1/*
2    SDL - Simple DirectMedia Layer
3    Copyright (C) 1997-2012 Sam Lantinga
4
5    This library is free software; you can redistribute it and/or
6    modify it under the terms of the GNU Lesser General Public
7    License as published by the Free Software Foundation; either
8    version 2.1 of the License, or (at your option) any later version.
9
10    This library is distributed in the hope that it will be useful,
11    but WITHOUT ANY WARRANTY; without even the implied warranty of
12    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
13    Lesser General Public License for more details.
14
15    You should have received a copy of the GNU Lesser General Public
16    License along with this library; if not, write to the Free Software
17    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
18
19    Carsten Griwodz
20    griff@kom.tu-darmstadt.de
21
22    based on linux/SDL_dspaudio.c by Sam Lantinga
23*/
24#include "SDL_config.h"
25
26/* Allow access to a raw mixing buffer */
27
28#include <errno.h>
29#include <unistd.h>
30#include <fcntl.h>
31#include <sys/time.h>
32#include <sys/ioctl.h>
33#include <sys/stat.h>
34
35#include "SDL_timer.h"
36#include "SDL_audio.h"
37#include "../SDL_audiomem.h"
38#include "../SDL_audio_c.h"
39#include "../SDL_audiodev_c.h"
40#include "SDL_paudio.h"
41
42#define DEBUG_AUDIO 1
43
44/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.
45 * I guess nobody ever uses audio... Shame over AIX header files.  */
46#include <sys/machine.h>
47#undef BIG_ENDIAN
48#include <sys/audio.h>
49
50/* The tag name used by paud audio */
51#define Paud_DRIVER_NAME         "paud"
52
53/* Open the audio device for playback, and don't block if busy */
54/* #define OPEN_FLAGS	(O_WRONLY|O_NONBLOCK) */
55#define OPEN_FLAGS	O_WRONLY
56
57/* Audio driver functions */
58static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec);
59static void Paud_WaitAudio(_THIS);
60static void Paud_PlayAudio(_THIS);
61static Uint8 *Paud_GetAudioBuf(_THIS);
62static void Paud_CloseAudio(_THIS);
63
64/* Audio driver bootstrap functions */
65
66static int Audio_Available(void)
67{
68	int fd;
69	int available;
70
71	available = 0;
72	fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
73	if ( fd >= 0 ) {
74		available = 1;
75		close(fd);
76	}
77	return(available);
78}
79
80static void Audio_DeleteDevice(SDL_AudioDevice *device)
81{
82	SDL_free(device->hidden);
83	SDL_free(device);
84}
85
86static SDL_AudioDevice *Audio_CreateDevice(int devindex)
87{
88	SDL_AudioDevice *this;
89
90	/* Initialize all variables that we clean on shutdown */
91	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
92	if ( this ) {
93		SDL_memset(this, 0, (sizeof *this));
94		this->hidden = (struct SDL_PrivateAudioData *)
95				SDL_malloc((sizeof *this->hidden));
96	}
97	if ( (this == NULL) || (this->hidden == NULL) ) {
98		SDL_OutOfMemory();
99		if ( this ) {
100			SDL_free(this);
101		}
102		return(0);
103	}
104	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
105	audio_fd = -1;
106
107	/* Set the function pointers */
108	this->OpenAudio = Paud_OpenAudio;
109	this->WaitAudio = Paud_WaitAudio;
110	this->PlayAudio = Paud_PlayAudio;
111	this->GetAudioBuf = Paud_GetAudioBuf;
112	this->CloseAudio = Paud_CloseAudio;
113
114	this->free = Audio_DeleteDevice;
115
116	return this;
117}
118
119AudioBootStrap Paud_bootstrap = {
120	Paud_DRIVER_NAME, "AIX Paudio",
121	Audio_Available, Audio_CreateDevice
122};
123
124/* This function waits until it is possible to write a full sound buffer */
125static void Paud_WaitAudio(_THIS)
126{
127    fd_set fdset;
128
129    /* See if we need to use timed audio synchronization */
130    if ( frame_ticks ) {
131        /* Use timer for general audio synchronization */
132        Sint32 ticks;
133
134        ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
135        if ( ticks > 0 ) {
136	    SDL_Delay(ticks);
137        }
138    } else {
139        audio_buffer  paud_bufinfo;
140
141        /* Use select() for audio synchronization */
142        struct timeval timeout;
143        FD_ZERO(&fdset);
144        FD_SET(audio_fd, &fdset);
145
146        if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) {
147#ifdef DEBUG_AUDIO
148            fprintf(stderr, "Couldn't get audio buffer information\n");
149#endif
150            timeout.tv_sec  = 10;
151            timeout.tv_usec = 0;
152        } else {
153	    long ms_in_buf = paud_bufinfo.write_buf_time;
154            timeout.tv_sec  = ms_in_buf/1000;
155	    ms_in_buf       = ms_in_buf - timeout.tv_sec*1000;
156            timeout.tv_usec = ms_in_buf*1000;
157#ifdef DEBUG_AUDIO
158            fprintf( stderr,
159		     "Waiting for write_buf_time=%ld,%ld\n",
160		     timeout.tv_sec,
161		     timeout.tv_usec );
162#endif
163	}
164
165#ifdef DEBUG_AUDIO
166        fprintf(stderr, "Waiting for audio to get ready\n");
167#endif
168        if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) {
169            const char *message = "Audio timeout - buggy audio driver? (disabled)";
170            /*
171	     * In general we should never print to the screen,
172             * but in this case we have no other way of letting
173             * the user know what happened.
