels_6_32bit(_THIS)
{
    SWIZ6(Uint32);
}
static __inline__ void
swizzle_alsa_channels_6_16bit(_THIS)
{
    SWIZ6(Uint16);
}
static __inline__ void
swizzle_alsa_channels_6_8bit(_THIS)
{
    SWIZ6(Uint8);
}

#undef SWIZ6


/*
 * Called right before feeding this->mixbuf to the hardware. Swizzle channels
 *  from Windows/Mac order to the format alsalib will want.
 */
static __inline__ void
swizzle_alsa_channels(_THIS)
{
    if (this->spec.channels == 6) {
        const Uint16 fmtsize = (this->spec.format & 0xFF);      /* bits/channel. */
        if (fmtsize == 16)
            swizzle_alsa_channels_6_16bit(this);
        else if (fmtsize == 8)
            swizzle_alsa_channels_6_8bit(this);
        else if (fmtsize == 32)
            swizzle_alsa_channels_6_32bit(this);
        else if (fmtsize == 64)
            swizzle_alsa_channels_6_64bit(this);
    }

    /* !!! FIXME: update this for 7.1 if needed, later. */
}


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

    swizzle_alsa_channels(this);

    sample_len = this->spec.samples;
    sample_buf = (signed short *) mixbuf;

    while (sample_len > 0) {
        status = ALSA_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 = ALSA_snd_pcm_resume(pcm_handle);
                } while (status == -EAGAIN);
            }
            if (status < 0) {
                status = ALSA_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) {
        ALSA_snd_pcm_drain(pcm_handle);
        ALSA_snd_pcm_close(pcm_handle);
        pcm_handle = NULL;
    }
}

static int
ALSA_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
    int status;
    snd_pcm_hw_params_t *hwparams;
    snd_pcm_sw_params_t *swparams;
    snd_pcm_format_t format;
    snd_pcm_uframes_t frames;
    SDL_AudioFormat test_format;

    /* Open the audio device */
    /* Name of device should depend on # channels in spec */
    status = ALSA_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",
                     ALSA_snd_strerror(status));
        return (-1);
    }

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

    /* SDL only uses interleaved sample output */
    status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
                                               SND_PCM_ACCESS_RW_INTERLEAVED);
    if (status < 0) {
        SDL_SetError("Couldn't set interleaved access: %s",
                     ALSA_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);) {
        status = 0;             /* if we can't support a format, it'll become -1. */
        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;
        case AUDIO_S32LSB:
            format = SND_PCM_FORMAT_S32_LE;
            break;
        case AUDIO_S32MSB:
            format = SND_PCM_FORMAT_S32_BE;
            break;
        case AUDIO_F32LSB:
            format = SND_PCM_FORMAT_FLOAT_LE;
            break;
        case AUDIO_F32MSB:
            format = SND_PCM_FORMAT_FLOAT_BE;
            break;
        default:
            status = -1;
            break;
        }
        if (status >= 0) {
            status = ALSA_snd_pcm_hw_params_set_format(pcm_handle,
                                                       hwparams, 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 = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
                                                 spec->channels);
    if (status < 0) {
        status = ALSA_snd_pcm_hw_params_get_channels(hwparams);
        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 = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
                                                  spec->freq, NULL);
    if (status < 0) {
        ALSA_CloseAudio(this);
        SDL_SetError("Couldn't set audio frequency: %s",
                     ALSA_snd_strerror(status));
        return (-1);
    }
    spec->freq = status;

    /* Set the buffer size, in samples */
    frames = spec->samples;
    frames = ALSA_snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams,
                                                         frames, NULL);
    spec->samples = frames;
    ALSA_snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, 2, NULL);

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

/* This is useful for debugging... */
/*
{ snd_pcm_sframes_t bufsize; int fragments;
   bufsize = ALSA_snd_pcm_hw_params_get_period_size(hwparams);
   fragments = ALSA_snd_pcm_hw_params_get_periods(hwparams);

   fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments);
}
*/

    /* Set the software parameters */
    snd_pcm_sw_params_alloca(&swparams);
    status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
    if (status < 0) {
        SDL_SetError("Couldn't get software config: %s",
                     ALSA_snd_strerror(status));
        ALSA_CloseAudio(this);
        return (-1);
    }
    status = ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle,swparams,0);
    if (status < 0) {
        SDL_SetError("Couldn't set start threshold: %s",
                     ALSA_snd_strerror(status));
        ALSA_CloseAudio(this);
        return (-1);
    }
    status = ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, frames);
    if (status < 0) {
        SDL_SetError("Couldn't set avail min: %s", ALSA_snd_strerror(status));
        ALSA_CloseAudio(this);
        return (-1);
    }
    status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);
    if (status < 0) {
        SDL_SetError("Couldn't set software audio parameters: %s",
                     ALSA_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);
    }
    SDL_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 */
    ALSA_snd_pcm_nonblock(pcm_handle, 0);

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

/* vi: set ts=4 sw=4 expandtab: */