source: mainline/uspace/srv/audio/hound/audio_device.c@ fa60cd69

lfn serial ticket/834-toolchain-update topic/msim-upgrade topic/simplify-dev-export
Last change on this file since fa60cd69 was fa60cd69, checked in by Jan Vesely <jano.vesely@…>, 12 years ago

hound: add connection class

This will enable N to M routing in the future
  • 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
45#include "audio_device.h"
46#include "log.h"
47
48#define BUFFER_PARTS 2
49
50static int device_sink_connection_callback(audio_sink_t *sink, bool new);
51static int device_source_connection_callback(audio_source_t *source, bool new);
52static void device_event_callback(ipc_callid_t iid, ipc_call_t *icall, void *arg);
53static int device_check_format(audio_sink_t* sink);
54static int get_buffer(audio_device_t *dev);
55static int release_buffer(audio_device_t *dev);
56
57
58int audio_device_init(audio_device_t *dev, service_id_t id, const char *name)
59{
60 assert(dev);
61 link_initialize(&dev->link);
62 dev->id = id;
63 dev->name = str_dup(name);
64 dev->sess = audio_pcm_open_service(id);
65 if (!dev->sess) {
66 log_debug("Failed to connect to device \"%s\"", name);
67 return ENOMEM;
68 }
69
70 audio_sink_init(&dev->sink, name, dev, device_sink_connection_callback,
71 device_check_format, &AUDIO_FORMAT_ANY);
72 audio_source_init(&dev->source, name, dev,
73 device_source_connection_callback, NULL, &AUDIO_FORMAT_ANY);
74
75 /* Init buffer members */
76 fibril_mutex_initialize(&dev->buffer.guard);
77 fibril_condvar_initialize(&dev->buffer.wc);
78 dev->buffer.base = NULL;
79 dev->buffer.position = NULL;
80 dev->buffer.size = 0;
81
82 log_verbose("Initialized device (%p) '%s' with id %u.",
83 dev, dev->name, dev->id);
84
85 return EOK;
86}
87void audio_device_fini(audio_device_t *dev)
88{
89 //TODO implement;
90}
91
92static int device_sink_connection_callback(audio_sink_t* sink, bool new)
93{
94 assert(sink);
95 audio_device_t *dev = sink->private_data;
96 if (new && list_count(&sink->connections) == 1) {
97 log_verbose("First connection on device sink '%s'", sink->name);
98
99 int ret = get_buffer(dev);
100 if (ret != EOK) {
101 log_error("Failed to get device buffer: %s",
102 str_error(ret));
103 return ret;
104 }
105 audio_pcm_register_event_callback(dev->sess,
106 device_event_callback, dev);\
107 // TODO set formats
108
109 /* Fill the buffer first */
110 audio_sink_mix_inputs(&dev->sink,
111 dev->buffer.base, dev->buffer.size);
112
113 log_verbose("Mixed inputs: %zu/(%u * %u)",
114 dev->buffer.size, BUFFER_PARTS, pcm_format_frame_size(&dev->sink.format));
115 const unsigned frames = dev->buffer.size /
116 (BUFFER_PARTS * pcm_format_frame_size(&dev->sink.format));
117 log_verbose("FRAME COUNT %u", frames);
118 ret = audio_pcm_start_playback_fragment(dev->sess, frames,
119 dev->sink.format.channels, dev->sink.format.sampling_rate,
120 dev->sink.format.sample_format);
121 if (ret != EOK) {
122 log_error("Failed to start playback: %s",
123 str_error(ret));
124 release_buffer(dev);
125 return ret;
126 }
127 }
128 if (list_count(&sink->connections) == 0) {
129 assert(!new);
130 log_verbose("No connections on device sink '%s'", sink->name);
131 int ret = audio_pcm_stop_playback(dev->sess);
132 if (ret != EOK) {
133 log_error("Failed to stop playback: %s",
134 str_error(ret));
135 return ret;
136 }
137 dev->sink.format = AUDIO_FORMAT_ANY;
138 ret = release_buffer(dev);
139 if (ret != EOK) {
140 log_error("Failed to release buffer: %s",
141 str_error(ret));
142 return ret;
143 }
144 }
145 return EOK;
146}
147
148static int device_source_connection_callback(audio_source_t *source, bool new)
