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![Kővágó, Zoltán](/assets/img/avatar_default.png)
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com> Message-id: ab9768e73dfe7b7305bd6a51629846e0d77622a5.1568927990.git.DirtY.iCE.hu@gmail.com Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
953 lines
24 KiB
C
953 lines
24 KiB
C
/*
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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 "qemu/osdep.h"
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#include <alsa/asoundlib.h>
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#include "qemu/main-loop.h"
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#include "qemu/module.h"
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#include "audio.h"
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#include "trace.h"
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#pragma GCC diagnostic ignored "-Waddress"
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#define AUDIO_CAP "alsa"
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#include "audio_int.h"
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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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AudioState *s;
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};
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typedef struct ALSAVoiceOut {
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HWVoiceOut hw;
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snd_pcm_t *handle;
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struct pollhlp pollhlp;
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Audiodev *dev;
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} ALSAVoiceOut;
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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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struct pollhlp pollhlp;
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Audiodev *dev;
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} ALSAVoiceIn;
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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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};
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struct alsa_params_obt {
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int freq;
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AudioFormat 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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};
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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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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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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
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}
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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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AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
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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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AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
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}
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static void alsa_fini_poll (struct pollhlp *hlp)
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{
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int i;
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struct pollfd *pfds = hlp->pfds;
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if (pfds) {
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for (i = 0; i < hlp->count; ++i) {
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qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
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}
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g_free (pfds);
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}
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hlp->pfds = NULL;
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hlp->count = 0;
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hlp->handle = NULL;
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}
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static void alsa_anal_close1 (snd_pcm_t **handlep)
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{
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int err = snd_pcm_close (*handlep);
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if (err) {
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alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
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}
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*handlep = NULL;
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}
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static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
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{
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alsa_fini_poll (hlp);
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alsa_anal_close1 (handlep);
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}
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static int alsa_recover (snd_pcm_t *handle)
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{
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int err = snd_pcm_prepare (handle);
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if (err < 0) {
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alsa_logerr (err, "Failed to prepare handle %p\n", handle);
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return -1;
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}
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return 0;
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}
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static int alsa_resume (snd_pcm_t *handle)
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{
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int err = snd_pcm_resume (handle);
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if (err < 0) {
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alsa_logerr (err, "Failed to resume handle %p\n", handle);
