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/*
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* QEMU ALSA audio driver
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*
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* Copyright (c) 2005 Vassili Karpov (malc)
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include <alsa/asoundlib.h> |
25 |
#include "vl.h" |
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|
27 |
#define AUDIO_CAP "alsa" |
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#include "audio_int.h" |
29 |
|
30 |
typedef struct ALSAVoiceOut { |
31 |
HWVoiceOut hw; |
32 |
void *pcm_buf;
|
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snd_pcm_t *handle; |
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} ALSAVoiceOut; |
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|
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typedef struct ALSAVoiceIn { |
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HWVoiceIn hw; |
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snd_pcm_t *handle; |
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void *pcm_buf;
|
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} ALSAVoiceIn; |
41 |
|
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static struct { |
43 |
int size_in_usec_in;
|
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int size_in_usec_out;
|
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const char *pcm_name_in; |
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const char *pcm_name_out; |
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unsigned int buffer_size_in; |
48 |
unsigned int period_size_in; |
49 |
unsigned int buffer_size_out; |
50 |
unsigned int period_size_out; |
51 |
unsigned int threshold; |
52 |
|
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int buffer_size_in_overriden;
|
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int period_size_in_overriden;
|
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|
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int buffer_size_out_overriden;
|
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int period_size_out_overriden;
|
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int verbose;
|
59 |
} conf = { |
60 |
#ifdef HIGH_LATENCY
|
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.size_in_usec_in = 1,
|
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.size_in_usec_out = 1,
|
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#endif
|
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.pcm_name_out = "hw:0,0",
|
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.pcm_name_in = "hw:0,0",
|
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#ifdef HIGH_LATENCY
|
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.buffer_size_in = 400000,
|
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.period_size_in = 400000 / 4, |
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.buffer_size_out = 400000,
|
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.period_size_out = 400000 / 4, |
71 |
#else
|
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#define DEFAULT_BUFFER_SIZE 1024 |
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#define DEFAULT_PERIOD_SIZE 256 |
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.buffer_size_in = DEFAULT_BUFFER_SIZE * 4,
|
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.period_size_in = DEFAULT_PERIOD_SIZE * 4,
|
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.buffer_size_out = DEFAULT_BUFFER_SIZE, |
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.period_size_out = DEFAULT_PERIOD_SIZE, |
78 |
.buffer_size_in_overriden = 0,
|
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.buffer_size_out_overriden = 0,
|
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.period_size_in_overriden = 0,
|
