summaryrefslogtreecommitdiffstats
path: root/kscd/libwm/audio/audio_alsa.c
blob: b1d4e938c37362146817f42a8d92a04d4b6704d6 (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
/*
 *  Driver for Advanced Linux Sound Architecture, http://alsa.jcu.cz
 *
 *  mpg123 comments:
 *  Code by Anders Semb Hermansen <ahermans@vf.telia.no>
 *  Cleanups by Jaroslav Kysela <perex@jcu.cz>
 *              Ville Syrjala <syrjala@sci.fi>
 *
 *  adopted for libworkman cdda audio backend from Alexander Kern alex.kern@gmx.de  
 *
 *  Adapted to support both ALSA V0.x and V1.x APIs for PCM calls 
 *  (Philip Nelson <teamdba@scotdb.com> 2004-03-15)
 *
 * This file comes under GPL license.
 */



#include <config.h>

#if defined(HAVE_ARTS_LIBASOUND2)

#include <alsa/asoundlib.h>
#include "audio.h"

char* device = NULL;
snd_pcm_t *handle;

snd_pcm_format_t format = SND_PCM_FORMAT_S16;    /* sample format */
int rate = 44100;                                /* stream rate */
int channels = 2;                                /* count of channels */
int buffer_time = 2000000;                       /* ring buffer length in us */
int period_time = 100000;                        /* period time in us */

snd_pcm_sframes_t buffer_size;
snd_pcm_sframes_t period_size;

int alsa_open(void);
int alsa_close(void);
int alsa_stop(void);
int alsa_play(struct cdda_block *blk);
int alsa_state(struct cdda_block *blk);
struct audio_oops* setup_alsa(const char *dev, const char *ctl);

static int set_hwparams(snd_pcm_hw_params_t *params,
                        snd_pcm_access_t accesspar)
{
       int err, dir, new_rate;

        /* choose all parameters */
        err = snd_pcm_hw_params_any(handle, params);
        if (err < 0) {
                ERRORLOG("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
                return err;
        }
        /* set the interleaved read/write format */
        err = snd_pcm_hw_params_set_access(handle, params, accesspar);
        if (err < 0) {
                ERRORLOG("Access type not available for playback: %s\n", snd_strerror(err));
                return err;
        }
        /* set the sample format */
        err = snd_pcm_hw_params_set_format(handle, params, format);
        if (err < 0) {
                ERRORLOG("Sample format not available for playback: %s\n", snd_strerror(err));
                return err;
        }
        /* set the count of channels */
        err = snd_pcm_hw_params_set_channels(handle, params, channels);
        if (err < 0) {
                ERRORLOG("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err));
                return err;
        }
        /* set the stream rate */
#if (SND_LIB_MAJOR < 1) 
        err = new_rate = snd_pcm_hw_params_set_rate_near(handle, params, rate, 0);
#else
        new_rate = rate;
        err = snd_pcm_hw_params_set_rate_near(handle, params, &new_rate, 0);
#endif
        if (err < 0) {
                ERRORLOG("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err));
                return err;
        }
        if (new_rate != rate) {
                ERRORLOG("Rate doesn't match (requested %iHz, get %iHz)\n", rate, new_rate);
                return -EINVAL;
        }
        /* set the buffer time */
#if (SND_LIB_MAJOR < 1)
         err = snd_pcm_hw_params_set_buffer_time_near(handle, params, buffer_time, &dir);
#else
        err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, &dir);
#endif
        if (err < 0) {
                ERRORLOG("Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err));
                return err;
        }
#if (SND_LIB_MAJOR < 1)
         buffer_size = snd_pcm_hw_params_get_buffer_size(params);
#else
        err = snd_pcm_hw_params_get_buffer_size(params, &buffer_size); 
        if (err < 0) {
                ERRORLOG("Unable to get buffer size : %s\n", snd_strerror(err));
                return err;
        }
#endif
        DEBUGLOG("buffersize %i\n", buffer_size);

        /* set the period time */
#if (SND_LIB_MAJOR < 1)
         err = snd_pcm_hw_params_set_period_time_near(handle, params, period_time, &dir);
#else 
        err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir);
#endif
        if (err < 0) {
                ERRORLOG("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err));
                return err;
        }

#if (SND_LIB_MAJOR < 1) 
        period_size = snd_pcm_hw_params_get_period_size(params, &dir);
#else
        err = snd_pcm_hw_params_get_period_size(params, &period_size, &dir);
        if (err < 0) {
                ERRORLOG("Unable to get hw period size: %s\n", snd_strerror(err));
        }
#endif

        DEBUGLOG("period_size %i\n", period_size);

        /* write the parameters to device */
        err = snd_pcm_hw_params(handle, params);
        if (err < 0) {
                ERRORLOG("Unable to set hw params for playback: %s\n", snd_strerror(err));
                return err;
        }
        return 0;
}

static int set_swparams(snd_pcm_sw_params_t *swparams)
{
        int err;

