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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 "qemu-common.h" |
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#include "audio.h" |
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|
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#define AUDIO_CAP "alsa" |
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#include "audio_int.h" |
30 |
|
31 |
typedef struct ALSAVoiceOut { |
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HWVoiceOut hw; |
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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; |
42 |
|
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static struct { |
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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; |
48 |
unsigned int buffer_size_in; |
49 |
unsigned int period_size_in; |
50 |
unsigned int buffer_size_out; |
51 |
unsigned int period_size_out; |
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unsigned int threshold; |
53 |
|
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int buffer_size_in_overridden;
|
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int period_size_in_overridden;
|
56 |
|
57 |
int buffer_size_out_overridden;
|
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int period_size_out_overridden;
|
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int verbose;
|
60 |
} conf = { |
61 |
.pcm_name_out = "default",
|
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.pcm_name_in = "default",
|
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}; |
64 |
|
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struct alsa_params_req {
|
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int freq;
|
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snd_pcm_format_t fmt; |
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int nchannels;
|
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int size_in_usec;
|
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unsigned int buffer_size; |
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unsigned int period_size; |
72 |
}; |
73 |
|
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struct alsa_params_obt {
|
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int freq;
|
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audfmt_e fmt; |
77 |
int endianness;
|
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int nchannels;
|
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snd_pcm_uframes_t samples; |
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}; |
81 |
|
82 |
static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) |
83 |
{ |
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va_list ap; |
85 |
|
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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); |
89 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
92 |
|
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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, |
97 |
... |
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) |
99 |
{ |
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va_list ap; |
101 |
|
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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); |
105 |
AUD_vlog (AUDIO_CAP, fmt, ap); |
106 |
va_end (ap); |
107 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
110 |
|
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static void alsa_anal_close (snd_pcm_t **handlep) |
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{ |
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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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} |
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*handlep = NULL;
|
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} |
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|
