view src/audio/paudio/SDL_paudio.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 1e431c2631ee
children f7b03b6838cb
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 Lesser General Public
    License as published by the Free Software Foundation; either
    version 2.1 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
    Lesser General Public License for more details.

    You should have received a copy of the GNU Lesser General Public
    License along with this library; if not, write to the Free Software
    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA

    Carsten Griwodz
    griff@kom.tu-darmstadt.de

    based on linux/SDL_dspaudio.c by Sam Lantinga
*/
#include "SDL_config.h"

/* Allow access to a raw mixing buffer */

#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/stat.h>

#include "SDL_timer.h"
#include "SDL_audio.h"
#include "SDL_stdinc.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_paudio.h"

#define DEBUG_AUDIO 0

/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.
 * I guess nobody ever uses audio... Shame over AIX header files.  */
#include <sys/machine.h>
#undef BIG_ENDIAN
#include <sys/audio.h>

/* The tag name used by paud audio */
#define PAUDIO_DRIVER_NAME         "paud"

/* Open the audio device for playback, and don't block if busy */
/* #define OPEN_FLAGS	(O_WRONLY|O_NONBLOCK) */
#define OPEN_FLAGS	O_WRONLY

/* Get the name of the audio device we use for output */

#ifndef _PATH_DEV_DSP
#define _PATH_DEV_DSP	"/dev/%caud%c/%c"
#endif

static char devsettings[][3] = {
    {'p', '0', '1'}, {'p', '0', '2'}, {'p', '0', '3'}, {'p', '0', '4'},
    {'p', '1', '1'}, {'p', '1', '2'}, {'p', '1', '3'}, {'p', '1', '4'},
    {'p', '2', '1'}, {'p', '2', '2'}, {'p', '2', '3'}, {'p', '2', '4'},
    {'p', '3', '1'}, {'p', '3', '2'}, {'p', '3', '3'}, {'p', '3', '4'},
    {'b', '0', '1'}, {'b', '0', '2'}, {'b', '0', '3'}, {'b', '0', '4'},
    {'b', '1', '1'}, {'b', '1', '2'}, {'b', '1', '3'}, {'b', '1', '4'},
    {'b', '2', '1'}, {'b', '2', '2'}, {'b', '2', '3'}, {'b', '2', '4'},
    {'b', '3', '1'}, {'b', '3', '2'}, {'b', '3', '3'}, {'b', '3', '4'},
    {'\0', '\0', '\0'}
};

static int
OpenUserDefinedDevice(char *path, int maxlen, int flags)
{
    const char *audiodev;
    int fd;

    /* Figure out what our audio device is */
    if ((audiodev = SDL_getenv("SDL_PATH_DSP")) == NULL) {
        audiodev = SDL_getenv("AUDIODEV");
    }
    if (audiodev == NULL) {
        return -1;
    }
    fd = open(audiodev, flags, 0);
    if (path != NULL) {
        SDL_strlcpy(path, audiodev, maxlen);
        path[maxlen - 1] = '\0';
    }
    return fd;
}

static int
OpenAudioPath(char *path, int maxlen, int flags, int classic)
{
    struct stat sb;
    int cycle = 0;
    int fd = OpenUserDefinedDevice(path, maxlen, flags);

    if (fd != -1) {
        return fd;
    }

    /* !!! FIXME: do we really need a table here? */
    while (devsettings[cycle][0] != '\0') {
        char audiopath[1024];
        SDL_snprintf(audiopath, SDL_arraysize(audiopath),
                     _PATH_DEV_DSP,
                     devsettings[cycle][0],
                     devsettings[cycle][1], devsettings[cycle][2]);

        if (stat(audiopath, &sb) == 0) {
            fd = open(audiopath, flags, 0);
            if (fd > 0) {
                if (path != NULL) {
                    SDL_strlcpy(path, audiopath, maxlen);
                }
                return fd;
            }
        }
    }
    return -1;
}

/* This function waits until it is possible to write a full sound buffer */
static void
PAUDIO_WaitDevice(_THIS)
{
    fd_set fdset;