        /* get the current swparams */
        err = snd_pcm_sw_params_current(handle, swparams);
        if (err < 0) {
                ERRORLOG("Unable to determine current swparams for playback: %s\n", snd_strerror(err));
                return err;
        }
        /* start the transfer when the buffer is full */
        err = snd_pcm_sw_params_set_start_threshold(handle, swparams, buffer_size);
        if (err < 0) {
                ERRORLOG("Unable to set start threshold mode for playback: %s\n", snd_strerror(err));
                return err;
        }
        /* allow the transfer when at least period_size samples can be processed */
        err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_size);
        if (err < 0) {
                ERRORLOG("Unable to set avail min for playback: %s\n", snd_strerror(err));
                return err;
        }
        /* align all transfers to 1 sample */
        err = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1);
        if (err < 0) {
                ERRORLOG("Unable to set transfer align for playback: %s\n", snd_strerror(err));
                return err;
        }
        /* write the parameters to the playback device */
        err = snd_pcm_sw_params(handle, swparams);
        if (err < 0) {
                ERRORLOG("Unable to set sw params for playback: %s\n", snd_strerror(err));
                return err;
        }
        return 0;
}

int alsa_open( void )
{
  int err;

  snd_pcm_hw_params_t *hwparams;
  snd_pcm_sw_params_t *swparams;

  DEBUGLOG("alsa_open\n");

  snd_pcm_hw_params_alloca(&hwparams);
  snd_pcm_sw_params_alloca(&swparams);

  if((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0/*SND_PCM_NONBLOCK*/)) < 0 ) {
    ERRORLOG("open failed: %s\n", snd_strerror(err));
    return -1;
  }

  if((err = set_hwparams(hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
    ERRORLOG("Setting of hwparams failed: %s\n", snd_strerror(err));
    return -1;
  }
  if((err = set_swparams(swparams)) < 0) {
    ERRORLOG("Setting of swparams failed: %s\n", snd_strerror(err));
    return -1;
  }

  return 0;
}

int alsa_close( void )
{
  int err;

  DEBUGLOG("alsa_close\n");
  
  err = alsa_stop();

#if (SND_LIB_MAJOR < 1)
   err = snd_pcm_close(handle);
#else
   err = snd_pcm_close(handle);
#endif

  free(device);

  return err;
}

/*
 * Play some audio and pass a status message upstream, if applicable.
 * Returns 0 on success.
 */
int
alsa_play(struct cdda_block *blk)
{
  signed short *ptr;
  int err = 0, frames;

  ptr = (signed short *)blk->buf;
  frames = blk->buflen / (channels * 2);
  DEBUGLOG("play %i frames, %i bytes\n", frames, blk->buflen);
  while (frames > 0) {
    err = snd_pcm_writei(handle, ptr, frames);

    if (err == -EAGAIN)
      continue;
    if(err == -EPIPE) {
      err = snd_pcm_prepare(handle);
      continue;
    } else if (err < 0)
      break;

    ptr += err * channels;
    frames -= err;
    DEBUGLOG("played %i, rest %i\n", err / channels, frames);
  }

  if (err < 0) {
    ERRORLOG("alsa_write failed: %s\n", snd_strerror(err));
    err = snd_pcm_prepare(handle);

    if (err < 0) {
      ERRORLOG("Unable to snd_pcm_prepare pcm stream: %s\n", snd_strerror(err));
    }
    blk->status = WM_CDM_CDDAERROR;
    return err;
  }

  return 0;
}

/*
 * Stop the audio immediately.
 */
int
alsa_stop( void )
{
  int err;

  DEBUGLOG("alsa_stop\n");

  err = snd_pcm_drop(handle);
  if (err < 0) {
    ERRORLOG("Unable to drop pcm stream: %s\n", snd_strerror(err));
  }
  
  err = snd_pcm_prepare(handle);
  if (err < 0) {
    ERRORLOG("Unable to snd_pcm_prepare pcm stream: %s\n", snd_strerror(err));
  }

  return err;
}

/*
 * Get the current audio state.
 */
int
alsa_state(struct cdda_block *blk)
{
  return -1; /* not implemented yet for ALSA */
}

static struct audio_oops alsa_oops = {
  .wmaudio_open    = alsa_open,
  .wmaudio_close   = alsa_close,
  .wmaudio_play    = alsa_play,
  .wmaudio_stop    = alsa_stop,
  .wmaudio_state   = alsa_state,
  .wmaudio_balance = NULL,
  .wmaudio_volume  = NULL
};

struct audio_oops*
setup_alsa(const char *dev, const char *ctl)
{
  static int init_complete = 0;

  if(dev && strlen(dev) > 0) {
    device = strdup(dev);
  } else {
    device = strdup("plughw:0,0"); /* playback device */
  }

  if(init_complete) {
    ERRORLOG("already initialized\n");
    return NULL;
  }
  if(!alsa_open())
    init_complete = 1;
  else
    return NULL;

  return &alsa_oops;
}

#endif /* ALSA */