view src/audio/ums/SDL_umsaudio.c @ 942:41a59de7f2ed

Here are patches for SDL12 and SDL_mixer for 4 or 6 channel surround sound on Linux using the Alsa driver. To use them, naturally you need a sound card that will do 4 or 6 channels and probably also a recent version of the Alsa drivers and library. Since the only SDL output driver that knows about surround sound is the Alsa driver, you���ll want to choose it, using: export SDL_AUDIODRIVER=alsa There are no syntactic changes to the programming API. No new library calls, no differences in arguments. There are two semantic changes: (1) For library calls with number of channels as an argument, formerly you could use only 1 or 2 for the number of channels. Now you can also use 4 or 6. (2) The two "left" and "right" arguments to Mix_SetPanning, for the case of 4 or 6 channels, no longer simply control the volumes of the left and right channels. Now the "left" argument is converted to an angle and Mix_SetPosition is called, and the "right" argu- ment is ignored. With two exceptions, so far as I know, the modified SDL12 and SDL_mixer work the same way as the original versions, when opened for 1 or 2 channel output. The two exceptions are bugs which I fixed. Well, the first, anyway, is a bug for sure. When rate conversions up or down by a factor of two are applied (in src/audio/SDL_audiocvt.c), streams with different numbers of channels (that is, mono and stereo) are treated the same way: either each sample is copied or every other sample is omitted. This is ok for mono, but for stereo, it is frames that should be copied or omitted, where by "frame" I mean a portion of the stream containing one sample for each channel. (In the SDL source, confusingly, sometimes frames are called "samples".) So for these rate conversions, stereo streams have to be treated differently, and they are, in my modified version. The other problem that might be characterized as a bug arises when SDL_mixer is passed a multichannel chunk which does not have an integral number of frames. Due to the way the effect_position code loops over frames, when the chunk ends with a partial frame, memory outside the chunk buffer will be accessed. In the case of stereo, it���s possible that because malloc may give more memory than requested, this potential problem never actually causes a segment fault. I don���t know. For 6 channel chunks, I do know, and it does cause segment faults. If SDL_mixer is passed defective chunks and this causes a segment fault, arguably, that���s not a bug in SDL_mixer. Still, whether or not it counts as a bug, it���s easy to protect against, so why not? I added code in mixer.c to discard any partial frame at the end of a chunk. Then what about when SDL or SDL_mixer is opened for 4 or 6 chan- nel output? What happens with the parts of the current library designed for stereo? I don���t know whether I���ve covered all the bases, but I���ve tried: (1) For playing 2 channel waves, or other cases where SDL knows it has to match up a 2 channel source with a 4 or 6 channel output, I���ve added code in SDL_audiocvt.c to make the necessary conversions. (2) For playing midis using timidity, I���ve converted timidity to do 4 or 6 channel output, upon request. (3) For playing mods using mikmod, I put ad hoc code in music.c to convert the stereo output that mikmod produces to 4 or 6 chan- nels. Obviously it would be better to change the mikmod code to mix down into 4 or 6 channels, but I have a hard time following the code in mikmod, so I didn���t do that. (4) For playing mp3s, I put ad hoc code in smpeg to copy channels in the case when 4 or 6 channel output is needed. (5) There seems to be no problem with .ogg files - stereo .oggs can be up converted as .wavs are. (6) The effect_position code in SDL_mixer is now generalized to in- clude the cases of 4 and 6 channel streams. I���ve done a very limited amount of compatibility testing for some of the games using SDL I happen to have. For details, see the file TESTS. I���ve put into a separate archive, Surround-SDL-testfiles.tgz, a couple of 6 channel wave files for testing and a 6 channel ogg file. If you have the right hardware and version of Alsa, you should be able to play the wave files with the Alsa utility aplay (and hear all channels, except maybe lfe, for chan-id.wav, since it���s rather faint). Don���t expect aplay to give good sound, though. There���s something wrong with the current version of aplay. The canyon.ogg file is to test loading of 6 channel oggs. After patching and compiling, you can play it with playmus. (My version of ogg123 will not play it, and I had to patch mplayer to get it to play 6 channel oggs.) Greg Lee <greg@ling.lll.hawaii.edu> Thus, July 1, 2004
author Sam Lantinga <slouken@libsdl.org>
date Sat, 21 Aug 2004 12:27:02 +0000
parents 74212992fb08
children c9b51268668f
line wrap: on
line source

/*
    AIX support for the SDL - Simple DirectMedia Layer
    Copyright (C) 2000  Carsten Griwodz

