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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 "qemu-char.h" |
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#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" |
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#endif
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|
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#define AUDIO_CAP "alsa" |
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#include "audio_int.h" |
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|
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struct pollhlp {
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snd_pcm_t *handle; |
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struct pollfd *pfds;
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int count;
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int mask;
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}; |
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|
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typedef struct ALSAVoiceOut { |
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HWVoiceOut hw; |
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int wpos;
|
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int pending;
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void *pcm_buf;
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snd_pcm_t *handle; |
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struct pollhlp pollhlp;
|
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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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struct pollhlp pollhlp;
|
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} ALSAVoiceIn; |
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|
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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; |
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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; |
69 |
|
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int buffer_size_in_overridden;
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int period_size_in_overridden;
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|
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int buffer_size_out_overridden;
|
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int period_size_out_overridden;
|
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int verbose;
|
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} conf = { |
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.buffer_size_out = 4096,
|
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.period_size_out = 1024,
|
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.pcm_name_out = "default",
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.pcm_name_in = "default",
|
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}; |
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|
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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; |
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}; |
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|
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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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}; |
100 |
|
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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; |
104 |
|
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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); |
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|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
111 |
|
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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, |
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... |
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) |
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{ |
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va_list ap; |
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|
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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); |
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|
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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
|
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} |
129 |
|
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static void alsa_fini_poll (struct pollhlp *hlp) |
131 |
{ |
132 |
int i;
|
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struct pollfd *pfds = hlp->pfds;
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|