    /* See if we need to use timed audio synchronization */
    if (this->hidden->frame_ticks) {
        /* Use timer for general audio synchronization */
        Sint32 ticks;

        ticks =
            ((Sint32) (this->hidden->next_frame - SDL_GetTicks())) -
            FUDGE_TICKS;
        if (ticks > 0) {
            SDL_Delay(ticks);
        }
    } else {
        audio_buffer paud_bufinfo;

        /* Use select() for audio synchronization */
        struct timeval timeout;
        FD_ZERO(&fdset);
        FD_SET(this->hidden->audio_fd, &fdset);

        if (ioctl(this->hidden->audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
#ifdef DEBUG_AUDIO
            fprintf(stderr, "Couldn't get audio buffer information\n");
#endif
            timeout.tv_sec = 10;
            timeout.tv_usec = 0;
        } else {
            long ms_in_buf = paud_bufinfo.write_buf_time;
            timeout.tv_sec = ms_in_buf / 1000;
            ms_in_buf = ms_in_buf - timeout.tv_sec * 1000;
            timeout.tv_usec = ms_in_buf * 1000;
#ifdef DEBUG_AUDIO
            fprintf(stderr,
                    "Waiting for write_buf_time=%ld,%ld\n",
                    timeout.tv_sec, timeout.tv_usec);
#endif
        }

#ifdef DEBUG_AUDIO
        fprintf(stderr, "Waiting for audio to get ready\n");
#endif
        if (select(this->hidden->audio_fd + 1, NULL, &fdset, NULL, &timeout)
            <= 0) {
            const char *message =
                "Audio timeout - buggy audio driver? (disabled)";
            /*
             * In general we should never print to the screen,
             * but in this case we have no other way of letting
             * the user know what happened.
             */
            fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);
            this->enabled = 0;
            /* Don't try to close - may hang */
            this->hidden->audio_fd = -1;
#ifdef DEBUG_AUDIO
            fprintf(stderr, "Done disabling audio\n");
#endif
        }
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Ready!\n");
#endif
    }
}

static void
PAUDIO_PlayDevice(_THIS)
{
    int written = 0;
    const Uint8 *mixbuf = this->hidden->mixbuf;
    const size_t mixlen = this->hidden->mixlen;

    /* Write the audio data, checking for EAGAIN on broken audio drivers */
    do {
        written = write(this->hidden->audio_fd, mixbuf, mixlen);
        if ((written < 0) && ((errno == 0) || (errno == EAGAIN))) {
            SDL_Delay(1);       /* Let a little CPU time go by */
        }
    } while ((written < 0) &&
             ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)));

    /* If timer synchronization is enabled, set the next write frame */
    if (this->hidden->frame_ticks) {
        this->hidden->next_frame += this->hidden->frame_ticks;
    }

    /* If we couldn't write, assume fatal error for now */
    if (written < 0) {
        this->enabled = 0;
    }
#ifdef DEBUG_AUDIO
    fprintf(stderr, "Wrote %d bytes of audio data\n", written);
#endif
}

static Uint8 *
PAUDIO_GetDeviceBuf(_THIS)
{
    return this->hidden->mixbuf;
}

static void
PAUDIO_CloseDevice(_THIS)
{
    if (this->hidden != NULL) {
        if (this->hidden->mixbuf != NULL) {
            SDL_FreeAudioMem(this->hidden->mixbuf);
            this->hidden->mixbuf = NULL;
        }
        if (this->hidden->audio_fd >= 0) {
            close(this->hidden->audio_fd);
            this->hidden->audio_fd = -1;
        }
        SDL_free(this->hidden);
        this->hidden = NULL;
    }
}

static int
PAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
    const char *workaround = SDL_getenv("SDL_DSP_NOSELECT");
    char audiodev[1024];
    const char *err = NULL;
    int format;
    int bytes_per_sample;
    SDL_AudioFormat test_format;
    audio_init paud_init;
    audio_buffer paud_bufinfo;
    audio_status paud_status;
    audio_control paud_control;
    audio_change paud_change;
    int fd = -1;

    /* 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 */
    fd = OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
    this->hidden->audio_fd = fd;
    if (fd < 0) {
        PAUDIO_CloseDevice(this);
        SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
        return 0;
    }

