1 /*
2  * CDDL HEADER START
3  *
4  * The contents of this file are subject to the terms of the
5  * Common Development and Distribution License (the "License").
6  * You may not use this file except in compliance with the License.
7  *
8  * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
9  * or http://www.opensolaris.org/os/licensing.
10  * See the License for the specific language governing permissions
11  * and limitations under the License.
12  *
13  * When distributing Covered Code, include this CDDL HEADER in each
14  * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
15  * If applicable, add the following below this CDDL HEADER, with the
16  * fields enclosed by brackets "[]" replaced with your own identifying
17  * information: Portions Copyright [yyyy] [name of copyright owner]
18  *
19  * CDDL HEADER END
20  */
21 /*
22  * Copyright (C) 4Front Technologies 1996-2008.
23  *
24  * Copyright 2010 Sun Microsystems, Inc.  All rights reserved.
25  * Use is subject to license terms.
26  */
27 
28 /*
29  * Purpose: Virtual mixing audio input routines
30  *
31  * This file contains the actual mixing and resampling engine for input.
32  */
33 
34 #include <sys/ddi.h>
35 #include <sys/sunddi.h>
36 #include <sys/sysmacros.h>
37 #include <sys/sdt.h>
38 #include "audio_impl.h"
39 
40 #define	DECL_AUDIO_IMPORT(NAME, TYPE, SWAP, SHIFT)			\
41 void									\
42 auimpl_import_##NAME(audio_engine_t *e, uint_t nfr, audio_stream_t *sp)	\
43 {									\
44 	int		nch = e->e_nchan;				\
45 	int32_t		*out = (void *)sp->s_cnv_src;			\
46 	TYPE		*in = (void *)e->e_data;			\
47 	int		ch = 0;						\
48 	int		vol = sp->s_gain_eff;				\
49 									\
50 	do {	/* for each channel */					\
51 		TYPE 	*ip;						\
52 		int32_t *op;						\
53 		int 	i;						\
54 		int 	incr = e->e_chincr[ch];				\
55 		uint_t	tidx = e->e_tidx;				\
56 									\
57 		/* get value and adjust next channel offset */		\
58 		op = out++;						\
59 		ip = in + e->e_choffs[ch] + (tidx * incr);		\
60 									\
61 		i = nfr;						\
62 									\
63 		do {	/* for each frame */				\
64 			int32_t	sample = (TYPE)SWAP(*ip);		\
65 			int32_t	scaled = sample SHIFT;			\
66 									\
67 			scaled *= vol;					\
68 			scaled /= AUDIO_VOL_SCALE;			\
69 									\
70 			*op = scaled;					\
71 			op += nch;					\
72 									\
73 			ip += incr;					\
74 			if (++tidx == e->e_nframes) {			\
75 				tidx = 0;				\
76 				ip = in + e->e_choffs[ch];		\
77 			}						\
78 		} while (--i);						\
79 		ch++;							\
80 	} while (ch < nch);						\
81 }
82 
83 DECL_AUDIO_IMPORT(16ne, int16_t, /* nop */, << 8)
84 DECL_AUDIO_IMPORT(16oe, int16_t, ddi_swap16, << 8)
85 DECL_AUDIO_IMPORT(32ne, int32_t, /* nop */, >> 8)
86 DECL_AUDIO_IMPORT(32oe, int32_t, ddi_swap32, >> 8)
87 DECL_AUDIO_IMPORT(24ne, int32_t, /* nop */, /* nop */)
88 DECL_AUDIO_IMPORT(24oe, int32_t, ddi_swap32, /* nop */)
89 
90 /*
91  * Produce capture data.  This takes data from the conversion buffer
92  * and copies it into the stream data buffer.
93  */
94 static void
95 auimpl_produce_data(audio_stream_t *sp, uint_t count)
96 {
97 	uint_t	nframes;
98 	uint_t	framesz;
99 	caddr_t	cnvsrc;
100 	caddr_t	data;
101 
102 	nframes = sp->s_nframes;
103 	framesz = sp->s_framesz;
104 
105 	ASSERT(sp->s_head >= sp->s_tail);
106 	ASSERT(sp->s_hidx < nframes);
107 	ASSERT(sp->s_tidx < nframes);
108 
109 	/*
110 	 * Copy data.  We deal properly with wraps.  Done as a
111 	 * do...while to minimize the number of tests.