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if (pfds) {
|
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for (i = 0; i < hlp->count; ++i) { |
137 |
qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); |
138 |
} |
139 |
qemu_free (pfds); |
140 |
} |
141 |
hlp->pfds = NULL;
|
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hlp->count = 0;
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hlp->handle = NULL;
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} |
145 |
|
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static void alsa_anal_close1 (snd_pcm_t **handlep) |
147 |
{ |
148 |
int err = snd_pcm_close (*handlep);
|
149 |
if (err) {
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alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
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} |
152 |
*handlep = NULL;
|
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} |
154 |
|
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static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp) |
156 |
{ |
157 |
alsa_fini_poll (hlp); |
158 |
alsa_anal_close1 (handlep); |
159 |
} |
160 |
|
161 |
static int alsa_recover (snd_pcm_t *handle) |
162 |
{ |
163 |
int err = snd_pcm_prepare (handle);
|
164 |
if (err < 0) { |
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alsa_logerr (err, "Failed to prepare handle %p\n", handle);
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return -1; |
167 |
} |
168 |
return 0; |
169 |
} |
170 |
|
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static int alsa_resume (snd_pcm_t *handle) |
172 |
{ |
173 |
int err = snd_pcm_resume (handle);
|
174 |
if (err < 0) { |
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alsa_logerr (err, "Failed to resume handle %p\n", handle);
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return -1; |
177 |
} |
178 |
return 0; |
179 |
} |
180 |
|
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static void alsa_poll_handler (void *opaque) |
182 |
{ |
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int err, count;
|
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snd_pcm_state_t state; |
185 |
struct pollhlp *hlp = opaque;
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unsigned short revents; |
187 |
|
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count = poll (hlp->pfds, hlp->count, 0);
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189 |
if (count < 0) { |
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dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
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return;
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} |
193 |
|
194 |
if (!count) {
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return;
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} |
197 |
|
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/* XXX: ALSA example uses initial count, not the one returned by
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poll, correct? */
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err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds, |
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hlp->count, &revents); |
202 |
if (err < 0) { |
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alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
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return;
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} |
206 |
|
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if (!(revents & hlp->mask)) {
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if (conf.verbose) {
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dolog ("revents = %d\n", revents);
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} |
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return;
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} |
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state = snd_pcm_state (hlp->handle); |
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switch (state) {
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case SND_PCM_STATE_XRUN:
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alsa_recover (hlp->handle); |
218 |
break;
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|
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case SND_PCM_STATE_SUSPENDED:
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alsa_resume (hlp->handle); |
222 |
break;
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|
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case SND_PCM_STATE_PREPARED:
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audio_run ("alsa run (prepared)");
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break;
