1229 lines
32 KiB
C
1229 lines
32 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-common.h"
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#include "qemu/main-loop.h"
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#include "audio.h"
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#include "trace.h"
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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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#define AUDIO_CAP "alsa"
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#include "audio_int.h"
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typedef struct ALSAConf {
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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;
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int buffer_size_in_overridden;
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int period_size_in_overridden;
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int buffer_size_out_overridden;
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int period_size_out_overridden;
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} ALSAConf;
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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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ALSAConf *conf;
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int count;
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int mask;
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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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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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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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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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};
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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 ("alsa run (prepared)");
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break;
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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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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 int alsa_write (SWVoiceOut *sw, void *buf, int len)
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{
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return audio_pcm_sw_write (sw, buf, len);
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}
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static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
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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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case AUD_FMT_U8:
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return SND_PCM_FORMAT_U8;
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case AUD_FMT_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 AUD_FMT_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 AUD_FMT_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 AUD_FMT_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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|
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default:
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dolog ("Internal logic error: Bad audio format %d\n", fmt);
|
|
#ifdef DEBUG_AUDIO
|
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abort ();
|
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#endif
|
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return SND_PCM_FORMAT_U8;
|
|
}
|
|
}
|
|
|
|
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *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 = AUD_FMT_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 = AUD_FMT_U8;
|
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break;
|
|
|
|
case SND_PCM_FORMAT_S16_LE:
|
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*endianness = 0;
|
|
*fmt = AUD_FMT_S16;
|
|
break;
|
|
|
|
case SND_PCM_FORMAT_U16_LE:
|
|
*endianness = 0;
|
|
*fmt = AUD_FMT_U16;
|
|
break;
|
|
|
|
case SND_PCM_FORMAT_S16_BE:
|
|
*endianness = 1;
|
|
*fmt = AUD_FMT_S16;
|
|
break;
|
|
|
|
case SND_PCM_FORMAT_U16_BE:
|
|
*endianness = 1;
|
|
*fmt = AUD_FMT_U16;
|
|
break;
|
|
|
|
case SND_PCM_FORMAT_S32_LE:
|
|
*endianness = 0;
|
|
*fmt = AUD_FMT_S32;
|
|
break;
|
|
|
|
case SND_PCM_FORMAT_U32_LE:
|
|
*endianness = 0;
|
|
*fmt = AUD_FMT_U32;
|
|
break;
|
|
|
|
case SND_PCM_FORMAT_S32_BE:
|
|
*endianness = 1;
|
|
*fmt = AUD_FMT_S32;
|
|
break;
|
|
|
|
case SND_PCM_FORMAT_U32_BE:
|
|
*endianness = 1;
|
|
*fmt = AUD_FMT_U32;
|
|
break;
|
|
|
|
default:
|
|
dolog ("Unrecognized audio format %d\n", alsafmt);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void alsa_dump_info (struct alsa_params_req *req,
|
|
struct alsa_params_obt *obt,
|
|
snd_pcm_format_t obtfmt)
|
|
{
|
|
dolog ("parameter | requested value | obtained value\n");
|
|
dolog ("format | %10d | %10d\n", req->fmt, obtfmt);
|
|
dolog ("channels | %10d | %10d\n",
|
|
req->nchannels, obt->nchannels);
|
|
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
|
|
dolog ("============================================\n");
|
|
dolog ("requested: buffer size %d period size %d\n",
|
|
req->buffer_size, req->period_size);
|
|
dolog ("obtained: samples %ld\n", obt->samples);
|
|
}
|
|
|
|
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
|
|
{
|
|
int err;
|
|
snd_pcm_sw_params_t *sw_params;
|
|
|
|
snd_pcm_sw_params_alloca (&sw_params);
|
|
|
|
err = snd_pcm_sw_params_current (handle, sw_params);
|
|
if (err < 0) {
|
|
dolog ("Could not fully initialize DAC\n");
|
|
alsa_logerr (err, "Failed to get current software parameters\n");
|
|
return;
|
|
}
|
|
|
|
err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
|
|
if (err < 0) {
|
|
dolog ("Could not fully initialize DAC\n");
|
|
alsa_logerr (err, "Failed to set software threshold to %ld\n",
|
|
threshold);
|
|
return;
|
|
}
|
|
|
|
err = snd_pcm_sw_params (handle, sw_params);
|
|
if (err < 0) {
|
|
dolog ("Could not fully initialize DAC\n");
|
|
alsa_logerr (err, "Failed to set software parameters\n");
|
|
return;
|
|
}
|
|
}
|
|
|
|
static int alsa_open (int in, struct alsa_params_req *req,
|
|
struct alsa_params_obt *obt, snd_pcm_t **handlep,
|
|
ALSAConf *conf)
|
|
{
|
|
snd_pcm_t *handle;
|
|
snd_pcm_hw_params_t *hw_params;
|
|
int err;
|
|
int size_in_usec;
|
|
unsigned int freq, nchannels;
|
|
const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
|
|
snd_pcm_uframes_t obt_buffer_size;
|
|
const char *typ = in ? "ADC" : "DAC";
|
|
snd_pcm_format_t obtfmt;
|
|
|
|
freq = req->freq;
|
|
nchannels = req->nchannels;
|
|
size_in_usec = req->size_in_usec;
|
|
|
|
snd_pcm_hw_params_alloca (&hw_params);
|
|
|
|
err = snd_pcm_open (
|
|
&handle,
|
|
pcm_name,
|
|
in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
|
|
SND_PCM_NONBLOCK
|
|
);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
|
|
return -1;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_any (handle, hw_params);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
|
|
goto err;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_set_access (
|
|
handle,
|
|
hw_params,
|
|
SND_PCM_ACCESS_RW_INTERLEAVED
|
|
);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to set access type\n");
|
|
goto err;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
|
|
}
|
|
|
|
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
|
|
goto err;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_set_channels_near (
|
|
handle,
|
|
hw_params,
|
|
&nchannels
|
|
);
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
|
|
req->nchannels);
|
|
goto err;
|
|
}
|
|
|
|
if (nchannels != 1 && nchannels != 2) {
|
|
alsa_logerr2 (err, typ,
|
|
"Can not handle obtained number of channels %d\n",
|
|
nchannels);
|
|
goto err;
|
|
}
|
|
|
|
if (req->buffer_size) {
|
|
unsigned long obt;
|
|
|
|
if (size_in_usec) {
|
|
int dir = 0;
|
|
unsigned int btime = req->buffer_size;
|
|
|
|
err = snd_pcm_hw_params_set_buffer_time_near (
|
|
handle,
|
|
hw_params,
|
|
&btime,
|
|
&dir
|
|
);
|
|
obt = btime;
|
|
}
|
|
else {
|
|
snd_pcm_uframes_t bsize = req->buffer_size;
|
|
|
|
err = snd_pcm_hw_params_set_buffer_size_near (
|
|
handle,
|
|
hw_params,
|
|
&bsize
|
|
);
|
|
obt = bsize;
|
|
}
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
|
|