    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

    Carsten Griwodz
    griff@kom.tu-darmstadt.de

    based on linux/SDL_dspaudio.c by Sam Lantinga
*/

#ifdef SAVE_RCSID
static char rcsid =
 "@(#) $Id$";
#endif

/* Allow access to a raw mixing buffer */

#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/types.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <sys/stat.h>
#include <sys/mman.h>

#include "SDL_audio.h"
#include "SDL_error.h"
#include "SDL_audio_c.h"
#include "SDL_audiodev_c.h"
#include "SDL_umsaudio.h"

/* The tag name used by UMS audio */
#define UMS_DRIVER_NAME         "ums"

#define DEBUG_AUDIO 1

/* Audio driver functions */
static int UMS_OpenAudio(_THIS, SDL_AudioSpec *spec);
static void UMS_PlayAudio(_THIS);
static Uint8 *UMS_GetAudioBuf(_THIS);
static void UMS_CloseAudio(_THIS);

static UMSAudioDevice_ReturnCode UADOpen(_THIS,  string device, string mode, long flags);
static UMSAudioDevice_ReturnCode UADClose(_THIS);
static UMSAudioDevice_ReturnCode UADGetBitsPerSample(_THIS, long* bits);
static UMSAudioDevice_ReturnCode UADSetBitsPerSample(_THIS, long bits);
static UMSAudioDevice_ReturnCode UADSetSampleRate(_THIS, long rate, long* set_rate);
static UMSAudioDevice_ReturnCode UADSetByteOrder(_THIS, string byte_order);
static UMSAudioDevice_ReturnCode UADSetAudioFormatType(_THIS, string fmt);
static UMSAudioDevice_ReturnCode UADSetNumberFormat(_THIS, string fmt);
static UMSAudioDevice_ReturnCode UADInitialize(_THIS);
static UMSAudioDevice_ReturnCode UADStart(_THIS);
static UMSAudioDevice_ReturnCode UADStop(_THIS);
static UMSAudioDevice_ReturnCode UADSetTimeFormat(_THIS,  UMSAudioTypes_TimeFormat fmt );
static UMSAudioDevice_ReturnCode UADWriteBuffSize(_THIS,  long* buff_size );
static UMSAudioDevice_ReturnCode UADWriteBuffRemain(_THIS,  long* buff_size );
static UMSAudioDevice_ReturnCode UADWriteBuffUsed(_THIS,  long* buff_size );
static UMSAudioDevice_ReturnCode UADSetDMABufferSize(_THIS,  long bytes, long* bytes_ret );
static UMSAudioDevice_ReturnCode UADSetVolume(_THIS,  long volume );
static UMSAudioDevice_ReturnCode UADSetBalance(_THIS,  long balance );
static UMSAudioDevice_ReturnCode UADSetChannels(_THIS,  long channels );
static UMSAudioDevice_ReturnCode UADPlayRemainingData(_THIS,  boolean block );
static UMSAudioDevice_ReturnCode UADEnableOutput(_THIS,  string output, long* left_gain, long* right_gain);
static UMSAudioDevice_ReturnCode UADWrite(_THIS,  UMSAudioTypes_Buffer* buff, long samples, long* samples_written);

/* Audio driver bootstrap functions */
static int Audio_Available(void)
{
    return 1;
}

static void Audio_DeleteDevice(_THIS)
{
    if(this->hidden->playbuf._buffer) free(this->hidden->playbuf._buffer);
    if(this->hidden->fillbuf._buffer) free(this->hidden->fillbuf._buffer);
    _somFree( this->hidden->umsdev );
    free(this->hidden);
    free(this);
}

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

    /*
     * Allocate and initialize management storage and private management
     * storage for this SDL-using library.
     */
    this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
    if ( this ) {
        memset(this, 0, (sizeof *this));
        this->hidden = (struct SDL_PrivateAudioData *)malloc((sizeof *this->hidden));
    }
    if ( (this == NULL) || (this->hidden == NULL) ) {
        SDL_OutOfMemory();
        if ( this ) {
            free(this);
        }
        return(0);
    }
    memset(this->hidden, 0, (sizeof *this->hidden));
#ifdef DEBUG_AUDIO
    fprintf(stderr, "Creating UMS Audio device\n");
#endif

    /*
     * Calls for UMS env initialization and audio object construction.
     */
    this->hidden->ev     = somGetGlobalEnvironment();
    this->hidden->umsdev = UMSAudioDeviceNew();

    /*
     * Set the function pointers.
     */
    this->OpenAudio   = UMS_OpenAudio;
    this->WaitAudio   = NULL;           /* we do blocking output */
    this->PlayAudio   = UMS_PlayAudio;
    this->GetAudioBuf = UMS_GetAudioBuf;
    this->CloseAudio  = UMS_CloseAudio;
    this->free        = Audio_DeleteDevice;