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.period_size_out_overriden = 0,
|
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#endif
|
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.threshold = 0,
|
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.verbose = 0
|
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}; |
86 |
|
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struct alsa_params_req {
|
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int freq;
|
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audfmt_e fmt; |
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int nchannels;
|
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unsigned int buffer_size; |
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unsigned int period_size; |
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}; |
94 |
|
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struct alsa_params_obt {
|
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int freq;
|
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audfmt_e fmt; |
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int nchannels;
|
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snd_pcm_uframes_t samples; |
100 |
}; |
101 |
|
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static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) |
103 |
{ |
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va_list ap; |
105 |
|
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va_start (ap, fmt); |
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AUD_vlog (AUDIO_CAP, fmt, ap); |
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va_end (ap); |
109 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
112 |
|
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static void GCC_FMT_ATTR (3, 4) alsa_logerr2 ( |
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int err,
|
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const char *typ, |
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const char *fmt, |
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... |
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) |
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{ |
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va_list ap; |
121 |
|
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AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
|
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|
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va_start (ap, fmt); |
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AUD_vlog (AUDIO_CAP, fmt, ap); |
126 |
va_end (ap); |
127 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
130 |
|
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static void alsa_anal_close (snd_pcm_t **handlep) |
132 |
{ |
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int err = snd_pcm_close (*handlep);
|
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if (err) {
|
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alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
|
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} |
137 |
*handlep = NULL;
|
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} |
139 |
|
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static int alsa_write (SWVoiceOut *sw, void *buf, int len) |
141 |
{ |
142 |
return audio_pcm_sw_write (sw, buf, len);
|
143 |
} |
144 |
|
145 |
static int aud_to_alsafmt (audfmt_e fmt) |
146 |
{ |
147 |
switch (fmt) {
|
148 |
case AUD_FMT_S8:
|
149 |
return SND_PCM_FORMAT_S8;
|
150 |
|
151 |
case AUD_FMT_U8:
|
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return SND_PCM_FORMAT_U8;
|
153 |
|
154 |
case AUD_FMT_S16:
|
155 |
return SND_PCM_FORMAT_S16_LE;
|
156 |
|
157 |
case AUD_FMT_U16:
|
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return SND_PCM_FORMAT_U16_LE;
|
159 |
|
160 |
default:
|
161 |
