Updated ALSA audio support for ALSA 0.9
authorSam Lantinga <slouken@libsdl.org>
Mon, 15 Apr 2002 07:38:54 +0000
changeset 354 30935e76acb5
parent 353 d47637068e52
child 355 b68a34dee49d
Updated ALSA audio support for ALSA 0.9
configure.in
src/audio/alsa/SDL_alsa_audio.c
src/audio/alsa/SDL_alsa_audio.h
src/audio/dma/SDL_dmaaudio.c
src/audio/dsp/SDL_dspaudio.c
--- a/configure.in	Mon Apr 15 04:53:41 2002 +0000
+++ b/configure.in	Mon Apr 15 07:38:54 2002 +0000
@@ -269,14 +269,25 @@
 [  --enable-alsa           support the ALSA audio API [default=yes]],
                   , enable_alsa=yes)
     if test x$enable_audio = xyes -a x$enable_alsa = xyes; then
+        AC_MSG_CHECKING(for ALSA audio support)
         have_alsa=no
-        AC_CHECK_HEADER(sys/asoundlib.h, have_alsa_hdr=yes)
-        AC_CHECK_LIB(asound, snd_pcm_open, have_alsa_lib=yes)
-        if test x$have_alsa_hdr = xyes -a x$have_alsa_lib = xyes; then
+        AC_TRY_COMPILE([
+          #include <alsa/asoundlib.h>
+        ],[
+          #if SND_LIB_VERSION < ((0<<16)|(9<<8)|0)
+          #error Your version of ALSA is too old
+          #endif
+          snd_pcm_t *pcm_handle;
+        ],[
+        AC_CHECK_LIB(asound, snd_pcm_open, have_alsa=yes)
+        ])
+        if test x$have_alsa = xyes; then
             CFLAGS="$CFLAGS -DALSA_SUPPORT"
             SYSTEM_LIBS="$SYSTEM_LIBS -lasound"
             AUDIO_SUBDIRS="$AUDIO_SUBDIRS alsa"
             AUDIO_DRIVERS="$AUDIO_DRIVERS alsa/libaudio_alsa.la"
+        else
+            AC_MSG_RESULT($have_alsa)
         fi
     fi
 }
--- a/src/audio/alsa/SDL_alsa_audio.c	Mon Apr 15 04:53:41 2002 +0000
+++ b/src/audio/alsa/SDL_alsa_audio.c	Mon Apr 15 07:38:54 2002 +0000
@@ -44,96 +44,41 @@
 /* The tag name used by ALSA audio */
 #define DRIVER_NAME         "alsa"
 
-/* default card and device numbers as listed in dev/snd */
-static int card_no = 0;
-static int device_no = 0;
-
-/* default channel communication parameters */
-#define DEFAULT_CPARAMS_RATE 22050
-#define DEFAULT_CPARAMS_VOICES 1
-#define DEFAULT_CPARAMS_FRAG_SIZE 512
-#define DEFAULT_CPARAMS_FRAGS_MIN 1
-#define DEFAULT_CPARAMS_FRAGS_MAX -1
-
-/* Open the audio device for playback, and don't block if busy */
-#define OPEN_FLAGS	(SND_PCM_OPEN_PLAYBACK|SND_PCM_OPEN_NONBLOCK)
+/* The default ALSA audio driver */
+#define DEFAULT_DEVICE	"plughw:0,0"
 
 /* Audio driver functions */
-static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void PCM_WaitAudio(_THIS);
-static void PCM_PlayAudio(_THIS);
-static Uint8 *PCM_GetAudioBuf(_THIS);
-static void PCM_CloseAudio(_THIS);
-
-/* PCM transfer channel parameters initialize function */
-static void init_pcm_cparams(snd_pcm_channel_params_t* cparams)
-{
-	memset(cparams,0,sizeof(snd_pcm_channel_params_t));
+static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec);
+static void ALSA_WaitAudio(_THIS);
+static void ALSA_PlayAudio(_THIS);
+static Uint8 *ALSA_GetAudioBuf(_THIS);
+static void ALSA_CloseAudio(_THIS);
 
