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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 |
|
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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; |
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unsigned int buffer_size_out; |
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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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.buffer_size_out = 1024,
|
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.pcm_name_out = "default",
|
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.pcm_name_in = "default",
|
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}; |
65 |
|
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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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int override_mask;
|
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unsigned int buffer_size; |
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unsigned int period_size; |
74 |
}; |
75 |
|
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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; |
82 |
}; |
83 |
|
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static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) |
85 |
{ |
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va_list ap; |
87 |
|
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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); |
91 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
94 |
|
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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, |
99 |
... |
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) |
101 |
{ |
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va_list ap; |
103 |
|
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AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
|
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|
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va_start (ap, fmt); |
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AUD_vlog (AUDIO_CAP, fmt, ap); |
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va_end (ap); |
109 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
112 |
|
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static void 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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} |
121 |
|
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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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} |
126 |
|
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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;
|
138 |
|
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case AUD_FMT_U16:
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return SND_PCM_FORMAT_U16_LE;
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|
142 |
case AUD_FMT_S32:
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return SND_PCM_FORMAT_S32_LE;
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|
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case AUD_FMT_U32:
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return SND_PCM_FORMAT_U32_LE;
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|
148 |
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 (); |
152 |
#endif
|
153 |
return SND_PCM_FORMAT_U8;
|
154 |
} |
155 |
} |
156 |
|
157 |
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, |
158 |
int *endianness)
|
159 |
{ |
160 |
switch (alsafmt) {
|
161 |
case SND_PCM_FORMAT_S8:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_S8; |
164 |
break;
|
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|
166 |
case SND_PCM_FORMAT_U8:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U8; |