174             */
175            fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);
176            this->enabled = 0;
177            /* Don't try to close - may hang */
178            audio_fd = -1;
179#ifdef DEBUG_AUDIO
180            fprintf(stderr, "Done disabling audio\n");
181#endif
182        }
183#ifdef DEBUG_AUDIO
184        fprintf(stderr, "Ready!\n");
185#endif
186    }
187}
188
189static void Paud_PlayAudio(_THIS)
190{
191	int written;
192
193	/* Write the audio data, checking for EAGAIN on broken audio drivers */
194	do {
195		written = write(audio_fd, mixbuf, mixlen);
196		if ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) ) {
197			SDL_Delay(1);	/* Let a little CPU time go by */
198		}
199	} while ( (written < 0) &&
200	          ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)) );
201
202	/* If timer synchronization is enabled, set the next write frame */
203	if ( frame_ticks ) {
204		next_frame += frame_ticks;
205	}
206
207	/* If we couldn't write, assume fatal error for now */
208	if ( written < 0 ) {
209		this->enabled = 0;
210	}
211#ifdef DEBUG_AUDIO
212	fprintf(stderr, "Wrote %d bytes of audio data\n", written);
213#endif
214}
215
216static Uint8 *Paud_GetAudioBuf(_THIS)
217{
218	return mixbuf;
219}
220
221static void Paud_CloseAudio(_THIS)
222{
223	if ( mixbuf != NULL ) {
224		SDL_FreeAudioMem(mixbuf);
225		mixbuf = NULL;
226	}
227	if ( audio_fd >= 0 ) {
228		close(audio_fd);
229		audio_fd = -1;
230	}
231}
232
233static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec)
234{
235	char          audiodev[1024];
236	int           format;
237	int           bytes_per_sample;
238	Uint16        test_format;
239	audio_init    paud_init;
240	audio_buffer  paud_bufinfo;
241	audio_status  paud_status;
242	audio_control paud_control;
243	audio_change  paud_change;
244
245	/* Reset the timer synchronization flag */
246	frame_ticks = 0.0;
247
248	/* Open the audio device */
249	audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
250	if ( audio_fd < 0 ) {
251		SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
252		return -1;
253	}
254
255	/*
256	 * We can't set the buffer size - just ask the device for the maximum
257	 * that we can have.
258	 */
259	if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) {
260		SDL_SetError("Couldn't get audio buffer information");
261		return -1;
262	}
263
264	mixbuf = NULL;
265
266	if ( spec->channels > 1 )
267	    spec->channels = 2;
268	else
269	    spec->channels = 1;
270
271	/*
272	 * Fields in the audio_init structure:
273	 *
274	 * Ignored by us:
275	 *
276	 * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?
277	 * paud.slot_number;         * slot number of the adapter
278	 * paud.device_id;           * adapter identification number
279	 *
280	 * Input:
281	 *
282	 * paud.srate;           * the sampling rate in Hz
283	 * paud.bits_per_sample; * 8, 16, 32, ...
