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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" |
26 |
|
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#define AUDIO_CAP "alsa" |
28 |
#include "audio_int.h" |
29 |
|
30 |
typedef struct ALSAVoiceOut { |
31 |
HWVoiceOut hw; |
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void *pcm_buf;
|
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snd_pcm_t *handle; |
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int can_pause;
|
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int was_enabled;
|
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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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int can_pause;
|
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} ALSAVoiceIn; |
44 |
|
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static struct { |
46 |
int size_in_usec_in;
|
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int size_in_usec_out;
|
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const char *pcm_name_in; |
49 |
const char *pcm_name_out; |
50 |
unsigned int buffer_size_in; |
51 |
unsigned int period_size_in; |
52 |
unsigned int buffer_size_out; |
53 |
unsigned int period_size_out; |
54 |
unsigned int threshold; |
55 |
|
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int buffer_size_in_overriden;
|
57 |
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;
|
61 |
} conf = { |
62 |
#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, |
73 |
#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, |
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.period_size_in = DEFAULT_PERIOD_SIZE, |
78 |
.buffer_size_out = DEFAULT_BUFFER_SIZE, |
79 |
.period_size_out = DEFAULT_PERIOD_SIZE, |
80 |
.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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}; |
87 |
|
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struct alsa_params_req {
|
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int freq;
|
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audfmt_e fmt; |
91 |
int nchannels;
|
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unsigned int buffer_size; |
93 |
unsigned int period_size; |
94 |
}; |
95 |
|
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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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int can_pause;
|
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snd_pcm_uframes_t samples; |
102 |
}; |
103 |
|
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static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) |
105 |
{ |
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va_list ap; |
107 |
|
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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); |
111 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
114 |
|
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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; |
123 |
|
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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); |
128 |
va_end (ap); |
129 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
132 |
|
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static void alsa_anal_close (snd_pcm_t **handlep) |
134 |
{ |
135 |
int err = snd_pcm_close (*handlep);
|
136 |
if (err) {
|
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alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
|
138 |
} |
139 |
*handlep = NULL;
|
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} |
141 |
|
142 |
static int alsa_write (SWVoiceOut *sw, void *buf, int len) |
143 |
{ |
144 |
return audio_pcm_sw_write (sw, buf, len);