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return -1;
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}
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return 0;
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}
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static void alsa_poll_handler (void *opaque)
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{
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int err, count;
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snd_pcm_state_t state;
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struct pollhlp *hlp = opaque;
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unsigned short revents;
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count = poll (hlp->pfds, hlp->count, 0);
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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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}
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if (!count) {
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return;
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}
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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);
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if (err < 0) {
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alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
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return;
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}
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if (!(revents & hlp->mask)) {
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trace_alsa_revents(revents);
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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_SETUP:
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alsa_recover (hlp->handle);
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break;
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case SND_PCM_STATE_XRUN:
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alsa_recover (hlp->handle);
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break;
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case SND_PCM_STATE_SUSPENDED:
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alsa_resume (hlp->handle);
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break;
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case SND_PCM_STATE_PREPARED:
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audio_run(hlp->s, "alsa run (prepared)");
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break;
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case SND_PCM_STATE_RUNNING:
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audio_run(hlp->s, "alsa run (running)");
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break;
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default:
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dolog ("Unexpected state %d\n", state);
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}
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}
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static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
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{
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int i, count, err;
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struct pollfd *pfds;
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count = snd_pcm_poll_descriptors_count (handle);
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if (count <= 0) {
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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;
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}
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pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
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if (!pfds) {
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dolog ("Could not initialize poll mode\n");
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return -1;
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}
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err = snd_pcm_poll_descriptors (handle, pfds, count);
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if (err < 0) {
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alsa_logerr (err, "Could not initialize poll mode\n"
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"Could not obtain poll descriptors\n");
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g_free (pfds);
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return -1;
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}
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for (i = 0; i < count; ++i) {
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if (pfds[i].events & POLLIN) {
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qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
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}
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if (pfds[i].events & POLLOUT) {
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trace_alsa_pollout(i, pfds[i].fd);
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qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
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}
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trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
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}
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hlp->pfds = pfds;
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hlp->count = count;
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hlp->handle = handle;
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hlp->mask = mask;
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return 0;
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}
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static int alsa_poll_out (HWVoiceOut *hw)
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{
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ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
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return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
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}
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static int alsa_poll_in (HWVoiceIn *hw)
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{