169 |
break;
|
170 |
|
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case SND_PCM_FORMAT_S16_LE:
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*endianness = 0;
|
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*fmt = AUD_FMT_S16; |
174 |
break;
|
175 |
|
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case SND_PCM_FORMAT_U16_LE:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U16; |
179 |
break;
|
180 |
|
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case SND_PCM_FORMAT_S16_BE:
|
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*endianness = 1;
|
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*fmt = AUD_FMT_S16; |
184 |
break;
|
185 |
|
186 |
case SND_PCM_FORMAT_U16_BE:
|
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*endianness = 1;
|
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*fmt = AUD_FMT_U16; |
189 |
break;
|
190 |
|
191 |
case SND_PCM_FORMAT_S32_LE:
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*endianness = 0;
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*fmt = AUD_FMT_S32; |
194 |
break;
|
195 |
|
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case SND_PCM_FORMAT_U32_LE:
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*endianness = 0;
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*fmt = AUD_FMT_U32; |
199 |
break;
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200 |
|
201 |
case SND_PCM_FORMAT_S32_BE:
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*endianness = 1;
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*fmt = AUD_FMT_S32; |
204 |
break;
|
205 |
|
206 |
case SND_PCM_FORMAT_U32_BE:
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*endianness = 1;
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*fmt = AUD_FMT_U32; |
209 |
break;
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210 |
|
211 |
default:
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dolog ("Unrecognized audio format %d\n", alsafmt);
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return -1; |
214 |
} |
215 |
|
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return 0; |
217 |
} |
218 |
|
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static void alsa_dump_info (struct alsa_params_req *req, |
220 |
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); |
226 |
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
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dolog ("============================================\n");
|
228 |
dolog ("requested: buffer size %d period size %d\n",
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req->buffer_size, req->period_size); |
230 |
dolog ("obtained: samples %ld\n", obt->samples);
|
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} |
232 |
|
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static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) |
234 |
{ |
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int err;
|
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snd_pcm_sw_params_t *sw_params; |
237 |
|
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snd_pcm_sw_params_alloca (&sw_params); |
239 |
|
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err = snd_pcm_sw_params_current (handle, sw_params); |
241 |
if (err < 0) { |
242 |
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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} |
246 |
|
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err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); |
248 |
if (err < 0) { |
249 |
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); |
252 |
return;
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253 |
} |
254 |
|
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err = snd_pcm_sw_params (handle, sw_params); |
256 |
if (err < 0) { |