-	cparams->channel = SND_PCM_CHANNEL_PLAYBACK;
-	cparams->mode = SND_PCM_MODE_BLOCK;
-	cparams->start_mode = SND_PCM_START_DATA; //_FULL
-	cparams->stop_mode  = SND_PCM_STOP_STOP;
-	cparams->format.format = SND_PCM_SFMT_S16_LE;
-	cparams->format.interleave = 1;
-	cparams->format.rate = DEFAULT_CPARAMS_RATE;
-	cparams->format.voices = DEFAULT_CPARAMS_VOICES;
-	cparams->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
-	cparams->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
-	cparams->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
+static const char *get_audio_device()
+{
+	const char *device;
+	
+	device = getenv("AUDIODEV");	/* Is there a standard variable name? */
+	if ( device == NULL ) {
+		device = DEFAULT_DEVICE;
+	}
+	return device;
 }
 
 /* Audio driver bootstrap functions */
 
 static int Audio_Available(void)
-/*
-	See if we can open a nonblocking channel.
-	Return value '1' means we can.
-	Return value '0' means we cannot.
-*/
 {
 	int available;
-	int rval;
+	int status;
 	snd_pcm_t *handle;
-	snd_pcm_channel_params_t cparams;
-#ifdef DEBUG_AUDIO
-	snd_pcm_channel_status_t cstatus;
-#endif
 
 	available = 0;
-	handle = NULL;
-
-	init_pcm_cparams(&cparams);
-	
-	rval = snd_pcm_open(&handle, card_no, device_no, OPEN_FLAGS);
-	if (rval >= 0)
-	{
-		rval = snd_pcm_plugin_params(handle, &cparams);
-
-#ifdef DEBUG_AUDIO
-		snd_pcm_plugin_status(handle, &cstatus);
-		printf("status after snd_pcm_plugin_params call = %d\n",cstatus.status);
-#endif
-		if (rval >= 0)
-		{
-			available = 1;
-		}
-		else
-		{
-	        	SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
-		}
-
-        if ((rval = snd_pcm_close(handle)) < 0)
-        {
-            SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
-			available = 0;
-        }
+	status = snd_pcm_open(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, 0);
+	if ( status >= 0 ) {
+		available = 1;
+        	snd_pcm_close(handle);
 	}
-	else
-	{
-       SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
-	}
-
 	return(available);
 }
 
@@ -162,14 +107,13 @@
 		return(0);
 	}
 	memset(this->hidden, 0, (sizeof *this->hidden));
-	audio_handle = NULL;
 
 	/* Set the function pointers */
-	this->OpenAudio = PCM_OpenAudio;
-	this->WaitAudio = PCM_WaitAudio;
-	this->PlayAudio = PCM_PlayAudio;
-	this->GetAudioBuf = PCM_GetAudioBuf;
-	this->CloseAudio = PCM_CloseAudio;
+	this->OpenAudio = ALSA_OpenAudio;
+	this->WaitAudio = ALSA_WaitAudio;
+	this->PlayAudio = ALSA_PlayAudio;
+	this->GetAudioBuf = ALSA_GetAudioBuf;
+	this->CloseAudio = ALSA_CloseAudio;
 
 	this->free = Audio_DeleteDevice;
 
@@ -177,14 +121,13 @@
 }
 
 AudioBootStrap ALSA_bootstrap = {
-	DRIVER_NAME, "ALSA PCM audio",
+	DRIVER_NAME, "ALSA 0.9 PCM audio",
 	Audio_Available, Audio_CreateDevice
 };
 
