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/*
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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" |
| 27 |
|
| 28 |
#define AUDIO_CAP "alsa" |
| 29 |
#include "audio_int.h" |
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|
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typedef struct ALSAVoiceOut { |
| 32 |
HWVoiceOut hw; |
| 33 |
void *pcm_buf;
|
| 34 |
snd_pcm_t *handle; |
| 35 |
} ALSAVoiceOut; |
| 36 |
|
| 37 |
typedef struct ALSAVoiceIn { |
| 38 |
HWVoiceIn hw; |
| 39 |
snd_pcm_t *handle; |
| 40 |
void *pcm_buf;
|
| 41 |
} ALSAVoiceIn; |
| 42 |
|
| 43 |
static struct { |
| 44 |
int size_in_usec_in;
|
| 45 |
int size_in_usec_out;
|
| 46 |
const char *pcm_name_in; |
| 47 |
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; |
| 52 |
unsigned int threshold; |
| 53 |
|
| 54 |
int buffer_size_in_overridden;
|
| 55 |
int period_size_in_overridden;
|
| 56 |
|
| 57 |
int buffer_size_out_overridden;
|
| 58 |
int period_size_out_overridden;
|
| 59 |
int verbose;
|
| 60 |
} conf = {
|
| 61 |
.pcm_name_out = "default",
|
| 62 |
.pcm_name_in = "default",
|
| 63 |
}; |
| 64 |
|
| 65 |
struct alsa_params_req {
|
| 66 |
int freq;
|
| 67 |
snd_pcm_format_t fmt; |
| 68 |
int nchannels;
|
| 69 |
int size_in_usec;
|
| 70 |
unsigned int buffer_size; |
| 71 |
unsigned int period_size; |
| 72 |
}; |
| 73 |
|
| 74 |
struct alsa_params_obt {
|
| 75 |
int freq;
|
| 76 |
audfmt_e fmt; |
| 77 |
int endianness;
|
| 78 |
int nchannels;
|
| 79 |
snd_pcm_uframes_t samples; |
| 80 |
}; |
| 81 |
|
| 82 |
static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) |
| 83 |
{
|
| 84 |
va_list ap; |
| 85 |
|
| 86 |
va_start (ap, fmt); |
| 87 |
AUD_vlog (AUDIO_CAP, fmt, ap); |
| 88 |
va_end (ap); |
| 89 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
| 92 |
|
| 93 |
static void GCC_FMT_ATTR (3, 4) alsa_logerr2 ( |
| 94 |
int err,
|
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const char *typ, |
| 96 |
const char *fmt, |
| 97 |
... |
| 98 |
) |
| 99 |
{
|
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va_list ap; |
| 101 |
|
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AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
|
| 103 |
|
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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) |
| 112 |
{
|
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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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} |
| 119 |
|
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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) {
|
| 128 |
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;
|
| 133 |
|
| 134 |
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;
|
| 139 |
|
| 140 |
case AUD_FMT_S32:
|
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return SND_PCM_FORMAT_S32_LE;
|
| 142 |
|
| 143 |
case AUD_FMT_U32:
|
| 144 |
return SND_PCM_FORMAT_U32_LE;
|
| 145 |
|
| 146 |
default:
|
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dolog ("Internal logic error: Bad audio format %d\n", fmt);
|
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#ifdef DEBUG_AUDIO
|
| 149 |
abort (); |
| 150 |
#endif
|
| 151 |
return SND_PCM_FORMAT_U8;
|
| 152 |
} |
| 153 |
} |
| 154 |
|
| 155 |
