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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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#if QEMU_GNUC_PREREQ(4, 3) |
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#pragma GCC diagnostic ignored "-Waddress" |
30 |
#endif
|
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|
32 |
#define AUDIO_CAP "alsa" |
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#include "audio_int.h" |
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|
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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; |
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|
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static struct { |
48 |
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; |
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unsigned int buffer_size_in; |
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unsigned int period_size_in; |
54 |
unsigned int buffer_size_out; |
55 |
unsigned int period_size_out; |
56 |
unsigned int threshold; |
57 |
|
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int buffer_size_in_overridden;
|
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int period_size_in_overridden;
|
60 |
|
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int buffer_size_out_overridden;
|
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int period_size_out_overridden;
|
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int verbose;
|
64 |
} 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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}; |
69 |
|
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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; |
77 |
unsigned int period_size; |
78 |
}; |
79 |
|
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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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}; |
87 |
|
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static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) |
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{ |
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va_list ap; |
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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); |
95 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
98 |
|
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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, |
103 |
... |
104 |
) |
105 |
{ |
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va_list ap; |
107 |
|
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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); |
113 |
|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
116 |
|
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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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} |
125 |
|
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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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} |
130 |
|
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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;
|
142 |
|
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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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148 |
|
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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);
|
154 |
#ifdef DEBUG_AUDIO
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abort (); |
156 |