    /*
     * We can't set the buffer size - just ask the device for the maximum
     * that we can have.
     */
    if (ioctl(fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
        PAUDIO_CloseDevice(this);
        SDL_SetError("Couldn't get audio buffer information");
        return 0;
    }

    if (this->spec.channels > 1)
        this->spec.channels = 2;
    else
        this->spec.channels = 1;

    /*
     * Fields in the audio_init structure:
     *
     * Ignored by us:
     *
     * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?
     * paud.slot_number;         * slot number of the adapter
     * paud.device_id;           * adapter identification number
     *
     * Input:
     *
     * paud.srate;           * the sampling rate in Hz
     * paud.bits_per_sample; * 8, 16, 32, ...
     * paud.bsize;           * block size for this rate
     * paud.mode;            * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX
     * paud.channels;        * 1=mono, 2=stereo
     * paud.flags;           * FIXED - fixed length data
     *                       * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)
     *                       * TWOS_COMPLEMENT - 2's complement data
     *                       * SIGNED - signed? comment seems wrong in sys/audio.h
     *                       * BIG_ENDIAN
     * paud.operation;       * PLAY, RECORD
     *
     * Output:
     *
     * paud.flags;           * PITCH            - pitch is supported
     *                       * INPUT            - input is supported
     *                       * OUTPUT           - output is supported
     *                       * MONITOR          - monitor is supported
     *                       * VOLUME           - volume is supported
     *                       * VOLUME_DELAY     - volume delay is supported
     *                       * BALANCE          - balance is supported
     *                       * BALANCE_DELAY    - balance delay is supported
     *                       * TREBLE           - treble control is supported
     *                       * BASS             - bass control is supported
     *                       * BESTFIT_PROVIDED - best fit returned
     *                       * LOAD_CODE        - DSP load needed
     * paud.rc;              * NO_PLAY         - DSP code can't do play requests
     *                       * NO_RECORD       - DSP code can't do record requests
     *                       * INVALID_REQUEST - request was invalid
     *                       * CONFLICT        - conflict with open's flags
     *                       * OVERLOADED      - out of DSP MIPS or memory
     * paud.position_resolution; * smallest increment for position
     */

    paud_init.srate = this->spec.freq;
    paud_init.mode = PCM;
    paud_init.operation = PLAY;
    paud_init.channels = this->spec.channels;

    /* Try for a closest match on audio format */
    format = 0;
    for (test_format = SDL_FirstAudioFormat(this->spec.format);
         !format && test_format;) {
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
        switch (test_format) {
        case AUDIO_U8:
            bytes_per_sample = 1;
            paud_init.bits_per_sample = 8;
            paud_init.flags = TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_S8:
            bytes_per_sample = 1;
            paud_init.bits_per_sample = 8;
            paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_S16LSB:
            bytes_per_sample = 2;
            paud_init.bits_per_sample = 16;
            paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_S16MSB:
            bytes_per_sample = 2;
            paud_init.bits_per_sample = 16;
            paud_init.flags = BIG_ENDIAN | SIGNED | TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_U16LSB:
            bytes_per_sample = 2;
            paud_init.bits_per_sample = 16;
            paud_init.flags = TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        case AUDIO_U16MSB:
            bytes_per_sample = 2;
            paud_init.bits_per_sample = 16;
            paud_init.flags = BIG_ENDIAN | TWOS_COMPLEMENT | FIXED;
            format = 1;
            break;
        default:
            break;
        }
        if (!format) {
            test_format = SDL_NextAudioFormat();
        }
    }
    if (format == 0) {
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Couldn't find any hardware audio formats\n");
#endif
        PAUDIO_CloseDevice(this);
        SDL_SetError("Couldn't find any hardware audio formats");
        return 0;
    }
    this->spec.format = test_format;

    /*
     * We know the buffer size and the max number of subsequent writes
     *  that can be pending. If more than one can pend, allow the application
     *  to do something like double buffering between our write buffer and
     *  the device's own buffer that we are filling with write() anyway.
     *
     * We calculate this->spec.samples like this because
     *  SDL_CalculateAudioSpec() will give put paud_bufinfo.write_buf_cap
     *  (or paud_bufinfo.write_buf_cap/2) into this->spec.size in return.
     */
    if (paud_bufinfo.request_buf_cap == 1) {
        this->spec.samples = paud_bufinfo.write_buf_cap
            / bytes_per_sample / this->spec.channels;
    } else {
        this->spec.samples = paud_bufinfo.write_buf_cap
            / bytes_per_sample / this->spec.channels / 2;
    }
    paud_init.bsize = bytes_per_sample * this->spec.channels;