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, |
| 156 |
int *endianness)
|
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{
|
| 158 |
switch (alsafmt) {
|
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case SND_PCM_FORMAT_S8:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_S8; |
| 162 |
break;
|
| 163 |
|
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case SND_PCM_FORMAT_U8:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U8; |
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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; |
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break;
|
| 173 |
|
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case SND_PCM_FORMAT_U16_LE:
|
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*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;
|
| 181 |
*fmt = AUD_FMT_S16; |
| 182 |
break;
|
| 183 |
|
| 184 |
case SND_PCM_FORMAT_U16_BE:
|
| 185 |
*endianness = 1;
|
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*fmt = AUD_FMT_U16; |
| 187 |
break;
|
| 188 |
|
| 189 |
case SND_PCM_FORMAT_S32_LE:
|
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*endianness = 0;
|
| 191 |
*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; |
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break;
|
| 208 |
|
| 209 |
default:
|
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dolog ("Unrecognized audio format %d\n", alsafmt);
|
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return -1; |
| 212 |
} |
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|
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return 0; |
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} |
| 216 |
|
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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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{
|
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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); |
| 224 |
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
|
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dolog ("============================================\n");
|
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dolog ("requested: buffer size %d period size %d\n",
|
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req->buffer_size, req->period_size); |
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dolog ("obtained: samples %ld\n", obt->samples);
|
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} |
| 230 |
|
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static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) |
| 232 |
{
|
| 233 |
int err;
|
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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");
|
| 241 |
alsa_logerr (err, "Failed to get current software parameters\n");
|
| 242 |
return;
|
| 243 |
} |
| 244 |
|
| 245 |
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",
|
| 249 |
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 |
if (size_in_usec) {
|
| 338 |
int dir = 0; |
| 339 |
unsigned int btime = req->buffer_size; |
| 340 |
|
| 341 |
err = snd_pcm_hw_params_set_buffer_time_near ( |
| 342 |
handle, |
| 343 |
hw_params, |
| 344 |
&btime, |
| 345 |
&dir |
| 346 |
); |
| 347 |
} |
| 348 |
else {
|
| 349 |
snd_pcm_uframes_t bsize = req->buffer_size; |
| 350 |
|
| 351 |