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static int alsa_write (SWVoiceOut *sw, void *buf, int len) |
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{ |
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return audio_pcm_sw_write (sw, buf, len);
|
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} |
124 |
|
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static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
|
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{ |
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switch (fmt) {
|
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case AUD_FMT_S8:
|
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return SND_PCM_FORMAT_S8;
|
130 |
|
131 |
case AUD_FMT_U8:
|
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return SND_PCM_FORMAT_U8;
|
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|
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case AUD_FMT_S16:
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return SND_PCM_FORMAT_S16_LE;
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|
137 |
case AUD_FMT_U16:
|
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return SND_PCM_FORMAT_U16_LE;
|
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|
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case AUD_FMT_S32:
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return SND_PCM_FORMAT_S32_LE;
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142 |
|
143 |
case AUD_FMT_U32:
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return SND_PCM_FORMAT_U32_LE;
|
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|
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default:
|
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dolog ("Internal logic error: Bad audio format %d\n", fmt);
|
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#ifdef DEBUG_AUDIO
|
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abort (); |
150 |
#endif
|
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return SND_PCM_FORMAT_U8;
|
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} |
153 |
} |
154 |
|
155 |
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, |
156 |
int *endianness)
|
157 |
{ |
158 |
switch (alsafmt) {
|
159 |
case SND_PCM_FORMAT_S8:
|
160 |
*endianness = 0;
|
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*fmt = AUD_FMT_S8; |
162 |
break;
|
163 |
|
164 |
case SND_PCM_FORMAT_U8:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U8; |
167 |
break;
|
168 |
|
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case SND_PCM_FORMAT_S16_LE:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_S16; |
172 |
break;
|
173 |
|
174 |
case SND_PCM_FORMAT_U16_LE:
|
175 |
*endianness = 0;
|
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*fmt = AUD_FMT_U16; |
177 |
break;
|
178 |
|
179 |
case SND_PCM_FORMAT_S16_BE:
|
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*endianness = 1;
|
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*fmt = AUD_FMT_S16; |
182 |
break;
|
183 |
|
184 |
case SND_PCM_FORMAT_U16_BE:
|
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*endianness = 1;
|
186 |
*fmt = AUD_FMT_U16; |
187 |
break;
|
188 |
|
189 |
case SND_PCM_FORMAT_S32_LE:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_S32; |
192 |
break;
|
193 |
|
194 |
case SND_PCM_FORMAT_U32_LE:
|
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*endianness = 0;
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*fmt = AUD_FMT_U32; |
197 |
break;
|
198 |
|
199 |
case SND_PCM_FORMAT_S32_BE:
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*endianness = 1;
|
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*fmt = AUD_FMT_S32; |
202 |
break;
|
203 |
|
204 |
case SND_PCM_FORMAT_U32_BE:
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*endianness = 1;
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*fmt = AUD_FMT_U32; |
207 |
break;
|
208 |
|
209 |
default:
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dolog ("Unrecognized audio format %d\n", alsafmt);
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return -1; |
212 |
} |
213 |
|
214 |
return 0; |
215 |
} |
216 |
|
217 |
static void alsa_dump_info (struct alsa_params_req *req, |
218 |
struct alsa_params_obt *obt)
|
219 |
{ |
220 |
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); |
224 |
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
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dolog ("============================================\n");
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226 |
dolog ("requested: buffer size %d period size %d\n",
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227 |
req->buffer_size, req->period_size); |
228 |
dolog ("obtained: samples %ld\n", obt->samples);
|
229 |
} |
230 |
|
231 |
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) |
232 |
{ |
233 |
int err;
|
234 |
snd_pcm_sw_params_t *sw_params; |
235 |
|
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snd_pcm_sw_params_alloca (&sw_params); |
237 |
|
238 |
err = snd_pcm_sw_params_current (handle, sw_params); |
239 |
if (err < 0) { |
240 |
dolog ("Could not fully initialize DAC\n");
|
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alsa_logerr (err, "Failed to get current software parameters\n");
|
242 |
return;
|
243 |
} |
244 |
|
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err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); |
246 |
if (err < 0) { |
247 |
dolog ("Could not fully initialize DAC\n");
|
248 |
alsa_logerr (err, "Failed to set software threshold to %ld\n",
|
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threshold); |
250 |
return;
|
251 |
} |
252 |
|
253 |
err = snd_pcm_sw_params (handle, sw_params); |
254 |
if (err < 0) { |
255 |
dolog ("Could not fully initialize DAC\n");
|
256 |
alsa_logerr (err, "Failed to set software parameters\n");
|
257 |
return;
|
258 |
} |
259 |
} |
260 |
|
261 |
static int alsa_open (int in, struct alsa_params_req *req, |
262 |
struct alsa_params_obt *obt, snd_pcm_t **handlep)
|
263 |
{ |
264 |
snd_pcm_t *handle; |
265 |
snd_pcm_hw_params_t *hw_params; |
266 |
int err;
|
267 |
int size_in_usec;
|
268 |
unsigned int freq, nchannels; |
269 |
const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; |
270 |
snd_pcm_uframes_t obt_buffer_size; |
271 |
const char *typ = in ? "ADC" : "DAC"; |
272 |
snd_pcm_format_t obtfmt; |
273 |
|
274 |
freq = req->freq; |
275 |
nchannels = req->nchannels; |
276 |
size_in_usec = req->size_in_usec; |
277 |
|
278 |
snd_pcm_hw_params_alloca (&hw_params); |
279 |
|
280 |
err = snd_pcm_open ( |
281 |
&handle, |
282 |
pcm_name, |
283 |
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, |
284 |
SND_PCM_NONBLOCK |
285 |
); |
286 |
if (err < 0) { |
287 |
alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
|
288 |
return -1; |
289 |
} |
290 |
|
291 |
err = snd_pcm_hw_params_any (handle, hw_params); |
292 |
if (err < 0) { |
293 |
alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
|
294 |
goto err;
|
295 |
} |
296 |
|
297 |
err = snd_pcm_hw_params_set_access ( |
298 |
handle, |
299 |
hw_params, |
300 |
SND_PCM_ACCESS_RW_INTERLEAVED |
301 |
); |
302 |
if (err < 0) { |
303 |
alsa_logerr2 (err, typ, "Failed to set access type\n");
|
304 |
goto err;
|
305 |
} |
306 |
|
307 |
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); |
308 |
if (err < 0 && conf.verbose) { |
309 |
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
|
310 |
} |
311 |
|
312 |
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
|
313 |
if (err < 0) { |
314 |
alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
|
315 |
goto err;
|
316 |
} |
317 |
|
318 |
err = snd_pcm_hw_params_set_channels_near ( |
319 |
handle, |
320 |
hw_params, |
321 |
&nchannels |
322 |
); |
323 |
if (err < 0) { |
324 |
alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
|
325 |
req->nchannels); |
326 |