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ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
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return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
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}
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static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
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{
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switch (fmt) {
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case AUDIO_FORMAT_S8:
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return SND_PCM_FORMAT_S8;
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case AUDIO_FORMAT_U8:
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return SND_PCM_FORMAT_U8;
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case AUDIO_FORMAT_S16:
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if (endianness) {
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return SND_PCM_FORMAT_S16_BE;
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}
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else {
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return SND_PCM_FORMAT_S16_LE;
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}
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case AUDIO_FORMAT_U16:
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if (endianness) {
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return SND_PCM_FORMAT_U16_BE;
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}
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else {
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return SND_PCM_FORMAT_U16_LE;
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}
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case AUDIO_FORMAT_S32:
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if (endianness) {
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return SND_PCM_FORMAT_S32_BE;
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}
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else {
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return SND_PCM_FORMAT_S32_LE;
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}
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case AUDIO_FORMAT_U32:
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if (endianness) {
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return SND_PCM_FORMAT_U32_BE;
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}
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else {
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return SND_PCM_FORMAT_U32_LE;
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}
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default:
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dolog ("Internal logic error: Bad audio format %d\n", fmt);
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#ifdef DEBUG_AUDIO
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abort ();
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#endif
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return SND_PCM_FORMAT_U8;
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}
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}
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static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
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int *endianness)
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{
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switch (alsafmt) {
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case SND_PCM_FORMAT_S8:
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*endianness = 0;
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*fmt = AUDIO_FORMAT_S8;
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break;
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case SND_PCM_FORMAT_U8:
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*endianness = 0;
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*fmt = AUDIO_FORMAT_U8;
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break;
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case SND_PCM_FORMAT_S16_LE:
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*endianness = 0;
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*fmt = AUDIO_FORMAT_S16;
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break;
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case SND_PCM_FORMAT_U16_LE:
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*endianness = 0;
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*fmt = AUDIO_FORMAT_U16;
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break;
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case SND_PCM_FORMAT_S16_BE:
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*endianness = 1;
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*fmt = AUDIO_FORMAT_S16;
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break;
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case SND_PCM_FORMAT_U16_BE:
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*endianness = 1;
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*fmt = AUDIO_FORMAT_U16;
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break;
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case SND_PCM_FORMAT_S32_LE:
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*endianness = 0;
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*fmt = AUDIO_FORMAT_S32;
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break;
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case SND_PCM_FORMAT_U32_LE:
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*endianness = 0;
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*fmt = AUDIO_FORMAT_U32;
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break;
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case SND_PCM_FORMAT_S32_BE:
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*endianness = 1;
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*fmt = AUDIO_FORMAT_S32;
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break;
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case SND_PCM_FORMAT_U32_BE:
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*endianness = 1;
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*fmt = AUDIO_FORMAT_U32;
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break;