 /* This function waits until it is possible to write a full sound buffer */
-static void PCM_WaitAudio(_THIS)
+static void ALSA_WaitAudio(_THIS)
 {
-
 	/* Check to see if the thread-parent process is still alive */
 	{ static int cnt = 0;
 		/* Note that this only works with thread implementations 
@@ -196,322 +139,177 @@
 			}
 		}
 	}
-
-	/* 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 
-	{
-    	/* Use select() for audio synchronization */
-		fd_set fdset;
-	    struct timeval timeout;
-    	FD_ZERO(&fdset);
-	    FD_SET(audio_fd, &fdset);
-    	timeout.tv_sec = 10;
-	    timeout.tv_usec = 0;
-#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\n", 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 snd_pcm_channel_status_t cstatus;
-
-static void PCM_PlayAudio(_THIS)
+static void ALSA_PlayAudio(_THIS)
 {
-    int written, rval;
-
-    /* Write the audio data, checking for EAGAIN (buffer full) and underrun */
-    do {
-		written = snd_pcm_plugin_write(audio_handle, pcm_buf, pcm_len);
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "written = %d pcm_len = %d\n",written,pcm_len);
-#endif
-		if (written != pcm_len)
-		{
-	        if (errno == EAGAIN) 
-			{
-            	SDL_Delay(1);   /* Let a little CPU time go by and try to write again */
-#ifdef DEBUG_AUDIO
-				fprintf(stderr, "errno == EAGAIN\n");
-#endif
-        	}
-			else
-			{
-		        if( (rval = snd_pcm_plugin_status(audio_handle, &cstatus)) < 0 )
-        		{
-		            SDL_SetError("snd_pcm_plugin_status failed: %s\n", snd_strerror(rval));
-        		    return;
-		        }
-				if ( (cstatus.status == SND_PCM_STATUS_UNDERRUN)
-					||(cstatus.status == SND_PCM_STATUS_READY) )
-				{
-#ifdef DEBUG_AUDIO
-					fprintf(stderr, "buffer underrun\n");
-#endif
-					if ( (rval = snd_pcm_plugin_prepare (audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0 )
-					{
-						SDL_SetError("snd_pcm_plugin_prepare failed: %s\n",snd_strerror(rval) );
-						return;
-					}
-					/* if we reach here, try to write again */
-				}
+	int           status;
+	int           sample_len;
+	signed short *sample_buf;
+       
+	sample_len = this->spec.samples;
+	sample_buf = (signed short *)mixbuf;
+	while ( sample_len > 0 ) {
+		status = snd_pcm_writei(pcm_handle, sample_buf, sample_len);
+		if ( status < 0 ) {
+			if ( status == -EAGAIN ) {
+				continue;
+			}
+			if ( status == -ESTRPIPE ) {
+				do {
+					status = snd_pcm_resume(pcm_handle);
+				} while ( status == -EAGAIN );
+			}
+			if ( status < 0 ) {
+				status = snd_pcm_prepare(pcm_handle);
+			}
+			if ( status < 0 ) {
+				/* Hmm, not much we can do - abort */
+				this->enabled = 0;
+				return;
 			}
 		}
-    } while ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) );
-
-    /* Set the next write frame */
-   if ( frame_ticks ) {
-	    next_frame += frame_ticks;
+		sample_buf += status * this->spec.channels;
+		sample_len -= status;
 	}
-
-    /* If we couldn't write, assume fatal error for now */
-    if ( written < 0 ) {
-        this->enabled = 0;
-    }
-	return;
 }
 
-static Uint8 *PCM_GetAudioBuf(_THIS)
+static Uint8 *ALSA_GetAudioBuf(_THIS)
 {
-	return(pcm_buf);
+	return(mixbuf);
 }
 
-static void PCM_CloseAudio(_THIS)
+static void ALSA_CloseAudio(_THIS)
 {
-	int rval;
-
-	if ( pcm_buf != NULL ) {
-		free(pcm_buf);
-		pcm_buf = NULL;
+	if ( mixbuf != NULL ) {
+		SDL_FreeAudioMem(mixbuf);
+		mixbuf = NULL;
 	}
-	if ( audio_handle != NULL ) {
-		if ((rval = snd_pcm_plugin_flush(audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0)
-		{
-        	SDL_SetError("snd_pcm_plugin_flush failed: %s\n",snd_strerror(rval));
-			return;
-		}
-		if ((rval = snd_pcm_close(audio_handle)) < 0)
-		{
-			SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
-			return;
-		}
-		audio_handle = NULL;
+	if ( pcm_handle ) {
+		snd_pcm_close(pcm_handle);
+		pcm_handle = NULL;
 	}
 }
 