goto err;
|
327 |
} |
328 |
|
329 |
if (nchannels != 1 && nchannels != 2) { |
330 |
alsa_logerr2 (err, typ, |
331 |
"Can not handle obtained number of channels %d\n",
|
332 |
nchannels); |
333 |
goto err;
|
334 |
} |
335 |
|
336 |
if (req->buffer_size) {
|
337 |
unsigned long obt; |
338 |
|
339 |
if (size_in_usec) {
|
340 |
int dir = 0; |
341 |
unsigned int btime = req->buffer_size; |
342 |
|
343 |
err = snd_pcm_hw_params_set_buffer_time_near ( |
344 |
handle, |
345 |
hw_params, |
346 |
&btime, |
347 |
&dir |
348 |
); |
349 |
obt = btime; |
350 |
} |
351 |
else {
|
352 |
snd_pcm_uframes_t bsize = req->buffer_size; |
353 |
|
354 |
err = snd_pcm_hw_params_set_buffer_size_near ( |
355 |
handle, |
356 |
hw_params, |
357 |
&bsize |
358 |
); |
359 |
obt = bsize; |
360 |
} |
361 |
if (err < 0) { |
362 |
alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
|
363 |
size_in_usec ? "time" : "size", req->buffer_size); |
364 |
goto err;
|
365 |
} |
366 |
|
367 |
if (obt - req->buffer_size)
|
368 |
dolog ("Requested buffer %s %u was rejected, using %lu\n",
|
369 |
size_in_usec ? "time" : "size", req->buffer_size, obt); |
370 |
} |
371 |
|
372 |
if (req->period_size) {
|
373 |
unsigned long obt; |
374 |
|
375 |
if (size_in_usec) {
|
376 |
int dir = 0; |
377 |
unsigned int ptime = req->period_size; |
378 |
|
379 |
err = snd_pcm_hw_params_set_period_time_near ( |
380 |
handle, |
381 |
hw_params, |
382 |
&ptime, |
383 |
&dir |
384 |
); |
385 |
obt = ptime; |
386 |
} |
387 |
else {
|
388 |
snd_pcm_uframes_t psize = req->period_size; |
389 |
|
390 |
err = snd_pcm_hw_params_set_buffer_size_near ( |
391 |
handle, |
392 |
hw_params, |
393 |
&psize |
394 |
); |
395 |
obt = psize; |
396 |
} |
397 |
|
398 |
if (err < 0) { |
399 |
alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
|
400 |
size_in_usec ? "time" : "size", req->period_size); |
401 |
goto err;
|
402 |
} |
403 |
|
404 |
if (obt - req->period_size)
|
405 |
dolog ("Requested period %s %u was rejected, using %lu\n",
|
406 |
size_in_usec ? "time" : "size", req->period_size, obt); |
407 |
} |
408 |
|
409 |
err = snd_pcm_hw_params (handle, hw_params); |
410 |
if (err < 0) { |
411 |
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
|
412 |
goto err;
|
413 |
} |
414 |
|
415 |
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); |
416 |
if (err < 0) { |
417 |
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
|
418 |
goto err;
|
419 |
} |
420 |
|
421 |
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); |
422 |
if (err < 0) { |
423 |
alsa_logerr2 (err, typ, "Failed to get format\n");
|
424 |
goto err;
|
425 |
} |
426 |
|
427 |
if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
|
428 |
dolog ("Invalid format was returned %d\n", obtfmt);
|
429 |
goto err;
|
430 |
} |
431 |
|
432 |
err = snd_pcm_prepare (handle); |
433 |
if (err < 0) { |
434 |
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
|
435 |
goto err;
|
436 |
} |
437 |
|
438 |
if (!in && conf.threshold) {
|
439 |
snd_pcm_uframes_t threshold; |
440 |
int bytes_per_sec;
|
441 |
|
442 |
bytes_per_sec = freq << (nchannels == 2);
|
443 |
|
444 |
switch (obt->fmt) {
|
445 |
case AUD_FMT_S8:
|
446 |
case AUD_FMT_U8:
|
447 |
break;
|
448 |
|
449 |
case AUD_FMT_S16:
|
450 |
case AUD_FMT_U16:
|
451 |
bytes_per_sec <<= 1;
|
452 |
break;
|
453 |
|
454 |
case AUD_FMT_S32:
|
455 |
case AUD_FMT_U32:
|
456 |
bytes_per_sec <<= 2;
|
457 |
break;
|
458 |
} |
459 |
|
460 |
threshold = (conf.threshold * bytes_per_sec) / 1000;
|
461 |
alsa_set_threshold (handle, threshold); |
462 |
} |
463 |
|
464 |
obt->nchannels = nchannels; |
465 |
obt->freq = freq; |
466 |
obt->samples = obt_buffer_size; |
467 |
|
468 |
*handlep = handle; |
469 |
|
470 |
if (conf.verbose &&
|
471 |
(obt->fmt != req->fmt || |
472 |
obt->nchannels != req->nchannels || |
473 |
obt->freq != req->freq)) { |
474 |
dolog ("Audio paramters for %s\n", typ);
|
475 |
alsa_dump_info (req, obt); |
476 |
} |
477 |
|
478 |
#ifdef DEBUG
|