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|
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case SND_PCM_STATE_RUNNING:
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audio_run ("alsa run (running)");
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break;
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|
232 |
default:
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dolog ("Unexpected state %d\n", state);
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} |
235 |
} |
236 |
|
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static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask) |
238 |
{ |
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int i, count, err;
|
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struct pollfd *pfds;
|
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|
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count = snd_pcm_poll_descriptors_count (handle); |
243 |
if (count <= 0) { |
244 |
dolog ("Could not initialize poll mode\n"
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"Invalid number of poll descriptors %d\n", count);
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return -1; |
247 |
} |
248 |
|
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pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds)); |
250 |
if (!pfds) {
|
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dolog ("Could not initialize poll mode\n");
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return -1; |
253 |
} |
254 |
|
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err = snd_pcm_poll_descriptors (handle, pfds, count); |
256 |
if (err < 0) { |
257 |
alsa_logerr (err, "Could not initialize poll mode\n"
|
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"Could not obtain poll descriptors\n");
|
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qemu_free (pfds); |
260 |
return -1; |
261 |
} |
262 |
|
263 |
for (i = 0; i < count; ++i) { |
264 |
if (pfds[i].events & POLLIN) {
|
265 |
err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, |
266 |
NULL, hlp);
|
267 |
} |
268 |
if (pfds[i].events & POLLOUT) {
|
269 |
if (conf.verbose) {
|
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dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
|
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} |
272 |
err = qemu_set_fd_handler (pfds[i].fd, NULL,
|
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alsa_poll_handler, hlp); |
274 |
} |
275 |
if (conf.verbose) {
|
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dolog ("Set handler events=%#x index=%d fd=%d err=%d\n",
|
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pfds[i].events, i, pfds[i].fd, err); |
278 |
} |
279 |
|
280 |
if (err) {
|
281 |
dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n",
|
282 |
pfds[i].events, i, pfds[i].fd, err); |
283 |
|
284 |
while (i--) {
|
285 |
qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); |
286 |
} |
287 |
qemu_free (pfds); |
288 |
return -1; |
289 |
} |
290 |
} |
291 |
hlp->pfds = pfds; |
292 |
hlp->count = count; |
293 |
hlp->handle = handle; |
294 |
hlp->mask = mask; |
295 |
return 0; |
296 |
} |
297 |
|
298 |
static int alsa_poll_out (HWVoiceOut *hw) |
299 |
{ |
300 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
301 |
|
302 |
return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
|
303 |
} |
304 |
|
305 |
static int alsa_poll_in (HWVoiceIn *hw) |
306 |
{ |
307 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
308 |
|
309 |
return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
|
310 |
} |
311 |
|
312 |
static int alsa_write (SWVoiceOut *sw, void *buf, int len) |
313 |
{ |
314 |
return audio_pcm_sw_write (sw, buf, len);
|
315 |
} |
316 |
|
317 |
static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
|
318 |
{ |
319 |
switch (fmt) {
|
320 |
case AUD_FMT_S8:
|
321 |
return SND_PCM_FORMAT_S8;
|
322 |
|
323 |
case AUD_FMT_U8:
|
324 |
return SND_PCM_FORMAT_U8;
|
325 |
|
326 |
case AUD_FMT_S16:
|
327 |
return SND_PCM_FORMAT_S16_LE;
|
328 |
|
329 |
case AUD_FMT_U16:
|
330 |
return SND_PCM_FORMAT_U16_LE;
|
331 |
|
332 |
case AUD_FMT_S32:
|
333 |
return SND_PCM_FORMAT_S32_LE;
|
334 |
|
335 |
case AUD_FMT_U32:
|
336 |
return SND_PCM_FORMAT_U32_LE;
|
337 |
|
338 |
default:
|
339 |
dolog ("Internal logic error: Bad audio format %d\n", fmt);
|
340 |
#ifdef DEBUG_AUDIO
|
341 |
abort (); |
342 |
#endif
|
343 |
return SND_PCM_FORMAT_U8;
|
344 |
} |
345 |
} |
346 |
|
347 |
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, |
348 |
int *endianness)
|
349 |
{ |
350 |
switch (alsafmt) {
|
351 |
case SND_PCM_FORMAT_S8:
|
352 |
*endianness = 0;
|
353 |
*fmt = AUD_FMT_S8; |
354 |
break;
|
355 |
|
356 |
case SND_PCM_FORMAT_U8:
|
357 |
*endianness = 0;
|
358 |