#endif
|
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return SND_PCM_FORMAT_U8;
|
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} |
159 |
} |
160 |
|
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static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, |
162 |
int *endianness)
|
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{ |
164 |
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; |
168 |
break;
|
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|
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case SND_PCM_FORMAT_U8:
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*endianness = 0;
|
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*fmt = AUD_FMT_U8; |
173 |
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; |
178 |
break;
|
179 |
|
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case SND_PCM_FORMAT_U16_LE:
|
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*endianness = 0;
|
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*fmt = AUD_FMT_U16; |
183 |
break;
|
184 |
|
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case SND_PCM_FORMAT_S16_BE:
|
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*endianness = 1;
|
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*fmt = AUD_FMT_S16; |
188 |
break;
|
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|
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case SND_PCM_FORMAT_U16_BE:
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*endianness = 1;
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*fmt = AUD_FMT_U16; |
193 |
break;
|
194 |
|
195 |
case SND_PCM_FORMAT_S32_LE:
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*endianness = 0;
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*fmt = AUD_FMT_S32; |
198 |
break;
|
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|
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case SND_PCM_FORMAT_U32_LE:
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*endianness = 0;
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*fmt = AUD_FMT_U32; |
203 |
break;
|
204 |
|
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case SND_PCM_FORMAT_S32_BE:
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*endianness = 1;
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*fmt = AUD_FMT_S32; |
208 |
break;
|
209 |
|
210 |
case SND_PCM_FORMAT_U32_BE:
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*endianness = 1;
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*fmt = AUD_FMT_U32; |
213 |
break;
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214 |
|
215 |
default:
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dolog ("Unrecognized audio format %d\n", alsafmt);
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return -1; |
218 |
} |
219 |
|
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return 0; |
221 |
} |
222 |
|
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static void alsa_dump_info (struct alsa_params_req *req, |
224 |
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); |
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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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} |
236 |
|
237 |
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) |
238 |
{ |
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int err;
|
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snd_pcm_sw_params_t *sw_params; |
241 |
|
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snd_pcm_sw_params_alloca (&sw_params); |
243 |
|
