src/audio/paudio/SDL_paudio.c
changeset 0 74212992fb08
child 1312 c9b51268668f
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/src/audio/paudio/SDL_paudio.c	Thu Apr 26 16:45:43 2001 +0000
@@ -0,0 +1,519 @@
+/*
+    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/time.h>
+#include <sys/ioctl.h>
+#include <sys/stat.h>
+
+#include "SDL_audio.h"
+#include "SDL_error.h"
+#include "SDL_audiomem.h"
+#include "SDL_audio_c.h"
+#include "SDL_timer.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)
+{
+	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_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;
+	Uint16        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;
+	}
+	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 = 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;
+}
+