src/audio/alsa/SDL_alsa_audio.c

/*
    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"

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)
{
    /*
     * !!! FIXME:
     * Eventually, this will deal with fallbacks, version changes, and
     *  missing symbols we can workaround. But for now, it doesn't.
     */

#if HAVE_DLVSYM
    *addr = dlvsym(alsa_handle, fn, "ALSA_0.9");
    if (*addr == NULL)
#endif
    {
        *addr = dlsym(alsa_handle, fn);
        if (*addr == NULL) {
            SDL_SetError("dlsym('%s') failed: %s", fn, strerror(errno));
            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;
}
#undef SDL_ALSA_SYM

#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC

static int library_load_count = 0;

static void
UnloadALSALibrary(void)
{
    if ((alsa_handle != NULL) && (--library_load_count == 0)) {
        dlclose(alsa_handle);
        alsa_handle = NULL;
    }
}

static int
LoadALSALibrary(void)
{
    int retval = 0;
    if (library_load_count++ == 0) {
        alsa_handle = dlopen(alsa_library, RTLD_NOW);
        if (alsa_handle == NULL) {
            library_load_count--;
            retval = -1;
            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)
{
}

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;
}


static int
ALSA_Available(void)
{
    int available = 0;

    if (LoadALSALibrary() >= 0) {
        available = 1;
        UnloadALSALibrary();
    }
    return (available);
}


/* This function waits until it is possible to write a full sound buffer */
static void
ALSA_WaitDevice(_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.
         */
        /* Check every 10 loops */
        if (this->hidden->parent && (((++cnt) % 10) == 0)) {
            if (kill(this->hidden->parent, 0) < 0) {
                this->enabled = 0;
            }
        }
    }
}


/* !!! FIXME: is there a channel swizzler in alsalib instead? */
/*
 * 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 *) this->hidden->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->hidden->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_PlayDevice(_THIS)
{
    int status;
    int sample_len;
    signed short *sample_buf;

    swizzle_alsa_channels(this);

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

    while (sample_len > 0) {
        status = ALSA_snd_pcm_writei(this->hidden->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(this->hidden->pcm_handle);
                } while (status == -EAGAIN);
            }
            if (status < 0) {
                status = ALSA_snd_pcm_prepare(this->hidden->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_GetDeviceBuf(_THIS)
{
    return (this->hidden->mixbuf);
}

static void
ALSA_CloseDevice(_THIS)
{
    if (this->hidden != NULL) {
        if (this->hidden->mixbuf != NULL) {
            SDL_FreeAudioMem(this->hidden->mixbuf);
            this->hidden->mixbuf = NULL;
        }
        if (this->hidden->pcm_handle) {
            ALSA_snd_pcm_drain(this->hidden->pcm_handle);
            ALSA_snd_pcm_close(this->hidden->pcm_handle);
            this->hidden->pcm_handle = NULL;
        }
        SDL_free(this->hidden);
        this->hidden = NULL;
        UnloadALSALibrary();
    }
}

static int
ALSA_OpenDevice(_THIS, const char *devname, int iscapture)
{
    int status = 0;
    snd_pcm_t *pcm_handle = NULL;
    snd_pcm_hw_params_t *hwparams = NULL;
    snd_pcm_sw_params_t *swparams = NULL;
    snd_pcm_format_t format = 0;
    snd_pcm_uframes_t frames = 0;
    SDL_AudioFormat test_format = 0;

    /* Initialize all variables that we clean on shutdown */
    this->hidden = (struct SDL_PrivateAudioData *)
                        SDL_malloc((sizeof *this->hidden));
    if (this->hidden == NULL) {
        SDL_OutOfMemory();
        return 0;
    }
    SDL_memset(this->hidden, 0, (sizeof *this->hidden));

    if (LoadALSALibrary() < 0) {
        ALSA_CloseDevice(this);
        return 0;
    }

    /* Open the audio device */
    /* Name of device should depend on # channels in spec */
    status = ALSA_snd_pcm_open(&pcm_handle,
                               get_audio_device(this->spec.channels),
                               SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);

    if (status < 0) {
        ALSA_CloseDevice(this);
        SDL_SetError("ALSA: Couldn't open audio device: %s",
                     ALSA_snd_strerror(status));
        return 0;
    }

    this->hidden->pcm_handle = pcm_handle;

    /* 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) {
        ALSA_CloseDevice(this);
        SDL_SetError("ALSA: Couldn't get hardware config: %s",
                     ALSA_snd_strerror(status));
        return 0;
    }

    /* 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) {
        ALSA_CloseDevice(this);
        SDL_SetError("ALSA: Couldn't set interleaved access: %s",
                     ALSA_snd_strerror(status));
        return 0;
    }

    /* Try for a closest match on audio format */
    status = -1;
    for (test_format = SDL_FirstAudioFormat(this->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) {
        ALSA_CloseDevice(this);
        SDL_SetError("ALSA: Couldn't find any hardware audio formats");
        return 0;
    }
    this->spec.format = test_format;

    /* Set the number of channels */
    status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
                                                 this->spec.channels);
    if (status < 0) {
        status = ALSA_snd_pcm_hw_params_get_channels(hwparams);
        if ((status <= 0) || (status > 2)) {
            ALSA_CloseDevice(this);
            SDL_SetError("ALSA: Couldn't set audio channels");
            return 0;
        }
        this->spec.channels = status;
    }

    /* Set the audio rate */
    status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
                                                  this->spec.freq, NULL);
    if (status < 0) {
        ALSA_CloseDevice(this);
        SDL_SetError("ALSA: Couldn't set audio frequency: %s",
                     ALSA_snd_strerror(status));
        return 0;
    }
    this->spec.freq = status;

    /* Set the buffer size, in samples */
    frames = this->spec.samples;
    frames = ALSA_snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams,
                                                         frames, NULL);
    this->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_CloseDevice(this);
        SDL_SetError("ALSA: Couldn't set hardware audio parameters: %s",
                     ALSA_snd_strerror(status));
        return 0;
    }

#if AUDIO_DEBUG
{
    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);
}
#endif

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

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

    /* Allocate mixing buffer */
    this->hidden->mixlen = this->spec.size;
    this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
    if (this->hidden->mixbuf == NULL) {
        ALSA_CloseDevice(this);
        SDL_OutOfMemory();
        return 0;
    }
    SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);

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

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

    /* We're ready to rock and roll. :-) */
    return 1;
}

static int
ALSA_Init(SDL_AudioDriverImpl *impl)
{
    /* Set the function pointers */
    impl->OpenDevice = ALSA_OpenDevice;
    impl->WaitDevice = ALSA_WaitDevice;
    impl->GetDeviceBuf = ALSA_GetDeviceBuf;
    impl->PlayDevice = ALSA_PlayDevice;
    impl->CloseDevice = ALSA_CloseDevice;
    impl->OnlyHasDefaultOutputDevice = 1;  /* !!! FIXME: Add device enum! */

    return 1;
}


AudioBootStrap ALSA_bootstrap = {
    DRIVER_NAME, "ALSA 0.9 PCM audio",
    ALSA_Available, ALSA_Init, 0
};

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