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err = snd_pcm_sw_params_current (handle, sw_params); |
245 |
if (err < 0) { |
246 |
dolog ("Could not fully initialize DAC\n");
|
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alsa_logerr (err, "Failed to get current software parameters\n");
|
248 |
return;
|
249 |
} |
250 |
|
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err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); |
252 |
if (err < 0) { |
253 |
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); |
256 |
return;
|
257 |
} |
258 |
|
259 |
err = snd_pcm_sw_params (handle, sw_params); |
260 |
if (err < 0) { |
261 |
dolog ("Could not fully initialize DAC\n");
|
262 |
alsa_logerr (err, "Failed to set software parameters\n");
|
263 |
return;
|
264 |
} |
265 |
} |
266 |
|
267 |
static int alsa_open (int in, struct alsa_params_req *req, |
268 |
struct alsa_params_obt *obt, snd_pcm_t **handlep)
|
269 |
{ |
270 |
snd_pcm_t *handle; |
271 |
snd_pcm_hw_params_t *hw_params; |
272 |
int err;
|
273 |
int size_in_usec;
|
274 |
unsigned int freq, nchannels; |
275 |
const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; |
276 |
snd_pcm_uframes_t obt_buffer_size; |
277 |
const char *typ = in ? "ADC" : "DAC"; |
278 |
snd_pcm_format_t obtfmt; |
279 |
|
280 |
freq = req->freq; |
281 |
nchannels = req->nchannels; |
282 |
size_in_usec = req->size_in_usec; |
283 |
|
284 |
snd_pcm_hw_params_alloca (&hw_params); |
285 |
|
286 |
err = snd_pcm_open ( |
287 |
&handle, |
288 |
pcm_name, |
289 |
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, |
290 |
SND_PCM_NONBLOCK |
291 |
); |
292 |
if (err < 0) { |
293 |
alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
|
294 |
return -1; |
295 |
} |
296 |
|
297 |
err = snd_pcm_hw_params_any (handle, hw_params); |
298 |
if (err < 0) { |
299 |
alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
|
300 |
goto err;
|
301 |
} |
302 |
|
303 |
err = snd_pcm_hw_params_set_access ( |
304 |
handle, |
305 |
hw_params, |
306 |
SND_PCM_ACCESS_RW_INTERLEAVED |
307 |
); |
308 |
if (err < 0) { |
309 |
alsa_logerr2 (err, typ, "Failed to set access type\n");
|
310 |
goto err;
|
311 |
} |
312 |
|
313 |
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); |
314 |
if (err < 0 && conf.verbose) { |
315 |
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
|
316 |
} |
317 |
|
318 |
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
|
319 |
if (err < 0) { |
320 |
alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
|
321 |
goto err;
|
322 |
} |
323 |
|
324 |
err = snd_pcm_hw_params_set_channels_near ( |
325 |
handle, |
326 |
hw_params, |
327 |
&nchannels |
328 |
); |
329 |
if (err < 0) { |
330 |
alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
|
331 |
req->nchannels); |
332 |
goto err;
|
333 |
} |
334 |
|
335 |
if (nchannels != 1 && nchannels != 2) { |
336 |
alsa_logerr2 (err, typ, |
337 |
"Can not handle obtained number of channels %d\n",
|
338 |
nchannels); |
339 |
goto err;
|
340 |
} |
341 |
|
342 |
if (req->buffer_size) {
|
343 |
unsigned long obt; |
344 |
|
345 |
if (size_in_usec) {
|
346 |
int dir = 0; |
347 |
unsigned int btime = req->buffer_size; |
348 |
|
349 |
err = snd_pcm_hw_params_set_buffer_time_near ( |
350 |
handle, |
351 |
hw_params, |
352 |
&btime, |
353 |
&dir |
354 |
); |
355 |
obt = btime; |
356 |
} |
357 |
else {
|
358 |
snd_pcm_uframes_t bsize = req->buffer_size; |
359 |
|
360 |
err = snd_pcm_hw_params_set_buffer_size_near ( |
361 |
handle, |
362 |
hw_params, |
363 |
&bsize |
364 |
); |
365 |
obt = bsize; |
366 |
} |
367 |
if (err < 0) { |
368 |
alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
|
369 |