#ifdef DEBUG_AUDIO
    fprintf(stderr, "done\n");
#endif
    return this;
}

AudioBootStrap UMS_bootstrap = {
	UMS_DRIVER_NAME, "AUX UMS audio",
	Audio_Available, Audio_CreateDevice
};

static Uint8 *UMS_GetAudioBuf(_THIS)
{
#ifdef DEBUG_AUDIO
    fprintf(stderr, "enter UMS_GetAudioBuf\n");
#endif
    return this->hidden->fillbuf._buffer;
/*
    long                      bufSize;
    UMSAudioDevice_ReturnCode rc;

    rc = UADSetTimeFormat(this, UMSAudioTypes_Bytes );
    rc = UADWriteBuffSize(this,  bufSize );
*/
}

static void UMS_CloseAudio(_THIS)
{
    UMSAudioDevice_ReturnCode rc;

#ifdef DEBUG_AUDIO
    fprintf(stderr, "enter UMS_CloseAudio\n");
#endif
    rc = UADPlayRemainingData(this, TRUE);
    rc = UADStop(this);
    rc = UADClose(this);
}

static void UMS_PlayAudio(_THIS)
{
    UMSAudioDevice_ReturnCode rc;
    long                      samplesToWrite;
    long                      samplesWritten;
    UMSAudioTypes_Buffer      swpbuf;

#ifdef DEBUG_AUDIO
    fprintf(stderr, "enter UMS_PlayAudio\n");
#endif
    samplesToWrite = this->hidden->playbuf._length/this->hidden->bytesPerSample;
    do
    {
        rc = UADWrite(this,  &this->hidden->playbuf,
		       samplesToWrite,
	               &samplesWritten );
	samplesToWrite -= samplesWritten;

	/* rc values: UMSAudioDevice_Success
	 *            UMSAudioDevice_Failure
	 *            UMSAudioDevice_Preempted
	 *            UMSAudioDevice_Interrupted
	 *            UMSAudioDevice_DeviceError
	 */
	if ( rc == UMSAudioDevice_DeviceError ) {
#ifdef DEBUG_AUDIO
	    fprintf(stderr, "Returning from PlayAudio with devices error\n");
#endif
	    return;
	}
    }
    while(samplesToWrite>0);

    SDL_LockAudio();
    memcpy( &swpbuf,                &this->hidden->playbuf, sizeof(UMSAudioTypes_Buffer) );
    memcpy( &this->hidden->playbuf, &this->hidden->fillbuf, sizeof(UMSAudioTypes_Buffer) );
    memcpy( &this->hidden->fillbuf, &swpbuf,                sizeof(UMSAudioTypes_Buffer) );
    SDL_UnlockAudio();

#ifdef DEBUG_AUDIO
    fprintf(stderr, "Wrote audio data and swapped buffer\n");
#endif
}

#if 0
// 	/* Set the DSP frequency */
// 	value = spec->freq;
// 	if ( ioctl(this->hidden->audio_fd, SOUND_PCM_WRITE_RATE, &value) < 0 ) {
// 		SDL_SetError("Couldn't set audio frequency");
// 		return(-1);
// 	}
// 	spec->freq = value;
#endif

static int UMS_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
    char*  audiodev = "/dev/paud0";
    long   lgain;
    long   rgain;
    long   outRate;
    long   outBufSize;
    long   bitsPerSample;
    long   samplesPerSec;
    long   success;
    Uint16 test_format;
    int    frag_spec;
    UMSAudioDevice_ReturnCode rc;

#ifdef DEBUG_AUDIO
    fprintf(stderr, "enter UMS_OpenAudio\n");
#endif
    rc = UADOpen(this, audiodev,"PLAY", UMSAudioDevice_BlockingIO);
    if ( rc != UMSAudioDevice_Success ) {
	SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
	return -1;
    }
 
    rc = UADSetAudioFormatType(this, "PCM"); 