-static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec)
+static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
 {
-	int rval;
-	snd_pcm_channel_params_t cparams;
-	snd_pcm_channel_setup_t  csetup;
-	int format;
-	Uint16 test_format;
-	int twidth;
-
-	/* initialize channel transfer parameters to default */
-	init_pcm_cparams(&cparams);
-
-	/* Reset the timer synchronization flag */
-	frame_ticks = 0.0;
+	int                  status;
+	snd_pcm_hw_params_t *params;
+	snd_pcm_format_t     format;
+	snd_pcm_uframes_t    frames;
+	Uint16               test_format;
 
 	/* Open the audio device */
-	
-	rval = snd_pcm_open(&audio_handle, card_no, device_no, OPEN_FLAGS);
-	if ( rval < 0 ) {
-		SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
+	status = snd_pcm_open(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, 0);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't open audio device: %s", snd_strerror(status));
+		return(-1);
+	}
+
+	/* Figure out what the hardware is capable of */
+	snd_pcm_hw_params_alloca(&params);
+	status = snd_pcm_hw_params_any(pcm_handle, params);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't get hardware config: %s", snd_strerror(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+
+	/* SDL only uses interleaved sample output */
+	status = snd_pcm_hw_params_set_access(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't set interleaved access: %s", snd_strerror(status));
+		ALSA_CloseAudio(this);
 		return(-1);
 	}
 
-#ifdef PLUGIN_DISABLE_MMAP /* This is gone in newer versions of ALSA? */
-    /* disable count status parameter */
-    if ((rval = snd_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP))<0)
-    {
-        SDL_SetError("snd_plugin_set_disable failed: %s\n", snd_strerror(rval));
-        return(-1);
-    }
-#endif
-
-	pcm_buf = NULL;
-
 	/* Try for a closest match on audio format */
-	format = 0;
+	status = -1;
 	for ( test_format = SDL_FirstAudioFormat(spec->format);
-						! format && test_format; ) 
-	{
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "Trying format 0x%4.4x spec->samples %d\n", test_format,spec->samples);
-#endif
-			/* if match found set format to equivalent ALSA format */
-        switch ( test_format ) {
+	      test_format && (status < 0); ) {
+		switch ( test_format ) {
 			case AUDIO_U8:
-				format = SND_PCM_SFMT_U8;
-				cparams.buf.block.frag_size = spec->samples * spec->channels;
+				format = SND_PCM_FORMAT_U8;
 				break;
 			case AUDIO_S8:
-				format = SND_PCM_SFMT_S8;
-				cparams.buf.block.frag_size = spec->samples * spec->channels;
+				format = SND_PCM_FORMAT_S8;
 				break;
 			case AUDIO_S16LSB:
-				format = SND_PCM_SFMT_S16_LE;
-				cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
+				format = SND_PCM_FORMAT_S16_LE;
 				break;
 			case AUDIO_S16MSB:
-				format = SND_PCM_SFMT_S16_BE;
-				cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
+				format = SND_PCM_FORMAT_S16_BE;
 				break;
 			case AUDIO_U16LSB:
-				format = SND_PCM_SFMT_U16_LE;
-				cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
+				format = SND_PCM_FORMAT_U16_LE;
 				break;
 			case AUDIO_U16MSB:
-				format = SND_PCM_SFMT_U16_BE;
-				cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
+				format = SND_PCM_FORMAT_U16_BE;
 				break;
 			default:
+				format = 0;
 				break;
 		}
-		if ( ! format ) {
+		if ( format != 0 ) {
+			status = snd_pcm_hw_params_set_format(pcm_handle, params, format);
+		}
+		if ( status < 0 ) {
 			test_format = SDL_NextAudioFormat();
 		}
 	}
-	if ( format == 0 ) {
+	if ( status < 0 ) {
 		SDL_SetError("Couldn't find any hardware audio formats");
+		ALSA_CloseAudio(this);
 		return(-1);
 	}
 	spec->format = test_format;
 