size_in_usec ? "time" : "size", req->buffer_size); |
370 |
goto err;
|
371 |
} |
372 |
|
373 |
if ((req->override_mask & 2) && (obt - req->buffer_size)) |
374 |
dolog ("Requested buffer %s %u was rejected, using %lu\n",
|
375 |
size_in_usec ? "time" : "size", req->buffer_size, obt); |
376 |
} |
377 |
|
378 |
if (req->period_size) {
|
379 |
unsigned long obt; |
380 |
|
381 |
if (size_in_usec) {
|
382 |
int dir = 0; |
383 |
unsigned int ptime = req->period_size; |
384 |
|
385 |
err = snd_pcm_hw_params_set_period_time_near ( |
386 |
handle, |
387 |
hw_params, |
388 |
&ptime, |
389 |
&dir |
390 |
); |
391 |
obt = ptime; |
392 |
} |
393 |
else {
|
394 |
int dir = 0; |
395 |
snd_pcm_uframes_t psize = req->period_size; |
396 |
|
397 |
err = snd_pcm_hw_params_set_period_size_near ( |
398 |
handle, |
399 |
hw_params, |
400 |
&psize, |
401 |
&dir |
402 |
); |
403 |
obt = psize; |
404 |
} |
405 |
|
406 |
if (err < 0) { |
407 |
alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
|
408 |
size_in_usec ? "time" : "size", req->period_size); |
409 |
goto err;
|
410 |
} |
411 |
|
412 |
if ((req->override_mask & 1) && (obt - req->period_size)) |
413 |
dolog ("Requested period %s %u was rejected, using %lu\n",
|
414 |
size_in_usec ? "time" : "size", req->period_size, obt); |
415 |
} |
416 |
|
417 |
err = snd_pcm_hw_params (handle, hw_params); |
418 |
if (err < 0) { |
419 |
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
|
420 |
goto err;
|
421 |
} |
422 |
|
423 |
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); |
424 |
if (err < 0) { |
425 |
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
|
426 |
goto err;
|
427 |
} |
428 |
|
429 |
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); |
430 |
if (err < 0) { |
431 |
alsa_logerr2 (err, typ, "Failed to get format\n");
|
432 |
goto err;
|
433 |
} |
434 |
|
435 |
if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
|
436 |
dolog ("Invalid format was returned %d\n", obtfmt);
|
437 |
goto err;
|
438 |
} |
439 |
|
440 |
err = snd_pcm_prepare (handle); |
441 |
if (err < 0) { |
442 |
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
|
443 |
goto err;
|
444 |
} |
445 |
|
446 |
if (!in && conf.threshold) {
|
447 |
snd_pcm_uframes_t threshold; |
448 |
int bytes_per_sec;
|
449 |
|
450 |
bytes_per_sec = freq << (nchannels == 2);
|
451 |
|
452 |
switch (obt->fmt) {
|
453 |
case AUD_FMT_S8:
|
454 |
case AUD_FMT_U8:
|
455 |
break;
|
456 |
|
457 |
case AUD_FMT_S16:
|
458 |
case AUD_FMT_U16:
|
459 |
bytes_per_sec <<= 1;
|
460 |
break;
|
461 |
|
462 |
case AUD_FMT_S32:
|
463 |
case AUD_FMT_U32:
|
464 |
bytes_per_sec <<= 2;
|
465 |
break;
|
466 |
} |
467 |
|
468 |
threshold = (conf.threshold * bytes_per_sec) / 1000;
|
469 |
alsa_set_threshold (handle, threshold); |
470 |
} |
471 |
|
472 |
obt->nchannels = nchannels; |
473 |
obt->freq = freq; |
474 |
obt->samples = obt_buffer_size; |
475 |
|
476 |
*handlep = handle; |
477 |
|
478 |
if (conf.verbose &&
|
479 |
(obt->fmt != req->fmt || |
480 |
obt->nchannels != req->nchannels || |
481 |
obt->freq != req->freq)) { |
482 |
dolog ("Audio paramters for %s\n", typ);
|
483 |
alsa_dump_info (req, obt); |
484 |
} |
485 |
|
486 |
#ifdef DEBUG
|
487 |
alsa_dump_info (req, obt); |
488 |
#endif
|
489 |
return 0; |
490 |
|
491 |
err:
|
492 |
alsa_anal_close (&handle); |
493 |
return -1; |
494 |
} |
495 |
|
496 |
static int alsa_recover (snd_pcm_t *handle) |
497 |
{ |
498 |
int err = snd_pcm_prepare (handle);
|
499 |
if (err < 0) { |
500 |
alsa_logerr (err, "Failed to prepare handle %p\n", handle);
|
501 |
return -1; |
502 |
} |
503 |
return 0; |
504 |
} |
505 |
|
506 |
static int alsa_resume (snd_pcm_t *handle) |