    SDL_CalculateAudioSpec(&this->spec);

    /*
     * The AIX paud device init can't modify the values of the audio_init
     * structure that we pass to it. So we don't need any recalculation
     * of this stuff and no reinit call as in linux dsp and dma code.
     *
     * /dev/paud supports all of the encoding formats, so we don't need
     * to do anything like reopening the device, either.
     */
    if (ioctl(fd, AUDIO_INIT, &paud_init) < 0) {
        switch (paud_init.rc) {
        case 1:
            err = "Couldn't set audio format: DSP can't do play requests";
            break;
        case 2:
            err = "Couldn't set audio format: DSP can't do record requests";
            break;
        case 4:
            err = "Couldn't set audio format: request was invalid";
            break;
        case 5:
            err = "Couldn't set audio format: conflict with open's flags";
            break;
        case 6:
            err = "Couldn't set audio format: out of DSP MIPS or memory";
            break;
        default:
            err = "Couldn't set audio format: not documented in sys/audio.h";
            break;
        }
    }

    if (err != NULL) {
        PAUDIO_CloseDevice(this);
        SDL_SetError("Paudio: %s", err);
        return 0;
    }

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

    /*
     * Set some paramters: full volume, first speaker that we can find.
     * Ignore the other settings for now.
     */
    paud_change.input = AUDIO_IGNORE;   /* the new input source */
    paud_change.output = OUTPUT_1;      /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */
    paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */
    paud_change.volume = 0x7fffffff;    /* volume level [0-0x7fffffff] */
    paud_change.volume_delay = AUDIO_IGNORE;    /* the new volume delay */
    paud_change.balance = 0x3fffffff;   /* the new balance */
    paud_change.balance_delay = AUDIO_IGNORE;   /* the new balance delay */
    paud_change.treble = AUDIO_IGNORE;  /* the new treble state */
    paud_change.bass = AUDIO_IGNORE;    /* the new bass state */
    paud_change.pitch = AUDIO_IGNORE;   /* the new pitch state */

    paud_control.ioctl_request = AUDIO_CHANGE;
    paud_control.request_info = (char *) &paud_change;
    if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Can't change audio display settings\n");
#endif
    }

    /*
     * Tell the device to expect data. Actual start will wait for
     * the first write() call.
     */
    paud_control.ioctl_request = AUDIO_START;
    paud_control.position = 0;
    if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
        PAUDIO_CloseDevice(this);
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Can't start audio play\n");
#endif
        SDL_SetError("Can't start audio play");
        return 0;
    }

    /* Check to see if we need to use select() workaround */
    if (workaround != NULL) {
        this->hidden->frame_ticks = (float) (this->spec.samples * 1000) /
            this->spec.freq;
        this->hidden->next_frame = SDL_GetTicks() + this->hidden->frame_ticks;
    }

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

static int
PAUDIO_Init(SDL_AudioDriverImpl * impl)
{
    /* !!! FIXME: not right for device enum? */
    int fd = OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
    if (fd < 0) {
        SDL_SetError("PAUDIO: Couldn't open audio device");
        return 0;
    }
    close(fd);

    /* Set the function pointers */
    impl->OpenDevice = DSP_OpenDevice;
    impl->PlayDevice = DSP_PlayDevice;
    impl->PlayDevice = DSP_WaitDevice;
    impl->GetDeviceBuf = DSP_GetDeviceBuf;
    impl->CloseDevice = DSP_CloseDevice;
    impl->OnlyHasDefaultOutputDevice = 1;       /* !!! FIXME: add device enum! */

    /* !!! FIXME: device enum might make this 1. */
    return 2;                   /* 2 == definitely has an audio device. */
}

AudioBootStrap PAUDIO_bootstrap = {
    PAUDIO_DRIVER_NAME, "AIX Paudio", PAUDIO_Init, 0
};

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