source: mainline/uspace/srv/audio/hound/audio_device.c@ 1df3018a

lfn serial ticket/834-toolchain-update topic/fix-logger-deadlock topic/msim-upgrade topic/simplify-dev-export
Last change on this file since 1df3018a was 1df3018a, checked in by Jan Vesely <jano.vesely@…>, 14 years ago

hound: Only few more TODOs left
  • Property mode set to 100644
File size: 7.9 KB
Line 
1/*
2 * Copyright (c) 2012 Jan Vesely
3 * All rights reserved.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions
7 * are met:
8 *
9 * - Redistributions of source code must retain the above copyright
10 * notice, this list of conditions and the following disclaimer.
11 * - Redistributions in binary form must reproduce the above copyright
12 * notice, this list of conditions and the following disclaimer in the
13 * documentation and/or other materials provided with the distribution.
14 * - The name of the author may not be used to endorse or promote products
15 * derived from this software without specific prior written permission.
16 *
17 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
18 * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
19 * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
20 * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
21 * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
22 * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
23 * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
24 * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
25 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
26 * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
27 */
28
29/**
30 * @addtogroup audio
31 * @brief HelenOS sound server.
32 * @{
33 */
34/** @file
35 */
36
37#include <assert.h>
38#include <async.h>
39#include <errno.h>
40#include <loc.h>
41#include <str.h>
42#include <str_error.h>
43
44#include <audio_pcm_iface.h>
45
46#include "audio_device.h"
47#include "log.h"
48
49#define BUFFER_BLOCKS 2
50
51static int device_sink_connection_callback(audio_sink_t *sink);
52static int device_source_connection_callback(audio_source_t *source);
53static void device_event_callback(ipc_callid_t iid, ipc_call_t *icall, void *arg);
54static int get_buffer(audio_device_t *dev);
55static int release_buffer(audio_device_t *dev);
56static int start_playback(audio_device_t *dev);
57static int stop_playback(audio_device_t *dev);
58static int start_recording(audio_device_t *dev);
59static int stop_recording(audio_device_t *dev);
60
61
62int audio_device_init(audio_device_t *dev, service_id_t id, const char *name)
63{
64 assert(dev);
65 link_initialize(&dev->link);
66 dev->id = id;
67 dev->name = str_dup(name);
68 dev->sess = loc_service_connect(EXCHANGE_SERIALIZE, id, 0);
69 if (!dev->sess) {
70 log_debug("Failed to connect to device \"%s\"", name);
71 return ENOMEM;
72 }
73
74 audio_sink_init(&dev->sink, name, dev, device_sink_connection_callback,
75 &AUDIO_FORMAT_ANY);
76 audio_source_init(&dev->source, name, dev,
77 device_source_connection_callback, NULL, &AUDIO_FORMAT_ANY);
78
79 /* Init buffer members */
80 fibril_mutex_initialize(&dev->buffer.guard);
81 fibril_condvar_initialize(&dev->buffer.wc);
82 dev->buffer.id = 0;
83 dev->buffer.base = NULL;
84 dev->buffer.position = NULL;
85 dev->buffer.size = 0;
86
87 log_verbose("Initialized device (%p) '%s' with id %u.",
88 dev, dev->name, dev->id);
89
90 return EOK;
91}
92void audio_device_fini(audio_device_t *dev)
93{
94 //TODO implement;
95}
96
97static int device_sink_connection_callback(audio_sink_t* sink)
98{
99 assert(sink);
100 audio_device_t *dev = sink->private_data;
101 if (list_count(&sink->sources) == 1) {
102 int ret = get_buffer(dev);
103 if (ret != EOK) {
104 log_error("Failed to get device buffer: %s",
105 str_error(ret));
106 return ret;
107 }
108 ret = start_playback(dev);
109 if (ret != EOK) {
110 log_error("Failed to start playback: %s",
111 str_error(ret));
112 release_buffer(dev);
113 return ret;
114 }
115 }
116 if (list_count(&sink->sources) == 0) {
117 int ret = stop_playback(dev);
118 if (ret != EOK) {
119 log_error("Failed to start playback: %s",
120 str_error(ret));
121 return ret;
122 }
123 dev->sink.format = AUDIO_FORMAT_ANY;
124 ret = release_buffer(dev);
125 if (ret != EOK) {
126 log_error("Failed to release buffer: %s",
127 str_error(ret));
128 return ret;
129 }
130 }
131 return EOK;
132}
133
134static int device_source_connection_callback(audio_source_t *source)