|
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} |
146 |
|
147 |
static int aud_to_alsafmt (audfmt_e fmt) |
148 |
{ |
149 |
switch (fmt) {
|
150 |
case AUD_FMT_S8:
|
151 |
return SND_PCM_FORMAT_S8;
|
152 |
|
153 |
case AUD_FMT_U8:
|
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return SND_PCM_FORMAT_U8;
|
155 |
|
156 |
case AUD_FMT_S16:
|
157 |
return SND_PCM_FORMAT_S16_LE;
|
158 |
|
159 |
case AUD_FMT_U16:
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return SND_PCM_FORMAT_U16_LE;
|
161 |
|
162 |
default:
|
163 |
dolog ("Internal logic error: Bad audio format %d\n", fmt);
|
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#ifdef DEBUG_AUDIO
|
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abort (); |
166 |
#endif
|
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return SND_PCM_FORMAT_U8;
|
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} |
169 |
} |
170 |
|
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static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness) |
172 |
{ |
173 |
switch (alsafmt) {
|
174 |
case SND_PCM_FORMAT_S8:
|
175 |
*endianness = 0;
|
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*fmt = AUD_FMT_S8; |
177 |
break;
|
178 |
|
179 |
case SND_PCM_FORMAT_U8:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U8; |
182 |
break;
|
183 |
|
184 |
case SND_PCM_FORMAT_S16_LE:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_S16; |
187 |
break;
|
188 |
|
189 |
case SND_PCM_FORMAT_U16_LE:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U16; |
192 |
break;
|
193 |
|
194 |
case SND_PCM_FORMAT_S16_BE:
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*endianness = 1;
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*fmt = AUD_FMT_S16; |
197 |
break;
|
198 |
|
199 |
case SND_PCM_FORMAT_U16_BE:
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*endianness = 1;
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*fmt = AUD_FMT_U16; |
202 |
break;
|
203 |
|
204 |
default:
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dolog ("Unrecognized audio format %d\n", alsafmt);
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return -1; |
207 |
} |
208 |
|
209 |
return 0; |
210 |
} |
211 |
|
212 |
#if defined DEBUG_MISMATCHES || defined DEBUG
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static void alsa_dump_info (struct alsa_params_req *req, |
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struct alsa_params_obt *obt)
|
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{ |
216 |
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); |
220 |
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
|
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dolog ("============================================\n");
|
222 |
dolog ("requested: buffer size %d period size %d\n",
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req->buffer_size, req->period_size); |
224 |
dolog ("obtained: samples %ld\n", obt->samples);
|
225 |
} |
226 |
#endif
|
227 |
|
228 |
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) |
229 |
{ |
230 |
int err;
|
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snd_pcm_sw_params_t *sw_params; |
232 |
|
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snd_pcm_sw_params_alloca (&sw_params); |
234 |
|
235 |
err = snd_pcm_sw_params_current (handle, sw_params); |
236 |
if (err < 0) { |
237 |
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;
|
240 |
} |
241 |
|
242 |
err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); |
243 |
if (err < 0) { |
244 |
dolog ("Could not fully initialize DAC\n");
|
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alsa_logerr (err, "Failed to set software threshold to %ld\n",
|
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threshold); |
