--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/src/audio/alsa/SDL_alsa_audio.c	Thu Apr 26 16:45:43 2001 +0000
@@ -0,0 +1,521 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997, 1998, 1999, 2000, 2001  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@devolution.com
+*/
+
+
+
+/* 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 <signal.h>
+#include <sys/types.h>
+#include <sys/time.h>
+
+#include "SDL_audio.h"
+#include "SDL_error.h"
+#include "SDL_audiomem.h"
+#include "SDL_audio_c.h"
+#include "SDL_timer.h"
+#include "SDL_alsa_audio.h"
+
+/* 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)
+
+/* 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));
+
+	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;
+}
+
+/* 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;
+	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;
+        }
+	}
+	else
+	{
+       SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
+	}
+
+	return(available);
+}
+
+static void Audio_DeleteDevice(SDL_AudioDevice *device)
+{
+	free(device->hidden);
+	free(device);
+}
+
+static SDL_AudioDevice *Audio_CreateDevice(int devindex)
+{
+	SDL_AudioDevice *this;
+
+	/* Initialize all variables that we clean on shutdown */
+	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));
+	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->free = Audio_DeleteDevice;
+
+	return this;
+}
+
+AudioBootStrap ALSA_bootstrap = {
+	DRIVER_NAME, "ALSA PCM audio",
+	Audio_Available, Audio_CreateDevice
+};
+
+/* This function waits until it is possible to write a full sound buffer */
+static void PCM_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 
+		   that use a different process id for each thread.
+		*/
+		if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
+			if ( kill(parent, 0) < 0 ) {
+				this->enabled = 0;
+			}
+		}
+	}
+
+	/* 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)
+{
+    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 */
+				}
+			}
+		}
+    } while ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) );
+
+    /* 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;
+    }
+	return;
+}
+
+static Uint8 *PCM_GetAudioBuf(_THIS)
+{
+	return(pcm_buf);
+}
+
+static void PCM_CloseAudio(_THIS)
+{
+	int rval;
+
+	if ( pcm_buf != NULL ) {
+		free(pcm_buf);
+		pcm_buf = 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;
+	}
+}
+
+static int PCM_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;
+
+	/* 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));
+		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;
+	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 ) {
+			case AUDIO_U8:
+				format = SND_PCM_SFMT_U8;
+				cparams.buf.block.frag_size = spec->samples * spec->channels;
+				break;
+			case AUDIO_S8:
+				format = SND_PCM_SFMT_S8;
+				cparams.buf.block.frag_size = spec->samples * spec->channels;
+				break;
+			case AUDIO_S16LSB:
+				format = SND_PCM_SFMT_S16_LE;
+				cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
+				break;
+			case AUDIO_S16MSB:
+				format = SND_PCM_SFMT_S16_BE;
+				cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
+				break;
+			case AUDIO_U16LSB:
+				format = SND_PCM_SFMT_U16_LE;
+				cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
+				break;
+			case AUDIO_U16MSB:
+				format = SND_PCM_SFMT_U16_BE;
+				cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
+				break;
+			default:
+				break;
+		}
+		if ( ! format ) {
+			test_format = SDL_NextAudioFormat();
+		}
+	}
+	if ( format == 0 ) {
+		SDL_SetError("Couldn't find any hardware audio formats");
+		return(-1);
+	}
+	spec->format = test_format;
+
+	/* Set the audio format */
+	cparams.format.format = format;
+
+	/* 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 ;
+
+	/* 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));
+		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
+
+    /*  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);
+	}
+	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;
+        }
+    }
+
+	/* Get the parent process id (we're the parent of the audio thread) */
+	parent = getpid();
+
+	/* We're ready to rock and roll. :-) */
+	return(0);
+}