*fmt = AUD_FMT_U8; |
359 |
break;
|
360 |
|
361 |
case SND_PCM_FORMAT_S16_LE:
|
362 |
*endianness = 0;
|
363 |
*fmt = AUD_FMT_S16; |
364 |
break;
|
365 |
|
366 |
case SND_PCM_FORMAT_U16_LE:
|
367 |
*endianness = 0;
|
368 |
*fmt = AUD_FMT_U16; |
369 |
break;
|
370 |
|
371 |
case SND_PCM_FORMAT_S16_BE:
|
372 |
*endianness = 1;
|
373 |
*fmt = AUD_FMT_S16; |
374 |
break;
|
375 |
|
376 |
case SND_PCM_FORMAT_U16_BE:
|
377 |
*endianness = 1;
|
378 |
*fmt = AUD_FMT_U16; |
379 |
break;
|
380 |
|
381 |
case SND_PCM_FORMAT_S32_LE:
|
382 |
*endianness = 0;
|
383 |
*fmt = AUD_FMT_S32; |
384 |
break;
|
385 |
|
386 |
case SND_PCM_FORMAT_U32_LE:
|
387 |
*endianness = 0;
|
388 |
*fmt = AUD_FMT_U32; |
389 |
break;
|
390 |
|
391 |
case SND_PCM_FORMAT_S32_BE:
|
392 |
*endianness = 1;
|
393 |
*fmt = AUD_FMT_S32; |
394 |
break;
|
395 |
|
396 |
case SND_PCM_FORMAT_U32_BE:
|
397 |
*endianness = 1;
|
398 |
*fmt = AUD_FMT_U32; |
399 |
break;
|
400 |
|
401 |
default:
|
402 |
dolog ("Unrecognized audio format %d\n", alsafmt);
|
403 |
return -1; |
404 |
} |
405 |
|
406 |
return 0; |
407 |
} |
408 |
|
409 |
static void alsa_dump_info (struct alsa_params_req *req, |
410 |
struct alsa_params_obt *obt)
|
411 |
{ |
412 |
dolog ("parameter | requested value | obtained value\n");
|
413 |
dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
|
414 |
dolog ("channels | %10d | %10d\n",
|
415 |
req->nchannels, obt->nchannels); |
416 |
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
|
417 |
dolog ("============================================\n");
|
418 |
dolog ("requested: buffer size %d period size %d\n",
|
419 |
req->buffer_size, req->period_size); |
420 |
dolog ("obtained: samples %ld\n", obt->samples);
|
421 |
} |
422 |
|
423 |
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) |
424 |
{ |
425 |
int err;
|
426 |
snd_pcm_sw_params_t *sw_params; |
427 |
|
428 |
snd_pcm_sw_params_alloca (&sw_params); |
429 |
|
430 |
err = snd_pcm_sw_params_current (handle, sw_params); |
431 |
if (err < 0) { |
432 |
dolog ("Could not fully initialize DAC\n");
|
433 |
alsa_logerr (err, "Failed to get current software parameters\n");
|
434 |
return;
|
435 |
} |
436 |
|
437 |
err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); |
438 |
if (err < 0) { |
439 |
dolog ("Could not fully initialize DAC\n");
|
440 |
alsa_logerr (err, "Failed to set software threshold to %ld\n",
|
441 |
threshold); |
442 |
return;
|
443 |
} |
444 |
|
445 |
err = snd_pcm_sw_params (handle, sw_params); |
446 |
if (err < 0) { |
447 |
dolog ("Could not fully initialize DAC\n");
|
448 |
alsa_logerr (err, "Failed to set software parameters\n");
|
449 |
return;
|
450 |
} |
451 |
} |
452 |
|
453 |
static int alsa_open (int in, struct alsa_params_req *req, |
454 |
struct alsa_params_obt *obt, snd_pcm_t **handlep)
|
455 |
{ |
456 |
snd_pcm_t *handle; |
457 |
snd_pcm_hw_params_t *hw_params; |
458 |
int err;
|
459 |
int size_in_usec;
|
460 |
unsigned int freq, nchannels; |
461 |
const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; |
462 |
snd_pcm_uframes_t obt_buffer_size; |
463 |
const char *typ = in ? "ADC" : "DAC"; |
464 |
snd_pcm_format_t obtfmt; |
465 |
|
466 |
freq = req->freq; |
467 |
nchannels = req->nchannels; |
468 |
size_in_usec = req->size_in_usec; |
469 |
|
470 |
snd_pcm_hw_params_alloca (&hw_params); |
471 |
|
472 |
err = snd_pcm_open ( |
473 |
&handle, |
474 |
pcm_name, |
475 |
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, |
476 |
SND_PCM_NONBLOCK |
477 |
); |
478 |
if (err < 0) { |
479 |
alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
|
480 |
return -1; |
481 |
} |
482 |
|
483 |
err = snd_pcm_hw_params_any (handle, hw_params); |
484 |
if (err < 0) { |
485 |
alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
|
486 |
goto err;
|
487 |
} |
488 |
|
489 |
err = snd_pcm_hw_params_set_access ( |
490 |
handle, |
491 |
hw_params, |
492 |
SND_PCM_ACCESS_RW_INTERLEAVED |
493 |
); |
494 |
if (err < 0) { |
495 |
alsa_logerr2 (err, typ, "Failed to set access type\n");
|
496 |
goto err;
|
497 |
} |
498 |
|
499 |
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); |
500 |
if (err < 0 && conf.verbose) { |
501 |
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
|
502 |
} |
503 |
|
504 |
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
|
505 |
if (err < 0) { |
506 |
alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
|
507 |
goto err;
|
508 |
} |
509 |
|
510 |
err = snd_pcm_hw_params_set_channels_near ( |
511 |
handle, |
512 |
hw_params, |
513 |
&nchannels |
514 |
); |
515 |
if (err < 0) { |
516 |
alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
|
517 |
req->nchannels); |
518 |
goto err;
|
519 |
} |
520 |
|
521 |
if (nchannels != 1 && nchannels != 2) { |
522 |
alsa_logerr2 (err, typ, |
523 |
