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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" |
26 |
#include "audio.h" |
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|
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#define AUDIO_CAP "alsa" |
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#include "audio_int.h" |
30 |
|
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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; |
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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 |
|
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int buffer_size_out_overridden;
|
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int period_size_out_overridden;
|
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int verbose;
|
60 |
} conf = { |
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.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; |
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}; |
73 |
|
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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 endianness;
|
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int nchannels;
|
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snd_pcm_uframes_t samples; |
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}; |
81 |
|
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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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} |
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|
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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;
|
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|
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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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|
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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 |
|
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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:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_S8; |
162 |
break;
|
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|
164 |
case SND_PCM_FORMAT_U8:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U8; |
167 |
break;
|
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|
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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:
|
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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;
|
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*fmt = AUD_FMT_S16; |
182 |
break;
|
183 |
|
184 |
case SND_PCM_FORMAT_U16_BE:
|
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*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;
|
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*fmt = AUD_FMT_S32; |
192 |
break;
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|
194 |
case SND_PCM_FORMAT_U32_LE:
|
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*endianness = 0;
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*fmt = AUD_FMT_U32; |
197 |
break;
|
198 |
|
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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)
|
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{ |
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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dolog ("requested: buffer size %d period size %d\n",
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req->buffer_size, req->period_size); |
228 |
dolog ("obtained: samples %ld\n", obt->samples);
|
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} |
230 |
|
231 |
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 |
|
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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");
|
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return;
|
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} |
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");
|
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alsa_logerr (err, "Failed to set software threshold to %ld\n",
|
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threshold); |
250 |
return;
|
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} |
252 |
|
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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 |
}; |