err = snd_pcm_hw_params_set_buffer_size_near ( |
| 352 |
handle, |
| 353 |
hw_params, |
| 354 |
&bsize |
| 355 |
); |
| 356 |
} |
| 357 |
if (err < 0) { |
| 358 |
alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
|
| 359 |
size_in_usec ? "time" : "size", req->buffer_size); |
| 360 |
goto err;
|
| 361 |
} |
| 362 |
} |
| 363 |
|
| 364 |
if (req->period_size) {
|
| 365 |
if (size_in_usec) {
|
| 366 |
int dir = 0; |
| 367 |
unsigned int ptime = req->period_size; |
| 368 |
|
| 369 |
err = snd_pcm_hw_params_set_period_time_near ( |
| 370 |
handle, |
| 371 |
hw_params, |
| 372 |
&ptime, |
| 373 |
&dir |
| 374 |
); |
| 375 |
} |
| 376 |
else {
|
| 377 |
snd_pcm_uframes_t psize = req->period_size; |
| 378 |
|
| 379 |
err = snd_pcm_hw_params_set_buffer_size_near ( |
| 380 |
handle, |
| 381 |
hw_params, |
| 382 |
&psize |
| 383 |
); |
| 384 |
} |
| 385 |
|
| 386 |
if (err < 0) { |
| 387 |
alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
|
| 388 |
size_in_usec ? "time" : "size", req->period_size); |
| 389 |
goto err;
|
| 390 |
} |
| 391 |
} |
| 392 |
|
| 393 |
err = snd_pcm_hw_params (handle, hw_params); |
| 394 |
if (err < 0) { |
| 395 |
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
|
| 396 |
goto err;
|
| 397 |
} |
| 398 |
|
| 399 |
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); |
| 400 |
if (err < 0) { |
| 401 |
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
|
| 402 |
goto err;
|
| 403 |
} |
| 404 |
|
| 405 |
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); |
| 406 |
if (err < 0) { |
| 407 |
alsa_logerr2 (err, typ, "Failed to get format\n");
|
| 408 |
goto err;
|
| 409 |
} |
| 410 |
|
| 411 |
if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
|
| 412 |
dolog ("Invalid format was returned %d\n", obtfmt);
|
| 413 |
goto err;
|
| 414 |
} |
| 415 |
|
| 416 |
err = snd_pcm_prepare (handle); |
| 417 |
if (err < 0) { |
| 418 |
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
|
| 419 |
goto err;
|
| 420 |
} |
| 421 |
|
| 422 |
if (!in && conf.threshold) {
|
| 423 |
snd_pcm_uframes_t threshold; |
| 424 |
int bytes_per_sec;
|
| 425 |
|
| 426 |
bytes_per_sec = freq << (nchannels == 2);
|
| 427 |
|
| 428 |
switch (obt->fmt) {
|
| 429 |
case AUD_FMT_S8:
|
| 430 |
case AUD_FMT_U8:
|
| 431 |
break;
|
| 432 |
|
| 433 |
case AUD_FMT_S16:
|
| 434 |
case AUD_FMT_U16:
|
| 435 |
bytes_per_sec <<= 1;
|
| 436 |
break;
|
| 437 |
|
| 438 |
case AUD_FMT_S32:
|
| 439 |
case AUD_FMT_U32:
|
| 440 |
bytes_per_sec <<= 2;
|
| 441 |
break;
|
| 442 |
} |
| 443 |
|
| 444 |
threshold = (conf.threshold * bytes_per_sec) / 1000;
|
| 445 |
alsa_set_threshold (handle, threshold); |
| 446 |
} |
| 447 |
|
| 448 |
obt->nchannels = nchannels; |
| 449 |
obt->freq = freq; |
| 450 |
obt->samples = obt_buffer_size; |
| 451 |
|
| 452 |
*handlep = handle; |
| 453 |
|
| 454 |
if (conf.verbose &&
|
| 455 |
(obt->fmt != req->fmt || |
| 456 |
obt->nchannels != req->nchannels || |
| 457 |
obt->freq != req->freq)) {
|
| 458 |
dolog ("Audio paramters for %s\n", typ);
|
| 459 |
alsa_dump_info (req, obt); |
| 460 |
} |
| 461 |
|
| 462 |
#ifdef DEBUG
|
| 463 |
alsa_dump_info (req, obt); |
| 464 |
#endif
|
| 465 |
return 0; |
| 466 |
|
| 467 |
err:
|
| 468 |
alsa_anal_close (&handle); |
| 469 |
return -1; |
| 470 |
} |
| 471 |
|
| 472 |
static int alsa_recover (snd_pcm_t *handle) |
| 473 |
{
|
| 474 |