112 	 */
113 	cnvsrc = sp->s_cnv_src;
114 	data = sp->s_data + (sp->s_hidx * framesz);
115 	do {
116 		unsigned nf;
117 		unsigned nb;
118 
119 		nf = min(nframes - sp->s_hidx, count);
120 		nb = nf * framesz;
121 
122 		bcopy(cnvsrc, data, nb);
123 		data += nb;
124 		cnvsrc += nb;
125 		sp->s_hidx += nf;
126 		sp->s_head += nf;
127 		count -= nf;
128 		sp->s_samples += nf;
129 		if (sp->s_hidx == nframes) {
130 			sp->s_hidx = 0;
131 			data = sp->s_data;
132 		}
133 	} while (count);
134 
135 	ASSERT(sp->s_tail <= sp->s_head);
136 	ASSERT(sp->s_hidx < nframes);
137 }
138 
139 void
140 auimpl_input_callback(void *arg)
141 {
142 	audio_engine_t	*e = arg;
143 	uint_t		fragfr = e->e_fragfr;
144 	audio_stream_t	*sp;
145 	audio_client_t	*c;
146 	audio_client_t	*clist = NULL;
147 	list_t		*l = &e->e_streams;
148 	uint64_t	h;
149 
150 	mutex_enter(&e->e_lock);
151 
152 	if (e->e_suspended || e->e_failed) {
153 		mutex_exit(&e->e_lock);
154 		return;
155 	}
156 
157 	if (e->e_need_start) {
158 		int rv;
159 		if ((rv = ENG_START(e)) != 0) {
160 			e->e_failed = B_TRUE;
161 			mutex_exit(&e->e_lock);
162 			audio_dev_warn(e->e_dev,
163 			    "failed starting input, rv = %d", rv);
164 			return;
165 		}
166 		e->e_need_start = B_FALSE;
167 	}
168 
169 	h = ENG_COUNT(e);
170 	ASSERT(h >= e->e_head);
171 	if (h < e->e_head) {
172 		/*
173 		 * This is a sign of a serious bug.  We should
174 		 * probably offline the device via FMA, if we ever
175 		 * support FMA for audio devices.
176 		 */
177 		e->e_failed = B_TRUE;
178 		ENG_STOP(e);
179 		mutex_exit(&e->e_lock);
180 		audio_dev_warn(e->e_dev,
181 		    "device malfunction: broken capture sample counter");
182 		return;
183 	}
184 	e->e_head = h;
185 	ASSERT(e->e_head >= e->e_tail);
186 
187 	if ((e->e_head - e->e_tail) > e->e_nframes) {
188 		/* no room for data, not much we can do */
189 		e->e_errors++;
190 		e->e_overruns++;
191 	}
192 
193 	/* consume all fragments in the buffer */
194 	while ((e->e_head - e->e_tail) > fragfr) {
195 
196 		/*
197 		 * Consider doing the SYNC outside of the lock.
198 		 */
199 		ENG_SYNC(e, fragfr);
200 
201 		for (sp = list_head(l); sp != NULL; sp = list_next(l, sp)) {
202 			int space;
203 			int count;
204 
205 			mutex_enter(&sp->s_lock);
206 			/* skip over streams paused or not running */
207 			if (sp->s_paused || !sp->s_running) {
208 				mutex_exit(&sp->s_lock);
209 				continue;
210 			}
211 			sp->s_cnv_src = sp->s_cnv_buf0;
212 			sp->s_cnv_dst = sp->s_cnv_buf1;
213 
214 			e->e_import(e, fragfr, sp);
215 
216 			/*
217 			 * Optionally convert fragment to requested sample
218 			 * format and rate.
219 			 */
220 			if (sp->s_converter != NULL) {
221 				count = sp->s_converter(sp, fragfr);
222 			} else {
223 				count = fragfr;
224 			}
225 
226 			ASSERT(sp->s_head >= sp->s_tail);
227 			space = sp->s_nframes - (sp->s_head - sp->s_tail);
228 			if (count > space) {
229 				e->e_stream_overruns++;
230 				e->e_errors++;
231 				sp->s_errors += count - space;
232 				count = space;
233 			}
234 
235 			auimpl_produce_data(sp, count);
236 
237 			/* wake blocked threads (blocking reads, etc.) */
238 			cv_broadcast(&sp->s_cv);
239 
240 			mutex_exit(&sp->s_lock);
241 
242 			/*
243 			 * Add client to notification list.  We'll
244 			 * process it after dropping the lock.
245 			 */
246 			c = sp->s_client;
247 
248 			if ((c->c_input != NULL) &&
249 			    (c->c_next_input == NULL)) {
250 				auclnt_hold(c);
251 				c->c_next_input = clist;
252 				clist = c;
253 			}
254 		}
255 
256 		/*
257 		 * Update the tail pointer, and the data pointer.
258 		 */
259 		e->e_tail += fragfr;
260 		e->e_tidx += fragfr;
261 		if (e->e_tidx >= e->e_nframes) {
262 			e->e_tidx -= e->e_nframes;
263 		}
264 	}
265 
266 	mutex_exit(&e->e_lock);
267 
268 	/*
269 	 * Notify client personalities.
270 	 */
271 
272 	while ((c = clist) != NULL) {
273 		clist = c->c_next_input;
274 		c->c_next_input = NULL;
275 		c->c_input(c);
276 		auclnt_release(c);
277 	}
278 }
279