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default:
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dolog ("Unrecognized audio format %d\n", alsafmt);
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return -1;
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}
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return 0;
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}
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static void alsa_dump_info (struct alsa_params_req *req,
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struct alsa_params_obt *obt,
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snd_pcm_format_t obtfmt,
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AudiodevAlsaPerDirectionOptions *apdo)
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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, obtfmt);
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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 len %" PRId32 " period len %" PRId32 "\n",
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apdo->buffer_length, apdo->period_length);
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dolog("obtained: samples %ld\n", obt->samples);
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}
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static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
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{
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int err;
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snd_pcm_sw_params_t *sw_params;
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snd_pcm_sw_params_alloca (&sw_params);
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err = snd_pcm_sw_params_current (handle, sw_params);
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if (err < 0) {
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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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}
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err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
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if (err < 0) {
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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);
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return;
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}
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err = snd_pcm_sw_params (handle, sw_params);
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if (err < 0) {
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dolog ("Could not fully initialize DAC\n");
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alsa_logerr (err, "Failed to set software parameters\n");
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return;
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}
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}
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static int alsa_open(bool in, struct alsa_params_req *req,
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struct alsa_params_obt *obt, snd_pcm_t **handlep,
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Audiodev *dev)
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{
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AudiodevAlsaOptions *aopts = &dev->u.alsa;
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AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
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snd_pcm_t *handle;
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snd_pcm_hw_params_t *hw_params;
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int err;
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unsigned int freq, nchannels;
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const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
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snd_pcm_uframes_t obt_buffer_size;
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const char *typ = in ? "ADC" : "DAC";
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snd_pcm_format_t obtfmt;
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freq = req->freq;
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nchannels = req->nchannels;
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snd_pcm_hw_params_alloca (&hw_params);
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err = snd_pcm_open (
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&handle,
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pcm_name,
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in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
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SND_PCM_NONBLOCK
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);
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if (err < 0) {
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alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
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return -1;
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}
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err = snd_pcm_hw_params_any (handle, hw_params);
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if (err < 0) {
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alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
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goto err;
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}
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err = snd_pcm_hw_params_set_access (
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handle,
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hw_params,
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SND_PCM_ACCESS_RW_INTERLEAVED
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);
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if (err < 0) {
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alsa_logerr2 (err, typ, "Failed to set access type\n");
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goto err;
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}
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err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
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if (err < 0) {
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alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