dolog ("Internal logic error: Bad audio format %d\n", fmt);
|
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#ifdef DEBUG_AUDIO
|
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abort (); |
164 |
#endif
|
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return SND_PCM_FORMAT_U8;
|
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} |
167 |
} |
168 |
|
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static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness) |
170 |
{ |
171 |
switch (alsafmt) {
|
172 |
case SND_PCM_FORMAT_S8:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_S8; |
175 |
break;
|
176 |
|
177 |
case SND_PCM_FORMAT_U8:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U8; |
180 |
break;
|
181 |
|
182 |
case SND_PCM_FORMAT_S16_LE:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_S16; |
185 |
break;
|
186 |
|
187 |
case SND_PCM_FORMAT_U16_LE:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U16; |
190 |
break;
|
191 |
|
192 |
case SND_PCM_FORMAT_S16_BE:
|
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*endianness = 1;
|
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*fmt = AUD_FMT_S16; |
195 |
break;
|
196 |
|
197 |
case SND_PCM_FORMAT_U16_BE:
|
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*endianness = 1;
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*fmt = AUD_FMT_U16; |
200 |
break;
|
201 |
|
202 |
default:
|
203 |
dolog ("Unrecognized audio format %d\n", alsafmt);
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return -1; |
205 |
} |
206 |
|
207 |
return 0; |
208 |
} |
209 |
|
210 |
#if defined DEBUG_MISMATCHES || defined DEBUG
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211 |
static void alsa_dump_info (struct alsa_params_req *req, |
212 |
struct alsa_params_obt *obt)
|
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{ |
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dolog ("parameter | requested value | obtained value\n");
|
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dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
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dolog ("channels | %10d | %10d\n",
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req->nchannels, obt->nchannels); |
218 |
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
|
219 |
dolog ("============================================\n");
|
220 |
dolog ("requested: buffer size %d period size %d\n",
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req->buffer_size, req->period_size); |
222 |
dolog ("obtained: samples %ld\n", obt->samples);
|
223 |
} |
224 |
#endif
|
225 |
|
226 |
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) |
227 |
{ |
228 |
int err;
|
229 |
snd_pcm_sw_params_t *sw_params; |
230 |
|
231 |
snd_pcm_sw_params_alloca (&sw_params); |
232 |
|
233 |
err = snd_pcm_sw_params_current (handle, sw_params); |
234 |
if (err < 0) { |
235 |
dolog ("Could not fully initialize DAC\n");
|
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alsa_logerr (err, "Failed to get current software parameters\n");
|
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return;
|
238 |
} |
239 |
|
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err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); |
241 |
if (err < 0) { |
242 |
dolog ("Could not fully initialize DAC\n");
|
243 |
alsa_logerr (err, "Failed to set software threshold to %ld\n",
|
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threshold); |
245 |
return;
|
246 |
} |
247 |
|
248 |
err = snd_pcm_sw_params (handle, sw_params); |
249 |
if (err < 0) { |
250 |
dolog ("Could not fully initialize DAC\n");
|
251 |
alsa_logerr (err, "Failed to set software parameters\n");
|
252 |
return;
|
253 |
} |
254 |
} |
255 |
|
256 |
static int alsa_open (int in, struct alsa_params_req *req, |
257 |
struct alsa_params_obt *obt, snd_pcm_t **handlep)
|
258 |
{ |
259 |
snd_pcm_t *handle; |