int err = snd_pcm_prepare (handle);
|
| 475 |
if (err < 0) { |
| 476 |
alsa_logerr (err, "Failed to prepare handle %p\n", handle);
|
| 477 |
return -1; |
| 478 |
} |
| 479 |
return 0; |
| 480 |
} |
| 481 |
|
| 482 |
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
|
| 483 |
{
|
| 484 |
snd_pcm_sframes_t avail; |
| 485 |
|
| 486 |
avail = snd_pcm_avail_update (handle); |
| 487 |
if (avail < 0) { |
| 488 |
if (avail == -EPIPE) {
|
| 489 |
if (!alsa_recover (handle)) {
|
| 490 |
avail = snd_pcm_avail_update (handle); |
| 491 |
} |
| 492 |
} |
| 493 |
|
| 494 |
if (avail < 0) { |
| 495 |
alsa_logerr (avail, |
| 496 |
"Could not obtain number of available frames\n");
|
| 497 |
return -1; |
| 498 |
} |
| 499 |
} |
| 500 |
|
| 501 |
return avail;
|
| 502 |
} |
| 503 |
|
| 504 |
static int alsa_run_out (HWVoiceOut *hw) |
| 505 |
{
|
| 506 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| 507 |
int rpos, live, decr;
|
| 508 |
int samples;
|
| 509 |
uint8_t *dst; |
| 510 |
st_sample_t *src; |
| 511 |
snd_pcm_sframes_t avail; |
| 512 |
|
| 513 |
live = audio_pcm_hw_get_live_out (hw); |
| 514 |
if (!live) {
|
| 515 |
return 0; |
| 516 |
} |
| 517 |
|
| 518 |
avail = alsa_get_avail (alsa->handle); |
| 519 |
if (avail < 0) { |
| 520 |
dolog ("Could not get number of available playback frames\n");
|
| 521 |
return 0; |
| 522 |
} |
| 523 |
|
| 524 |
decr = audio_MIN (live, avail); |
| 525 |
samples = decr; |
| 526 |
rpos = hw->rpos; |
| 527 |
while (samples) {
|
| 528 |
int left_till_end_samples = hw->samples - rpos;
|
| 529 |
int len = audio_MIN (samples, left_till_end_samples);
|
| 530 |
snd_pcm_sframes_t written; |
| 531 |
|
| 532 |
src = hw->mix_buf + rpos; |
| 533 |
dst = advance (alsa->pcm_buf, rpos << hw->info.shift); |
| 534 |
|
| 535 |
hw->clip (dst, src, len); |
| 536 |
|
| 537 |
while (len) {
|
| 538 |
written = snd_pcm_writei (alsa->handle, dst, len); |
| 539 |
|
| 540 |
if (written <= 0) { |
| 541 |
switch (written) {
|
| 542 |
case 0: |
| 543 |
if (conf.verbose) {
|
| 544 |
dolog ("Failed to write %d frames (wrote zero)\n", len);
|
| 545 |
} |
| 546 |
goto exit;
|
| 547 |
|
| 548 |
case -EPIPE:
|
| 549 |
if (alsa_recover (alsa->handle)) {
|
| 550 |
alsa_logerr (written, "Failed to write %d frames\n",
|
| 551 |
len); |
| 552 |
goto exit;
|
| 553 |
} |
| 554 |
if (conf.verbose) {
|
| 555 |
dolog ("Recovering from playback xrun\n");
|
| 556 |
} |
| 557 |
continue;
|
| 558 |
|
| 559 |
case -EAGAIN:
|
| 560 |
goto exit;
|
| 561 |
|
| 562 |
default:
|
| 563 |
alsa_logerr (written, "Failed to write %d frames to %p\n",
|
| 564 |
len, dst); |
| 565 |
goto exit;
|
| 566 |
} |
| 567 |
} |
| 568 |
|
| 569 |
rpos = (rpos + written) % hw->samples; |
| 570 |
samples -= written; |
| 571 |
len -= written; |
| 572 |
dst = advance (dst, written << hw->info.shift); |
| 573 |
src += written; |
| 574 |
} |
| 575 |
} |
| 576 |
|
| 577 |
exit:
|
| 578 |
hw->rpos = rpos; |
| 579 |
return decr;
|
| 580 |
} |
| 581 |
|
| 582 |
static void alsa_fini_out (HWVoiceOut *hw) |
| 583 |
{
|
| 584 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| 585 |
|
| 586 |
ldebug ("alsa_fini\n");
|
| 587 |
alsa_anal_close (&alsa->handle); |
| 588 |
|
| 589 |
if (alsa->pcm_buf) {
|
| 590 |
qemu_free (alsa->pcm_buf); |
| 591 |
alsa->pcm_buf = NULL;
|
| 592 |
} |
| 593 |
} |
| 594 |
|
| 595 |
static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) |