"Can not handle obtained number of channels %d\n",
|
524 |
nchannels); |
525 |
goto err;
|
526 |
} |
527 |
|
528 |
if (req->buffer_size) {
|
529 |
unsigned long obt; |
530 |
|
531 |
if (size_in_usec) {
|
532 |
int dir = 0; |
533 |
unsigned int btime = req->buffer_size; |
534 |
|
535 |
err = snd_pcm_hw_params_set_buffer_time_near ( |
536 |
handle, |
537 |
hw_params, |
538 |
&btime, |
539 |
&dir |
540 |
); |
541 |
obt = btime; |
542 |
} |
543 |
else {
|
544 |
snd_pcm_uframes_t bsize = req->buffer_size; |
545 |
|
546 |
err = snd_pcm_hw_params_set_buffer_size_near ( |
547 |
handle, |
548 |
hw_params, |
549 |
&bsize |
550 |
); |
551 |
obt = bsize; |
552 |
} |
553 |
if (err < 0) { |
554 |
alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
|
555 |
size_in_usec ? "time" : "size", req->buffer_size); |
556 |
goto err;
|
557 |
} |
558 |
|
559 |
if ((req->override_mask & 2) && (obt - req->buffer_size)) |
560 |
dolog ("Requested buffer %s %u was rejected, using %lu\n",
|
561 |
size_in_usec ? "time" : "size", req->buffer_size, obt); |
562 |
} |
563 |
|
564 |
if (req->period_size) {
|
565 |
unsigned long obt; |
566 |
|
567 |
if (size_in_usec) {
|
568 |
int dir = 0; |
569 |
unsigned int ptime = req->period_size; |
570 |
|
571 |
err = snd_pcm_hw_params_set_period_time_near ( |
572 |
handle, |
573 |
hw_params, |
574 |
&ptime, |
575 |
&dir |
576 |
); |
577 |
obt = ptime; |
578 |
} |
579 |
else {
|
580 |
int dir = 0; |
581 |
snd_pcm_uframes_t psize = req->period_size; |
582 |
|
583 |
err = snd_pcm_hw_params_set_period_size_near ( |
584 |
handle, |
585 |
hw_params, |
586 |
&psize, |
587 |
&dir |
588 |
); |
589 |
obt = psize; |
590 |
} |
591 |
|
592 |
if (err < 0) { |
593 |
alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
|
594 |
size_in_usec ? "time" : "size", req->period_size); |
595 |
goto err;
|
596 |
} |
597 |
|
598 |
if (((req->override_mask & 1) && (obt - req->period_size))) |
599 |
dolog ("Requested period %s %u was rejected, using %lu\n",
|
600 |
size_in_usec ? "time" : "size", req->period_size, obt); |
601 |
} |
602 |
|
603 |
err = snd_pcm_hw_params (handle, hw_params); |
604 |
if (err < 0) { |
605 |
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
|
606 |
goto err;
|
607 |
} |
608 |
|
609 |
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); |
610 |
if (err < 0) { |
611 |
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
|
612 |
goto err;
|
613 |
} |
614 |
|
615 |
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); |
616 |
if (err < 0) { |
617 |
alsa_logerr2 (err, typ, "Failed to get format\n");
|
618 |
goto err;
|
619 |
} |
620 |
|
621 |
if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
|
622 |
dolog ("Invalid format was returned %d\n", obtfmt);
|
623 |
goto err;
|
624 |
} |
625 |
|
626 |
err = snd_pcm_prepare (handle); |
627 |
if (err < 0) { |
628 |
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
|
629 |
goto err;
|
630 |
} |
631 |
|
632 |
if (!in && conf.threshold) {
|
633 |
snd_pcm_uframes_t threshold; |
634 |
int bytes_per_sec;
|
635 |
|
636 |
bytes_per_sec = freq << (nchannels == 2);
|
637 |
|
638 |
switch (obt->fmt) {
|
639 |
case AUD_FMT_S8:
|
640 |
case AUD_FMT_U8:
|
641 |
break;
|
642 |
|
643 |
case AUD_FMT_S16:
|
644 |
case AUD_FMT_U16:
|
645 |
bytes_per_sec <<= 1;
|
646 |
break;
|
647 |
|
648 |
case AUD_FMT_S32:
|
649 |
case AUD_FMT_U32:
|
650 |
bytes_per_sec <<= 2;
|
651 |
break;
|
652 |
} |
653 |
|
654 |
threshold = (conf.threshold * bytes_per_sec) / 1000;
|
655 |
alsa_set_threshold (handle, threshold); |
656 |
} |
657 |
|
658 |
obt->nchannels = nchannels; |
659 |
obt->freq = freq; |
660 |
obt->samples = obt_buffer_size; |
661 |
|
662 |
*handlep = handle; |
663 |
|
664 |
if (conf.verbose &&
|
665 |
(obt->fmt != req->fmt || |
666 |
obt->nchannels != req->nchannels || |
667 |
obt->freq != req->freq)) { |
668 |
dolog ("Audio paramters for %s\n", typ);
|
669 |
alsa_dump_info (req, obt); |
670 |
} |
671 |
|
672 |
#ifdef DEBUG
|
673 |
alsa_dump_info (req, obt); |
674 |
#endif
|
675 |
return 0; |
676 |
|
677 |
err:
|
678 |
alsa_anal_close1 (&handle); |
679 |
return -1; |
680 |
} |
681 |
|
682 |
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
|
683 |
{ |
684 |
snd_pcm_sframes_t avail; |
685 |
|
686 |
avail = snd_pcm_avail_update (handle); |
687 |
if (avail < 0) { |
688 |
if (avail == -EPIPE) {
|
689 |
if (!alsa_recover (handle)) {
|
690 |
avail = snd_pcm_avail_update (handle); |
691 |
} |
692 |
} |
693 |
|
694 |
if (avail < 0) { |
695 |
alsa_logerr (avail, |
696 |
"Could not obtain number of available frames\n");
|
697 |
return -1; |
698 |
} |
699 |
} |
700 |
|
701 |
return avail;
|
702 |
} |
703 |
|
704 |
static void alsa_write_pending (ALSAVoiceOut *alsa) |
705 |
{ |
706 |
HWVoiceOut *hw = &alsa->hw; |
707 |
|
708 |
while (alsa->pending) {
|
709 |
