src/audio/paudio/SDL_paudio.c
author Ryan C. Gordon <icculus@icculus.org>
Thu, 24 Aug 2006 12:10:46 +0000
changeset 1982 3b4ce57c6215
parent 1895 c121d94672cb
child 2049 5f6550e5184f
child 3798 c8b3d3d13ed1
permissions -rw-r--r--
First shot at new audio data types (int32 and float32). Notable changes: - Converters between types are autogenerated. Instead of making multiple passes over the data with seperate filters for endianess, size, signedness, etc, converting between data types is always one specialized filter. This simplifies SDL_BuildAudioCVT(), which otherwise had a million edge cases with the new types, and makes the actually conversions more CPU cache friendly. Left a stub for adding specific optimized versions of these routines (SSE/MMX/Altivec, assembler, etc) - Autogenerated converters are built by SDL/src/audio/sdlgenaudiocvt.pl. This does not need to be run unless tweaking the code, and thus doesn't need integration into the build system. - Went through all the drivers and tried to weed out all the "Uint16" references that are better specified with the new SDL_AudioFormat typedef. - Cleaned out a bunch of hardcoded bitwise magic numbers and replaced them with new SDL_AUDIO_* macros. - Added initial float32 and int32 support code. Theoretically, existing drivers will push these through converters to get the data they want to feed to the hardware. Still TODO: - Optimize and debug new converters. - Update the CoreAudio backend to accept float32 data directly. - Other backends, too? - SDL_LoadWAV() needs to be updated to support int32 and float32 .wav files (both of which exist and can be generated by 'sox' for testing purposes). - Update the mixer to handle new datatypes. - Optionally update SDL_sound and SDL_mixer, etc.

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2006 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/stat.h>

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

#define DEBUG_AUDIO 1

/* 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 Paud_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

/* Audio driver functions */
static int Paud_OpenAudio(_THIS, SDL_AudioSpec * spec);
static void Paud_WaitAudio(_THIS);
static void Paud_PlayAudio(_THIS);
static Uint8 *Paud_GetAudioBuf(_THIS);
static void Paud_CloseAudio(_THIS);

/* Audio driver bootstrap functions */

static int
Audio_Available(void)
{
    int fd;
    int available;

    available = 0;
    fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
    if (fd >= 0) {
        available = 1;
        close(fd);
    }
    return (available);
}

static void
Audio_DeleteDevice(SDL_AudioDevice * device)
{
    SDL_free(device->hidden);
    SDL_free(device);
}

static SDL_AudioDevice *
Audio_CreateDevice(int devindex)
{
    SDL_AudioDevice *this;

    /* Initialize all variables that we clean on shutdown */
    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));
    audio_fd = -1;

    /* Set the function pointers */
    this->OpenAudio = Paud_OpenAudio;
    this->WaitAudio = Paud_WaitAudio;
    this->PlayAudio = Paud_PlayAudio;
    this->GetAudioBuf = Paud_GetAudioBuf;
    this->CloseAudio = Paud_CloseAudio;

    this->free = Audio_DeleteDevice;

    return this;
}

AudioBootStrap Paud_bootstrap = {
    Paud_DRIVER_NAME, "AIX Paudio",
    Audio_Available, Audio_CreateDevice
};

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

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

        ticks = ((Sint32) (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(audio_fd, &fdset);

        if (ioctl(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(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 */
            audio_fd = -1;
#ifdef DEBUG_AUDIO
            fprintf(stderr, "Done disabling audio\n");
#endif
        }
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Ready!\n");
#endif
    }
}

static void
Paud_PlayAudio(_THIS)
{
    int written;

    /* Write the audio data, checking for EAGAIN on broken audio drivers */
    do {
        written = write(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 (frame_ticks) {
        next_frame += 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 *
Paud_GetAudioBuf(_THIS)
{
    return mixbuf;
}

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

static int
Paud_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
    char audiodev[1024];
    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;

    /* Reset the timer synchronization flag */
    frame_ticks = 0.0;

    /* Open the audio device */
    audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
    if (audio_fd < 0) {
        SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
        return -1;
    }

    /*
     * We can't set the buffer size - just ask the device for the maximum
     * that we can have.
     */
    if (ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
        SDL_SetError("Couldn't get audio buffer information");
        return -1;
    }

    mixbuf = NULL;

    if (spec->channels > 1)
        spec->channels = 2;
    else
        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 = spec->freq;
    paud_init.mode = PCM;
    paud_init.operation = PLAY;
    paud_init.channels = spec->channels;

    /* Try for a closest match on audio format */
    format = 0;
    for (test_format = SDL_FirstAudioFormat(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
        SDL_SetError("Couldn't find any hardware audio formats");
        return -1;
    }
    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 spec->samples like this because SDL_CalculateAudioSpec()
     * will give put paud_bufinfo.write_buf_cap (or paud_bufinfo.write_buf_cap/2)
     * into spec->size in return.
     */
    if (paud_bufinfo.request_buf_cap == 1) {
        spec->samples = paud_bufinfo.write_buf_cap
            / bytes_per_sample / spec->channels;
    } else {
        spec->samples = paud_bufinfo.write_buf_cap
            / bytes_per_sample / spec->channels / 2;
    }
    paud_init.bsize = bytes_per_sample * spec->channels;

    SDL_CalculateAudioSpec(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(audio_fd, AUDIO_INIT, &paud_init) < 0) {
        switch (paud_init.rc) {
        case 1:
            SDL_SetError
                ("Couldn't set audio format: DSP can't do play requests");
            return -1;
            break;
        case 2:
            SDL_SetError
                ("Couldn't set audio format: DSP can't do record requests");
            return -1;
            break;
        case 4:
            SDL_SetError("Couldn't set audio format: request was invalid");
            return -1;
            break;
        case 5:
            SDL_SetError
                ("Couldn't set audio format: conflict with open's flags");
            return -1;
            break;
        case 6:
            SDL_SetError
                ("Couldn't set audio format: out of DSP MIPS or memory");
            return -1;
            break;
        default:
            SDL_SetError
                ("Couldn't set audio format: not documented in sys/audio.h");
            return -1;
            break;
        }
    }

    /* Allocate mixing buffer */
    mixlen = spec->size;
    mixbuf = (Uint8 *) SDL_AllocAudioMem(mixlen);
    if (mixbuf == NULL) {
        return -1;
    }
    SDL_memset(mixbuf, spec->silence, 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(audio_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(audio_fd, AUDIO_CONTROL, &paud_control) < 0) {
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Can't start audio play\n");
#endif
        SDL_SetError("Can't start audio play");
        return -1;
    }

    /* Check to see if we need to use select() workaround */
    {
        char *workaround;
        workaround = SDL_getenv("SDL_DSP_NOSELECT");
        if (workaround) {
            frame_ticks = (float) (spec->samples * 1000) / spec->freq;
            next_frame = SDL_GetTicks() + frame_ticks;
        }
    }

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

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

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