| 596 |
{
|
| 597 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| 598 |
struct alsa_params_req req;
|
| 599 |
struct alsa_params_obt obt;
|
| 600 |
snd_pcm_t *handle; |
| 601 |
audsettings_t obt_as; |
| 602 |
|
| 603 |
req.fmt = aud_to_alsafmt (as->fmt); |
| 604 |
req.freq = as->freq; |
| 605 |
req.nchannels = as->nchannels; |
| 606 |
req.period_size = conf.period_size_out; |
| 607 |
req.buffer_size = conf.buffer_size_out; |
| 608 |
req.size_in_usec = conf.size_in_usec_in; |
| 609 |
|
| 610 |
if (alsa_open (0, &req, &obt, &handle)) { |
| 611 |
return -1; |
| 612 |
} |
| 613 |
|
| 614 |
obt_as.freq = obt.freq; |
| 615 |
obt_as.nchannels = obt.nchannels; |
| 616 |
obt_as.fmt = obt.fmt; |
| 617 |
obt_as.endianness = obt.endianness; |
| 618 |
|
| 619 |
audio_pcm_init_info (&hw->info, &obt_as); |
| 620 |
hw->samples = obt.samples; |
| 621 |
|
| 622 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
|
| 623 |
if (!alsa->pcm_buf) {
|
| 624 |
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
|
| 625 |
hw->samples, 1 << hw->info.shift);
|
| 626 |
alsa_anal_close (&handle); |
| 627 |
return -1; |
| 628 |
} |
| 629 |
|
| 630 |
alsa->handle = handle; |
| 631 |
return 0; |
| 632 |
} |
| 633 |
|
| 634 |
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause) |
| 635 |
{
|
| 636 |
int err;
|
| 637 |
|
| 638 |
if (pause) {
|
| 639 |
err = snd_pcm_drop (handle); |
| 640 |
if (err < 0) { |
| 641 |
alsa_logerr (err, "Could not stop %s\n", typ);
|
| 642 |
return -1; |
| 643 |
} |
| 644 |
} |
| 645 |
else {
|
| 646 |
err = snd_pcm_prepare (handle); |
| 647 |
if (err < 0) { |
| 648 |
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
|
| 649 |
return -1; |
| 650 |
} |
| 651 |
} |
| 652 |
|
| 653 |
return 0; |
| 654 |
} |
| 655 |
|
| 656 |
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) |
| 657 |
{
|
| 658 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| 659 |
|
| 660 |
switch (cmd) {
|
| 661 |
case VOICE_ENABLE:
|
| 662 |
ldebug ("enabling voice\n");
|
| 663 |
return alsa_voice_ctl (alsa->handle, "playback", 0); |
| 664 |
|
| 665 |
case VOICE_DISABLE:
|
| 666 |
ldebug ("disabling voice\n");
|
| 667 |
return alsa_voice_ctl (alsa->handle, "playback", 1); |
| 668 |
} |
| 669 |
|
| 670 |
return -1; |
| 671 |
} |
| 672 |
|
| 673 |
static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) |
| 674 |
{
|
| 675 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| 676 |
struct alsa_params_req req;
|
| 677 |
struct alsa_params_obt obt;
|
| 678 |
snd_pcm_t *handle; |
| 679 |
audsettings_t obt_as; |
| 680 |
|
| 681 |
req.fmt = aud_to_alsafmt (as->fmt); |
| 682 |
req.freq = as->freq; |
| 683 |
req.nchannels = as->nchannels; |
| 684 |
req.period_size = conf.period_size_in; |
| 685 |
req.buffer_size = conf.buffer_size_in; |
| 686 |
req.size_in_usec = conf.size_in_usec_in; |
| 687 |
|
| 688 |
if (alsa_open (1, &req, &obt, &handle)) { |
| 689 |
return -1; |
| 690 |
} |
| 691 |
|
| 692 |
obt_as.freq = obt.freq; |
| 693 |
obt_as.nchannels = obt.nchannels; |
| 694 |
obt_as.fmt = obt.fmt; |
| 695 |
obt_as.endianness = obt.endianness; |
| 696 |
|
| 697 |
audio_pcm_init_info (&hw->info, &obt_as); |
| 698 |
hw->samples = obt.samples; |
| 699 |
|
| 700 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
| 701 |
if (!alsa->pcm_buf) {
|
| 702 |
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
