view src/audio/alsa/SDL_alsa_audio.c @ 3099:82e60908fab1

Date: Mon, 23 Mar 2009 09:17:24 +0200 From: "Mike Gorchak" Subject: New QNX patches Please apply patch qnx4.diff, which is attached. What has been done: 1)Added back OpenGL ES renderer for QNX target. Added few corrections to OpenGL ES renderer to let it work under QNX. OpenGL ES renderer do not support textures under QNX, so I think some additional work must be done. 2) Added GL_OES_query_matrix extension to SDL_opengles.h header file, which required by OpenGL ES 1.1 specification. 3) Added attribute clearing at the entrance of function SDL_GL_GetAttribure(). Added error checking into the function SDL_GL_GetAttribure(), because some attributes can't be obtained in OpenGL ES 1.0. 4) Porting testdyngles to OpenGL ES 1.0 (1.1 has glColor4ub() and glColor4f() functions, but 1.0 has glColor4f() only). 5) Added error checking after obtaining attributes using SDL_GL_GetAttribute() function to the testgl2 and testgles. 6) Small correction to testmultiaudio with printing errors. 7) Added software and accelerated OpenGL ES 1.0 support into the QNX GF driver. Please remove ./src/audio/nto directory - it will not be used anymore. Please create ./src/audio/qsa directory and add content of the archive qsa.tar.gz into this directory. I rewrote some sound code, added support for multiple audio cards, enumeration, etc. Added initial support for capture. As far as I can understand SDL 1.3 is not supporting audio capture right now ? Sam, Am I right ? Or audio capture must be supported through the PlayDevice routine ? And last, please put file SDL_gf_opengles.c to the ./src/video/qnxgf directory. It is OpenGL ES 1.1 emulation layer for some functions, which are not supported by OpenGL ES 1.0.
author Sam Lantinga <slouken@libsdl.org>
date Tue, 24 Mar 2009 10:33:12 +0000
parents b21348d47cab
children 4e83cdb58134
line wrap: on
line source

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2009 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 void
UnloadALSALibrary(void)
{
    if (alsa_handle != NULL) {
        dlclose(alsa_handle);
        alsa_handle = NULL;
    }
}

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


/* 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 && errno == ESRCH) {
                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;
    }
}

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));

    /* 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 void
ALSA_Deinitialize(void)
{
    UnloadALSALibrary();
}

static int
ALSA_Init(SDL_AudioDriverImpl * impl)
{
    if (LoadALSALibrary() < 0) {
        return 0;
    }

    /* 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->Deinitialize = ALSA_Deinitialize;
    impl->OnlyHasDefaultOutputDevice = 1;       /* !!! FIXME: Add device enum! */

    return 1;                   /* !!! FIXME: return 2 once device enum is implemented. */
}


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

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