284	 * paud.bsize;           * block size for this rate
285	 * paud.mode;            * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX
286	 * paud.channels;        * 1=mono, 2=stereo
287	 * paud.flags;           * FIXED - fixed length data
288	 *                       * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)
289	 *                       * TWOS_COMPLEMENT - 2's complement data
290	 *                       * SIGNED - signed? comment seems wrong in sys/audio.h
291	 *                       * BIG_ENDIAN
292	 * paud.operation;       * PLAY, RECORD
293	 *
294	 * Output:
295	 *
296	 * paud.flags;           * PITCH            - pitch is supported
297	 *                       * INPUT            - input is supported
298	 *                       * OUTPUT           - output is supported
299	 *                       * MONITOR          - monitor is supported
300	 *                       * VOLUME           - volume is supported
301	 *                       * VOLUME_DELAY     - volume delay is supported
302	 *                       * BALANCE          - balance is supported
303	 *                       * BALANCE_DELAY    - balance delay is supported
304	 *                       * TREBLE           - treble control is supported
305	 *                       * BASS             - bass control is supported
306	 *                       * BESTFIT_PROVIDED - best fit returned
307	 *                       * LOAD_CODE        - DSP load needed
308	 * paud.rc;              * NO_PLAY         - DSP code can't do play requests
309	 *                       * NO_RECORD       - DSP code can't do record requests
310	 *                       * INVALID_REQUEST - request was invalid
311	 *                       * CONFLICT        - conflict with open's flags
312	 *                       * OVERLOADED      - out of DSP MIPS or memory
313	 * paud.position_resolution; * smallest increment for position
314	 */
315
316        paud_init.srate = spec->freq;
317	paud_init.mode = PCM;
318	paud_init.operation = PLAY;
319	paud_init.channels = spec->channels;
320
321	/* Try for a closest match on audio format */
322	format = 0;
323	for ( test_format = SDL_FirstAudioFormat(spec->format);
324						! format && test_format; ) {
325#ifdef DEBUG_AUDIO
326		fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
327#endif
328		switch ( test_format ) {
329			case AUDIO_U8:
330			    bytes_per_sample = 1;
331			    paud_init.bits_per_sample = 8;
332			    paud_init.flags = TWOS_COMPLEMENT | FIXED;
333			    format = 1;
334			    break;
335			case AUDIO_S8:
336			    bytes_per_sample = 1;
337			    paud_init.bits_per_sample = 8;
338			    paud_init.flags = SIGNED |
339					      TWOS_COMPLEMENT | FIXED;
340			    format = 1;
341			    break;
342			case AUDIO_S16LSB:
343			    bytes_per_sample = 2;
344			    paud_init.bits_per_sample = 16;
345			    paud_init.flags = SIGNED |
346					      TWOS_COMPLEMENT | FIXED;
347			    format = 1;
348			    break;
349			case AUDIO_S16MSB:
350			    bytes_per_sample = 2;
351			    paud_init.bits_per_sample = 16;
352			    paud_init.flags = BIG_ENDIAN |
353					      SIGNED |
354					      TWOS_COMPLEMENT | FIXED;
355			    format = 1;
356			    break;
357			case AUDIO_U16LSB:
358			    bytes_per_sample = 2;
359			    paud_init.bits_per_sample = 16;
360			    paud_init.flags = TWOS_COMPLEMENT | FIXED;
361			    format = 1;
362			    break;
363			case AUDIO_U16MSB:
364			    bytes_per_sample = 2;
365			    paud_init.bits_per_sample = 16;
366			    paud_init.flags = BIG_ENDIAN |
367					      TWOS_COMPLEMENT | FIXED;
368			    format = 1;
369			    break;
370			default:
371				break;
372		}
373		if ( ! format ) {
374			test_format = SDL_NextAudioFormat();
375		}
376	}
377	if ( format == 0 ) {
378#ifdef DEBUG_AUDIO
379            fprintf(stderr, "Couldn't find any hardware audio formats\n");
380#endif
381	    SDL_SetError("Couldn't find any hardware audio formats");
382	    return -1;
383	}
384	spec->format = test_format;
385
386	/*
387	 * We know the buffer size and the max number of subsequent writes
388	 * that can be pending. If more than one can pend, allow the application
389	 * to do something like double buffering between our write buffer and
390	 * the device's own buffer that we are filling with write() anyway.