|
| 703 |
hw->samples, 1 << hw->info.shift);
|
| 704 |
alsa_anal_close (&handle); |
| 705 |
return -1; |
| 706 |
} |
| 707 |
|
| 708 |
alsa->handle = handle; |
| 709 |
return 0; |
| 710 |
} |
| 711 |
|
| 712 |
static void alsa_fini_in (HWVoiceIn *hw) |
| 713 |
{
|
| 714 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| 715 |
|
| 716 |
alsa_anal_close (&alsa->handle); |
| 717 |
|
| 718 |
if (alsa->pcm_buf) {
|
| 719 |
qemu_free (alsa->pcm_buf); |
| 720 |
alsa->pcm_buf = NULL;
|
| 721 |
} |
| 722 |
} |
| 723 |
|
| 724 |
static int alsa_run_in (HWVoiceIn *hw) |
| 725 |
{
|
| 726 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| 727 |
int hwshift = hw->info.shift;
|
| 728 |
int i;
|
| 729 |
int live = audio_pcm_hw_get_live_in (hw);
|
| 730 |
int dead = hw->samples - live;
|
| 731 |
int decr;
|
| 732 |
struct {
|
| 733 |
int add;
|
| 734 |
int len;
|
| 735 |
} bufs[2] = {
|
| 736 |
{ hw->wpos, 0 },
|
| 737 |
{ 0, 0 }
|
| 738 |
}; |
| 739 |
snd_pcm_sframes_t avail; |
| 740 |
snd_pcm_uframes_t read_samples = 0;
|
| 741 |
|
| 742 |
if (!dead) {
|
| 743 |
return 0; |
| 744 |
} |
| 745 |
|
| 746 |
avail = alsa_get_avail (alsa->handle); |
| 747 |
if (avail < 0) { |
| 748 |
dolog ("Could not get number of captured frames\n");
|
| 749 |
return 0; |
| 750 |
} |
| 751 |
|
| 752 |
if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
|
| 753 |
avail = hw->samples; |
| 754 |
} |
| 755 |
|
| 756 |
decr = audio_MIN (dead, avail); |
| 757 |
if (!decr) {
|
| 758 |
return 0; |
| 759 |
} |
| 760 |
|
| 761 |
if (hw->wpos + decr > hw->samples) {
|
| 762 |
bufs[0].len = (hw->samples - hw->wpos);
|
| 763 |
bufs[1].len = (decr - (hw->samples - hw->wpos));
|
| 764 |
} |
| 765 |
else {
|
| 766 |
bufs[0].len = decr;
|
| 767 |
} |
| 768 |
|
| 769 |
for (i = 0; i < 2; ++i) { |
| 770 |
void *src;
|
| 771 |
st_sample_t *dst; |
| 772 |
snd_pcm_sframes_t nread; |
| 773 |
snd_pcm_uframes_t len; |
| 774 |
|
| 775 |
len = bufs[i].len; |
| 776 |
|
| 777 |
src = advance (alsa->pcm_buf, bufs[i].add << hwshift); |
| 778 |
dst = hw->conv_buf + bufs[i].add; |
| 779 |
|
| 780 |
while (len) {
|
| 781 |
nread = snd_pcm_readi (alsa->handle, src, len); |
| 782 |
|
| 783 |
if (nread <= 0) { |
| 784 |
switch (nread) {
|
| 785 |
case 0: |
| 786 |
if (conf.verbose) {
|
| 787 |
dolog ("Failed to read %ld frames (read zero)\n", len);
|
| 788 |
} |
| 789 |
goto exit;
|
| 790 |
|
| 791 |
case -EPIPE:
|
| 792 |
if (alsa_recover (alsa->handle)) {
|
| 793 |
alsa_logerr (nread, "Failed to read %ld frames\n", len);
|
| 794 |
goto exit;
|
| 795 |
} |
| 796 |
if (conf.verbose) {
|
| 797 |
dolog ("Recovering from capture xrun\n");
|
| 798 |
} |
| 799 |
continue;
|
| 800 |
|
| 801 |
case -EAGAIN:
|
| 802 |
goto exit;
|
| 803 |
|
| 804 |
default:
|
| 805 |
alsa_logerr ( |
| 806 |
nread, |
| 807 |
"Failed to read %ld frames from %p\n",
|
| 808 |
len, |
| 809 |
src |
| 810 |
); |
| 811 |
goto exit;
|
| 812 |
} |
| 813 |
} |
| 814 |
|
| 815 |
hw->conv (dst, src, nread, &nominal_volume); |
| 816 |
|
| 817 |
src = advance (src, nread << hwshift); |
| 818 |
dst += nread; |
| 819 |
|
| 820 |
read_samples += nread; |
| 821 |
len -= nread; |
| 822 |
} |
| 823 |
} |
| 824 |
|
| 825 |
exit:
|
| 826 |
hw->wpos = (hw->wpos + read_samples) % hw->samples; |
| 827 |
return read_samples;
|
| 828 |
} |
| 829 |
|
| 830 |
static int alsa_read (SWVoiceIn *sw, void *buf, int size) |