260 |
snd_pcm_hw_params_t *hw_params; |
261 |
int err, freq, nchannels;
|
262 |
const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; |
263 |
unsigned int period_size, buffer_size; |
264 |
snd_pcm_uframes_t obt_buffer_size; |
265 |
const char *typ = in ? "ADC" : "DAC"; |
266 |
|
267 |
freq = req->freq; |
268 |
period_size = req->period_size; |
269 |
buffer_size = req->buffer_size; |
270 |
nchannels = req->nchannels; |
271 |
|
272 |
snd_pcm_hw_params_alloca (&hw_params); |
273 |
|
274 |
err = snd_pcm_open ( |
275 |
&handle, |
276 |
pcm_name, |
277 |
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, |
278 |
SND_PCM_NONBLOCK |
279 |
); |
280 |
if (err < 0) { |
281 |
alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
|
282 |
return -1; |
283 |
} |
284 |
|
285 |
err = snd_pcm_hw_params_any (handle, hw_params); |
286 |
if (err < 0) { |
287 |
alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
|
288 |
goto err;
|
289 |
} |
290 |
|
291 |
err = snd_pcm_hw_params_set_access ( |
292 |
handle, |
293 |
hw_params, |
294 |
SND_PCM_ACCESS_RW_INTERLEAVED |
295 |
); |
296 |
if (err < 0) { |
297 |
alsa_logerr2 (err, typ, "Failed to set access type\n");
|
298 |
goto err;
|
299 |
} |
300 |
|
301 |
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); |
302 |
if (err < 0) { |
303 |
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
|
304 |
goto err;
|
305 |
} |
306 |
|
307 |
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
|
308 |
if (err < 0) { |
309 |
alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
|
310 |
goto err;
|
311 |
} |
312 |
|
313 |
err = snd_pcm_hw_params_set_channels_near ( |
314 |
handle, |
315 |
hw_params, |
316 |
&nchannels |
317 |
); |
318 |
if (err < 0) { |
319 |
alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
|
320 |
req->nchannels); |
321 |
goto err;
|
322 |
} |
323 |
|
324 |
if (nchannels != 1 && nchannels != 2) { |
325 |
alsa_logerr2 (err, typ, |
326 |
"Can not handle obtained number of channels %d\n",
|
327 |
nchannels); |
328 |
goto err;
|
329 |
} |
330 |
|
331 |
if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
|
332 |
if (!buffer_size) {
|
333 |
buffer_size = DEFAULT_BUFFER_SIZE; |
334 |
period_size= DEFAULT_PERIOD_SIZE; |
335 |
} |
336 |
} |
337 |
|
338 |
if (buffer_size) {
|
339 |
if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
|
340 |
if (period_size) {
|
341 |
err = snd_pcm_hw_params_set_period_time_near ( |
342 |
handle, |
343 |
hw_params, |
344 |
&period_size, |
345 |
0
|
346 |
); |
347 |
if (err < 0) { |
348 |
alsa_logerr2 (err, typ, |
349 |
"Failed to set period time %d\n",
|
350 |
req->period_size); |
351 |
goto err;
|
352 |
} |
353 |
} |
354 |
|
355 |
err = snd_pcm_hw_params_set_buffer_time_near ( |
356 |
handle, |
357 |
hw_params, |
358 |
&buffer_size, |
359 |
0
|
360 |
); |
361 |
|
362 |
if (err < 0) { |
363 |
alsa_logerr2 (err, typ, |
364 |
"Failed to set buffer time %d\n",
|
365 |
req->buffer_size); |
366 |
goto err;
|
367 |
} |
368 |
} |
369 |
else {
|
370 |
int dir;
|
371 |
snd_pcm_uframes_t minval; |
372 |
|
373 |
if (period_size) {
|
374 |
minval = period_size; |
375 |
dir = 0;
|
376 |
|
377 |
err = snd_pcm_hw_params_get_period_size_min ( |
378 |
hw_params, |
379 |
&minval, |
380 |
&dir |
381 |
); |
382 |
if (err < 0) { |
383 |
alsa_logerr ( |
384 |
err, |
385 |
"Could not get minmal period size for %s\n",
|
386 |
typ |
387 |
); |
388 |
} |
389 |
else {
|
390 |
if (period_size < minval) {
|
391 |
if ((in && conf.period_size_in_overriden)
|
392 |
|| (!in && conf.period_size_out_overriden)) { |
393 |
dolog ("%s period size(%d) is less "
|
394 |
"than minmal period size(%ld)\n",
|
395 |
typ, |
396 |
period_size, |
397 |
minval); |
398 |
} |
399 |
period_size = minval; |
400 |
} |
401 |
} |
402 |
|
403 |
err = snd_pcm_hw_params_set_period_size ( |
404 |
handle, |
405 |
hw_params, |
406 |
period_size, |
407 |
0
|
408 |
); |
409 |
if (err < 0) { |