    success = 0;
    test_format = SDL_FirstAudioFormat(spec->format);
    do
    {
#ifdef DEBUG_AUDIO
        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
        switch ( test_format )
        {
        case AUDIO_U8:
/* from the mac code: better ? */
/* sample_bits = spec->size / spec->samples / spec->channels * 8; */
	    success       = 1;
            bitsPerSample = 8;
            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
            rc = UADSetByteOrder(this, "MSB");       /* irrelevant */
            rc = UADSetNumberFormat(this, "UNSIGNED");
            break;
        case AUDIO_S8:
	    success       = 1;
            bitsPerSample = 8;
            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
            rc = UADSetByteOrder(this, "MSB");       /* irrelevant */
            rc = UADSetNumberFormat(this, "SIGNED");
            break;
        case AUDIO_S16LSB:
	    success       = 1;
            bitsPerSample = 16;
            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
            rc = UADSetByteOrder(this, "LSB");
            rc = UADSetNumberFormat(this, "SIGNED");
            break;
        case AUDIO_S16MSB:
	    success       = 1;
            bitsPerSample = 16;
            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
            rc = UADSetByteOrder(this, "MSB");
            rc = UADSetNumberFormat(this, "SIGNED");
            break;
        case AUDIO_U16LSB:
	    success       = 1;
            bitsPerSample = 16;
            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
            rc = UADSetByteOrder(this, "LSB");
            rc = UADSetNumberFormat(this, "UNSIGNED");
            break;
        case AUDIO_U16MSB:
	    success       = 1;
            bitsPerSample = 16;
            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
            rc = UADSetByteOrder(this, "MSB");
            rc = UADSetNumberFormat(this, "UNSIGNED");
            break;
        default:
            break;
        }
        if ( ! success ) {
            test_format = SDL_NextAudioFormat();
        }
    }
    while ( ! success && test_format );

    if ( success == 0 ) {
        SDL_SetError("Couldn't find any hardware audio formats");
        return -1;
    }

    spec->format = test_format;

    for ( frag_spec = 0; (0x01<<frag_spec) < spec->size; ++frag_spec );
    if ( (0x01<<frag_spec) != spec->size ) {
        SDL_SetError("Fragment size must be a power of two");
        return -1;
    }
    if ( frag_spec > 2048 ) frag_spec = 2048;

    this->hidden->bytesPerSample   = (bitsPerSample / 8) * spec->channels;
    samplesPerSec                  = this->hidden->bytesPerSample * outRate;

    this->hidden->playbuf._length  = 0;
    this->hidden->playbuf._maximum = spec->size;
    this->hidden->playbuf._buffer  = (unsigned char*)malloc(spec->size);
    this->hidden->fillbuf._length  = 0;
    this->hidden->fillbuf._maximum = spec->size;
    this->hidden->fillbuf._buffer  = (unsigned char*)malloc(spec->size);

    rc = UADSetBitsPerSample(this,  bitsPerSample );
    rc = UADSetDMABufferSize(this,  frag_spec, &outBufSize );
    rc = UADSetChannels(this, spec->channels);      /* functions reduces to mono or stereo */

    lgain = 100; /*maximum left input gain*/
    rgain = 100; /*maimum right input gain*/
    rc = UADEnableOutput(this, "LINE_OUT",&lgain,&rgain);
    rc = UADInitialize(this);
    rc = UADStart(this);
    rc = UADSetVolume(this, 100);
    rc = UADSetBalance(this, 0);

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


static UMSAudioDevice_ReturnCode UADGetBitsPerSample(_THIS, long* bits)
{
    return UMSAudioDevice_get_bits_per_sample( this->hidden->umsdev,
					       this->hidden->ev,
					       bits );
}

static UMSAudioDevice_ReturnCode UADSetBitsPerSample(_THIS, long bits)
{
    return UMSAudioDevice_set_bits_per_sample( this->hidden->umsdev,
					       this->hidden->ev,
					       bits );
}

static UMSAudioDevice_ReturnCode UADSetSampleRate(_THIS, long rate, long* set_rate)
{
    /* from the mac code: sample rate = spec->freq << 16; */
    return UMSAudioDevice_set_sample_rate( this->hidden->umsdev,
					   this->hidden->ev,
					   rate,
					   set_rate );
}

static UMSAudioDevice_ReturnCode UADSetByteOrder(_THIS, string byte_order)
{
    return UMSAudioDevice_set_byte_order( this->hidden->umsdev,
					  this->hidden->ev,
					  byte_order );
}

static UMSAudioDevice_ReturnCode UADSetAudioFormatType(_THIS, string fmt)
{
    /* possible PCM, A_LAW or MU_LAW */
    return UMSAudioDevice_set_audio_format_type( this->hidden->umsdev,
						 this->hidden->ev,
						 fmt );
}

static UMSAudioDevice_ReturnCode UADSetNumberFormat(_THIS, string fmt)
{
    /* possible SIGNED, UNSIGNED, or TWOS_COMPLEMENT */
    return UMSAudioDevice_set_number_format( this->hidden->umsdev,
					     this->hidden->ev,
					     fmt );
}