135{
136 assert(source);
137 audio_device_t *dev = source->private_data;
138 if (source->connected_sink) {
139 int ret = get_buffer(dev);
140 if (ret != EOK) {
141 log_error("Failed to get device buffer: %s",
142 str_error(ret));
143 return ret;
144 }
145 ret = start_recording(dev);
146 if (ret != EOK) {
147 log_error("Failed to start recording: %s",
148 str_error(ret));
149 release_buffer(dev);
150 return ret;
151 }
152 } else { /* Disconnected */
153 int ret = stop_recording(dev);
154 if (ret != EOK) {
155 log_error("Failed to start recording: %s",
156 str_error(ret));
157 return ret;
158 }
159 source->format = AUDIO_FORMAT_ANY;
160 ret = release_buffer(dev);
161 if (ret != EOK) {
162 log_error("Failed to release buffer: %s",
163 str_error(ret));
164 return ret;
165 }
166 }
167
168 return EOK;
169}
170
171static void device_event_callback(ipc_callid_t iid, ipc_call_t *icall, void *arg)
172{
173 /* Answer initial request */
174 async_answer_0(iid, EOK);
175 audio_device_t *dev = arg;
176 assert(dev);
177 while (1) {
178 ipc_call_t call;
179 ipc_callid_t callid = async_get_call(&call);
180 async_answer_0(callid, EOK);
181 if (IPC_GET_IMETHOD(call) != IPC_FIRST_USER_METHOD) {
182 log_debug("Unknown event.\n");
183 continue;
184 }
185 // Assume playback for now
186 if (dev->buffer.position) {
187 dev->buffer.position += dev->buffer.size / BUFFER_BLOCKS;
188 }
189 if (!dev->buffer.position ||
190 dev->buffer.position >= dev->buffer.base + dev->buffer.size)
191 {
192 dev->buffer.position = dev->buffer.base;
193 }
194 audio_sink_mix_inputs(
195 &dev->sink, dev->buffer.base,
196 dev->buffer.size / BUFFER_BLOCKS);
197
198 }
199 //TODO implement
200}
201
202static int get_buffer(audio_device_t *dev)
203{
204 assert(dev);
205 if (!dev->sess) {
206 log_debug("No connection to device");
207 return EIO;
208 }
209 if (dev->buffer.base) {
210 log_debug("We already have a buffer");
211 return EBUSY;
212 }
213
214 dev->buffer.size = 0;
215
216 async_exch_t *exch = async_exchange_begin(dev->sess);
217 const int ret = audio_pcm_get_buffer(exch, &dev->buffer.base,
218 &dev->buffer.size, &dev->buffer.id, device_event_callback, dev);
219 async_exchange_end(exch);
220 return ret;
221}
222
223#define CHECK_BUFFER_AND_CONNECTION() \
224do { \
225 assert(dev); \
226 if (!dev->sess) { \
227 log_debug("No connection to device"); \
228 return EIO; \
229 } \
230 if (!dev->buffer.base) { \
231 log_debug("We don't have a buffer"); \
232 return ENOENT; \
233 } \
234} while (0)
235
236
237static int release_buffer(audio_device_t *dev)
238{
239 CHECK_BUFFER_AND_CONNECTION();
240
241 async_exch_t *exch = async_exchange_begin(dev->sess);
242 const int ret = audio_pcm_release_buffer(exch, dev->buffer.id);
243 async_exchange_end(exch);
244 if (ret == EOK) {
245 dev->buffer.base = NULL;
246 dev->buffer.size = 0;
247 dev->buffer.position = NULL;
248 }
249 return ret;
250}
251
252static int start_playback(audio_device_t *dev)
253{
254 CHECK_BUFFER_AND_CONNECTION();
255
256 /* Fill the buffer first */
257 audio_sink_mix_inputs(&dev->sink, dev->buffer.base, dev->buffer.size);
258
259 async_exch_t *exch = async_exchange_begin(dev->sess);
260 const int ret = audio_pcm_start_playback(exch, dev->buffer.id,
261 BUFFER_BLOCKS, dev->sink.format.channels,
262 dev->sink.format.sampling_rate, dev->sink.format.sample_format);
263 async_exchange_end(exch);
264 return ret;
265}
266
267static int stop_playback(audio_device_t *dev)
268{
269 CHECK_BUFFER_AND_CONNECTION();
270
271 async_exch_t *exch = async_exchange_begin(dev->sess);
272 const int ret = audio_pcm_stop_playback(exch, dev->buffer.id);
273 async_exchange_end(exch);
274 return ret;
275}
276
277static int start_recording(audio_device_t *dev)
278{
279 CHECK_BUFFER_AND_CONNECTION();
280
281 async_exch_t *exch = async_exchange_begin(dev->sess);
282 const int ret = audio_pcm_start_record(exch, dev->buffer.id,
283 BUFFER_BLOCKS, dev->sink.format.channels,
284 dev->sink.format.sampling_rate, dev->sink.format.sample_format);
285 async_exchange_end(exch);
286 return ret;
287}
288
289static int stop_recording(audio_device_t *dev)
290{
291 CHECK_BUFFER_AND_CONNECTION();
292
293 async_exch_t *exch = async_exchange_begin(dev->sess);
294 const int ret = audio_pcm_stop_record(exch, dev->buffer.id);
295 async_exchange_end(exch);
296 return ret;
297}
298
299/**
300 * @}
301 */
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