247 |
return;
|
248 |
} |
249 |
|
250 |
err = snd_pcm_sw_params (handle, sw_params); |
251 |
if (err < 0) { |
252 |
dolog ("Could not fully initialize DAC\n");
|
253 |
alsa_logerr (err, "Failed to set software parameters\n");
|
254 |
return;
|
255 |
} |
256 |
} |
257 |
|
258 |
static int alsa_open (int in, struct alsa_params_req *req, |
259 |
struct alsa_params_obt *obt, snd_pcm_t **handlep)
|
260 |
{ |
261 |
snd_pcm_t *handle; |
262 |
snd_pcm_hw_params_t *hw_params; |
263 |
int err, freq, nchannels;
|
264 |
const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; |
265 |
unsigned int period_size, buffer_size; |
266 |
snd_pcm_uframes_t obt_buffer_size; |
267 |
const char *typ = in ? "ADC" : "DAC"; |
268 |
|
269 |
freq = req->freq; |
270 |
period_size = req->period_size; |
271 |
buffer_size = req->buffer_size; |
272 |
nchannels = req->nchannels; |
273 |
|
274 |
snd_pcm_hw_params_alloca (&hw_params); |
275 |
|
276 |
err = snd_pcm_open ( |
277 |
&handle, |
278 |
pcm_name, |
279 |
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, |
280 |
SND_PCM_NONBLOCK |
281 |
); |
282 |
if (err < 0) { |
283 |
alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
|
284 |
return -1; |
285 |
} |
286 |
|
287 |
err = snd_pcm_hw_params_any (handle, hw_params); |
288 |
if (err < 0) { |
289 |
alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
|
290 |
goto err;
|
291 |
} |
292 |
|
293 |
err = snd_pcm_hw_params_set_access ( |
294 |
handle, |
295 |
hw_params, |
296 |
SND_PCM_ACCESS_RW_INTERLEAVED |
297 |
); |
298 |
if (err < 0) { |
299 |
alsa_logerr2 (err, typ, "Failed to set access type\n");
|
300 |
goto err;
|
301 |
} |
302 |
|
303 |
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); |
304 |
if (err < 0) { |
305 |
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
|
306 |
goto err;
|
307 |
} |
308 |
|
309 |
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
|
310 |
if (err < 0) { |
311 |
alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
|
312 |
goto err;
|
313 |
} |
314 |
|
315 |
err = snd_pcm_hw_params_set_channels_near ( |
316 |
handle, |
317 |
hw_params, |
318 |
&nchannels |
319 |
); |
320 |
if (err < 0) { |
321 |
alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
|
322 |
req->nchannels); |
323 |
goto err;
|
324 |
} |
325 |
|
326 |
if (nchannels != 1 && nchannels != 2) { |
327 |
alsa_logerr2 (err, typ, |
328 |
"Can not handle obtained number of channels %d\n",
|
329 |
nchannels); |
330 |
goto err;
|
331 |
} |
332 |
|
333 |
if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
|
334 |
if (!buffer_size) {
|
335 |
buffer_size = DEFAULT_BUFFER_SIZE; |
336 |
period_size= DEFAULT_PERIOD_SIZE; |
337 |
} |
338 |
} |
339 |
|
340 |
if (buffer_size) {
|
341 |
if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
|
342 |
if (period_size) {
|
343 |
err = snd_pcm_hw_params_set_period_time_near ( |
344 |
handle, |
345 |
hw_params, |
346 |
&period_size, |
347 |
0
|
348 |
); |
349 |
if (err < 0) { |
350 |
alsa_logerr2 (err, typ, |
351 |
"Failed to set period time %d\n",
|
352 |
req->period_size); |
353 |
goto err;
|
354 |
} |
355 |
} |
356 |
|
357 |
err = snd_pcm_hw_params_set_buffer_time_near ( |
358 |
handle, |
359 |
hw_params, |
360 |
&buffer_size, |
361 |
0
|
362 |
); |
363 |
|
364 |
if (err < 0) { |
365 |
alsa_logerr2 (err, typ, |
366 |
"Failed to set buffer time %d\n",
|
367 |
req->buffer_size); |
368 |
goto err;
|
369 |
} |
370 |
} |
371 |
else {
|
372 |
int dir;
|
373 |
snd_pcm_uframes_t minval; |
374 |
|
375 |
if (period_size) {
|
376 |
minval = period_size; |
377 |
dir = 0;
|
378 |
|
379 |
err = snd_pcm_hw_params_get_period_size_min ( |
380 |
hw_params, |
381 |
&minval, |
382 |
&dir |
383 |
); |
384 |
if (err < 0) { |
385 |
alsa_logerr ( |
386 |
err, |
387 |
