aboutsummaryrefslogtreecommitdiffhomepage
path: root/audio/out/ao_openal.c
blob: 715ffddceb877b0e5e5932d5929579038c044ce2 (plain)
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
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
/*
 * OpenAL audio output driver for MPlayer
 *
 * Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
 *
 * This file is part of mpv.
 *
 * mpv is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * mpv 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 Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with mpv.  If not, see <http://www.gnu.org/licenses/>.
 */

#include "config.h"

#include <stdlib.h>
#include <stdio.h>
#include <inttypes.h>

#ifdef __APPLE__
#ifndef AL_FORMAT_MONO_FLOAT32
#define AL_FORMAT_MONO_FLOAT32 0x10010
#endif
#ifndef AL_FORMAT_STEREO_FLOAT32
#define AL_FORMAT_STEREO_FLOAT32 0x10011
#endif
#ifndef AL_FORMAT_MONO_DOUBLE_EXT
#define AL_FORMAT_MONO_DOUBLE_EXT 0x10012
#endif
#include <OpenAL/MacOSX_OALExtensions.h>
#else
#ifdef OPENAL_AL_H
#include <OpenAL/alc.h>
#include <OpenAL/al.h>
#include <OpenAL/alext.h>
#else
#include <AL/alc.h>
#include <AL/al.h>
#include <AL/alext.h>
#endif
#endif // __APPLE__

#include "common/msg.h"

#include "ao.h"
#include "internal.h"
#include "audio/format.h"
#include "osdep/timer.h"
#include "options/m_option.h"

#define MAX_CHANS MP_NUM_CHANNELS
#define NUM_BUF 128
#define CHUNK_SAMPLES 256
static ALuint buffers[MAX_CHANS][NUM_BUF];
static ALuint sources[MAX_CHANS];

static int cur_buf[MAX_CHANS];
static int unqueue_buf[MAX_CHANS];

static struct ao *ao_data;

struct priv {
    ALenum al_format;
    int chunk_size;
};

static void reset(struct ao *ao);

static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
    switch (cmd) {
    case AOCONTROL_GET_VOLUME:
    case AOCONTROL_SET_VOLUME: {
        ALfloat volume;
        ao_control_vol_t *vol = (ao_control_vol_t *)arg;
        if (cmd == AOCONTROL_SET_VOLUME) {
            volume = (vol->left + vol->right) / 200.0;
            alListenerf(AL_GAIN, volume);
        }
        alGetListenerf(AL_GAIN, &volume);
        vol->left = vol->right = volume * 100;
        return CONTROL_TRUE;
    }
    case AOCONTROL_HAS_SOFT_VOLUME:
        return CONTROL_TRUE;
    }
    return CONTROL_UNKNOWN;
}

struct speaker {
    int id;
    float pos[3];
};

static const struct speaker speaker_pos[] = {
    {MP_SPEAKER_ID_FL,   {-0.500,  0, -0.866}}, // -30 deg
    {MP_SPEAKER_ID_FR,   { 0.500,  0, -0.866}}, //  30 deg
    {MP_SPEAKER_ID_FC,   {     0,  0,     -1}}, //   0 deg
    {MP_SPEAKER_ID_LFE,  {     0, -1,      0}}, //   below
    {MP_SPEAKER_ID_BL,   {-0.609,  0,  0.793}}, // -142.5 deg
    {MP_SPEAKER_ID_BR,   { 0.609,  0,  0.793}}, //  142.5 deg
    {MP_SPEAKER_ID_BC,   {     0,  0,      1}}, //  180 deg
    {MP_SPEAKER_ID_SL,   {-0.985,  0,  0.174}}, // -100 deg
    {MP_SPEAKER_ID_SR,   { 0.985,  0,  0.174}}, //  100 deg
    {-1},
};

static ALenum get_al_format(int format)
{
    switch (format) {
    case AF_FORMAT_U8P: return AL_FORMAT_MONO8;
    case AF_FORMAT_S16P: return  AL_FORMAT_MONO16;
    case AF_FORMAT_FLOATP:
        if (alIsExtensionPresent((ALchar*)"AL_EXT_float32") == AL_TRUE)
            return AL_FORMAT_MONO_FLOAT32;
        break;
    case AF_FORMAT_DOUBLEP:
        if (alIsExtensionPresent((ALchar*)"AL_EXT_double") == AL_TRUE)
            return AL_FORMAT_MONO_DOUBLE_EXT;
        break;
    }
    return AL_FALSE;
}

// close audio device
static void uninit(struct ao *ao)
{
    ALCcontext *ctx = alcGetCurrentContext();
    ALCdevice *dev = alcGetContextsDevice(ctx);
    reset(ao);
    alcMakeContextCurrent(NULL);
    alcDestroyContext(ctx);
    alcCloseDevice(dev);
    ao_data = NULL;
}