410 |
alsa_logerr2 (err, typ, "Failed to set period size %d\n",
|
411 |
req->period_size); |
412 |
goto err;
|
413 |
} |
414 |
} |
415 |
|
416 |
minval = buffer_size; |
417 |
err = snd_pcm_hw_params_get_buffer_size_min ( |
418 |
hw_params, |
419 |
&minval |
420 |
); |
421 |
if (err < 0) { |
422 |
alsa_logerr (err, "Could not get minmal buffer size for %s\n",
|
423 |
typ); |
424 |
} |
425 |
else {
|
426 |
if (buffer_size < minval) {
|
427 |
if ((in && conf.buffer_size_in_overriden)
|
428 |
|| (!in && conf.buffer_size_out_overriden)) { |
429 |
dolog ( |
430 |
"%s buffer size(%d) is less "
|
431 |
"than minimal buffer size(%ld)\n",
|
432 |
typ, |
433 |
buffer_size, |
434 |
minval |
435 |
); |
436 |
} |
437 |
buffer_size = minval; |
438 |
} |
439 |
} |
440 |
|
441 |
err = snd_pcm_hw_params_set_buffer_size ( |
442 |
handle, |
443 |
hw_params, |
444 |
buffer_size |
445 |
); |
446 |
if (err < 0) { |
447 |
alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
|
448 |
req->buffer_size); |
449 |
goto err;
|
450 |
} |
451 |
} |
452 |
} |
453 |
else {
|
454 |
dolog ("warning: Buffer size is not set\n");
|
455 |
} |
456 |
|
457 |
err = snd_pcm_hw_params (handle, hw_params); |
458 |
if (err < 0) { |
459 |
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
|
460 |
goto err;
|
461 |
} |
462 |
|
463 |
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); |
464 |
if (err < 0) { |
465 |
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
|
466 |
goto err;
|
467 |
} |
468 |
|
469 |
err = snd_pcm_prepare (handle); |
470 |
if (err < 0) { |
471 |
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
|
472 |
goto err;
|
473 |
} |
474 |
|
475 |
if (!in && conf.threshold) {
|
476 |
snd_pcm_uframes_t threshold; |
477 |
int bytes_per_sec;
|
478 |
|
479 |
bytes_per_sec = freq |
480 |
<< (nchannels == 2)
|
481 |
<< (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16); |
482 |
|
483 |
threshold = (conf.threshold * bytes_per_sec) / 1000;
|
484 |
alsa_set_threshold (handle, threshold); |
485 |
} |
486 |
|
487 |
obt->fmt = req->fmt; |
488 |
obt->nchannels = nchannels; |
489 |
obt->freq = freq; |
490 |
obt->samples = obt_buffer_size; |
491 |
*handlep = handle; |
492 |
|
493 |
#if defined DEBUG_MISMATCHES || defined DEBUG
|
494 |
if (obt->fmt != req->fmt ||
|
495 |
obt->nchannels != req->nchannels || |
496 |
obt->freq != req->freq) { |
497 |
dolog ("Audio paramters mismatch for %s\n", typ);
|
498 |
alsa_dump_info (req, obt); |
499 |
} |
500 |
#endif
|
501 |
|
502 |
#ifdef DEBUG
|
503 |
alsa_dump_info (req, obt); |
504 |
#endif
|
505 |
return 0; |
506 |
|
507 |
err:
|
508 |
alsa_anal_close (&handle); |
509 |
return -1; |
510 |
} |
511 |
|
512 |
static int alsa_recover (snd_pcm_t *handle) |
513 |
{ |
514 |
int err = snd_pcm_prepare (handle);
|
515 |
if (err < 0) { |
516 |
alsa_logerr (err, "Failed to prepare handle %p\n", handle);
|
517 |
return -1; |
518 |
} |
519 |
return 0; |
520 |
} |
521 |
|
522 |
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
|
523 |
{ |
524 |
snd_pcm_sframes_t avail; |
525 |
|
526 |
avail = snd_pcm_avail_update (handle); |
527 |
if (avail < 0) { |
528 |
if (avail == -EPIPE) {
|
529 |
if (!alsa_recover (handle)) {
|
530 |
avail = snd_pcm_avail_update (handle); |
531 |
} |
532 |
} |
533 |
|
534 |
if (avail < 0) { |
535 |
alsa_logerr (avail, |
536 |
"Could not obtain number of available frames\n");
|
537 |
return -1; |
538 |
} |
539 |
} |
540 |
|
541 |
return avail;
|
542 |
} |
543 |
|
544 |
static int alsa_run_out (HWVoiceOut *hw) |
545 |
{ |
546 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
547 |
int rpos, live, decr;
|
548 |
int samples;
|
549 |
uint8_t *dst; |
550 |
st_sample_t *src; |
551 |
snd_pcm_sframes_t avail; |
552 |
|
553 |
live = audio_pcm_hw_get_live_out (hw); |
554 |
if (!live) {
|
555 |
return 0; |
556 |
} |
557 |
|
558 |
avail = alsa_get_avail (alsa->handle); |
559 |
if (avail < 0) { |
560 |