| 831 |
{
|
| 832 |
return audio_pcm_sw_read (sw, buf, size);
|
| 833 |
} |
| 834 |
|
| 835 |
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) |
| 836 |
{
|
| 837 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| 838 |
|
| 839 |
switch (cmd) {
|
| 840 |
case VOICE_ENABLE:
|
| 841 |
ldebug ("enabling voice\n");
|
| 842 |
return alsa_voice_ctl (alsa->handle, "capture", 0); |
| 843 |
|
| 844 |
case VOICE_DISABLE:
|
| 845 |
ldebug ("disabling voice\n");
|
| 846 |
return alsa_voice_ctl (alsa->handle, "capture", 1); |
| 847 |
} |
| 848 |
|
| 849 |
return -1; |
| 850 |
} |
| 851 |
|
| 852 |
static void *alsa_audio_init (void) |
| 853 |
{
|
| 854 |
return &conf;
|
| 855 |
} |
| 856 |
|
| 857 |
static void alsa_audio_fini (void *opaque) |
| 858 |
{
|
| 859 |
(void) opaque;
|
| 860 |
} |
| 861 |
|
| 862 |
static struct audio_option alsa_options[] = { |
| 863 |
{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
|
| 864 |
"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
| 865 |
{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
|
| 866 |
"DAC period size (0 to go with system default)",
|
| 867 |
&conf.period_size_out_overridden, 0},
|
| 868 |
{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
|
| 869 |
"DAC buffer size (0 to go with system default)",
|
| 870 |
&conf.buffer_size_out_overridden, 0},
|
| 871 |
|
| 872 |
{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
|
| 873 |
"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
| 874 |
{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
|
| 875 |
"ADC period size (0 to go with system default)",
|
| 876 |
&conf.period_size_in_overridden, 0},
|
| 877 |
{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
|
| 878 |
"ADC buffer size (0 to go with system default)",
|
| 879 |
&conf.buffer_size_in_overridden, 0},
|
| 880 |
|
| 881 |
{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
|
| 882 |
"(undocumented)", NULL, 0}, |
| 883 |
|
| 884 |
{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
|
| 885 |
"DAC device name (for instance dmix)", NULL, 0}, |
| 886 |
|
| 887 |
{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
|
| 888 |
"ADC device name", NULL, 0}, |
| 889 |
|
| 890 |
{"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
|
| 891 |
"Behave in a more verbose way", NULL, 0}, |
| 892 |
|
| 893 |
{NULL, 0, NULL, NULL, NULL, 0}
|
| 894 |
}; |
| 895 |
|
| 896 |
static struct audio_pcm_ops alsa_pcm_ops = { |
| 897 |
alsa_init_out, |
| 898 |
alsa_fini_out, |
| 899 |
alsa_run_out, |
| 900 |
alsa_write, |
| 901 |
alsa_ctl_out, |
| 902 |
|
| 903 |
alsa_init_in, |
| 904 |
alsa_fini_in, |
| 905 |
alsa_run_in, |
| 906 |
alsa_read, |
| 907 |
alsa_ctl_in |
| 908 |
}; |
| 909 |
|
| 910 |
struct audio_driver alsa_audio_driver = {
|
| 911 |
INIT_FIELD (name = ) "alsa",
|
| 912 |
INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
|
| 913 |
INIT_FIELD (options = ) alsa_options, |
| 914 |
INIT_FIELD (init = ) alsa_audio_init, |
| 915 |
INIT_FIELD (fini = ) alsa_audio_fini, |
| 916 |
INIT_FIELD (pcm_ops = ) &alsa_pcm_ops, |
| 917 |
INIT_FIELD (can_be_default = ) 1,
|
| 918 |
INIT_FIELD (max_voices_out = ) INT_MAX, |
| 919 |
INIT_FIELD (max_voices_in = ) INT_MAX, |
| 920 |
INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
|
| 921 |
INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
|
| 922 |
}; |