257 |
dolog ("Could not fully initialize DAC\n");
|
258 |
alsa_logerr (err, "Failed to set software parameters\n");
|
259 |
return;
|
260 |
} |
261 |
} |
262 |
|
263 |
static int alsa_open (int in, struct alsa_params_req *req, |
264 |
struct alsa_params_obt *obt, snd_pcm_t **handlep)
|
265 |
{ |
266 |
snd_pcm_t *handle; |
267 |
snd_pcm_hw_params_t *hw_params; |
268 |
int err;
|
269 |
int size_in_usec;
|
270 |
unsigned int freq, nchannels; |
271 |
const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; |
272 |
snd_pcm_uframes_t obt_buffer_size; |
273 |
const char *typ = in ? "ADC" : "DAC"; |
274 |
snd_pcm_format_t obtfmt; |
275 |
|
276 |
freq = req->freq; |
277 |
nchannels = req->nchannels; |
278 |
size_in_usec = req->size_in_usec; |
279 |
|
280 |
snd_pcm_hw_params_alloca (&hw_params); |
281 |
|
282 |
err = snd_pcm_open ( |
283 |
&handle, |
284 |
pcm_name, |
285 |
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, |
286 |
SND_PCM_NONBLOCK |
287 |
); |
288 |
if (err < 0) { |
289 |
alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
|
290 |
return -1; |
291 |
} |
292 |
|
293 |
err = snd_pcm_hw_params_any (handle, hw_params); |
294 |
if (err < 0) { |
295 |
alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
|
296 |
goto err;
|
297 |
} |
298 |
|
299 |
err = snd_pcm_hw_params_set_access ( |
300 |
handle, |
301 |
hw_params, |
302 |
SND_PCM_ACCESS_RW_INTERLEAVED |
303 |
); |
304 |
if (err < 0) { |
305 |
alsa_logerr2 (err, typ, "Failed to set access type\n");
|
306 |
goto err;
|
307 |
} |
308 |
|
309 |
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); |
310 |
if (err < 0 && conf.verbose) { |
311 |
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
|
312 |
} |
313 |
|
314 |
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
|
315 |
if (err < 0) { |
316 |
alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
|
317 |
goto err;
|
318 |
} |
319 |
|
320 |
err = snd_pcm_hw_params_set_channels_near ( |
321 |
handle, |
322 |
hw_params, |
323 |
&nchannels |
324 |
); |
325 |
if (err < 0) { |
326 |
alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
|
327 |
req->nchannels); |
328 |
goto err;
|
329 |
} |
330 |
|
331 |
if (nchannels != 1 && nchannels != 2) { |
332 |
alsa_logerr2 (err, typ, |
333 |
"Can not handle obtained number of channels %d\n",
|
334 |
nchannels); |
335 |
goto err;
|
336 |
} |
337 |
|
338 |
if (req->buffer_size) {
|
339 |
unsigned long obt; |
340 |
|
341 |
if (size_in_usec) {
|
342 |
int dir = 0; |
343 |
unsigned int btime = req->buffer_size; |
344 |
|
345 |
err = snd_pcm_hw_params_set_buffer_time_near ( |
346 |
handle, |
347 |
hw_params, |
348 |
&btime, |
349 |
&dir |
350 |
); |
351 |
obt = btime; |
352 |
} |
353 |
else {
|
354 |
snd_pcm_uframes_t bsize = req->buffer_size; |
355 |
|
356 |
err = snd_pcm_hw_params_set_buffer_size_near ( |
357 |
handle, |
358 |
hw_params, |
359 |
&bsize |
360 |
); |
361 |
obt = bsize; |
362 |
} |
363 |
if (err < 0) { |
364 |
alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
|
365 |
size_in_usec ? "time" : "size", req->buffer_size); |
366 |
goto err;
|
367 |
} |
368 |
|
369 |
if ((req->override_mask & 2) && (obt - req->buffer_size)) |
370 |
dolog ("Requested buffer %s %u was rejected, using %lu\n",
|
371 |
size_in_usec ? "time" : "size", req->buffer_size, obt); |
372 |
} |
373 |
|
374 |
if (req->period_size) {
|
375 |
unsigned long obt; |
376 |
|
377 |
if (size_in_usec) {
|
378 |
int dir = 0; |
379 |
unsigned int ptime = req->period_size; |
380 |
|
381 |
err = snd_pcm_hw_params_set_period_time_near ( |
382 |
handle, |
383 |
hw_params, |
384 |
&ptime, |
385 |
&dir |
386 |
); |
387 |
obt = ptime; |
388 |
} |
389 |
else {
|
390 |
int dir = 0; |
391 |
snd_pcm_uframes_t psize = req->period_size; |
392 |
|
393 |
err = snd_pcm_hw_params_set_period_size_near ( |
394 |
handle, |
395 |
hw_params, |
396 |
&psize, |
397 |