-	/* Set the audio format */
-	cparams.format.format = format;
+	/* Set the number of channels */
+	status = snd_pcm_hw_params_set_channels(pcm_handle, params, spec->channels);
+	if ( status < 0 ) {
+		status = snd_pcm_hw_params_get_channels(params);
+		if ( (status <= 0) || (status > 2) ) {
+			SDL_SetError("Couldn't set audio channels");
+			ALSA_CloseAudio(this);
+			return(-1);
+		}
+		spec->channels = status;
+	}
 
-	/* Set mono or stereo audio (currently only two channels supported) */
-	cparams.format.voices = spec->channels;
-	
-	#ifdef DEBUG_AUDIO
-	printf("intializing channels %d\n", cparams.format.voices);
-	#endif
-	
-	/* Set rate */
-	cparams.format.rate = spec->freq ;
+	/* Set the audio rate */
+	status = snd_pcm_hw_params_set_rate_near(pcm_handle, params, spec->freq, NULL);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't set audio frequency: %s", snd_strerror(status));
+		ALSA_CloseAudio(this);
+		return(-1);
+	}
+	spec->freq = status;
 
-	/* Setup the transfer parameters according to cparams */
-	rval = snd_pcm_plugin_params(audio_handle, &cparams);
-	if (rval < 0) {
-		SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
+	/* Set the buffer size, in samples */
+	frames = spec->samples;
+	frames = snd_pcm_hw_params_set_period_size_near(pcm_handle, params, frames, NULL);
+	spec->samples = frames;
+	snd_pcm_hw_params_set_periods_near(pcm_handle, params, 2, NULL);
+
+	/* "set" the hardware with the desired parameters */
+	status = snd_pcm_hw_params(pcm_handle, params);
+	if ( status < 0 ) {
+		SDL_SetError("Couldn't set audio parameters: %s", snd_strerror(status));
+		ALSA_CloseAudio(this);
 		return(-1);
 	}
 
-    /*  Make sure channel is setup right one last time */
-    memset( &csetup, 0, sizeof( csetup ) );
-    csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
-    if ( snd_pcm_plugin_setup( audio_handle, &csetup ) < 0 )
-    {
-        SDL_SetError("Unable to setup playback channel\n" );
-        return(-1);
-    }
-
-#ifdef DEBUG_AUDIO
-    else
-    {
-        fprintf(stderr,"requested format: %d\n",cparams.format.format);
-        fprintf(stderr,"requested frag size: %d\n",cparams.buf.block.frag_size);
-        fprintf(stderr,"requested max frags: %d\n\n",cparams.buf.block.frags_max);
-
-        fprintf(stderr,"real format: %d\n", csetup.format.format );
-        fprintf(stderr,"real frag size : %d\n", csetup.buf.block.frag_size );
-		fprintf(stderr,"real max frags : %d\n", csetup.buf.block.frags_max );
-    }
-#endif // DEBUG_AUDIO
+	/* Calculate the final parameters for this audio specification */
+	SDL_CalculateAudioSpec(spec);
 