391	 *
392	 * We calculate spec->samples like this because SDL_CalculateAudioSpec()
393	 * will give put paud_bufinfo.write_buf_cap (or paud_bufinfo.write_buf_cap/2)
394	 * into spec->size in return.
395	 */
396	if ( paud_bufinfo.request_buf_cap == 1 )
397	{
398	    spec->samples = paud_bufinfo.write_buf_cap
399			  / bytes_per_sample
400			  / spec->channels;
401	}
402	else
403	{
404	    spec->samples = paud_bufinfo.write_buf_cap
405			  / bytes_per_sample
406			  / spec->channels
407			  / 2;
408	}
409        paud_init.bsize = bytes_per_sample * spec->channels;
410
411	SDL_CalculateAudioSpec(spec);
412
413	/*
414	 * The AIX paud device init can't modify the values of the audio_init
415	 * structure that we pass to it. So we don't need any recalculation
416	 * of this stuff and no reinit call as in linux dsp and dma code.
417	 *
418	 * /dev/paud supports all of the encoding formats, so we don't need
419	 * to do anything like reopening the device, either.
420	 */
421	if ( ioctl(audio_fd, AUDIO_INIT, &paud_init) < 0 ) {
422	    switch ( paud_init.rc )
423	    {
424	    case 1 :
425		SDL_SetError("Couldn't set audio format: DSP can't do play requests");
426		return -1;
427		break;
428	    case 2 :
429		SDL_SetError("Couldn't set audio format: DSP can't do record requests");
430		return -1;
431		break;
432	    case 4 :
433		SDL_SetError("Couldn't set audio format: request was invalid");
434		return -1;
435		break;
436	    case 5 :
437		SDL_SetError("Couldn't set audio format: conflict with open's flags");
438		return -1;
439		break;
440	    case 6 :
441		SDL_SetError("Couldn't set audio format: out of DSP MIPS or memory");
442		return -1;
443		break;
444	    default :
445		SDL_SetError("Couldn't set audio format: not documented in sys/audio.h");
446		return -1;
447		break;
448	    }
449	}
450
451	/* Allocate mixing buffer */
452	mixlen = spec->size;
453	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
454	if ( mixbuf == NULL ) {
455		return -1;
456	}
457	SDL_memset(mixbuf, spec->silence, spec->size);
458
459	/*
460	 * Set some paramters: full volume, first speaker that we can find.
461	 * Ignore the other settings for now.
462	 */
463	paud_change.input = AUDIO_IGNORE;         /* the new input source */
464        paud_change.output = OUTPUT_1;            /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */
465        paud_change.monitor = AUDIO_IGNORE;       /* the new monitor state */
466        paud_change.volume = 0x7fffffff;          /* volume level [0-0x7fffffff] */
467        paud_change.volume_delay = AUDIO_IGNORE;  /* the new volume delay */
468        paud_change.balance = 0x3fffffff;         /* the new balance */
469        paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */
470        paud_change.treble = AUDIO_IGNORE;        /* the new treble state */
471        paud_change.bass = AUDIO_IGNORE;          /* the new bass state */
472        paud_change.pitch = AUDIO_IGNORE;         /* the new pitch state */
473
474	paud_control.ioctl_request = AUDIO_CHANGE;
475	paud_control.request_info = (char*)&paud_change;
476	if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) {
477#ifdef DEBUG_AUDIO
478            fprintf(stderr, "Can't change audio display settings\n" );
479#endif
480	}
481
482	/*
483	 * Tell the device to expect data. Actual start will wait for
484	 * the first write() call.
485	 */
486	paud_control.ioctl_request = AUDIO_START;
487	paud_control.position = 0;
488	if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) {
489#ifdef DEBUG_AUDIO
490            fprintf(stderr, "Can't start audio play\n" );
491#endif
492	    SDL_SetError("Can't start audio play");
493	    return -1;
494	}
495
496        /* Check to see if we need to use select() workaround */
497        { char *workaround;
498                workaround = SDL_getenv("SDL_DSP_NOSELECT");
499                if ( workaround ) {
500                        frame_ticks = (float)(spec->samples*1000)/spec->freq;
501                        next_frame = SDL_GetTicks()+frame_ticks;
502                }
503        }
504
505	/* Get the parent process id (we're the parent of the audio thread) */
506	parent = getpid();
507
508	/* We're ready to rock and roll. :-) */
509	return 0;
510}
511
512