int left_till_end_samples = hw->samples - alsa->wpos;
|
710 |
int len = audio_MIN (alsa->pending, left_till_end_samples);
|
711 |
char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
|
712 |
|
713 |
while (len) {
|
714 |
snd_pcm_sframes_t written; |
715 |
|
716 |
written = snd_pcm_writei (alsa->handle, src, len); |
717 |
|
718 |
if (written <= 0) { |
719 |
switch (written) {
|
720 |
case 0: |
721 |
if (conf.verbose) {
|
722 |
dolog ("Failed to write %d frames (wrote zero)\n", len);
|
723 |
} |
724 |
return;
|
725 |
|
726 |
case -EPIPE:
|
727 |
if (alsa_recover (alsa->handle)) {
|
728 |
alsa_logerr (written, "Failed to write %d frames\n",
|
729 |
len); |
730 |
return;
|
731 |
} |
732 |
if (conf.verbose) {
|
733 |
dolog ("Recovering from playback xrun\n");
|
734 |
} |
735 |
continue;
|
736 |
|
737 |
case -ESTRPIPE:
|
738 |
/* stream is suspended and waiting for an
|
739 |
application recovery */
|
740 |
if (alsa_resume (alsa->handle)) {
|
741 |
alsa_logerr (written, "Failed to write %d frames\n",
|
742 |
len); |
743 |
return;
|
744 |
} |
745 |
if (conf.verbose) {
|
746 |
dolog ("Resuming suspended output stream\n");
|
747 |
} |
748 |
continue;
|
749 |
|
750 |
case -EAGAIN:
|
751 |
return;
|
752 |
|
753 |
default:
|
754 |
alsa_logerr (written, "Failed to write %d frames from %p\n",
|
755 |
len, src); |
756 |
return;
|
757 |
} |
758 |
} |
759 |
|
760 |
alsa->wpos = (alsa->wpos + written) % hw->samples; |
761 |
alsa->pending -= written; |
762 |
len -= written; |
763 |
} |
764 |
} |
765 |
} |
766 |
|
767 |
static int alsa_run_out (HWVoiceOut *hw, int live) |
768 |
{ |
769 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
770 |
int decr;
|
771 |
snd_pcm_sframes_t avail; |
772 |
|
773 |
avail = alsa_get_avail (alsa->handle); |
774 |
if (avail < 0) { |
775 |
dolog ("Could not get number of available playback frames\n");
|
776 |
return 0; |
777 |
} |
778 |
|
779 |
decr = audio_MIN (live, avail); |
780 |
decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending); |
781 |
alsa->pending += decr; |
782 |
alsa_write_pending (alsa); |
783 |
return decr;
|
784 |
} |
785 |
|
786 |
static void alsa_fini_out (HWVoiceOut *hw) |
787 |
{ |
788 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
789 |
|
790 |
ldebug ("alsa_fini\n");
|
791 |
alsa_anal_close (&alsa->handle, &alsa->pollhlp); |
792 |
|
793 |
if (alsa->pcm_buf) {
|
794 |
qemu_free (alsa->pcm_buf); |
795 |
alsa->pcm_buf = NULL;
|
796 |
} |
797 |
} |
798 |
|
799 |
static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as) |
800 |
{ |
801 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
802 |
struct alsa_params_req req;
|
803 |
struct alsa_params_obt obt;
|
804 |
snd_pcm_t *handle; |
805 |
struct audsettings obt_as;
|
806 |
|
807 |
req.fmt = aud_to_alsafmt (as->fmt); |
808 |
req.freq = as->freq; |
809 |
req.nchannels = as->nchannels; |
810 |
req.period_size = conf.period_size_out; |
811 |
req.buffer_size = conf.buffer_size_out; |
812 |
req.size_in_usec = conf.size_in_usec_out; |
813 |
req.override_mask = |
814 |
(conf.period_size_out_overridden ? 1 : 0) | |
815 |
(conf.buffer_size_out_overridden ? 2 : 0); |
816 |
|
817 |
if (alsa_open (0, &req, &obt, &handle)) { |
818 |
return -1; |
819 |
} |
820 |
|
821 |
obt_as.freq = obt.freq; |
822 |
obt_as.nchannels = obt.nchannels; |
823 |
obt_as.fmt = obt.fmt; |
824 |
obt_as.endianness = obt.endianness; |
825 |
|
826 |
audio_pcm_init_info (&hw->info, &obt_as); |
827 |
hw->samples = obt.samples; |
828 |
|
829 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
|
830 |
if (!alsa->pcm_buf) {
|
831 |
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
|
832 |
hw->samples, 1 << hw->info.shift);
|
833 |
alsa_anal_close1 (&handle); |
834 |
return -1; |
835 |
} |
836 |
|
837 |
alsa->handle = handle; |
838 |
return 0; |
839 |
} |
840 |
|
841 |
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause) |
842 |
{ |
843 |
int err;
|
844 |
|
845 |
if (pause) {
|
846 |
err = snd_pcm_drop (handle); |
847 |
if (err < 0) { |
848 |
alsa_logerr (err, "Could not stop %s\n", typ);
|
849 |
return -1; |
850 |
} |
851 |
} |
852 |
else {
|
853 |
err = snd_pcm_prepare (handle); |
854 |
if (err < 0) { |
855 |
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
|
856 |
return -1; |
857 |
} |
858 |
} |
859 |
|
860 |
return 0; |
861 |
} |
862 |
|
863 |
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) |
864 |
{ |
865 |
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
866 |
|
867 |
switch (cmd) {
|
868 |
case VOICE_ENABLE:
|
869 |
{ |
870 |
va_list ap; |
871 |
int poll_mode;
|
872 |
|
873 |
va_start (ap, cmd); |
874 |
poll_mode = va_arg (ap, int);
|
875 |
va_end (ap); |
876 |
|
877 |
ldebug ("enabling voice\n");
|
878 |
if (poll_mode && alsa_poll_out (hw)) {
|
879 |
poll_mode = 0;
|
880 |
} |
881 |