dolog ("Could not get number of available playback frames\n");
|
561 |
return 0; |
562 |
} |
563 |
|
564 |
decr = audio_MIN (live, avail); |
565 |
samples = decr; |
566 |
rpos = hw->rpos; |
567 |
while (samples) {
|
568 |
int left_till_end_samples = hw->samples - rpos;
|
569 |
int len = audio_MIN (samples, left_till_end_samples);
|
570 |
snd_pcm_sframes_t written; |
571 |
|
572 |
src = hw->mix_buf + rpos; |
573 |
dst = advance (alsa->pcm_buf, rpos << hw->info.shift); |
574 |
|
575 |
hw->clip (dst, src, len); |
576 |
|
577 |
while (len) {
|
578 |
written = snd_pcm_writei (alsa->handle, dst, len); |
579 |
|
580 |
if (written <= 0) { |
581 |
switch (written) {
|
582 |
case 0: |
583 |
if (conf.verbose) {
|
584 |
dolog ("Failed to write %d frames (wrote zero)\n", len);
|
585 |
} |
586 |
goto exit;
|
587 |
|
588 |
case -EPIPE:
|
589 |
if (alsa_recover (alsa->handle)) {
|
590 |
alsa_logerr (written, "Failed to write %d frames\n",
|
591 |
len); |
592 |
goto exit;
|
593 |
} |
594 |
if (conf.verbose) {
|
595 |
dolog ("Recovering from playback xrun\n");
|
596 |
} |
597 |
continue;
|
598 |
|
599 |
case -EAGAIN:
|
600 |
goto exit;
|
601 |
|
602 |
default:
|
603 |
alsa_logerr (written, "Failed to write %d frames to %p\n",
|
604 |
len, dst); |
605 |
goto exit;
|
606 |
} |
607 |
} |
608 |
|
609 |
mixeng_clear (src, written); |
610 |
rpos = (rpos + written) % hw->samples; |
611 |
samples -= written; |
612 |
len -= written; |
613 |
dst = advance (dst, written << hw->info.shift); |
614 |
src += written; |
615 |
} |
616 |
} |
617 |
|
618 |
exit:
|
619 |
hw->rpos = rpos; |
620 |
return decr;
|
621 |
} |
622 |
|
623 |
static void alsa_fini_out (HWVoiceOut *hw) |
624 |
{ |
625 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
626 |
|
627 |
ldebug ("alsa_fini\n");
|
628 |
alsa_anal_close (&alsa->handle); |
629 |
|
630 |
if (alsa->pcm_buf) {
|
631 |
qemu_free (alsa->pcm_buf); |
632 |
alsa->pcm_buf = NULL;
|
633 |
} |
634 |
} |
635 |
|
636 |
static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) |
637 |
{ |
638 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
639 |
struct alsa_params_req req;
|
640 |
struct alsa_params_obt obt;
|
641 |
audfmt_e effective_fmt; |
642 |
int endianness;
|
643 |
int err;
|
644 |
snd_pcm_t *handle; |
645 |
audsettings_t obt_as; |
646 |
|
647 |
req.fmt = aud_to_alsafmt (as->fmt); |
648 |
req.freq = as->freq; |
649 |
req.nchannels = as->nchannels; |
650 |
req.period_size = conf.period_size_out; |
651 |
req.buffer_size = conf.buffer_size_out; |
652 |
|
653 |
if (alsa_open (0, &req, &obt, &handle)) { |
654 |
return -1; |
655 |
} |
656 |
|
657 |
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); |
658 |
if (err) {
|
659 |
alsa_anal_close (&handle); |
660 |
return -1; |
661 |
} |
662 |
|
663 |
obt_as.freq = obt.freq; |
664 |
obt_as.nchannels = obt.nchannels; |
665 |
obt_as.fmt = effective_fmt; |
666 |
|
667 |
audio_pcm_init_info ( |
668 |
&hw->info, |
669 |
&obt_as, |
670 |
audio_need_to_swap_endian (endianness) |
671 |
); |
672 |
hw->samples = obt.samples; |
673 |
|
674 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
|
675 |
if (!alsa->pcm_buf) {
|
676 |
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
|
677 |
hw->samples, 1 << hw->info.shift);
|
678 |
alsa_anal_close (&handle); |
679 |
return -1; |
680 |
} |
681 |
|
682 |
alsa->handle = handle; |
683 |
return 0; |
684 |
} |
685 |
|
686 |
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause) |
687 |
{ |
688 |
int err;
|
689 |
|
690 |
if (pause) {
|
691 |
err = snd_pcm_drop (handle); |
692 |
if (err < 0) { |
693 |
alsa_logerr (err, "Could not stop %s\n", typ);
|
694 |
return -1; |
695 |
} |
696 |
} |
697 |
else {
|
698 |
err = snd_pcm_prepare (handle); |
699 |
if (err < 0) { |
700 |
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
|
701 |
return -1; |
702 |
} |
703 |
} |
704 |
|
705 |
return 0; |
706 |
} |
707 |
|
708 |