149{
150 assert(source);
151 audio_device_t *dev = source->private_data;
152 if (new && list_count(&source->connections)) {
153 int ret = get_buffer(dev);
154 if (ret != EOK) {
155 log_error("Failed to get device buffer: %s",
156 str_error(ret));
157 return ret;
158 }
159 const unsigned frames = dev->buffer.size /
160 (BUFFER_PARTS * pcm_format_frame_size(&dev->sink.format));
161 ret = audio_pcm_start_capture_fragment(dev->sess, frames,
162 dev->sink.format.channels, dev->sink.format.sampling_rate,
163 dev->sink.format.sample_format);
164 if (ret != EOK) {
165 log_error("Failed to start recording: %s",
166 str_error(ret));
167 release_buffer(dev);
168 return ret;
169 }
170 }
171 if (list_count(&source->connections) == 0) { /* Disconnected */
172 assert(!new);
173 int ret = audio_pcm_stop_capture(dev->sess);
174 if (ret != EOK) {
175 log_error("Failed to start recording: %s",
176 str_error(ret));
177 return ret;
178 }
179 source->format = AUDIO_FORMAT_ANY;
180 ret = release_buffer(dev);
181 if (ret != EOK) {
182 log_error("Failed to release buffer: %s",
183 str_error(ret));
184 return ret;
185 }
186 audio_pcm_unregister_event_callback(dev->sess);
187 }
188
189 return EOK;
190}
191
192static void device_event_callback(ipc_callid_t iid, ipc_call_t *icall, void *arg)
193{
194 /* Answer initial request */
195 async_answer_0(iid, EOK);
196 audio_device_t *dev = arg;
197 assert(dev);
198 while (1) {
199 ipc_call_t call;
200 ipc_callid_t callid = async_get_call(&call);
201 async_answer_0(callid, EOK);
202 switch(IPC_GET_IMETHOD(call)) {
203 case PCM_EVENT_FRAMES_PLAYED: {
204 struct timeval time1;
205 getuptime(&time1);
206 //TODO add underrun protection.
207 if (dev->buffer.position) {
208 dev->buffer.position +=
209 (dev->buffer.size / BUFFER_PARTS);
210 }
211 if ((!dev->buffer.position) ||
212 (dev->buffer.position >=
213 (dev->buffer.base + dev->buffer.size)))
214 {
215 dev->buffer.position = dev->buffer.base;
216 }
217 audio_sink_mix_inputs(&dev->sink, dev->buffer.position,
218 dev->buffer.size / BUFFER_PARTS);
219 struct timeval time2;
220 getuptime(&time2);
221 log_verbose("Time to mix sources: %li\n",
222 tv_sub(&time2, &time1));
223 break;
224 }
225 case PCM_EVENT_PLAYBACK_TERMINATED:
226 log_verbose("Playback terminated!");
227 return;
228 case PCM_EVENT_FRAMES_CAPTURED:
229 //TODO implement
230 break;
231 case PCM_EVENT_CAPTURE_TERMINATED:
232 log_verbose("Recording terminated!");
233 return;
234 }
235
236 }
237}
238static int device_check_format(audio_sink_t* sink)
239{
240 assert(sink);
241 audio_device_t *dev = sink->private_data;
242 assert(dev);
243 log_verbose("Checking format on sink %s", sink->name);
244 return audio_pcm_test_format(dev->sess, &sink->format.channels,
245 &sink->format.sampling_rate, &sink->format.sample_format);
246}
247
248static int get_buffer(audio_device_t *dev)
249{
250 assert(dev);
251 if (!dev->sess) {
252 log_debug("No connection to device");
253 return EIO;
254 }
255 if (dev->buffer.base) {
256 log_debug("We already have a buffer");
257 return EBUSY;
258 }
259
260 dev->buffer.size = 0;
261
262 return audio_pcm_get_buffer(dev->sess, &dev->buffer.base,
263 &dev->buffer.size);
264}
265
266static int release_buffer(audio_device_t *dev)
267{
268 assert(dev);
269 assert(dev->buffer.base);
270
271 const int ret = audio_pcm_release_buffer(dev->sess);
272 if (ret == EOK) {
273 dev->buffer.base = NULL;
274 dev->buffer.size = 0;
275 dev->buffer.position = NULL;
276 } else {
277 log_debug("Failed to release buffer: %s", str_error(ret));
278 }
279 return ret;
280}
281/**
282 * @}
283 */
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