&dir |
398 |
); |
399 |
obt = psize; |
400 |
} |
401 |
|
402 |
if (err < 0) { |
403 |
alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
|
404 |
size_in_usec ? "time" : "size", req->period_size); |
405 |
goto err;
|
406 |
} |
407 |
|
408 |
if ((req->override_mask & 1) && (obt - req->period_size)) |
409 |
dolog ("Requested period %s %u was rejected, using %lu\n",
|
410 |
size_in_usec ? "time" : "size", req->period_size, obt); |
411 |
} |
412 |
|
413 |
err = snd_pcm_hw_params (handle, hw_params); |
414 |
if (err < 0) { |
415 |
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
|
416 |
goto err;
|
417 |
} |
418 |
|
419 |
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); |
420 |
if (err < 0) { |
421 |
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
|
422 |
goto err;
|
423 |
} |
424 |
|
425 |
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); |
426 |
if (err < 0) { |
427 |
alsa_logerr2 (err, typ, "Failed to get format\n");
|
428 |
goto err;
|
429 |
} |
430 |
|
431 |
if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
|
432 |
dolog ("Invalid format was returned %d\n", obtfmt);
|
433 |
goto err;
|
434 |
} |
435 |
|
436 |
err = snd_pcm_prepare (handle); |
437 |
if (err < 0) { |
438 |
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
|
439 |
goto err;
|
440 |
} |
441 |
|
442 |
if (!in && conf.threshold) {
|
443 |
snd_pcm_uframes_t threshold; |
444 |
int bytes_per_sec;
|
445 |
|
446 |
bytes_per_sec = freq << (nchannels == 2);
|
447 |
|
448 |
switch (obt->fmt) {
|
449 |
case AUD_FMT_S8:
|
450 |
case AUD_FMT_U8:
|
451 |
break;
|
452 |
|
453 |
case AUD_FMT_S16:
|
454 |
case AUD_FMT_U16:
|
455 |
bytes_per_sec <<= 1;
|
456 |
break;
|
457 |
|
458 |
case AUD_FMT_S32:
|
459 |
case AUD_FMT_U32:
|
460 |
bytes_per_sec <<= 2;
|
461 |
break;
|
462 |
} |
463 |
|
464 |
threshold = (conf.threshold * bytes_per_sec) / 1000;
|
465 |
alsa_set_threshold (handle, threshold); |
466 |
} |
467 |
|
468 |
obt->nchannels = nchannels; |
469 |
obt->freq = freq; |
470 |
obt->samples = obt_buffer_size; |
471 |
|
472 |
*handlep = handle; |
473 |
|
474 |
if (conf.verbose &&
|
475 |
(obt->fmt != req->fmt || |
476 |
obt->nchannels != req->nchannels || |
477 |
obt->freq != req->freq)) { |
478 |
dolog ("Audio paramters for %s\n", typ);
|
479 |
alsa_dump_info (req, obt); |
480 |
} |
481 |
|
482 |
#ifdef DEBUG
|
483 |
alsa_dump_info (req, obt); |
484 |
#endif
|
485 |
return 0; |
486 |
|
487 |
err:
|
488 |
alsa_anal_close (&handle); |
489 |
return -1; |
490 |
} |
491 |
|
492 |
static int alsa_recover (snd_pcm_t *handle) |
493 |
{ |
494 |
int err = snd_pcm_prepare (handle);
|
495 |
if (err < 0) { |
496 |
alsa_logerr (err, "Failed to prepare handle %p\n", handle);
|
497 |
return -1; |
498 |
} |
499 |
return 0; |
500 |
} |
501 |
|
502 |
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
|
503 |
{ |
504 |
snd_pcm_sframes_t avail; |
505 |
|
506 |
avail = snd_pcm_avail_update (handle); |
507 |
if (avail < 0) { |
508 |
if (avail == -EPIPE) {
|
509 |
if (!alsa_recover (handle)) {
|
510 |
avail = snd_pcm_avail_update (handle); |
511 |
} |
512 |
} |
513 |
|
514 |
if (avail < 0) { |
515 |
alsa_logerr (avail, |
516 |
"Could not obtain number of available frames\n");
|
517 |
return -1; |
518 |
} |
519 |
} |
520 |
|
521 |
return avail;
|
522 |
} |
523 |
|
524 |
static int alsa_run_out (HWVoiceOut *hw) |
525 |
{ |
526 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
527 |
int rpos, live, decr;
|
528 |
int samples;
|
529 |
uint8_t *dst; |
530 |
struct st_sample *src;
|
531 |
snd_pcm_sframes_t avail; |
532 |
|
533 |
live = audio_pcm_hw_get_live_out (hw); |
534 |
if (!live) {
|
535 |
return 0; |
536 |
} |
537 |
|
538 |
avail = alsa_get_avail (alsa->handle); |
539 |
if (avail < 0) { |
540 |
dolog ("Could not get number of available playback frames\n");
|
541 |
return 0; |
542 |
} |
543 |
|
544 |