507 |
{ |
508 |
int err = snd_pcm_resume (handle);
|
509 |
if (err < 0) { |
510 |
alsa_logerr (err, "Failed to resume handle %p\n", handle);
|
511 |
return -1; |
512 |
} |
513 |
return 0; |
514 |
} |
515 |
|
516 |
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
|
517 |
{ |
518 |
snd_pcm_sframes_t avail; |
519 |
|
520 |
avail = snd_pcm_avail_update (handle); |
521 |
if (avail < 0) { |
522 |
if (avail == -EPIPE) {
|
523 |
if (!alsa_recover (handle)) {
|
524 |
avail = snd_pcm_avail_update (handle); |
525 |
} |
526 |
} |
527 |
|
528 |
if (avail < 0) { |
529 |
alsa_logerr (avail, |
530 |
"Could not obtain number of available frames\n");
|
531 |
return -1; |
532 |
} |
533 |
} |
534 |
|
535 |
return avail;
|
536 |
} |
537 |
|
538 |
static int alsa_run_out (HWVoiceOut *hw) |
539 |
{ |
540 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
541 |
int rpos, live, decr;
|
542 |
int samples;
|
543 |
uint8_t *dst; |
544 |
struct st_sample *src;
|
545 |
snd_pcm_sframes_t avail; |
546 |
|
547 |
live = audio_pcm_hw_get_live_out (hw); |
548 |
if (!live) {
|
549 |
return 0; |
550 |
} |
551 |
|
552 |
avail = alsa_get_avail (alsa->handle); |
553 |
if (avail < 0) { |
554 |
dolog ("Could not get number of available playback frames\n");
|
555 |
return 0; |
556 |
} |
557 |
|
558 |
decr = audio_MIN (live, avail); |
559 |
samples = decr; |
560 |
rpos = hw->rpos; |
561 |
while (samples) {
|
562 |
int left_till_end_samples = hw->samples - rpos;
|
563 |
int len = audio_MIN (samples, left_till_end_samples);
|
564 |
snd_pcm_sframes_t written; |
565 |
|
566 |
src = hw->mix_buf + rpos; |
567 |
dst = advance (alsa->pcm_buf, rpos << hw->info.shift); |
568 |
|
569 |
hw->clip (dst, src, len); |
570 |
|
571 |
while (len) {
|
572 |
written = snd_pcm_writei (alsa->handle, dst, len); |
573 |
|
574 |
if (written <= 0) { |
575 |
switch (written) {
|
576 |
case 0: |
577 |
if (conf.verbose) {
|
578 |
dolog ("Failed to write %d frames (wrote zero)\n", len);
|
579 |
} |
580 |
goto exit;
|
581 |
|
582 |
case -EPIPE:
|
583 |
if (alsa_recover (alsa->handle)) {
|
584 |
alsa_logerr (written, "Failed to write %d frames\n",
|
585 |
len); |
586 |
goto exit;
|
587 |
} |
588 |
if (conf.verbose) {
|
589 |
dolog ("Recovering from playback xrun\n");
|
590 |
} |
591 |
continue;
|
592 |
|
593 |
case -ESTRPIPE:
|
594 |
/* stream is suspended and waiting for an
|
595 |
application recovery */
|
596 |
if (alsa_resume (alsa->handle)) {
|
597 |
alsa_logerr (written, "Failed to write %d frames\n",
|
598 |
len); |
599 |
goto exit;
|
600 |
} |
601 |
if (conf.verbose) {
|
602 |
dolog ("Resuming suspended output stream\n");
|
603 |
} |
604 |
continue;
|
605 |
|
606 |
case -EAGAIN:
|
607 |
goto exit;
|
608 |
|
609 |
default:
|
610 |
alsa_logerr (written, "Failed to write %d frames to %p\n",
|
611 |
len, dst); |
612 |
goto exit;
|
613 |
} |
614 |
} |
615 |
|
616 |
rpos = (rpos + written) % hw->samples; |
617 |
samples -= written; |
618 |
len -= written; |
619 |
dst = advance (dst, written << hw->info.shift); |
620 |
src += written; |
621 |
} |
622 |
} |
623 |
|
624 |
exit:
|
625 |
hw->rpos = rpos; |
626 |
return decr;
|
627 |
} |
628 |
|
629 |
static void alsa_fini_out (HWVoiceOut *hw) |
630 |
{ |
631 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
632 |
|
633 |
ldebug ("alsa_fini\n");
|
634 |
alsa_anal_close (&alsa->handle); |
635 |
|
636 |
if (alsa->pcm_buf) {
|
637 |
qemu_free (alsa->pcm_buf); |
638 |
alsa->pcm_buf = NULL;
|
639 |
} |
640 |
} |
641 |
|
642 |
static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as) |
643 |
{ |
644 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