"Could not get minmal period size for %s\n",
|
388 |
typ |
389 |
); |
390 |
} |
391 |
else {
|
392 |
if (period_size < minval) {
|
393 |
if ((in && conf.period_size_in_overriden)
|
394 |
|| (!in && conf.period_size_out_overriden)) { |
395 |
dolog ("%s period size(%d) is less "
|
396 |
"than minmal period size(%ld)\n",
|
397 |
typ, |
398 |
period_size, |
399 |
minval); |
400 |
} |
401 |
period_size = minval; |
402 |
} |
403 |
} |
404 |
|
405 |
err = snd_pcm_hw_params_set_period_size ( |
406 |
handle, |
407 |
hw_params, |
408 |
period_size, |
409 |
0
|
410 |
); |
411 |
if (err < 0) { |
412 |
alsa_logerr2 (err, typ, "Failed to set period size %d\n",
|
413 |
req->period_size); |
414 |
goto err;
|
415 |
} |
416 |
} |
417 |
|
418 |
minval = buffer_size; |
419 |
err = snd_pcm_hw_params_get_buffer_size_min ( |
420 |
hw_params, |
421 |
&minval |
422 |
); |
423 |
if (err < 0) { |
424 |
alsa_logerr (err, "Could not get minmal buffer size for %s\n",
|
425 |
typ); |
426 |
} |
427 |
else {
|
428 |
if (buffer_size < minval) {
|
429 |
if ((in && conf.buffer_size_in_overriden)
|
430 |
|| (!in && conf.buffer_size_out_overriden)) { |
431 |
dolog ( |
432 |
"%s buffer size(%d) is less "
|
433 |
"than minimal buffer size(%ld)\n",
|
434 |
typ, |
435 |
buffer_size, |
436 |
minval |
437 |
); |
438 |
} |
439 |
buffer_size = minval; |
440 |
} |
441 |
} |
442 |
|
443 |
err = snd_pcm_hw_params_set_buffer_size ( |
444 |
handle, |
445 |
hw_params, |
446 |
buffer_size |
447 |
); |
448 |
if (err < 0) { |
449 |
alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
|
450 |
req->buffer_size); |
451 |
goto err;
|
452 |
} |
453 |
} |
454 |
} |
455 |
else {
|
456 |
dolog ("warning: Buffer size is not set\n");
|
457 |
} |
458 |
|
459 |
err = snd_pcm_hw_params (handle, hw_params); |
460 |
if (err < 0) { |
461 |
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
|
462 |
goto err;
|
463 |
} |
464 |
|
465 |
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); |
466 |
if (err < 0) { |
467 |
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
|
468 |
goto err;
|
469 |
} |
470 |
|
471 |
err = snd_pcm_prepare (handle); |
472 |
if (err < 0) { |
473 |
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
|
474 |
goto err;
|
475 |
} |
476 |
|
477 |
obt->can_pause = snd_pcm_hw_params_can_pause (hw_params); |
478 |
if (obt->can_pause < 0) { |
479 |
alsa_logerr (err, "Could not get pause capability for %s\n", typ);
|
480 |
obt->can_pause = 0;
|
481 |
} |
482 |
|
483 |
if (!in && conf.threshold) {
|
484 |
snd_pcm_uframes_t threshold; |
485 |
int bytes_per_sec;
|
486 |
|
487 |
bytes_per_sec = freq |
488 |
<< (nchannels == 2)
|
489 |
<< (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16); |
490 |
|
491 |
threshold = (conf.threshold * bytes_per_sec) / 1000;
|
492 |
alsa_set_threshold (handle, threshold); |
493 |
} |
494 |
|
495 |
obt->fmt = req->fmt; |
496 |
obt->nchannels = nchannels; |
497 |
obt->freq = freq; |
498 |
obt->samples = obt_buffer_size; |
499 |
*handlep = handle; |
500 |
|
501 |
#if defined DEBUG_MISMATCHES || defined DEBUG
|
502 |
if (obt->fmt != req->fmt ||
|
503 |
obt->nchannels != req->nchannels || |
504 |
obt->freq != req->freq) { |
505 |
dolog ("Audio paramters mismatch for %s\n", typ);
|
506 |
alsa_dump_info (req, obt); |
507 |
} |
508 |
#endif
|
509 |
|
510 |
#ifdef DEBUG
|
511 |
alsa_dump_info (req, obt); |
512 |
#endif
|
513 |
return 0; |
514 |
|
515 |
err:
|
516 |
alsa_anal_close (&handle); |
517 |
return -1; |
518 |
} |
519 |
|
520 |
static int alsa_recover (snd_pcm_t *handle) |
521 |
{ |
522 |
int err = snd_pcm_prepare (handle);
|
523 |
if (err < 0) { |
524 |
alsa_logerr (err, "Failed to prepare handle %p\n", handle);
|
525 |
return -1; |
526 |
} |
527 |
return 0; |
528 |
} |
529 |
|
530 |