decr = audio_MIN (live, avail); |
545 |
samples = decr; |
546 |
rpos = hw->rpos; |
547 |
while (samples) {
|
548 |
int left_till_end_samples = hw->samples - rpos;
|
549 |
int len = audio_MIN (samples, left_till_end_samples);
|
550 |
snd_pcm_sframes_t written; |
551 |
|
552 |
src = hw->mix_buf + rpos; |
553 |
dst = advance (alsa->pcm_buf, rpos << hw->info.shift); |
554 |
|
555 |
hw->clip (dst, src, len); |
556 |
|
557 |
while (len) {
|
558 |
written = snd_pcm_writei (alsa->handle, dst, len); |
559 |
|
560 |
if (written <= 0) { |
561 |
switch (written) {
|
562 |
case 0: |
563 |
if (conf.verbose) {
|
564 |
dolog ("Failed to write %d frames (wrote zero)\n", len);
|
565 |
} |
566 |
goto exit;
|
567 |
|
568 |
case -EPIPE:
|
569 |
if (alsa_recover (alsa->handle)) {
|
570 |
alsa_logerr (written, "Failed to write %d frames\n",
|
571 |
len); |
572 |
goto exit;
|
573 |
} |
574 |
if (conf.verbose) {
|
575 |
dolog ("Recovering from playback xrun\n");
|
576 |
} |
577 |
continue;
|
578 |
|
579 |
case -EAGAIN:
|
580 |
goto exit;
|
581 |
|
582 |
default:
|
583 |
alsa_logerr (written, "Failed to write %d frames to %p\n",
|
584 |
len, dst); |
585 |
goto exit;
|
586 |
} |
587 |
} |
588 |
|
589 |
rpos = (rpos + written) % hw->samples; |
590 |
samples -= written; |
591 |
len -= written; |
592 |
dst = advance (dst, written << hw->info.shift); |
593 |
src += written; |
594 |
} |
595 |
} |
596 |
|
597 |
exit:
|
598 |
hw->rpos = rpos; |
599 |
return decr;
|
600 |
} |
601 |
|
602 |
static void alsa_fini_out (HWVoiceOut *hw) |
603 |
{ |
604 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
605 |
|
606 |
ldebug ("alsa_fini\n");
|
607 |
alsa_anal_close (&alsa->handle); |
608 |
|
609 |
if (alsa->pcm_buf) {
|
610 |
qemu_free (alsa->pcm_buf); |
611 |
alsa->pcm_buf = NULL;
|
612 |
} |
613 |
} |
614 |
|
615 |
static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as) |
616 |
{ |
617 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
618 |
struct alsa_params_req req;
|
619 |
struct alsa_params_obt obt;
|
620 |
snd_pcm_t *handle; |
621 |
struct audsettings obt_as;
|
622 |
|
623 |
req.fmt = aud_to_alsafmt (as->fmt); |
624 |
req.freq = as->freq; |
625 |
req.nchannels = as->nchannels; |
626 |
req.period_size = conf.period_size_out; |
627 |
req.buffer_size = conf.buffer_size_out; |
628 |
req.size_in_usec = conf.size_in_usec_out; |
629 |
req.override_mask = !!conf.period_size_out_overridden |
630 |
| (!!conf.buffer_size_out_overridden << 1);
|
631 |
|
632 |
if (alsa_open (0, &req, &obt, &handle)) { |
633 |
return -1; |
634 |
} |
635 |
|
636 |
obt_as.freq = obt.freq; |
637 |
obt_as.nchannels = obt.nchannels; |
638 |
obt_as.fmt = obt.fmt; |
639 |
obt_as.endianness = obt.endianness; |
640 |
|
641 |
audio_pcm_init_info (&hw->info, &obt_as); |
642 |
hw->samples = obt.samples; |
643 |
|
644 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
|
645 |
if (!alsa->pcm_buf) {
|
646 |
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
|
647 |
hw->samples, 1 << hw->info.shift);
|
648 |
alsa_anal_close (&handle); |
649 |
return -1; |
650 |
} |
651 |
|
652 |
alsa->handle = handle; |
653 |
return 0; |
654 |
} |
655 |
|
656 |
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause) |
657 |
{ |
658 |
int err;
|
659 |
|
660 |
if (pause) {
|
661 |
err = snd_pcm_drop (handle); |
662 |
if (err < 0) { |
663 |
alsa_logerr (err, "Could not stop %s\n", typ);
|
664 |
return -1; |
665 |
} |
666 |
} |
667 |
else {
|
668 |
err = snd_pcm_prepare (handle); |
669 |
if (err < 0) { |
670 |
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
|
671 |
return -1; |
672 |
} |
673 |
} |
674 |
|
675 |
return 0; |
676 |
} |
677 |
|
678 |
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) |
679 |
{ |
680 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
681 |
|
682 |
switch (cmd) {
|
683 |
case VOICE_ENABLE:
|
684 |