645 |
struct alsa_params_req req;
|
646 |
struct alsa_params_obt obt;
|
647 |
snd_pcm_t *handle; |
648 |
struct audsettings obt_as;
|
649 |
|
650 |
req.fmt = aud_to_alsafmt (as->fmt); |
651 |
req.freq = as->freq; |
652 |
req.nchannels = as->nchannels; |
653 |
req.period_size = conf.period_size_out; |
654 |
req.buffer_size = conf.buffer_size_out; |
655 |
req.size_in_usec = conf.size_in_usec_out; |
656 |
req.override_mask = |
657 |
(conf.period_size_out_overridden ? 1 : 0) | |
658 |
(conf.buffer_size_out_overridden ? 2 : 0); |
659 |
|
660 |
if (alsa_open (0, &req, &obt, &handle)) { |
661 |
return -1; |
662 |
} |
663 |
|
664 |
obt_as.freq = obt.freq; |
665 |
obt_as.nchannels = obt.nchannels; |
666 |
obt_as.fmt = obt.fmt; |
667 |
obt_as.endianness = obt.endianness; |
668 |
|
669 |
audio_pcm_init_info (&hw->info, &obt_as); |
670 |
hw->samples = obt.samples; |
671 |
|
672 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
|
673 |
if (!alsa->pcm_buf) {
|
674 |
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
|
675 |
hw->samples, 1 << hw->info.shift);
|
676 |
alsa_anal_close (&handle); |
677 |
return -1; |
678 |
} |
679 |
|
680 |
alsa->handle = handle; |
681 |
return 0; |
682 |
} |
683 |
|
684 |
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause) |
685 |
{ |
686 |
int err;
|
687 |
|
688 |
if (pause) {
|
689 |
err = snd_pcm_drop (handle); |
690 |
if (err < 0) { |
691 |
alsa_logerr (err, "Could not stop %s\n", typ);
|
692 |
return -1; |
693 |
} |
694 |
} |
695 |
else {
|
696 |
err = snd_pcm_prepare (handle); |
697 |
if (err < 0) { |
698 |
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
|
699 |
return -1; |
700 |
} |
701 |
} |
702 |
|
703 |
return 0; |
704 |
} |
705 |
|
706 |
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) |
707 |
{ |
708 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
709 |
|
710 |
switch (cmd) {
|
711 |
case VOICE_ENABLE:
|
712 |
ldebug ("enabling voice\n");
|
713 |
return alsa_voice_ctl (alsa->handle, "playback", 0); |
714 |
|
715 |
case VOICE_DISABLE:
|
716 |
ldebug ("disabling voice\n");
|
717 |
return alsa_voice_ctl (alsa->handle, "playback", 1); |
718 |
} |
719 |
|
720 |
return -1; |
721 |
} |
722 |
|
723 |
static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as) |
724 |
{ |
725 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
726 |
struct alsa_params_req req;
|
727 |
struct alsa_params_obt obt;
|
728 |
snd_pcm_t *handle; |
729 |
struct audsettings obt_as;
|
730 |
|
731 |
req.fmt = aud_to_alsafmt (as->fmt); |
732 |
req.freq = as->freq; |
733 |
req.nchannels = as->nchannels; |
734 |
req.period_size = conf.period_size_in; |
735 |
req.buffer_size = conf.buffer_size_in; |
736 |
req.size_in_usec = conf.size_in_usec_in; |
737 |
req.override_mask = |
738 |
(conf.period_size_in_overridden ? 1 : 0) | |
739 |
(conf.buffer_size_in_overridden ? 2 : 0); |
740 |
|
741 |
if (alsa_open (1, &req, &obt, &handle)) { |
742 |
return -1; |
743 |
} |
744 |
|
745 |
obt_as.freq = obt.freq; |
746 |
obt_as.nchannels = obt.nchannels; |
747 |
obt_as.fmt = obt.fmt; |
748 |
obt_as.endianness = obt.endianness; |
749 |
|
750 |
audio_pcm_init_info (&hw->info, &obt_as); |
751 |
hw->samples = obt.samples; |
752 |
|
753 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
754 |
if (!alsa->pcm_buf) {
|
755 |
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
|
756 |
hw->samples, 1 << hw->info.shift);
|
757 |
alsa_anal_close (&handle); |
758 |
return -1; |
759 |
} |
760 |
|
761 |
alsa->handle = handle; |
762 |
return 0; |
763 |
} |
764 |
|
765 |
static void alsa_fini_in (HWVoiceIn *hw) |
766 |
{ |
767 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