static int init(struct ao *ao)
{
    float position[3] = {0, 0, 0};
    float direction[6] = {0, 0, -1, 0, 1, 0};
    ALCdevice *dev = NULL;
    ALCcontext *ctx = NULL;
    ALCint freq = 0;
    ALCint attribs[] = {ALC_FREQUENCY, ao->samplerate, 0, 0};
    int i;
    struct priv *p = ao->priv;
    if (ao_data) {
        MP_FATAL(ao, "Not reentrant!\n");
        return -1;
    }
    ao_data = ao;
    struct mp_chmap_sel sel = {0};
    for (i = 0; speaker_pos[i].id != -1; i++)
        mp_chmap_sel_add_speaker(&sel, speaker_pos[i].id);
    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
        goto err_out;
    struct speaker speakers[MAX_CHANS];
    for (i = 0; i < ao->channels.num; i++) {
        speakers[i].id = -1;
        for (int n = 0; speaker_pos[n].id >= 0; n++) {
            if (speaker_pos[n].id == ao->channels.speaker[i])
                speakers[i] = speaker_pos[n];
        }
        if (speakers[i].id < 0) {
            MP_FATAL(ao, "Unknown channel layout\n");
            goto err_out;
        }
    }
    char *dev_name = ao->device;
    dev = alcOpenDevice(dev_name && dev_name[0] ? dev_name : NULL);
    if (!dev) {
        MP_FATAL(ao, "could not open device\n");
        goto err_out;
    }
    ctx = alcCreateContext(dev, attribs);
    alcMakeContextCurrent(ctx);
    alListenerfv(AL_POSITION, position);
    alListenerfv(AL_ORIENTATION, direction);
    alGenSources(ao->channels.num, sources);
    for (i = 0; i < ao->channels.num; i++) {
        cur_buf[i] = 0;
        unqueue_buf[i] = 0;
        alGenBuffers(NUM_BUF, buffers[i]);
        alSourcefv(sources[i], AL_POSITION, speakers[i].pos);
        alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
    }
    alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
    if (alcGetError(dev) == ALC_NO_ERROR && freq)
        ao->samplerate = freq;

    p->al_format = AL_FALSE;
    int try_formats[AF_FORMAT_COUNT];
    af_get_best_sample_formats(ao->format, try_formats);
    for (int n = 0; try_formats[n]; n++) {
        p->al_format = get_al_format(try_formats[n]);
        if (p->al_format != AL_FALSE) {
            ao->format = try_formats[n];
            break;
        }
    }

    if (p->al_format == AL_FALSE) {
        MP_FATAL(ao, "Can't find appropriate sample format.\n");
        uninit(ao);
        goto err_out;
    }

    p->chunk_size = CHUNK_SAMPLES * af_fmt_to_bytes(ao->format);
    ao->period_size = CHUNK_SAMPLES;
    return 0;

err_out:
    ao_data = NULL;
    return -1;
}

static void drain(struct ao *ao)
{
    ALint state;
    alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
    while (state == AL_PLAYING) {
        mp_sleep_us(10000);
        alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
    }
}

static void unqueue_buffers(void)
{
    ALint p;
    int s;
    for (s = 0; s < ao_data->channels.num; s++) {
        int till_wrap = NUM_BUF - unqueue_buf[s];
        alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
        if (p >= till_wrap) {
            alSourceUnqueueBuffers(sources[s], till_wrap,
                                   &buffers[s][unqueue_buf[s]]);
            unqueue_buf[s] = 0;
            p -= till_wrap;
        }
        if (p) {
            alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]);
            unqueue_buf[s] += p;
        }
    }
}

/**
 * \brief stop playing and empty buffers (for seeking/pause)
 */
static void reset(struct ao *ao)
{
    alSourceStopv(ao->channels.num, sources);
    unqueue_buffers();
}

/**
 * \brief stop playing, keep buffers (for pause)
 */
static void audio_pause(struct ao *ao)
{
    alSourcePausev(ao->channels.num, sources);
}

/**
 * \brief resume playing, after audio_pause()
 */
static void audio_resume(struct ao *ao)
{
    alSourcePlayv(ao->channels.num, sources);
}

static int get_space(struct ao *ao)
{
    ALint queued;
    unqueue_buffers();
    alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
    queued = NUM_BUF - queued - 3;
    if (queued < 0)
        return 0;
    return queued * CHUNK_SAMPLES;
}

/**
 * \brief write data into buffer and reset underrun flag
 */
static int play(struct ao *ao, void **data, int samples, int flags)
{
    struct priv *p = ao->priv;
    ALint state;
    int num = samples / CHUNK_SAMPLES;
    for (int i = 0; i < num; i++) {
        for (int ch = 0; ch < ao->channels.num; ch++) {
            char *d = data[ch];
            d += i * p->chunk_size;
            alBufferData(buffers[ch][cur_buf[ch]], p->al_format, d,
                         p->chunk_size, ao->samplerate);
            alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
            cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
        }
    }
    alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
    if (state != AL_PLAYING) // checked here in case of an underrun
        alSourcePlayv(ao->channels.num, sources);
    return num * CHUNK_SAMPLES;
}

static double get_delay(struct ao *ao)
{
    ALint queued;
    unqueue_buffers();
    alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
    return queued * CHUNK_SAMPLES / (double)ao->samplerate;
}

#define OPT_BASE_STRUCT struct priv

const struct ao_driver audio_out_openal = {
    .description = "OpenAL audio output",
    .name      = "openal",
    .init      = init,
    .uninit    = uninit,
    .control   = control,
    .get_space = get_space,
    .play      = play,
    .get_delay = get_delay,
    .pause     = audio_pause,
    .resume    = audio_resume,
    .reset     = reset,
    .drain     = drain,
    .priv_size = sizeof(struct priv),
};