hw->poll_mode = poll_mode; |
882 |
return alsa_voice_ctl (alsa->handle, "playback", 0); |
883 |
} |
884 |
|
885 |
case VOICE_DISABLE:
|
886 |
ldebug ("disabling voice\n");
|
887 |
return alsa_voice_ctl (alsa->handle, "playback", 1); |
888 |
} |
889 |
|
890 |
return -1; |
891 |
} |
892 |
|
893 |
static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as) |
894 |
{ |
895 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
896 |
struct alsa_params_req req;
|
897 |
struct alsa_params_obt obt;
|
898 |
snd_pcm_t *handle; |
899 |
struct audsettings obt_as;
|
900 |
|
901 |
req.fmt = aud_to_alsafmt (as->fmt); |
902 |
req.freq = as->freq; |
903 |
req.nchannels = as->nchannels; |
904 |
req.period_size = conf.period_size_in; |
905 |
req.buffer_size = conf.buffer_size_in; |
906 |
req.size_in_usec = conf.size_in_usec_in; |
907 |
req.override_mask = |
908 |
(conf.period_size_in_overridden ? 1 : 0) | |
909 |
(conf.buffer_size_in_overridden ? 2 : 0); |
910 |
|
911 |
if (alsa_open (1, &req, &obt, &handle)) { |
912 |
return -1; |
913 |
} |
914 |
|
915 |
obt_as.freq = obt.freq; |
916 |
obt_as.nchannels = obt.nchannels; |
917 |
obt_as.fmt = obt.fmt; |
918 |
obt_as.endianness = obt.endianness; |
919 |
|
920 |
audio_pcm_init_info (&hw->info, &obt_as); |
921 |
hw->samples = obt.samples; |
922 |
|
923 |
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
924 |
if (!alsa->pcm_buf) {
|
925 |
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
|
926 |
hw->samples, 1 << hw->info.shift);
|
927 |
alsa_anal_close1 (&handle); |
928 |
return -1; |
929 |
} |
930 |
|
931 |
alsa->handle = handle; |
932 |
return 0; |
933 |
} |
934 |
|
935 |
static void alsa_fini_in (HWVoiceIn *hw) |
936 |
{ |
937 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
938 |
|
939 |
alsa_anal_close (&alsa->handle, &alsa->pollhlp); |
940 |
|
941 |
if (alsa->pcm_buf) {
|
942 |
qemu_free (alsa->pcm_buf); |
943 |
alsa->pcm_buf = NULL;
|
944 |
} |
945 |
} |
946 |
|
947 |
static int alsa_run_in (HWVoiceIn *hw) |
948 |
{ |
949 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
950 |
int hwshift = hw->info.shift;
|
951 |
int i;
|
952 |
int live = audio_pcm_hw_get_live_in (hw);
|
953 |
int dead = hw->samples - live;
|
954 |
int decr;
|
955 |
struct {
|
956 |
int add;
|
957 |
int len;
|
958 |
} bufs[2] = {
|
959 |
{ .add = hw->wpos, .len = 0 },
|
960 |
{ .add = 0, .len = 0 } |
961 |
}; |
962 |
snd_pcm_sframes_t avail; |
963 |
snd_pcm_uframes_t read_samples = 0;
|
964 |
|
965 |
if (!dead) {
|
966 |
return 0; |
967 |
} |
968 |
|
969 |
avail = alsa_get_avail (alsa->handle); |
970 |
if (avail < 0) { |
971 |
dolog ("Could not get number of captured frames\n");
|
972 |
return 0; |
973 |
} |
974 |
|
975 |
if (!avail) {
|
976 |
snd_pcm_state_t state; |
977 |
|
978 |
state = snd_pcm_state (alsa->handle); |
979 |
switch (state) {
|
980 |
case SND_PCM_STATE_PREPARED:
|
981 |
avail = hw->samples; |
982 |
break;
|
983 |
case SND_PCM_STATE_SUSPENDED:
|
984 |
/* stream is suspended and waiting for an application recovery */
|
985 |
if (alsa_resume (alsa->handle)) {
|
986 |
dolog ("Failed to resume suspended input stream\n");
|
987 |
return 0; |
988 |
} |
989 |
if (conf.verbose) {
|
990 |
dolog ("Resuming suspended input stream\n");
|
991 |
} |
992 |
break;
|
993 |
default:
|
994 |
if (conf.verbose) {
|
995 |
dolog ("No frames available and ALSA state is %d\n", state);
|
996 |
} |
997 |
return 0; |
998 |
} |
999 |
} |
1000 |
|
1001 |
decr = audio_MIN (dead, avail); |
1002 |
if (!decr) {
|
1003 |
return 0; |
1004 |
} |
1005 |
|
1006 |
if (hw->wpos + decr > hw->samples) {
|
1007 |
bufs[0].len = (hw->samples - hw->wpos);
|
1008 |
bufs[1].len = (decr - (hw->samples - hw->wpos));
|
1009 |
} |
1010 |
else {
|
1011 |
bufs[0].len = decr;
|
1012 |
} |
1013 |
|
1014 |
for (i = 0; i < 2; ++i) { |
1015 |
void *src;
|
1016 |
struct st_sample *dst;
|
1017 |
snd_pcm_sframes_t nread; |
1018 |
snd_pcm_uframes_t len; |
1019 |
|
1020 |
len = bufs[i].len; |
1021 |
|
1022 |
src = advance (alsa->pcm_buf, bufs[i].add << hwshift); |
1023 |
dst = hw->conv_buf + bufs[i].add; |
1024 |
|
1025 |
while (len) {
|
1026 |
nread = snd_pcm_readi (alsa->handle, src, len); |
1027 |
|
1028 |
if (nread <= 0) { |
1029 |
switch (nread) {
|
1030 |
case 0: |
1031 |
if (conf.verbose) {
|
1032 |
dolog ("Failed to read %ld frames (read zero)\n", len);
|
1033 |
} |
1034 |
goto exit;
|
1035 |
|
1036 |
case -EPIPE:
|
1037 |
if (alsa_recover (alsa->handle)) {
|
1038 |
alsa_logerr (nread, "Failed to read %ld frames\n", len);
|
1039 |
goto exit;
|
1040 |
} |
1041 |
if (conf.verbose) {
|
1042 |
dolog ("Recovering from capture xrun\n");
|
1043 |
} |
1044 |
continue;
|
1045 |
|
1046 |
case -EAGAIN:
|
1047 |
goto exit;
|
1048 |
|
1049 |
default:
|
1050 |
alsa_logerr ( |
1051 |
nread, |
1052 |