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) |
709 |
{ |
710 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
711 |
|
712 |
switch (cmd) {
|
713 |
case VOICE_ENABLE:
|
714 |
ldebug ("enabling voice\n");
|
715 |
return alsa_voice_ctl (alsa->handle, "playback", 0); |
716 |
|
717 |
case VOICE_DISABLE:
|
718 |
ldebug ("disabling voice\n");
|
719 |
return alsa_voice_ctl (alsa->handle, "playback", 1); |
720 |
} |
721 |
|
722 |
return -1; |
723 |
} |
724 |
|
725 |
static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) |
726 |
{ |
727 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
728 |
struct alsa_params_req req;
|
729 |
struct alsa_params_obt obt;
|
730 |
int endianness;
|
731 |
int err;
|
732 |
audfmt_e effective_fmt; |
733 |
snd_pcm_t *handle; |
734 |
audsettings_t obt_as; |
735 |
|
736 |
req.fmt = aud_to_alsafmt (as->fmt); |
737 |
req.freq = as->freq; |
738 |
req.nchannels = as->nchannels; |
739 |
req.period_size = conf.period_size_in; |
740 |
req.buffer_size = conf.buffer_size_in; |
741 |
|
742 |
if (alsa_open (1, &req, &obt, &handle)) { |
743 |
return -1; |
744 |
} |
745 |
|
746 |
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); |
747 |
if (err) {
|
748 |
alsa_anal_close (&handle); |
749 |
return -1; |
750 |
} |
751 |
|
752 |
obt_as.freq = obt.freq; |
753 |
obt_as.nchannels = obt.nchannels; |
754 |
obt_as.fmt = effective_fmt; |
755 |
|
756 |
audio_pcm_init_info ( |
757 |
&hw->info, |
758 |
&obt_as, |
759 |
audio_need_to_swap_endian (endianness) |
760 |
); |
761 |
hw->samples = obt.samples; |
762 |
|
763 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
764 |
if (!alsa->pcm_buf) {
|
765 |
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
|
766 |
hw->samples, 1 << hw->info.shift);
|
767 |
alsa_anal_close (&handle); |
768 |
return -1; |
769 |
} |
770 |
|
771 |
alsa->handle = handle; |
772 |
return 0; |
773 |
} |
774 |
|
775 |
static void alsa_fini_in (HWVoiceIn *hw) |
776 |
{ |
777 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
778 |
|
779 |
alsa_anal_close (&alsa->handle); |
780 |
|
781 |
if (alsa->pcm_buf) {
|
782 |
qemu_free (alsa->pcm_buf); |
783 |
alsa->pcm_buf = NULL;
|
784 |
} |
785 |
} |
786 |
|
787 |
static int alsa_run_in (HWVoiceIn *hw) |
788 |
{ |
789 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
790 |
int hwshift = hw->info.shift;
|
791 |
int i;
|
792 |
int live = audio_pcm_hw_get_live_in (hw);
|
793 |
int dead = hw->samples - live;
|
794 |
int decr;
|
795 |
struct {
|
796 |
int add;
|
797 |
int len;
|
798 |
} bufs[2] = {
|
799 |
{ hw->wpos, 0 },
|
800 |
{ 0, 0 } |
801 |
}; |
802 |
snd_pcm_sframes_t avail; |
803 |
snd_pcm_uframes_t read_samples = 0;
|
804 |
|
805 |
if (!dead) {
|
806 |
return 0; |
807 |
} |
808 |
|
809 |
avail = alsa_get_avail (alsa->handle); |
810 |
if (avail < 0) { |
811 |
dolog ("Could not get number of captured frames\n");
|
812 |
return 0; |
813 |
} |
814 |
|
815 |
if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
|
816 |
avail = hw->samples; |
817 |
} |
818 |
|
819 |
decr = audio_MIN (dead, avail); |
820 |
if (!decr) {
|
821 |
return 0; |
822 |
} |
823 |
|
824 |
if (hw->wpos + decr > hw->samples) {
|
825 |
bufs[0].len = (hw->samples - hw->wpos);
|
826 |
bufs[1].len = (decr - (hw->samples - hw->wpos));
|
827 |
} |
828 |
else {
|
829 |
bufs[0].len = decr;
|
830 |
} |
831 |
|
832 |
for (i = 0; i < 2; ++i) { |
833 |
void *src;
|
834 |
st_sample_t *dst; |
835 |
snd_pcm_sframes_t nread; |
836 |
snd_pcm_uframes_t len; |
837 |
|
838 |
len = bufs[i].len; |
839 |
|
840 |
src = advance (alsa->pcm_buf, bufs[i].add << hwshift); |
841 |
dst = hw->conv_buf + bufs[i].add; |
842 |
|
843 |
while (len) {
|
844 |
nread = snd_pcm_readi (alsa->handle, src, len); |
845 |
|
846 |
if (nread <= 0) { |
847 |
switch (nread) {
|
848 |
case 0: |
849 |
if (conf.verbose) {
|
850 |
dolog ("Failed to read %ld frames (read zero)\n", len);
|
851 |
} |
852 |
goto exit;
|
853 |
|
854 |
case -EPIPE:
|
855 |
if (alsa_recover (alsa->handle)) {
|
856 |
alsa_logerr (nread, "Failed to read %ld frames\n", len);
|
857 |
goto exit;
|
858 |
} |
859 |
if (conf.verbose) {
|
860 |
dolog ("Recovering from capture xrun\n");
|
861 |
} |
862 |
continue;
|
863 |
|
864 |
case -EAGAIN:
|
865 |
goto exit;
|
866 |
|
867 |
default:
|
868 |
alsa_logerr ( |
869 |
nread, |
870 |
"Failed to read %ld frames from %p\n",
|
871 |
len, |
872 |
src |
873 |
); |
874 |
goto exit;
|
875 |
} |
876 |
} |
877 |
|
878 |
hw->conv (dst, src, nread, &nominal_volume); |
879 |
|
880 |
src = advance (src, nread << hwshift); |
881 |
dst += nread; |
882 |
|
883 |
read_samples += nread; |
884 |
len -= nread; |
885 |
} |
886 |
} |
887 |
|
888 |
exit:
|
889 |
hw->wpos = (hw->wpos + read_samples) % hw->samples; |
890 |
return read_samples;
|
891 |
} |
892 |
|
893 |
static int alsa_read (SWVoiceIn *sw, void *buf, int size) |
894 |
{ |
895 |
return audio_pcm_sw_read (sw, buf, size);
|
896 |
} |
897 |
|
898 |
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) |
899 |
{ |
900 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
901 |
|
902 |
switch (cmd) {
|
903 |
case VOICE_ENABLE:
|
904 |
ldebug ("enabling voice\n");
|
905 |
return alsa_voice_ctl (alsa->handle, "capture", 0); |
906 |
|
907 |
case VOICE_DISABLE:
|
908 |
ldebug ("disabling voice\n");
|
909 |
return alsa_voice_ctl (alsa->handle, "capture", 1); |
910 |
} |
911 |
|
912 |
return -1; |
913 |
} |
914 |
|
915 |
static void *alsa_audio_init (void) |
916 |
{ |
917 |
return &conf;
|
918 |
} |
919 |
|
920 |
static void alsa_audio_fini (void *opaque) |
921 |
{ |
922 |
(void) opaque;
|
923 |
} |
924 |
|
925 |
static struct audio_option alsa_options[] = { |
926 |
{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
|
927 |
"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
928 |
{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
|
929 |
"DAC period size", &conf.period_size_out_overriden, 0}, |
930 |
{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
|
931 |
"DAC buffer size", &conf.buffer_size_out_overriden, 0}, |
932 |
|
933 |
{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
|
934 |
"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
935 |
{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
|
936 |
"ADC period size", &conf.period_size_in_overriden, 0}, |
937 |
{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
|
938 |
"ADC buffer size", &conf.buffer_size_in_overriden, 0}, |
939 |
|
940 |
{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
|
941 |
"(undocumented)", NULL, 0}, |
942 |
|
943 |
{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
|
944 |
"DAC device name (for instance dmix)", NULL, 0}, |
945 |
|
946 |
{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
|
947 |
"ADC device name", NULL, 0}, |
948 |
|
949 |
{"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
|
950 |
"Behave in a more verbose way", NULL, 0}, |
951 |
|
952 |
{NULL, 0, NULL, NULL, NULL, 0} |
953 |
}; |
954 |
|
955 |
static struct audio_pcm_ops alsa_pcm_ops = { |
956 |
alsa_init_out, |
957 |
alsa_fini_out, |
958 |
alsa_run_out, |
959 |
alsa_write, |
960 |
alsa_ctl_out, |
961 |
|
962 |
alsa_init_in, |
963 |
alsa_fini_in, |
964 |
alsa_run_in, |
965 |
alsa_read, |
966 |
alsa_ctl_in |
967 |
}; |
968 |
|
969 |
struct audio_driver alsa_audio_driver = {
|
970 |
INIT_FIELD (name = ) "alsa",
|
971 |
INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
|
972 |
INIT_FIELD (options = ) alsa_options, |
973 |
INIT_FIELD (init = ) alsa_audio_init, |
974 |
INIT_FIELD (fini = ) alsa_audio_fini, |
975 |
INIT_FIELD (pcm_ops = ) &alsa_pcm_ops, |
976 |
INIT_FIELD (can_be_default = ) 1,
|
977 |
INIT_FIELD (max_voices_out = ) INT_MAX, |
978 |
INIT_FIELD (max_voices_in = ) INT_MAX, |
979 |
INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
|
980 |
INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
|
981 |
}; |