static int alsa_run_out (HWVoiceOut *hw) |
531 |
{ |
532 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
533 |
int rpos, live, decr;
|
534 |
int samples;
|
535 |
uint8_t *dst; |
536 |
st_sample_t *src; |
537 |
snd_pcm_sframes_t avail; |
538 |
|
539 |
live = audio_pcm_hw_get_live_out (hw); |
540 |
if (!live) {
|
541 |
return 0; |
542 |
} |
543 |
|
544 |
avail = snd_pcm_avail_update (alsa->handle); |
545 |
if (avail < 0) { |
546 |
if (avail == -EPIPE) {
|
547 |
if (!alsa_recover (alsa->handle)) {
|
548 |
avail = snd_pcm_avail_update (alsa->handle); |
549 |
if (avail >= 0) { |
550 |
goto ok;
|
551 |
} |
552 |
} |
553 |
} |
554 |
|
555 |
alsa_logerr (avail, "Could not get amount free space\n");
|
556 |
return 0; |
557 |
} |
558 |
|
559 |
ok:
|
560 |
decr = audio_MIN (live, avail); |
561 |
samples = decr; |
562 |
rpos = hw->rpos; |
563 |
while (samples) {
|
564 |
int left_till_end_samples = hw->samples - rpos;
|
565 |
int convert_samples = audio_MIN (samples, left_till_end_samples);
|
566 |
snd_pcm_sframes_t written; |
567 |
|
568 |
src = hw->mix_buf + rpos; |
569 |
dst = advance (alsa->pcm_buf, rpos << hw->info.shift); |
570 |
|
571 |
hw->clip (dst, src, convert_samples); |
572 |
|
573 |
again:
|
574 |
written = snd_pcm_writei (alsa->handle, dst, convert_samples); |
575 |
|
576 |
if (written < 0) { |
577 |
switch (written) {
|
578 |
case -EPIPE:
|
579 |
if (!alsa_recover (alsa->handle)) {
|
580 |
goto again;
|
581 |
} |
582 |
dolog ( |
583 |
"Failed to write %d frames to %p, handle %p not prepared\n",
|
584 |
convert_samples, |
585 |
dst, |
586 |
alsa->handle |
587 |
); |
588 |
goto exit;
|
589 |
|
590 |
case -EAGAIN:
|
591 |
goto again;
|
592 |
|
593 |
default:
|
594 |
alsa_logerr (written, "Failed to write %d frames to %p\n",
|
595 |
convert_samples, dst); |
596 |
goto exit;
|
597 |
} |
598 |
} |
599 |
|
600 |
mixeng_clear (src, written); |
601 |
rpos = (rpos + written) % hw->samples; |
602 |
samples -= written; |
603 |
} |
604 |
|
605 |
exit:
|
606 |
hw->rpos = rpos; |
607 |
return decr;
|
608 |
} |
609 |
|
610 |
static void alsa_fini_out (HWVoiceOut *hw) |
611 |
{ |
612 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
613 |
|
614 |
ldebug ("alsa_fini\n");
|
615 |
alsa_anal_close (&alsa->handle); |
616 |
|
617 |
if (alsa->pcm_buf) {
|
618 |
qemu_free (alsa->pcm_buf); |
619 |
alsa->pcm_buf = NULL;
|
620 |
} |
621 |
} |
622 |
|
623 |
static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) |
624 |
{ |
625 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
626 |
struct alsa_params_req req;
|
627 |
struct alsa_params_obt obt;
|
628 |
audfmt_e effective_fmt; |
629 |
int endianness;
|
630 |
int err;
|
631 |
snd_pcm_t *handle; |
632 |
audsettings_t obt_as; |
633 |
|
634 |
req.fmt = aud_to_alsafmt (as->fmt); |
635 |
req.freq = as->freq; |
636 |
req.nchannels = as->nchannels; |
637 |
req.period_size = conf.period_size_out; |
638 |
req.buffer_size = conf.buffer_size_out; |
639 |
|
640 |
if (alsa_open (0, &req, &obt, &handle)) { |
641 |
return -1; |
642 |
} |
643 |
|
644 |
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); |
645 |
if (err) {
|
646 |
alsa_anal_close (&handle); |
647 |
return -1; |
648 |
} |
649 |
|
650 |
obt_as.freq = obt.freq; |
651 |
obt_as.nchannels = obt.nchannels; |
652 |
obt_as.fmt = effective_fmt; |
653 |
|
654 |
audio_pcm_init_info ( |
655 |
&hw->info, |
656 |
&obt_as, |
657 |
audio_need_to_swap_endian (endianness) |
658 |
); |
659 |
alsa->can_pause = obt.can_pause; |
660 |
hw->samples = obt.samples; |
661 |
|
662 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
|
663 |
if (!alsa->pcm_buf) {
|
664 |
dolog ("Could not allocate DAC buffer (%d bytes)\n",
|
665 |
hw->samples << hw->info.shift); |
666 |
alsa_anal_close (&handle); |
667 |
return -1; |
668 |
} |
669 |
|