768 |
|
769 |
alsa_anal_close (&alsa->handle); |
770 |
|
771 |
if (alsa->pcm_buf) {
|
772 |
qemu_free (alsa->pcm_buf); |
773 |
alsa->pcm_buf = NULL;
|
774 |
} |
775 |
} |
776 |
|
777 |
static int alsa_run_in (HWVoiceIn *hw) |
778 |
{ |
779 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
780 |
int hwshift = hw->info.shift;
|
781 |
int i;
|
782 |
int live = audio_pcm_hw_get_live_in (hw);
|
783 |
int dead = hw->samples - live;
|
784 |
int decr;
|
785 |
struct {
|
786 |
int add;
|
787 |
int len;
|
788 |
} bufs[2] = {
|
789 |
{ .add = hw->wpos, .len = 0 },
|
790 |
{ .add = 0, .len = 0 } |
791 |
}; |
792 |
snd_pcm_sframes_t avail; |
793 |
snd_pcm_uframes_t read_samples = 0;
|
794 |
|
795 |
if (!dead) {
|
796 |
return 0; |
797 |
} |
798 |
|
799 |
avail = alsa_get_avail (alsa->handle); |
800 |
if (avail < 0) { |
801 |
dolog ("Could not get number of captured frames\n");
|
802 |
return 0; |
803 |
} |
804 |
|
805 |
if (!avail) {
|
806 |
snd_pcm_state_t state; |
807 |
|
808 |
state = snd_pcm_state (alsa->handle); |
809 |
switch (state) {
|
810 |
case SND_PCM_STATE_PREPARED:
|
811 |
avail = hw->samples; |
812 |
break;
|
813 |
case SND_PCM_STATE_SUSPENDED:
|
814 |
/* stream is suspended and waiting for an application recovery */
|
815 |
if (alsa_resume (alsa->handle)) {
|
816 |
dolog ("Failed to resume suspended input stream\n");
|
817 |
return 0; |
818 |
} |
819 |
if (conf.verbose) {
|
820 |
dolog ("Resuming suspended input stream\n");
|
821 |
} |
822 |
break;
|
823 |
default:
|
824 |
if (conf.verbose) {
|
825 |
dolog ("No frames available and ALSA state is %d\n", state);
|
826 |
} |
827 |
return 0; |
828 |
} |
829 |
} |
830 |
|
831 |
decr = audio_MIN (dead, avail); |
832 |
if (!decr) {
|
833 |
return 0; |
834 |
} |
835 |
|
836 |
if (hw->wpos + decr > hw->samples) {
|
837 |
bufs[0].len = (hw->samples - hw->wpos);
|
838 |
bufs[1].len = (decr - (hw->samples - hw->wpos));
|
839 |
} |
840 |
else {
|
841 |
bufs[0].len = decr;
|
842 |
} |
843 |
|
844 |
for (i = 0; i < 2; ++i) { |
845 |
void *src;
|
846 |
struct st_sample *dst;
|
847 |
snd_pcm_sframes_t nread; |
848 |
snd_pcm_uframes_t len; |
849 |
|
850 |
len = bufs[i].len; |
851 |
|
852 |
src = advance (alsa->pcm_buf, bufs[i].add << hwshift); |
853 |
dst = hw->conv_buf + bufs[i].add; |
854 |
|
855 |
while (len) {
|
856 |
nread = snd_pcm_readi (alsa->handle, src, len); |
857 |
|
858 |
if (nread <= 0) { |
859 |
switch (nread) {
|
860 |
case 0: |
861 |
if (conf.verbose) {
|
862 |
dolog ("Failed to read %ld frames (read zero)\n", len);
|
863 |
} |
864 |
goto exit;
|
865 |
|
866 |
case -EPIPE:
|
867 |
if (alsa_recover (alsa->handle)) {
|
868 |
alsa_logerr (nread, "Failed to read %ld frames\n", len);
|
869 |
goto exit;
|
870 |
} |
871 |
if (conf.verbose) {
|
872 |
dolog ("Recovering from capture xrun\n");
|
873 |
} |
874 |
continue;
|
875 |
|
876 |
case -EAGAIN:
|
877 |
goto exit;
|
878 |
|
879 |
default:
|
880 |
alsa_logerr ( |
881 |
nread, |
882 |
"Failed to read %ld frames from %p\n",
|
883 |
len, |
884 |
src |
885 |
); |
886 |
goto exit;
|
887 |
} |
888 |
} |
889 |
|
890 |
hw->conv (dst, src, nread, &nominal_volume); |
891 |
|
892 |
src = advance (src, nread << hwshift); |
893 |
dst += nread; |
894 |
|
895 |
read_samples += nread; |
896 |
len -= nread; |
897 |
} |
898 |
} |
899 |
|
900 |
exit:
|
901 |
hw->wpos = (hw->wpos + read_samples) % hw->samples; |
902 |
return read_samples;
|
903 |
} |
904 |
|
905 |
static int alsa_read (SWVoiceIn *sw, void *buf, int size) |
906 |
{ |
907 |
return audio_pcm_sw_read (sw, buf, size);
|
908 |
} |
909 |
|
910 |
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) |