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}
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err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
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if (err < 0) {
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alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
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goto err;
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}
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err = snd_pcm_hw_params_set_channels_near (
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handle,
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hw_params,
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&nchannels
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);
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if (err < 0) {
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alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
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req->nchannels);
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goto err;
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}
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if (nchannels != 1 && nchannels != 2) {
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alsa_logerr2 (err, typ,
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"Can not handle obtained number of channels %d\n",
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nchannels);
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goto err;
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}
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if (apdo->buffer_length) {
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int dir = 0;
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unsigned int btime = apdo->buffer_length;
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err = snd_pcm_hw_params_set_buffer_time_near(
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handle, hw_params, &btime, &dir);
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if (err < 0) {
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alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
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apdo->buffer_length);
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goto err;
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}
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if (apdo->has_buffer_length && btime != apdo->buffer_length) {
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dolog("Requested buffer time %" PRId32
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" was rejected, using %u\n", apdo->buffer_length, btime);
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}
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}
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if (apdo->period_length) {
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int dir = 0;
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unsigned int ptime = apdo->period_length;
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err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
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&dir);
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if (err < 0) {
|
|
alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
|
|
apdo->period_length);
|
|
goto err;
|
|
}
|
|
|
|
if (apdo->has_period_length && ptime != apdo->period_length) {
|
|
dolog("Requested period time %" PRId32 " was rejected, using %d\n",
|
|
apdo->period_length, ptime);
|
|
}
|
|
}
|
|
|
|
err = snd_pcm_hw_params (handle, hw_params);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
|
|
goto err;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to get buffer size\n");
|
|
goto err;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to get format\n");
|
|
goto err;
|
|
}
|
|
|
|
if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
|
|
dolog ("Invalid format was returned %d\n", obtfmt);
|
|
goto err;
|
|
}
|
|
|
|
err = snd_pcm_prepare (handle);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
|
|
goto err;
|
|
}
|
|
|
|
if (!in && aopts->has_threshold && aopts->threshold) {
|
|
struct audsettings as = { .freq = freq };
|
|
alsa_set_threshold(
|
|
handle,
|
|
audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
|
|
&as, aopts->threshold));
|
|
}
|
|
|
|
obt->nchannels = nchannels;
|
|
obt->freq = freq;
|
|
obt->samples = obt_buffer_size;
|
|
|
|
*handlep = handle;
|
|
|
|
if (obtfmt != req->fmt ||
|
|
obt->nchannels != req->nchannels ||
|
|
obt->freq != req->freq) {
|
|
dolog ("Audio parameters for %s\n", typ);
|
|
alsa_dump_info(req, obt, obtfmt, apdo);
|
|
}
|
|
|
|
#ifdef DEBUG
|
|
alsa_dump_info(req, obt, obtfmt, pdo);
|
|
#endif
|
|
return 0;
|
|
|
|
err:
|
|
alsa_anal_close1 (&handle);
|
|
return -1;
|
|
}
|
|
|
|
static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
|
|
{
|
|
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
|
|
size_t pos = 0;
|
|
size_t len_frames = len >> hw->info.shift;
|
|
|
|
while (len_frames) {
|
|
char *src = advance(buf, pos);
|
|
snd_pcm_sframes_t written;
|
|
|
|
written = snd_pcm_writei(alsa->handle, src, len_frames);
|
|
|
|
if (written <= 0) {
|
|
switch (written) {
|
|
case 0:
|
|
trace_alsa_wrote_zero(len_frames);
|
|
return pos;
|
|
|
|
case -EPIPE:
|
|
if (alsa_recover(alsa->handle)) {
|
|
alsa_logerr(written, "Failed to write %zu frames\n",
|
|
len_frames);
|
|
return pos;
|
|
}
|
|
trace_alsa_xrun_out();
|
|
continue;
|
|
|
|
case -ESTRPIPE:
|
|
/*
|
|
* stream is suspended and waiting for an application
|
|
* recovery
|
|
*/
|
|
if (alsa_resume(alsa->handle)) {
|
|
alsa_logerr(written, "Failed to write %zu frames\n",
|
|
len_frames);
|
|
return pos;
|
|
}
|
|
trace_alsa_resume_out();
|
|
continue;
|
|
|
|
case -EAGAIN:
|
|
return pos;
|
|
|
|
default:
|
|
alsa_logerr(written, "Failed to write %zu frames from %p\n",
|
|
len, src);
|
|
return pos;
|
|
}
|
|
}
|
|
|
|
pos += written << hw->info.shift;
|
|
if (written < len_frames) {
|
|
break;
|
|
}
|
|
len_frames -= written;
|
|
}
|
|
|
|
return pos;
|
|
}
|
|
|
|
static void alsa_fini_out (HWVoiceOut *hw)
|
|
{
|
|
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
|
|
|
|
ldebug ("alsa_fini\n");
|
|
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
|
|
}
|
|
|
|
static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
|
|
void *drv_opaque)
|
|
{
|
|
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
|