670 |
alsa->handle = handle; |
671 |
alsa->was_enabled = 0;
|
672 |
return 0; |
673 |
} |
674 |
|
675 |
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) |
676 |
{ |
677 |
int err;
|
678 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
679 |
|
680 |
switch (cmd) {
|
681 |
case VOICE_ENABLE:
|
682 |
ldebug ("enabling voice\n");
|
683 |
audio_pcm_info_clear_buf (&hw->info, alsa->pcm_buf, hw->samples); |
684 |
if (alsa->can_pause) {
|
685 |
/* Why this was_enabled madness is needed at all?? */
|
686 |
if (alsa->was_enabled) {
|
687 |
err = snd_pcm_pause (alsa->handle, 0);
|
688 |
if (err < 0) { |
689 |
alsa_logerr (err, "Failed to resume playing\n");
|
690 |
/* not fatal really */
|
691 |
} |
692 |
} |
693 |
else {
|
694 |
alsa->was_enabled = 1;
|
695 |
} |
696 |
} |
697 |
break;
|
698 |
|
699 |
case VOICE_DISABLE:
|
700 |
ldebug ("disabling voice\n");
|
701 |
if (alsa->can_pause) {
|
702 |
err = snd_pcm_pause (alsa->handle, 1);
|
703 |
if (err < 0) { |
704 |
alsa_logerr (err, "Failed to stop playing\n");
|
705 |
/* not fatal really */
|
706 |
} |
707 |
} |
708 |
break;
|
709 |
} |
710 |
return 0; |
711 |
} |
712 |
|
713 |
static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) |
714 |
{ |
715 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
716 |
struct alsa_params_req req;
|
717 |
struct alsa_params_obt obt;
|
718 |
int endianness;
|
719 |
int err;
|
720 |
audfmt_e effective_fmt; |
721 |
snd_pcm_t *handle; |
722 |
audsettings_t obt_as; |
723 |
|
724 |
req.fmt = aud_to_alsafmt (as->fmt); |
725 |
req.freq = as->freq; |
726 |
req.nchannels = as->nchannels; |
727 |
req.period_size = conf.period_size_in; |
728 |
req.buffer_size = conf.buffer_size_in; |
729 |
|
730 |
if (alsa_open (1, &req, &obt, &handle)) { |
731 |
return -1; |
732 |
} |
733 |
|
734 |
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); |
735 |
if (err) {
|
736 |
alsa_anal_close (&handle); |
737 |
return -1; |
738 |
} |
739 |
|
740 |
obt_as.freq = obt.freq; |
741 |
obt_as.nchannels = obt.nchannels; |
742 |
obt_as.fmt = effective_fmt; |
743 |
|
744 |
audio_pcm_init_info ( |
745 |
&hw->info, |
746 |
&obt_as, |
747 |
audio_need_to_swap_endian (endianness) |
748 |
); |
749 |
alsa->can_pause = obt.can_pause; |
750 |
hw->samples = obt.samples; |
751 |
|
752 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
753 |
if (!alsa->pcm_buf) {
|
754 |
dolog ("Could not allocate ADC buffer (%d bytes)\n",
|
755 |
hw->samples << hw->info.shift); |
756 |
alsa_anal_close (&handle); |
757 |
return -1; |
758 |
} |
759 |
|
760 |
alsa->handle = handle; |
761 |
return 0; |
762 |
} |
763 |
|
764 |
static void alsa_fini_in (HWVoiceIn *hw) |
765 |
{ |
766 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
767 |
|
768 |
alsa_anal_close (&alsa->handle); |
769 |
|
770 |
if (alsa->pcm_buf) {
|
771 |
qemu_free (alsa->pcm_buf); |
772 |
alsa->pcm_buf = NULL;
|
773 |
} |
774 |
} |
775 |
|
776 |
static int alsa_run_in (HWVoiceIn *hw) |
777 |
{ |
778 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
779 |
int hwshift = hw->info.shift;
|
780 |
int i;
|
781 |
int live = audio_pcm_hw_get_live_in (hw);
|
782 |
int dead = hw->samples - live;
|
783 |
struct {
|
784 |
int add;
|
785 |
int len;
|
786 |
} bufs[2] = {
|
787 |
{ hw->wpos, 0 },
|
788 |
{ 0, 0 } |
789 |
}; |
790 |
|
791 |
snd_pcm_uframes_t read_samples = 0;
|
792 |
|
793 |
if (!dead) {
|
794 |
return 0; |
795 |
} |
796 |
|
797 |
if (hw->wpos + dead > hw->samples) {
|
798 |
bufs[0].len = (hw->samples - hw->wpos);
|
799 |
bufs[1].len = (dead - (hw->samples - hw->wpos));
|
800 |
} |
801 |
else {
|
802 |
bufs[0].len = dead;
|
803 |
} |
804 |
|
805 |
|
806 |
for (i = 0; i < 2; ++i) { |
807 |
void *src;
|
808 |
st_sample_t *dst; |
809 |
snd_pcm_sframes_t nread; |
810 |
snd_pcm_uframes_t len; |