479 |
alsa_dump_info (req, obt); |
480 |
#endif
|
481 |
return 0; |
482 |
|
483 |
err:
|
484 |
alsa_anal_close (&handle); |
485 |
return -1; |
486 |
} |
487 |
|
488 |
static int alsa_recover (snd_pcm_t *handle) |
489 |
{ |
490 |
int err = snd_pcm_prepare (handle);
|
491 |
if (err < 0) { |
492 |
alsa_logerr (err, "Failed to prepare handle %p\n", handle);
|
493 |
return -1; |
494 |
} |
495 |
return 0; |
496 |
} |
497 |
|
498 |
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
|
499 |
{ |
500 |
snd_pcm_sframes_t avail; |
501 |
|
502 |
avail = snd_pcm_avail_update (handle); |
503 |
if (avail < 0) { |
504 |
if (avail == -EPIPE) {
|
505 |
if (!alsa_recover (handle)) {
|
506 |
avail = snd_pcm_avail_update (handle); |
507 |
} |
508 |
} |
509 |
|
510 |
if (avail < 0) { |
511 |
alsa_logerr (avail, |
512 |
"Could not obtain number of available frames\n");
|
513 |
return -1; |
514 |
} |
515 |
} |
516 |
|
517 |
return avail;
|
518 |
} |
519 |
|
520 |
static int alsa_run_out (HWVoiceOut *hw) |
521 |
{ |
522 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
523 |
int rpos, live, decr;
|
524 |
int samples;
|
525 |
uint8_t *dst; |
526 |
st_sample_t *src; |
527 |
snd_pcm_sframes_t avail; |
528 |
|
529 |
live = audio_pcm_hw_get_live_out (hw); |
530 |
if (!live) {
|
531 |
return 0; |
532 |
} |
533 |
|
534 |
avail = alsa_get_avail (alsa->handle); |
535 |
if (avail < 0) { |
536 |
dolog ("Could not get number of available playback frames\n");
|
537 |
return 0; |
538 |
} |
539 |
|
540 |
decr = audio_MIN (live, avail); |
541 |
samples = decr; |
542 |
rpos = hw->rpos; |
543 |
while (samples) {
|
544 |
int left_till_end_samples = hw->samples - rpos;
|
545 |
int len = audio_MIN (samples, left_till_end_samples);
|
546 |
snd_pcm_sframes_t written; |
547 |
|
548 |
src = hw->mix_buf + rpos; |
549 |
dst = advance (alsa->pcm_buf, rpos << hw->info.shift); |
550 |
|
551 |
hw->clip (dst, src, len); |
552 |
|
553 |
while (len) {
|
554 |
written = snd_pcm_writei (alsa->handle, dst, len); |
555 |
|
556 |
if (written <= 0) { |
557 |
switch (written) {
|
558 |
case 0: |
559 |
if (conf.verbose) {
|
560 |
dolog ("Failed to write %d frames (wrote zero)\n", len);
|
561 |
} |
562 |
goto exit;
|
563 |
|
564 |
case -EPIPE:
|
565 |
if (alsa_recover (alsa->handle)) {
|
566 |
alsa_logerr (written, "Failed to write %d frames\n",
|
567 |
len); |
568 |
goto exit;
|
569 |
} |
570 |
if (conf.verbose) {
|
571 |
dolog ("Recovering from playback xrun\n");
|
572 |
} |
573 |
continue;
|
574 |
|
575 |
case -EAGAIN:
|
576 |
goto exit;
|
577 |
|
578 |
default:
|
579 |
alsa_logerr (written, "Failed to write %d frames to %p\n",
|
580 |
len, dst); |
581 |
goto exit;
|
582 |
} |
583 |
} |
584 |
|
585 |
rpos = (rpos + written) % hw->samples; |
586 |
samples -= written; |
587 |
len -= written; |
588 |
dst = advance (dst, written << hw->info.shift); |
589 |
src += written; |
590 |
} |
591 |
} |
592 |
|
593 |
exit:
|
594 |
hw->rpos = rpos; |
595 |
return decr;
|
596 |
} |
597 |
|
598 |
static void alsa_fini_out (HWVoiceOut *hw) |
599 |
{ |
600 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
601 |
|
602 |
ldebug ("alsa_fini\n");
|
603 |
alsa_anal_close (&alsa->handle); |
604 |
|
605 |
if (alsa->pcm_buf) {
|
606 |
qemu_free (alsa->pcm_buf); |
607 |
alsa->pcm_buf = NULL;
|
608 |
} |
609 |
} |
610 |
|
611 |
static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) |
612 |
{ |
613 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
614 |
struct alsa_params_req req;
|
615 |
struct alsa_params_obt obt;
|
616 |
snd_pcm_t *handle; |
617 |
audsettings_t obt_as; |
618 |
|
619 |
req.fmt = aud_to_alsafmt (as->fmt); |
620 |
req.freq = as->freq; |
621 |
req.nchannels = as->nchannels; |
622 |
req.period_size = conf.period_size_out; |
623 |
req.buffer_size = conf.buffer_size_out; |
624 |
req.size_in_usec = conf.size_in_usec_out; |
625 |
|
626 |
if (alsa_open (0, &req, &obt, &handle)) { |
627 |
return -1; |
628 |
} |
629 |
|
630 |
obt_as.freq = obt.freq; |