ldebug ("enabling voice\n");
|
685 |
return alsa_voice_ctl (alsa->handle, "playback", 0); |
686 |
|
687 |
case VOICE_DISABLE:
|
688 |
ldebug ("disabling voice\n");
|
689 |
return alsa_voice_ctl (alsa->handle, "playback", 1); |
690 |
} |
691 |
|
692 |
return -1; |
693 |
} |
694 |
|
695 |
static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as) |
696 |
{ |
697 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
698 |
struct alsa_params_req req;
|
699 |
struct alsa_params_obt obt;
|
700 |
snd_pcm_t *handle; |
701 |
struct audsettings obt_as;
|
702 |
|
703 |
req.fmt = aud_to_alsafmt (as->fmt); |
704 |
req.freq = as->freq; |
705 |
req.nchannels = as->nchannels; |
706 |
req.period_size = conf.period_size_in; |
707 |
req.buffer_size = conf.buffer_size_in; |
708 |
req.size_in_usec = conf.size_in_usec_in; |
709 |
req.override_mask = !!conf.period_size_in_overridden |
710 |
| (!!conf.buffer_size_in_overridden << 1);
|
711 |
|
712 |
if (alsa_open (1, &req, &obt, &handle)) { |
713 |
return -1; |
714 |
} |
715 |
|
716 |
obt_as.freq = obt.freq; |
717 |
obt_as.nchannels = obt.nchannels; |
718 |
obt_as.fmt = obt.fmt; |
719 |
obt_as.endianness = obt.endianness; |
720 |
|
721 |
audio_pcm_init_info (&hw->info, &obt_as); |
722 |
hw->samples = obt.samples; |
723 |
|
724 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
725 |
if (!alsa->pcm_buf) {
|
726 |
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
|
727 |
hw->samples, 1 << hw->info.shift);
|
728 |
alsa_anal_close (&handle); |
729 |
return -1; |
730 |
} |
731 |
|
732 |
alsa->handle = handle; |
733 |
return 0; |
734 |
} |
735 |
|
736 |
static void alsa_fini_in (HWVoiceIn *hw) |
737 |
{ |
738 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
739 |
|
740 |
alsa_anal_close (&alsa->handle); |
741 |
|
742 |
if (alsa->pcm_buf) {
|
743 |
qemu_free (alsa->pcm_buf); |
744 |
alsa->pcm_buf = NULL;
|
745 |
} |
746 |
} |
747 |
|
748 |
static int alsa_run_in (HWVoiceIn *hw) |
749 |
{ |
750 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
751 |
int hwshift = hw->info.shift;
|
752 |
int i;
|
753 |
int live = audio_pcm_hw_get_live_in (hw);
|
754 |
int dead = hw->samples - live;
|
755 |
int decr;
|
756 |
struct {
|
757 |
int add;
|
758 |
int len;
|
759 |
} bufs[2] = {
|
760 |
{ hw->wpos, 0 },
|
761 |
{ 0, 0 } |
762 |
}; |
763 |
snd_pcm_sframes_t avail; |
764 |
snd_pcm_uframes_t read_samples = 0;
|
765 |
|
766 |
if (!dead) {
|
767 |
return 0; |
768 |
} |
769 |
|
770 |
avail = alsa_get_avail (alsa->handle); |
771 |
if (avail < 0) { |
772 |
dolog ("Could not get number of captured frames\n");
|
773 |
return 0; |
774 |
} |
775 |
|
776 |
if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
|
777 |
avail = hw->samples; |
778 |
} |
779 |
|
780 |
decr = audio_MIN (dead, avail); |
781 |
if (!decr) {
|
782 |
return 0; |
783 |
} |
784 |
|
785 |
if (hw->wpos + decr > hw->samples) {
|
786 |
bufs[0].len = (hw->samples - hw->wpos);
|
787 |
bufs[1].len = (decr - (hw->samples - hw->wpos));
|
788 |
} |
789 |
else {
|
790 |
bufs[0].len = decr;
|
791 |
} |
792 |
|
793 |
for (i = 0; i < 2; ++i) { |
794 |
void *src;
|
795 |
struct st_sample *dst;
|
796 |
snd_pcm_sframes_t nread; |
797 |
snd_pcm_uframes_t len; |
798 |
|
799 |
len = bufs[i].len; |
800 |
|
801 |
src = advance (alsa->pcm_buf, bufs[i].add << hwshift); |
802 |
dst = hw->conv_buf + bufs[i].add; |
803 |
|
804 |
while (len) {
|
805 |
nread = snd_pcm_readi (alsa->handle, src, len); |
806 |
|
807 |
if (nread <= 0) { |
808 |
switch (nread) {
|
809 |
case 0: |
810 |
if (conf.verbose) {
|
811 |
dolog ("Failed to read %ld frames (read zero)\n", len);
|
812 |
} |
813 |
goto exit;
|
814 |
|
815 |
case -EPIPE:
|
816 |
if (alsa_recover (alsa->handle)) {
|
817 |
alsa_logerr (nread, "Failed to read %ld frames\n", len);
|
818 |
goto exit;
|
819 |
} |
820 |
if (conf.verbose) {
|
821 |
dolog ("Recovering from capture xrun\n");