|
struct alsa_params_req req;
|
|
struct alsa_params_obt obt;
|
|
snd_pcm_t *handle;
|
|
struct audsettings obt_as;
|
|
Audiodev *dev = drv_opaque;
|
|
|
|
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
|
|
req.freq = as->freq;
|
|
req.nchannels = as->nchannels;
|
|
|
|
if (alsa_open(0, &req, &obt, &handle, dev)) {
|
|
return -1;
|
|
}
|
|
|
|
obt_as.freq = obt.freq;
|
|
obt_as.nchannels = obt.nchannels;
|
|
obt_as.fmt = obt.fmt;
|
|
obt_as.endianness = obt.endianness;
|
|
|
|
audio_pcm_init_info (&hw->info, &obt_as);
|
|
hw->samples = obt.samples;
|
|
|
|
alsa->pollhlp.s = hw->s;
|
|
alsa->handle = handle;
|
|
alsa->dev = dev;
|
|
return 0;
|
|
}
|
|
|
|
#define VOICE_CTL_PAUSE 0
|
|
#define VOICE_CTL_PREPARE 1
|
|
#define VOICE_CTL_START 2
|
|
|
|
static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
|
|
{
|
|
int err;
|
|
|
|
if (ctl == VOICE_CTL_PAUSE) {
|
|
err = snd_pcm_drop (handle);
|
|
if (err < 0) {
|
|
alsa_logerr (err, "Could not stop %s\n", typ);
|
|
return -1;
|
|
}
|
|
}
|
|
else {
|
|
err = snd_pcm_prepare (handle);
|
|
if (err < 0) {
|
|
alsa_logerr (err, "Could not prepare handle for %s\n", typ);
|
|
return -1;
|
|
}
|
|
if (ctl == VOICE_CTL_START) {
|
|
err = snd_pcm_start(handle);
|
|
if (err < 0) {
|
|
alsa_logerr (err, "Could not start handle for %s\n", typ);
|
|
return -1;
|
|
}
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
|
|
{
|
|
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
|
|
AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
|
|
|
|
switch (cmd) {
|
|
case VOICE_ENABLE:
|
|
{
|
|
bool poll_mode = apdo->try_poll;
|
|
|
|
ldebug ("enabling voice\n");
|
|
if (poll_mode && alsa_poll_out (hw)) {
|
|
poll_mode = 0;
|
|
}
|
|
hw->poll_mode = poll_mode;
|
|
return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
|
|
}
|
|
|
|
case VOICE_DISABLE:
|
|
ldebug ("disabling voice\n");
|
|
if (hw->poll_mode) {
|
|
hw->poll_mode = 0;
|
|
alsa_fini_poll (&alsa->pollhlp);
|
|
}
|
|
return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
|
|
{
|
|
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
|
|
struct alsa_params_req req;
|
|
struct alsa_params_obt obt;
|
|
snd_pcm_t *handle;
|
|
struct audsettings obt_as;
|
|
Audiodev *dev = drv_opaque;
|
|
|
|
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
|
|
req.freq = as->freq;
|
|
req.nchannels = as->nchannels;
|
|
|
|
if (alsa_open(1, &req, &obt, &handle, dev)) {
|
|
return -1;
|
|
}
|
|
|
|
obt_as.freq = obt.freq;
|
|
obt_as.nchannels = obt.nchannels;
|
|
obt_as.fmt = obt.fmt;
|
|
obt_as.endianness = obt.endianness;
|
|
|
|
audio_pcm_init_info (&hw->info, &obt_as);
|
|
hw->samples = obt.samples;
|
|
|
|
alsa->pollhlp.s = hw->s;
|
|
alsa->handle = handle;
|
|
alsa->dev = dev;
|
|
return 0;
|
|
}
|
|
|
|
static void alsa_fini_in (HWVoiceIn *hw)
|
|
{
|
|
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
|
|
|
|
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
|
|
}
|
|
|
|
static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
|
|
{
|
|
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
|
|
size_t pos = 0;
|
|
|
|
while (len) {
|
|
void *dst = advance(buf, pos);
|
|
snd_pcm_sframes_t nread;
|
|
|
|
nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
|
|
|
|
if (nread <= 0) {
|
|
switch (nread) {
|
|
case 0:
|
|
trace_alsa_read_zero(len);
|
|
return pos;;
|
|
|
|
case -EPIPE:
|
|
if (alsa_recover(alsa->handle)) {
|
|
alsa_logerr(nread, "Failed to read %zu frames\n", len);
|
|
return pos;
|
|
}
|
|
trace_alsa_xrun_in();
|
|
continue;
|
|
|
|
case -EAGAIN:
|
|
return pos;
|
|
|
|
default:
|
|
alsa_logerr(nread, "Failed to read %zu frames to %p\n",
|
|
len, dst);
|
|
return pos;;
|
|
}
|
|
}
|
|
|
|
pos += nread << hw->info.shift;
|
|
len -= nread << hw->info.shift;
|
|
}
|
|
|
|
return pos;
|
|
}
|
|
|
|
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
|
|
{
|
|
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
|
|
AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
|
|
|
|
switch (cmd) {
|
|
case VOICE_ENABLE:
|
|
{
|
|
bool poll_mode = apdo->try_poll;
|
|
|
|
ldebug ("enabling voice\n");
|
|
if (poll_mode && alsa_poll_in (hw)) {
|
|
poll_mode = 0;
|
|
}
|
|
hw->poll_mode = poll_mode;
|
|
|
|
return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
|
|
}
|
|
|
|
case VOICE_DISABLE:
|
|
ldebug ("disabling voice\n");
|
|
if (hw->poll_mode) {
|
|
hw->poll_mode = 0;
|
|
alsa_fini_poll (&alsa->pollhlp);
|
|
}
|
|
return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
|
|
{
|
|
if (!apdo->has_try_poll) {
|
|
apdo->try_poll = true;
|
|
apdo->has_try_poll = true;
|
|
}
|
|
}
|
|
|
|
static void *alsa_audio_init(Audiodev *dev)
|
|
{
|
|
AudiodevAlsaOptions *aopts;
|
|
assert(dev->driver == AUDIODEV_DRIVER_ALSA);
|
|
|
|
aopts = &dev->u.alsa;
|
|
alsa_init_per_direction(aopts->in);
|
|
alsa_init_per_direction(aopts->out);
|
|
|
|
/*
|
|
* need to define them, as otherwise alsa produces no sound
|
|
* doesn't set has_* so alsa_open can identify it wasn't set by the user
|
|
*/
|
|
if (!dev->u.alsa.out->has_period_length) {
|
|
/* 1024 frames assuming 44100Hz */
|
|
dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
|
|
}
|
|
if (!dev->u.alsa.out->has_buffer_length) {
|
|
/* 4096 frames assuming 44100Hz */
|
|
dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
|
|
}
|
|
|
|
/*
|
|
* OptsVisitor sets unspecified optional fields to zero, but do not depend
|
|
* on it...
|
|
*/
|
|
if (!dev->u.alsa.in->has_period_length) {
|
|
dev->u.alsa.in->period_length = 0;
|
|
}
|
|
if (!dev->u.alsa.in->has_buffer_length) {
|
|
dev->u.alsa.in->buffer_length = 0;
|
|
}
|
|
|
|
return dev;
|
|
}
|
|
|
|
static void alsa_audio_fini (void *opaque)
|
|
{
|
|
}
|
|
|
|
static struct audio_pcm_ops alsa_pcm_ops = {
|
|
.init_out = alsa_init_out,
|
|
.fini_out = alsa_fini_out,
|
|
.write = alsa_write,
|
|
.ctl_out = alsa_ctl_out,
|
|
|
|
.init_in = alsa_init_in,
|
|
.fini_in = alsa_fini_in,
|
|
.read = alsa_read,
|
|
.ctl_in = alsa_ctl_in,
|
|
};
|
|
|
|
static struct audio_driver alsa_audio_driver = {
|
|
.name = "alsa",
|
|
.descr = "ALSA http://www.alsa-project.org",
|
|
.init = alsa_audio_init,
|
|
.fini = alsa_audio_fini,
|
|
.pcm_ops = &alsa_pcm_ops,
|
|
.can_be_default = 1,
|
|
.max_voices_out = INT_MAX,
|
|
.max_voices_in = INT_MAX,
|
|
.voice_size_out = sizeof (ALSAVoiceOut),
|
|
.voice_size_in = sizeof (ALSAVoiceIn)
|
|
};
|
|
|
|
static void register_audio_alsa(void)
|
|
{
|
|
audio_driver_register(&alsa_audio_driver);
|
|
}
|
|
type_init(register_audio_alsa);
|