-    /*  Allocate memory to the audio buffer and initialize with silence
-        (Note that buffer size must be a multiple of fragment size, so find closest multiple)
-    */
-    
-    twidth = snd_pcm_format_width(format);
-    if (twidth < 0) {
-        printf("snd_pcm_format_width failed\n");
-        twidth = 0;
-    }
-#ifdef DEBUG_AUDIO
-    printf("format is %d bits wide\n",twidth);
-#endif      
-    
-    pcm_len = csetup.buf.block.frag_size * (twidth/8) * csetup.format.voices ;
-    
-#ifdef DEBUG_AUDIO    
-    printf("pcm_len set to %d\n", pcm_len);
-#endif
-    
-    if (pcm_len == 0)
-    {
-        pcm_len = csetup.buf.block.frag_size;
-    }
-    
-    pcm_buf = (Uint8*)malloc(pcm_len);
-    if (pcm_buf == NULL) {
-        SDL_SetError("pcm_buf malloc failed\n");
-        return(-1);
-    }
-    memset(pcm_buf,spec->silence,pcm_len);
-
-#ifdef DEBUG_AUDIO
-	fprintf(stderr,"pcm_buf malloced and silenced.\n");
-#endif
-
-    /* get the file descriptor */
-    if( (audio_fd = snd_pcm_file_descriptor(audio_handle, device_no)) < 0)
-    {
-       fprintf(stderr, "snd_pcm_file_descriptor failed with error code: %d\n", audio_fd);
-    }
-
-	/* Trigger audio playback */
-	rval = snd_pcm_plugin_prepare( audio_handle, SND_PCM_CHANNEL_PLAYBACK);
-	if (rval < 0) {
-       SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror (rval));
-       return(-1);
+	/* Allocate mixing buffer */
+	mixlen = spec->size;
+	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
+	if ( mixbuf == NULL ) {
+		ALSA_CloseAudio(this);
+		return(-1);
 	}
-	rval =  snd_pcm_playback_go(audio_handle);
-    if (rval < 0) {
-       SDL_SetError("snd_pcm_playback_go failed: %s\n", snd_strerror (rval));
-       return(-1);
-    }
-
-    /* Check to see if we need to use select() workaround */
-    { char *workaround;
-        workaround = getenv("SDL_DSP_NOSELECT");
-        if ( workaround ) {
-            frame_ticks = (float)(spec->samples*1000)/spec->freq;
-            next_frame = SDL_GetTicks()+frame_ticks;
-        }
-    }
+	memset(mixbuf, spec->silence, spec->size);
 
 	/* Get the parent process id (we're the parent of the audio thread) */
 	parent = getpid();
--- a/src/audio/alsa/SDL_alsa_audio.h	Mon Apr 15 04:53:41 2002 +0000
+++ b/src/audio/alsa/SDL_alsa_audio.h	Mon Apr 15 07:38:54 2002 +0000
@@ -24,38 +24,27 @@
 #define _ALSA_PCM_audio_h
 
 #include "SDL_sysaudio.h"
-#include <sys/asoundlib.h>
+#include <alsa/asoundlib.h>
 
 /* Hidden "this" pointer for the video functions */
 #define _THIS	SDL_AudioDevice *this
 
 struct SDL_PrivateAudioData {
 	/* The audio device handle */
-	 snd_pcm_t *audio_handle;
-
-	/* The audio file descriptor */
-	int audio_fd;
+	snd_pcm_t *pcm_handle;
 
 	/* The parent process id, to detect when application quits */
 	pid_t parent;
 
 	/* Raw mixing buffer */
-	Uint8 *pcm_buf;
-	int    pcm_len;
-
-	/* Support for audio timing using a timer, in addition to select() */
-	float frame_ticks;
-	float next_frame;
+	Uint8 *mixbuf;
+	int    mixlen;
 };
-#define FUDGE_TICKS	10	/* The scheduler overhead ticks per frame */
 
 /* Old variable names */
-#define audio_handle	(this->hidden->audio_handle)
-#define audio_fd		(this->hidden->audio_fd)
+#define pcm_handle		(this->hidden->pcm_handle)
 #define parent			(this->hidden->parent)
-#define pcm_buf			(this->hidden->pcm_buf)
-#define pcm_len			(this->hidden->pcm_len)
-#define frame_ticks		(this->hidden->frame_ticks)
-#define next_frame		(this->hidden->next_frame)
+#define mixbuf			(this->hidden->mixbuf)
+#define mixlen			(this->hidden->mixlen)
 
 #endif /* _ALSA_PCM_audio_h */
--- a/src/audio/dma/SDL_dmaaudio.c	Mon Apr 15 04:53:41 2002 +0000
+++ b/src/audio/dma/SDL_dmaaudio.c	Mon Apr 15 07:38:54 2002 +0000
@@ -377,6 +377,7 @@
 				}
 				break;
 			default:
+				format = 0;
 				break;
 		}
 		if ( ! format ) {
--- a/src/audio/dsp/SDL_dspaudio.c	Mon Apr 15 04:53:41 2002 +0000
+++ b/src/audio/dsp/SDL_dspaudio.c	Mon Apr 15 07:38:54 2002 +0000
@@ -394,6 +394,7 @@
 				}
 				break;
 			default:
+				format = 0;
 				break;
 		}
 		if ( ! format ) {