|
822 |
} |
823 |
continue;
|
824 |
|
825 |
case -EAGAIN:
|
826 |
goto exit;
|
827 |
|
828 |
default:
|
829 |
alsa_logerr ( |
830 |
nread, |
831 |
"Failed to read %ld frames from %p\n",
|
832 |
len, |
833 |
src |
834 |
); |
835 |
goto exit;
|
836 |
} |
837 |
} |
838 |
|
839 |
hw->conv (dst, src, nread, &nominal_volume); |
840 |
|
841 |
src = advance (src, nread << hwshift); |
842 |
dst += nread; |
843 |
|
844 |
read_samples += nread; |
845 |
len -= nread; |
846 |
} |
847 |
} |
848 |
|
849 |
exit:
|
850 |
hw->wpos = (hw->wpos + read_samples) % hw->samples; |
851 |
return read_samples;
|
852 |
} |
853 |
|
854 |
static int alsa_read (SWVoiceIn *sw, void *buf, int size) |
855 |
{ |
856 |
return audio_pcm_sw_read (sw, buf, size);
|
857 |
} |
858 |
|
859 |
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) |
860 |
{ |
861 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
862 |
|
863 |
switch (cmd) {
|
864 |
case VOICE_ENABLE:
|
865 |
ldebug ("enabling voice\n");
|
866 |
return alsa_voice_ctl (alsa->handle, "capture", 0); |
867 |
|
868 |
case VOICE_DISABLE:
|
869 |
ldebug ("disabling voice\n");
|
870 |
return alsa_voice_ctl (alsa->handle, "capture", 1); |
871 |
} |
872 |
|
873 |
return -1; |
874 |
} |
875 |
|
876 |
static void *alsa_audio_init (void) |
877 |
{ |
878 |
return &conf;
|
879 |
} |
880 |
|
881 |
static void alsa_audio_fini (void *opaque) |
882 |
{ |
883 |
(void) opaque;
|
884 |
} |
885 |
|
886 |
static struct audio_option alsa_options[] = { |
887 |
{"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
|
888 |
"DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
889 |
{"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
|
890 |
"DAC period size (0 to go with system default)",
|
891 |
&conf.period_size_out_overridden, 0},
|
892 |
{"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
|
893 |
"DAC buffer size (0 to go with system default)",
|
894 |
&conf.buffer_size_out_overridden, 0},
|
895 |
|
896 |
{"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
|
897 |
"ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0}, |
898 |
{"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
|
899 |
"ADC period size (0 to go with system default)",
|
900 |
&conf.period_size_in_overridden, 0},
|
901 |
{"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
|
902 |
"ADC buffer size (0 to go with system default)",
|
903 |
&conf.buffer_size_in_overridden, 0},
|
904 |
|
905 |
{"THRESHOLD", AUD_OPT_INT, &conf.threshold,
|
906 |
"(undocumented)", NULL, 0}, |
907 |
|
908 |
{"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
|
909 |
"DAC device name (for instance dmix)", NULL, 0}, |
910 |
|
911 |
{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
|
912 |
"ADC device name", NULL, 0}, |
913 |
|
914 |
{"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
|
915 |
"Behave in a more verbose way", NULL, 0}, |
916 |
|
917 |
{NULL, 0, NULL, NULL, NULL, 0} |
918 |
}; |
919 |
|
920 |
static struct audio_pcm_ops alsa_pcm_ops = { |
921 |
alsa_init_out, |
922 |
alsa_fini_out, |
923 |
alsa_run_out, |
924 |
alsa_write, |
925 |
alsa_ctl_out, |
926 |
|
927 |
alsa_init_in, |
928 |
alsa_fini_in, |
929 |
alsa_run_in, |
930 |
alsa_read, |
931 |
alsa_ctl_in |
932 |
}; |
933 |
|
934 |
struct audio_driver alsa_audio_driver = {
|
935 |
INIT_FIELD (name = ) "alsa",
|
936 |
INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
|
937 |
INIT_FIELD (options = ) alsa_options, |
938 |
INIT_FIELD (init = ) alsa_audio_init, |
939 |
INIT_FIELD (fini = ) alsa_audio_fini, |
940 |
INIT_FIELD (pcm_ops = ) &alsa_pcm_ops, |
941 |
INIT_FIELD (can_be_default = ) 1,
|
942 |
INIT_FIELD (max_voices_out = ) INT_MAX, |
943 |
INIT_FIELD (max_voices_in = ) INT_MAX, |
944 |
INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
|
945 |
INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
|
946 |
}; |