911 |
{ |
912 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
913 |
|
914 |
switch (cmd) {
|
915 |
case VOICE_ENABLE:
|
916 |
ldebug ("enabling voice\n");
|
917 |
return alsa_voice_ctl (alsa->handle, "capture", 0); |
918 |
|
919 |
case VOICE_DISABLE:
|
920 |
ldebug ("disabling voice\n");
|
921 |
return alsa_voice_ctl (alsa->handle, "capture", 1); |
922 |
} |
923 |
|
924 |
return -1; |
925 |
} |
926 |
|
927 |
static void *alsa_audio_init (void) |
928 |
{ |
929 |
return &conf;
|
930 |
} |
931 |
|
932 |
static void alsa_audio_fini (void *opaque) |
933 |
{ |
934 |
(void) opaque;
|
935 |
} |
936 |
|
937 |
static struct audio_option alsa_options[] = { |
938 |
{ |
939 |
.name = "DAC_SIZE_IN_USEC",
|
940 |
.tag = AUD_OPT_BOOL, |
941 |
.valp = &conf.size_in_usec_out, |
942 |
.descr = "DAC period/buffer size in microseconds (otherwise in frames)"
|
943 |
}, |
944 |
{ |
945 |
.name = "DAC_PERIOD_SIZE",
|
946 |
.tag = AUD_OPT_INT, |
947 |
.valp = &conf.period_size_out, |
948 |
.descr = "DAC period size (0 to go with system default)",
|
949 |
.overriddenp = &conf.period_size_out_overridden |
950 |
}, |
951 |
{ |
952 |
.name = "DAC_BUFFER_SIZE",
|
953 |
.tag = AUD_OPT_INT, |
954 |
.valp = &conf.buffer_size_out, |
955 |
.descr = "DAC buffer size (0 to go with system default)",
|
956 |
.overriddenp = &conf.buffer_size_out_overridden |
957 |
}, |
958 |
{ |
959 |
.name = "ADC_SIZE_IN_USEC",
|
960 |
.tag = AUD_OPT_BOOL, |
961 |
.valp = &conf.size_in_usec_in, |
962 |
.descr = |
963 |
"ADC period/buffer size in microseconds (otherwise in frames)"
|
964 |
}, |
965 |
{ |
966 |
.name = "ADC_PERIOD_SIZE",
|
967 |
.tag = AUD_OPT_INT, |
968 |
.valp = &conf.period_size_in, |
969 |
.descr = "ADC period size (0 to go with system default)",
|
970 |
.overriddenp = &conf.period_size_in_overridden |
971 |
}, |
972 |
{ |
973 |
.name = "ADC_BUFFER_SIZE",
|
974 |
.tag = AUD_OPT_INT, |
975 |
.valp = &conf.buffer_size_in, |
976 |
.descr = "ADC buffer size (0 to go with system default)",
|
977 |
.overriddenp = &conf.buffer_size_in_overridden |
978 |
}, |
979 |
{ |
980 |
.name = "THRESHOLD",
|
981 |
.tag = AUD_OPT_INT, |
982 |
.valp = &conf.threshold, |
983 |
.descr = "(undocumented)"
|
984 |
}, |
985 |
{ |
986 |
.name = "DAC_DEV",
|
987 |
.tag = AUD_OPT_STR, |
988 |
.valp = &conf.pcm_name_out, |
989 |
.descr = "DAC device name (for instance dmix)"
|
990 |
}, |
991 |
{ |
992 |
.name = "ADC_DEV",
|
993 |
.tag = AUD_OPT_STR, |
994 |
.valp = &conf.pcm_name_in, |
995 |
.descr = "ADC device name"
|
996 |
}, |
997 |
{ |
998 |
.name = "VERBOSE",
|
999 |
.tag = AUD_OPT_BOOL, |
1000 |
.valp = &conf.verbose, |
1001 |
.descr = "Behave in a more verbose way"
|
1002 |
}, |
1003 |
{ /* End of list */ }
|
1004 |
}; |
1005 |
|
1006 |
static struct audio_pcm_ops alsa_pcm_ops = { |
1007 |
.init_out = alsa_init_out, |
1008 |
.fini_out = alsa_fini_out, |
1009 |
.run_out = alsa_run_out, |
1010 |
.write = alsa_write, |
1011 |
.ctl_out = alsa_ctl_out, |
1012 |
|
1013 |
.init_in = alsa_init_in, |
1014 |
.fini_in = alsa_fini_in, |
1015 |
.run_in = alsa_run_in, |
1016 |
.read = alsa_read, |
1017 |
.ctl_in = alsa_ctl_in |
1018 |
}; |
1019 |
|
1020 |
struct audio_driver alsa_audio_driver = {
|
1021 |
.name = "alsa",
|
1022 |
.descr = "ALSA http://www.alsa-project.org",
|
1023 |
.options = alsa_options, |
1024 |
.init = alsa_audio_init, |
1025 |
.fini = alsa_audio_fini, |
1026 |
.pcm_ops = &alsa_pcm_ops, |
1027 |
.can_be_default = 1,
|
1028 |
.max_voices_out = INT_MAX, |
1029 |
.max_voices_in = INT_MAX, |
1030 |
.voice_size_out = sizeof (ALSAVoiceOut),
|
1031 |
.voice_size_in = sizeof (ALSAVoiceIn)
|
1032 |
}; |