631 |
obt_as.nchannels = obt.nchannels; |
632 |
obt_as.fmt = obt.fmt; |
633 |
obt_as.endianness = obt.endianness; |
634 |
|
635 |
audio_pcm_init_info (&hw->info, &obt_as); |
636 |
hw->samples = obt.samples; |
637 |
|
638 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
|
639 |
if (!alsa->pcm_buf) {
|
640 |
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
|
641 |
hw->samples, 1 << hw->info.shift);
|
642 |
alsa_anal_close (&handle); |
643 |
return -1; |
644 |
} |
645 |
|
646 |
alsa->handle = handle; |
647 |
return 0; |
648 |
} |
649 |
|
650 |
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause) |
651 |
{ |
652 |
int err;
|
653 |
|
654 |
if (pause) {
|
655 |
err = snd_pcm_drop (handle); |
656 |
if (err < 0) { |
657 |
alsa_logerr (err, "Could not stop %s\n", typ);
|
658 |
return -1; |
659 |
} |
660 |
} |
661 |
else {
|
662 |
err = snd_pcm_prepare (handle); |
663 |
if (err < 0) { |
664 |
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
|
665 |
return -1; |
666 |
} |
667 |
} |
668 |
|
669 |
return 0; |
670 |
} |
671 |
|
672 |
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) |
673 |
{ |
674 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
675 |
|
676 |
switch (cmd) {
|
677 |
case VOICE_ENABLE:
|
678 |
ldebug ("enabling voice\n");
|
679 |
return alsa_voice_ctl (alsa->handle, "playback", 0); |
680 |
|
681 |
case VOICE_DISABLE:
|
682 |
ldebug ("disabling voice\n");
|
683 |
return alsa_voice_ctl (alsa->handle, "playback", 1); |
684 |
} |
685 |
|
686 |
return -1; |
687 |
} |
688 |
|
689 |
static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) |
690 |
{ |
691 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
692 |
struct alsa_params_req req;
|
693 |
struct alsa_params_obt obt;
|
694 |
snd_pcm_t *handle; |
695 |
audsettings_t obt_as; |
696 |
|
697 |
req.fmt = aud_to_alsafmt (as->fmt); |
698 |
req.freq = as->freq; |
699 |
req.nchannels = as->nchannels; |
700 |
req.period_size = conf.period_size_in; |
701 |
req.buffer_size = conf.buffer_size_in; |
702 |
req.size_in_usec = conf.size_in_usec_in; |
703 |
|
704 |
if (alsa_open (1, &req, &obt, &handle)) { |
705 |
return -1; |
706 |
} |
707 |
|
708 |
obt_as.freq = obt.freq; |
709 |
obt_as.nchannels = obt.nchannels; |
710 |
obt_as.fmt = obt.fmt; |
711 |
obt_as.endianness = obt.endianness; |
712 |
|
713 |
audio_pcm_init_info (&hw->info, &obt_as); |
714 |
hw->samples = obt.samples; |
715 |
|
716 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
717 |
if (!alsa->pcm_buf) {
|
718 |
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
|
719 |
hw->samples, 1 << hw->info.shift);
|
720 |
alsa_anal_close (&handle); |
721 |
return -1; |
722 |
} |
723 |
|
724 |
alsa->handle = handle; |
725 |
return 0; |
726 |
} |
727 |
|
728 |
static void alsa_fini_in (HWVoiceIn *hw) |
729 |
{ |
730 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
731 |
|
732 |
alsa_anal_close (&alsa->handle); |
733 |
|
734 |
if (alsa->pcm_buf) {
|
735 |
qemu_free (alsa->pcm_buf); |
736 |
alsa->pcm_buf = NULL;
|
737 |
} |
738 |
} |
739 |
|
740 |
static int alsa_run_in (HWVoiceIn *hw) |
741 |
{ |
742 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
743 |
int hwshift = hw->info.shift;
|
744 |
int i;
|
745 |
int live = audio_pcm_hw_get_live_in (hw);
|
746 |
int dead = hw->samples - live;
|
747 |
int decr;
|
748 |
struct {
|
749 |
int add;
|
750 |
int len;
|
751 |
} bufs[2] = {
|
752 |
{ hw->wpos, 0 },
|
753 |
{ 0, 0 } |
754 |
}; |
755 |
snd_pcm_sframes_t avail; |
756 |
snd_pcm_uframes_t read_samples = 0;
|
757 |
|
758 |
if (!dead) {
|
759 |
return 0; |
760 |
} |
761 |
|
762 |
avail = alsa_get_avail (alsa->handle); |
763 |
if (avail < 0) { |
764 |
dolog ("Could not get number of captured frames\n");
|
765 |
return 0; |
766 |
} |
767 |
|
768 |
if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
|
769 |
avail = hw->samples; |
770 |
} |
771 |
|
772 |
decr = audio_MIN (dead, avail); |
773 |
if (!decr) {
|
774 |
return 0; |
775 |
} |
776 |
|
777 |