811 |
|
812 |
len = bufs[i].len; |
813 |
|
814 |
src = advance (alsa->pcm_buf, bufs[i].add << hwshift); |
815 |
dst = hw->conv_buf + bufs[i].add; |
816 |
|
817 |
while (len) {
|
818 |
nread = snd_pcm_readi (alsa->handle, src, len); |
819 |
|
820 |
if (nread < 0) { |
821 |
switch (nread) {
|
822 |
case -EPIPE:
|
823 |
if (!alsa_recover (alsa->handle)) {
|
824 |
continue;
|
825 |
} |
826 |
dolog ( |
827 |
"Failed to read %ld frames from %p, "
|
828 |
"handle %p not prepared\n",
|
829 |
len, |
830 |
src, |
831 |
alsa->handle |
832 |
); |
833 |
goto exit;
|
834 |
|
835 |
case -EAGAIN:
|
836 |
continue;
|
837 |
|
838 |
default:
|
839 |
alsa_logerr ( |
840 |
nread, |
841 |
"Failed to read %ld frames from %p\n",
|
842 |
len, |
843 |
src |
844 |
); |
845 |
goto exit;
|
846 |
} |
847 |
} |
848 |
|
849 |
hw->conv (dst, src, nread, &nominal_volume); |
850 |
|
851 |
src = advance (src, nread << hwshift); |
852 |
dst += nread; |
853 |
|
854 |
read_samples += nread; |
855 |
len -= nread; |
856 |
} |
857 |
} |
858 |
|
859 |
exit:
|
860 |
hw->wpos = (hw->wpos + read_samples) % hw->samples; |
861 |
return read_samples;
|
862 |
} |
863 |
|
864 |
static int alsa_read (SWVoiceIn *sw, void *buf, int size) |
865 |
{ |
866 |
return audio_pcm_sw_read (sw, buf, size);
|
867 |
} |
868 |
|
869 |
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) |
870 |
{ |
871 |
(void) hw;
|
872 |
(void) cmd;
|
873 |
return 0; |
874 |
} |
875 |
|
876 |
static void *alsa_audio_init (void) |
877 |
{ |
878 |
return &conf;
|
879 |
} |
880 |
|
881 |
static void alsa_audio_fini (void *opaque) |
882 |
{ |
883 |
(void) opaque;
|
884 |
} |
885 |
|
886 |
static struct audio_option alsa_options[] = { |
887 |
{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
|
888 |
"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
889 |
{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
|
890 |
"DAC period size", &conf.period_size_out_overriden, 0}, |
891 |
{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
|
892 |
"DAC buffer size", &conf.buffer_size_out_overriden, 0}, |
893 |
|
894 |
{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
|
895 |
"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
896 |
{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
|
897 |
"ADC period size", &conf.period_size_in_overriden, 0}, |
898 |
{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
|
899 |
"ADC buffer size", &conf.buffer_size_in_overriden, 0}, |
900 |
|
901 |
{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
|
902 |
"(undocumented)", NULL, 0}, |
903 |
|
904 |
{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
|
905 |
"DAC device name (for instance dmix)", NULL, 0}, |
906 |
|
907 |
{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
|
908 |
"ADC device name", NULL, 0}, |
909 |
{NULL, 0, NULL, NULL, NULL, 0} |
910 |
}; |
911 |
|
912 |
static struct audio_pcm_ops alsa_pcm_ops = { |
913 |
alsa_init_out, |
914 |
alsa_fini_out, |
915 |
alsa_run_out, |
916 |
alsa_write, |
917 |
alsa_ctl_out, |
918 |
|
919 |
alsa_init_in, |
920 |
alsa_fini_in, |
921 |
alsa_run_in, |
922 |
alsa_read, |
923 |
alsa_ctl_in |
924 |
}; |
925 |
|
926 |
struct audio_driver alsa_audio_driver = {
|
927 |
INIT_FIELD (name = ) "alsa",
|
928 |
INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
|
929 |
INIT_FIELD (options = ) alsa_options, |
930 |
INIT_FIELD (init = ) alsa_audio_init, |
931 |
INIT_FIELD (fini = ) alsa_audio_fini, |
932 |
INIT_FIELD (pcm_ops = ) &alsa_pcm_ops, |
933 |
INIT_FIELD (can_be_default = ) 1,
|
934 |
INIT_FIELD (max_voices_out = ) INT_MAX, |
935 |
INIT_FIELD (max_voices_in = ) INT_MAX, |
936 |
INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
|
937 |
INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
|
938 |
}; |