aboutsummaryrefslogtreecommitdiffstats
path: root/src/output/OpenALOutputPlugin.cxx
blob: e68032d75f3bd9af2a8e4464283aee70446c8d61 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
/*
 * Copyright (C) 2003-2013 The Music Player Daemon Project
 * http://www.musicpd.org
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program 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 General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with this program; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#include "config.h"
#include "OpenALOutputPlugin.hxx"
#include "OutputAPI.hxx"

#include <glib.h>

#ifndef HAVE_OSX
#include <AL/al.h>
#include <AL/alc.h>
#else
#include <OpenAL/al.h>
#include <OpenAL/alc.h>
#endif

#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "openal"

/* should be enough for buffer size = 2048 */
#define NUM_BUFFERS 16

struct OpenALOutput {
	struct audio_output base;

	const char *device_name;
	ALCdevice *device;
	ALCcontext *context;
	ALuint buffers[NUM_BUFFERS];
	unsigned filled;
	ALuint source;
	ALenum format;
	ALuint frequency;

	bool Initialize(const config_param *param, GError **error_r) {
		return ao_base_init(&base, &openal_output_plugin, param,
				    error_r);
	}

	void Deinitialize() {
		ao_base_finish(&base);
	}
};

static inline GQuark
openal_output_quark(void)
{
	return g_quark_from_static_string("openal_output");
}

static ALenum
openal_audio_format(struct audio_format *audio_format)
{
	/* note: cannot map SAMPLE_FORMAT_S8 to AL_FORMAT_STEREO8 or
	   AL_FORMAT_MONO8 since OpenAL expects unsigned 8 bit
	   samples, while MPD uses signed samples */

	switch (audio_format->format) {
	case SAMPLE_FORMAT_S16:
		if (audio_format->channels == 2)
			return AL_FORMAT_STEREO16;
		if (audio_format->channels == 1)
			return AL_FORMAT_MONO16;

		/* fall back to mono */
		audio_format->channels = 1;
		return openal_audio_format(audio_format);

	default:
		/* fall back to 16 bit */
		audio_format->format = SAMPLE_FORMAT_S16;
		return openal_audio_format(audio_format);
	}
}

G_GNUC_PURE
static inline ALint
openal_get_source_i(const OpenALOutput *od, ALenum param)
{
	ALint value;
	alGetSourcei(od->source, param, &value);
	return value;
}

G_GNUC_PURE
static inline bool
openal_has_processed(const OpenALOutput *od)
{
	return openal_get_source_i(od, AL_BUFFERS_PROCESSED) > 0;
}

G_GNUC_PURE
static inline ALint
openal_is_playing(const OpenALOutput *od)
{
	return openal_get_source_i(od, AL_SOURCE_STATE) == AL_PLAYING;
}

static bool
openal_setup_context(OpenALOutput *od,
		     GError **error)
{
	od->device = alcOpenDevice(od->device_name);

	if (od->device == nullptr) {
		g_set_error(error, openal_output_quark(), 0,
			    "Error opening OpenAL device \"%s\"\n",
			    od->device_name);
		return false;
	}

	od->context = alcCreateContext(od->device, nullptr);

	if (od->context == nullptr) {
		g_set_error(error, openal_output_quark(), 0,
			    "Error creating context for \"%s\"\n",
			    od->device_name);
		alcCloseDevice(od->device);
		return false;
	}

	return true;
}

static struct audio_output *
openal_init(const config_param *param, GError **error_r)
{
	const char *device_name = config_get_block_string(param, "device", nullptr);

	if (device_name == nullptr) {
		device_name = alcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
	}

	OpenALOutput *od = new OpenALOutput();
	if (!od->Initialize(param, error_r)) {
		delete od;
		return nullptr;
	}

	od->device_name = device_name;

	return &od->base;
}

static void
openal_finish(struct audio_output *ao)
{
	OpenALOutput *od = (OpenALOutput *)ao;

	od->Deinitialize();
	delete od;
}

static bool
openal_open(struct audio_output *ao, struct audio_format *audio_format,
	    GError **error)
{
	OpenALOutput *od = (OpenALOutput *)ao;

	od->format = openal_audio_format(audio_format);

	if (!openal_setup_context(od, error)) {
		return false;
	}

	alcMakeContextCurrent(od->context);
	alGenBuffers(NUM_BUFFERS, od->buffers);

	if (alGetError() != AL_NO_ERROR) {
		g_set_error(error, openal_output_quark(), 0,
			    "Failed to generate buffers");
		return false;
	}

	alGenSources(1, &od->source);

	if (alGetError() != AL_NO_ERROR) {
		g_set_error(error, openal_output_quark(), 0,
			    "Failed to generate source");
		alDeleteBuffers(NUM_BUFFERS, od->buffers);
		return false;
	}

	od->filled = 0;
	od->frequency = audio_format->sample_rate;

	return true;
}

static void
openal_close(struct audio_output *ao)
{
	OpenALOutput *od = (OpenALOutput *)ao;

	alcMakeContextCurrent(od->context);
	alDeleteSources(1, &od->source);
	alDeleteBuffers(NUM_BUFFERS, od->buffers);
	alcDestroyContext(od->context);
	alcCloseDevice(od->device);
}

static unsigned
openal_delay(struct audio_output *ao)
{
	OpenALOutput *od = (OpenALOutput *)ao;

	return od->filled < NUM_BUFFERS || openal_has_processed(od)
		? 0
		/* we don't know exactly how long we must wait for the
		   next buffer to finish, so this is a random
		   guess: */
		: 50;
}

static size_t
openal_play(struct audio_output *ao, const void *chunk, size_t size,
	    G_GNUC_UNUSED GError **error)
{
	OpenALOutput *od = (OpenALOutput *)ao;
	ALuint buffer;

	if (alcGetCurrentContext() != od->context) {
		alcMakeContextCurrent(od->context);
	}

	if (od->filled < NUM_BUFFERS) {
		/* fill all buffers */
		buffer = od->buffers[od->filled];
		od->filled++;
	} else {
		/* wait for processed buffer */
		while (!openal_has_processed(od))
			g_usleep(10);

		alSourceUnqueueBuffers(od->source, 1, &buffer);
	}

	alBufferData(buffer, od->format, chunk, size, od->frequency);
	alSourceQueueBuffers(od->source, 1, &buffer);

	if (!openal_is_playing(od))
		alSourcePlay(od->source);

	return size;
}

static void
openal_cancel(struct audio_output *ao)
{
	OpenALOutput *od = (OpenALOutput *)ao;

	od->filled = 0;
	alcMakeContextCurrent(od->context);
	alSourceStop(od->source);

	/* force-unqueue all buffers */
	alSourcei(od->source, AL_BUFFER, 0);
	od->filled = 0;
}

const struct audio_output_plugin openal_output_plugin = {
	"openal",
	nullptr,
	openal_init,
	openal_finish,
	nullptr,
	nullptr,
	openal_open,
	openal_close,
	openal_delay,
	nullptr,
	openal_play,
	nullptr,
	openal_cancel,
	nullptr,
	nullptr,
};