static UMSAudioDevice_ReturnCode UADInitialize(_THIS)
{
    return UMSAudioDevice_initialize( this->hidden->umsdev,
				      this->hidden->ev );
}

static UMSAudioDevice_ReturnCode UADStart(_THIS)
{
    return UMSAudioDevice_start( this->hidden->umsdev,
				 this->hidden->ev );
}

static UMSAudioDevice_ReturnCode UADSetTimeFormat(_THIS,  UMSAudioTypes_TimeFormat fmt )
{
    /*
     * Switches the time format to the new format, immediately.
     * possible UMSAudioTypes_Msecs, UMSAudioTypes_Bytes or UMSAudioTypes_Samples
     */
    return UMSAudioDevice_set_time_format( this->hidden->umsdev,
					   this->hidden->ev,
					   fmt );
}

static UMSAudioDevice_ReturnCode UADWriteBuffSize(_THIS,  long* buff_size )
{
    /*
     * returns write buffer size in the current time format
     */
    return UMSAudioDevice_write_buff_size( this->hidden->umsdev,
                                           this->hidden->ev,
					   buff_size );
}

static UMSAudioDevice_ReturnCode UADWriteBuffRemain(_THIS,  long* buff_size )
{
    /*
     * returns amount of available space in the write buffer
     * in the current time format
     */
    return UMSAudioDevice_write_buff_remain( this->hidden->umsdev,
                                             this->hidden->ev,
					     buff_size );
}

static UMSAudioDevice_ReturnCode UADWriteBuffUsed(_THIS,  long* buff_size )
{
    /*
     * returns amount of filled space in the write buffer
     * in the current time format
     */
    return UMSAudioDevice_write_buff_used( this->hidden->umsdev,
                                           this->hidden->ev,
					   buff_size );
}

static UMSAudioDevice_ReturnCode UADSetDMABufferSize(_THIS,  long bytes, long* bytes_ret )
{
    /*
     * Request a new DMA buffer size, maximum requested size 2048.
     * Takes effect with next initialize() call.
     * Devices may or may not support DMA.
     */
    return UMSAudioDevice_set_DMA_buffer_size( this->hidden->umsdev,
					       this->hidden->ev,
					       bytes,
					       bytes_ret );
}

static UMSAudioDevice_ReturnCode UADSetVolume(_THIS,  long volume )
{
    /*
     * Set the volume.
     * Takes effect immediately.
     */
    return UMSAudioDevice_set_volume( this->hidden->umsdev,
				      this->hidden->ev,
				      volume );
}

static UMSAudioDevice_ReturnCode UADSetBalance(_THIS,  long balance )
{
    /*
     * Set the balance.
     * Takes effect immediately.
     */
    return UMSAudioDevice_set_balance( this->hidden->umsdev,
				       this->hidden->ev,
				       balance );
}

static UMSAudioDevice_ReturnCode UADSetChannels(_THIS,  long channels )
{
    /*
     * Set mono or stereo.
     * Takes effect with next initialize() call.
     */
    if ( channels != 1 ) channels = 2;
    return UMSAudioDevice_set_number_of_channels( this->hidden->umsdev,
				                  this->hidden->ev,
				                  channels );
}

static UMSAudioDevice_ReturnCode UADOpen(_THIS,  string device, string mode, long flags)
{
    return UMSAudioDevice_open( this->hidden->umsdev,
				this->hidden->ev,
				device,
				mode,
				flags );
}

static UMSAudioDevice_ReturnCode UADWrite(_THIS,  UMSAudioTypes_Buffer* buff,
                                           long samples,
					   long* samples_written)
{
    return UMSAudioDevice_write( this->hidden->umsdev,
				 this->hidden->ev,
				 buff,
				 samples,
				 samples_written );
}

static UMSAudioDevice_ReturnCode UADPlayRemainingData(_THIS,  boolean block )
{
    return UMSAudioDevice_play_remaining_data( this->hidden->umsdev,
					       this->hidden->ev,
					       block);
}

static UMSAudioDevice_ReturnCode UADStop(_THIS)
{
    return UMSAudioDevice_stop( this->hidden->umsdev,
				this->hidden->ev );
}

static UMSAudioDevice_ReturnCode UADClose(_THIS)
{
    return UMSAudioDevice_close( this->hidden->umsdev,
				 this->hidden->ev );
}

static UMSAudioDevice_ReturnCode UADEnableOutput(_THIS,  string output, long* left_gain, long* right_gain)
{
    return UMSAudioDevice_enable_output( this->hidden->umsdev,
					 this->hidden->ev,
					 output,
					 left_gain,
					 right_gain );
}