"Failed to read %ld frames from %p\n",
|
1053 |
len, |
1054 |
src |
1055 |
); |
1056 |
goto exit;
|
1057 |
} |
1058 |
} |
1059 |
|
1060 |
hw->conv (dst, src, nread, &nominal_volume); |
1061 |
|
1062 |
src = advance (src, nread << hwshift); |
1063 |
dst += nread; |
1064 |
|
1065 |
read_samples += nread; |
1066 |
len -= nread; |
1067 |
} |
1068 |
} |
1069 |
|
1070 |
exit:
|
1071 |
hw->wpos = (hw->wpos + read_samples) % hw->samples; |
1072 |
return read_samples;
|
1073 |
} |
1074 |
|
1075 |
static int alsa_read (SWVoiceIn *sw, void *buf, int size) |
1076 |
{ |
1077 |
return audio_pcm_sw_read (sw, buf, size);
|
1078 |
} |
1079 |
|
1080 |
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) |
1081 |
{ |
1082 |
va_list ap; |
1083 |
int poll_mode;
|
1084 |
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
1085 |
|
1086 |
va_start (ap, cmd); |
1087 |
poll_mode = va_arg (ap, int);
|
1088 |
va_end (ap); |
1089 |
|
1090 |
switch (cmd) {
|
1091 |
case VOICE_ENABLE:
|
1092 |
ldebug ("enabling voice\n");
|
1093 |
if (poll_mode && alsa_poll_in (hw)) {
|
1094 |
poll_mode = 0;
|
1095 |
} |
1096 |
hw->poll_mode = poll_mode; |
1097 |
|
1098 |
return alsa_voice_ctl (alsa->handle, "capture", 0); |
1099 |
|
1100 |
case VOICE_DISABLE:
|
1101 |
ldebug ("disabling voice\n");
|
1102 |
if (hw->poll_mode) {
|
1103 |
hw->poll_mode = 0;
|
1104 |
alsa_fini_poll (&alsa->pollhlp); |
1105 |
} |
1106 |
return alsa_voice_ctl (alsa->handle, "capture", 1); |
1107 |
} |
1108 |
|
1109 |
return -1; |
1110 |
} |
1111 |
|
1112 |
static void *alsa_audio_init (void) |
1113 |
{ |
1114 |
return &conf;
|
1115 |
} |
1116 |
|
1117 |
static void alsa_audio_fini (void *opaque) |
1118 |
{ |
1119 |
(void) opaque;
|
1120 |
} |
1121 |
|
1122 |
static struct audio_option alsa_options[] = { |
1123 |
{ |
1124 |
.name = "DAC_SIZE_IN_USEC",
|
1125 |
.tag = AUD_OPT_BOOL, |
1126 |
.valp = &conf.size_in_usec_out, |
1127 |
.descr = "DAC period/buffer size in microseconds (otherwise in frames)"
|
1128 |
}, |
1129 |
{ |
1130 |
.name = "DAC_PERIOD_SIZE",
|
1131 |
.tag = AUD_OPT_INT, |
1132 |
.valp = &conf.period_size_out, |
1133 |
.descr = "DAC period size (0 to go with system default)",
|
1134 |
.overriddenp = &conf.period_size_out_overridden |
1135 |
}, |
1136 |
{ |
1137 |
.name = "DAC_BUFFER_SIZE",
|
1138 |
.tag = AUD_OPT_INT, |
1139 |
.valp = &conf.buffer_size_out, |
1140 |
.descr = "DAC buffer size (0 to go with system default)",
|
1141 |
.overriddenp = &conf.buffer_size_out_overridden |
1142 |
}, |
1143 |
{ |
1144 |
.name = "ADC_SIZE_IN_USEC",
|
1145 |
.tag = AUD_OPT_BOOL, |
1146 |
.valp = &conf.size_in_usec_in, |
1147 |
.descr = |
1148 |
"ADC period/buffer size in microseconds (otherwise in frames)"
|
1149 |
}, |
1150 |
{ |
1151 |
.name = "ADC_PERIOD_SIZE",
|
1152 |
.tag = AUD_OPT_INT, |
1153 |
.valp = &conf.period_size_in, |
1154 |
.descr = "ADC period size (0 to go with system default)",
|
1155 |
.overriddenp = &conf.period_size_in_overridden |
1156 |
}, |
1157 |
{ |
1158 |
.name = "ADC_BUFFER_SIZE",
|
1159 |
.tag = AUD_OPT_INT, |
1160 |
.valp = &conf.buffer_size_in, |
1161 |
.descr = "ADC buffer size (0 to go with system default)",
|
1162 |
.overriddenp = &conf.buffer_size_in_overridden |
1163 |
}, |
1164 |
{ |
1165 |
.name = "THRESHOLD",
|
1166 |
.tag = AUD_OPT_INT, |
1167 |
.valp = &conf.threshold, |
1168 |
.descr = "(undocumented)"
|
1169 |
}, |
1170 |
{ |
1171 |
.name = "DAC_DEV",
|
1172 |
.tag = AUD_OPT_STR, |
1173 |
.valp = &conf.pcm_name_out, |
1174 |
.descr = "DAC device name (for instance dmix)"
|
1175 |
}, |
1176 |
{ |
1177 |
.name = "ADC_DEV",
|
1178 |
.tag = AUD_OPT_STR, |
1179 |
.valp = &conf.pcm_name_in, |
1180 |
.descr = "ADC device name"
|
1181 |
}, |
1182 |
{ |
1183 |
.name = "VERBOSE",
|
1184 |
.tag = AUD_OPT_BOOL, |
1185 |
.valp = &conf.verbose, |
1186 |
.descr = "Behave in a more verbose way"
|
1187 |
}, |
1188 |
{ /* End of list */ }
|
1189 |
}; |
1190 |
|
1191 |
static struct audio_pcm_ops alsa_pcm_ops = { |
1192 |
.init_out = alsa_init_out, |
1193 |
.fini_out = alsa_fini_out, |
1194 |
.run_out = alsa_run_out, |
1195 |
.write = alsa_write, |
1196 |
.ctl_out = alsa_ctl_out, |
1197 |
|
1198 |
.init_in = alsa_init_in, |
1199 |
.fini_in = alsa_fini_in, |
1200 |
.run_in = alsa_run_in, |
1201 |
.read = alsa_read, |
1202 |
.ctl_in = alsa_ctl_in, |
1203 |
}; |
1204 |
|
1205 |
struct audio_driver alsa_audio_driver = {
|
1206 |
.name = "alsa",
|
1207 |
.descr = "ALSA http://www.alsa-project.org",
|
1208 |
.options = alsa_options, |
1209 |
.init = alsa_audio_init, |
1210 |
.fini = alsa_audio_fini, |
1211 |
.pcm_ops = &alsa_pcm_ops, |
1212 |
.can_be_default = 1,
|
1213 |
.max_voices_out = INT_MAX, |
1214 |
.max_voices_in = INT_MAX, |
1215 |
.voice_size_out = sizeof (ALSAVoiceOut),
|
1216 |
.voice_size_in = sizeof (ALSAVoiceIn)
|
1217 |
}; |