if (hw->wpos + decr > hw->samples) {
|
778 |
bufs[0].len = (hw->samples - hw->wpos);
|
779 |
bufs[1].len = (decr - (hw->samples - hw->wpos));
|
780 |
} |
781 |
else {
|
782 |
bufs[0].len = decr;
|
783 |
} |
784 |
|
785 |
for (i = 0; i < 2; ++i) { |
786 |
void *src;
|
787 |
st_sample_t *dst; |
788 |
snd_pcm_sframes_t nread; |
789 |
snd_pcm_uframes_t len; |
790 |
|
791 |
len = bufs[i].len; |
792 |
|
793 |
src = advance (alsa->pcm_buf, bufs[i].add << hwshift); |
794 |
dst = hw->conv_buf + bufs[i].add; |
795 |
|
796 |
while (len) {
|
797 |
nread = snd_pcm_readi (alsa->handle, src, len); |
798 |
|
799 |
if (nread <= 0) { |
800 |
switch (nread) {
|
801 |
case 0: |
802 |
if (conf.verbose) {
|
803 |
dolog ("Failed to read %ld frames (read zero)\n", len);
|
804 |
} |
805 |
goto exit;
|
806 |
|
807 |
case -EPIPE:
|
808 |
if (alsa_recover (alsa->handle)) {
|
809 |
alsa_logerr (nread, "Failed to read %ld frames\n", len);
|
810 |
goto exit;
|
811 |
} |
812 |
if (conf.verbose) {
|
813 |
dolog ("Recovering from capture xrun\n");
|
814 |
} |
815 |
continue;
|
816 |
|
817 |
case -EAGAIN:
|
818 |
goto exit;
|
819 |
|
820 |
default:
|
821 |
alsa_logerr ( |
822 |
nread, |
823 |
"Failed to read %ld frames from %p\n",
|
824 |
len, |
825 |
src |
826 |
); |
827 |
goto exit;
|
828 |
} |
829 |
} |
830 |
|
831 |
hw->conv (dst, src, nread, &nominal_volume); |
832 |
|
833 |
src = advance (src, nread << hwshift); |
834 |
dst += nread; |
835 |
|
836 |
read_samples += nread; |
837 |
len -= nread; |
838 |
} |
839 |
} |
840 |
|
841 |
exit:
|
842 |
hw->wpos = (hw->wpos + read_samples) % hw->samples; |
843 |
return read_samples;
|
844 |
} |
845 |
|
846 |
static int alsa_read (SWVoiceIn *sw, void *buf, int size) |
847 |
{ |
848 |
return audio_pcm_sw_read (sw, buf, size);
|
849 |
} |
850 |
|
851 |
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) |
852 |
{ |
853 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
854 |
|
855 |
switch (cmd) {
|
856 |
case VOICE_ENABLE:
|
857 |
ldebug ("enabling voice\n");
|
858 |
return alsa_voice_ctl (alsa->handle, "capture", 0); |
859 |
|
860 |
case VOICE_DISABLE:
|
861 |
ldebug ("disabling voice\n");
|
862 |
return alsa_voice_ctl (alsa->handle, "capture", 1); |
863 |
} |
864 |
|
865 |
return -1; |
866 |
} |
867 |
|
868 |
static void *alsa_audio_init (void) |
869 |
{ |
870 |
return &conf;
|
871 |
} |
872 |
|
873 |
static void alsa_audio_fini (void *opaque) |
874 |
{ |
875 |
(void) opaque;
|
876 |
} |
877 |
|
878 |
static struct audio_option alsa_options[] = { |
879 |
{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
|
880 |
"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
881 |
{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
|
882 |
"DAC period size (0 to go with system default)",
|
883 |
&conf.period_size_out_overridden, 0},
|
884 |
{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
|
885 |
"DAC buffer size (0 to go with system default)",
|
886 |
&conf.buffer_size_out_overridden, 0},
|
887 |
|
888 |
{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
|
889 |
"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
890 |
{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
|
891 |
"ADC period size (0 to go with system default)",
|
892 |
&conf.period_size_in_overridden, 0},
|
893 |
{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
|
894 |
"ADC buffer size (0 to go with system default)",
|
895 |
&conf.buffer_size_in_overridden, 0},
|
896 |
|
897 |
{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
|
898 |
"(undocumented)", NULL, 0}, |
899 |
|
900 |
{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
|
901 |
"DAC device name (for instance dmix)", NULL, 0}, |
902 |
|
903 |
{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
|
904 |
"ADC device name", NULL, 0}, |
905 |
|
906 |
{"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
|
907 |
"Behave in a more verbose way", NULL, 0}, |
908 |
|
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 |
}; |