size_in_usec ? "time" : "size", req->buffer_size);
|
|
goto err;
|
|
}
|
|
|
|
if ((req->override_mask & 2) && (obt - req->buffer_size))
|
|
dolog ("Requested buffer %s %u was rejected, using %lu\n",
|
|
size_in_usec ? "time" : "size", req->buffer_size, obt);
|
|
}
|
|
|
|
if (req->period_size) {
|
|
unsigned long obt;
|
|
|
|
if (size_in_usec) {
|
|
int dir = 0;
|
|
unsigned int ptime = req->period_size;
|
|
|
|
err = snd_pcm_hw_params_set_period_time_near (
|
|
handle,
|
|
hw_params,
|
|
&ptime,
|
|
&dir
|
|
);
|
|
obt = ptime;
|
|
}
|
|
else {
|
|
int dir = 0;
|
|
snd_pcm_uframes_t psize = req->period_size;
|
|
|
|
err = snd_pcm_hw_params_set_period_size_near (
|
|
handle,
|
|
hw_params,
|
|
&psize,
|
|
&dir
|
|
);
|
|
obt = psize;
|
|
}
|
|
|
|
if (err < 0) {
|
|
alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
|
|
size_in_usec ? "time" : "size", req->period_size);
|
|
goto err;
|
|
}
|
|
|
|
if (((req->override_mask & 1) && (obt - req->period_size)))
|
|
dolog ("Requested period %s %u was rejected, using %lu\n",
|
|
size_in_usec ? "time" : "size", req->period_size, obt);
|
|
}
|
|
|
|
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 && conf->threshold) {
|
|
snd_pcm_uframes_t threshold;
|
|
int bytes_per_sec;
|
|
|
|
bytes_per_sec = freq << (nchannels == 2);
|
|
|
|
switch (obt->fmt) {
|
|
case AUD_FMT_S8:
|
|
case AUD_FMT_U8:
|
|
break;
|
|
|
|
case AUD_FMT_S16:
|
|
case AUD_FMT_U16:
|
|
bytes_per_sec <<= 1;
|
|
break;
|
|
|
|
case AUD_FMT_S32:
|
|
case AUD_FMT_U32:
|
|
bytes_per_sec <<= 2;
|
|
break;
|
|
}
|
|
|
|
threshold = (conf->threshold * bytes_per_sec) / 1000;
|
|
alsa_set_threshold (handle, 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);
|
|
}
|
|
|
|
#ifdef DEBUG
|
|
alsa_dump_info (req, obt, obtfmt);
|
|
#endif
|
|
return 0;
|
|
|
|
err:
|
|
alsa_anal_close1 (&handle);
|
|
return -1;
|
|
}
|
|
|
|
static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
|
|
{
|
|
snd_pcm_sframes_t avail;
|
|
|
|
avail = snd_pcm_avail_update (handle);
|
|
if (avail < 0) {
|
|
if (avail == -EPIPE) {
|
|
if (!alsa_recover (handle)) {
|
|
avail = snd_pcm_avail_update (handle);
|
|
}
|
|
}
|
|
|
|
if (avail < 0) {
|
|
alsa_logerr (avail,
|
|
"Could not obtain number of available frames\n");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
return avail;
|
|
}
|
|
|
|
static void alsa_write_pending (ALSAVoiceOut *alsa)
|
|
{
|
|
HWVoiceOut *hw = &alsa->hw;
|
|
|
|
while (alsa->pending) {
|
|
int left_till_end_samples = hw->samples - alsa->wpos;
|
|
int len = audio_MIN (alsa->pending, left_till_end_samples);
|
|
char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
|
|
|
|
while (len) {
|
|
snd_pcm_sframes_t written;
|
|
|
|
written = snd_pcm_writei (alsa->handle, src, len);
|
|
|
|
if (written <= 0) {
|
|
switch (written) {
|
|
case 0:
|
|
trace_alsa_wrote_zero(len);
|
|
return;
|
|
|
|
case -EPIPE:
|
|
if (alsa_recover (alsa->handle)) {
|
|
alsa_logerr (written, "Failed to write %d frames\n",
|
|
len);
|
|
return;
|
|
}
|
|
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 %d frames\n",
|
|
len);
|
|
return;
|
|
}
|
|
trace_alsa_resume_out();
|
|
continue;
|
|
|
|
case -EAGAIN:
|
|
return;
|
|
|
|
default:
|
|
alsa_logerr (written, "Failed to write %d frames from %p\n",
|
|
len, src);
|
|
return;
|
|
}
|
|
}
|
|
|
|
alsa->wpos = (alsa->wpos + written) % hw->samples;
|
|
alsa->pending -= written;
|
|
len -= written;
|
|
}
|
|
}
|
|
}
|
|
|
|
static int alsa_run_out (HWVoiceOut *hw, int live)
|
|
{
|
|
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
|
|
int decr;
|
|
snd_pcm_sframes_t avail;
|
|
|
|
avail = alsa_get_avail (alsa->handle);
|
|
if (avail < 0) {
|
|
dolog ("Could not get number of available playback frames\n");
|
|
return 0;
|
|
}
|
|
|
|
decr = audio_MIN (live, avail);
|
|
decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
|
|
alsa->pending += decr;
|
|
alsa_write_pending (alsa);
|
|
return decr;
|
|
}
|
|
|
|
static void alsa_fini_out (HWVoiceOut *hw)
|
|
{
|
|
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
|
|
|
|
ldebug ("alsa_fini\n");
|
|
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
|
|
|
|
g_free(alsa->pcm_buf);
|
|
alsa->pcm_buf = NULL;
|
|
}
|
|
|
|
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;
|
|
ALSAConf *conf = drv_opaque;
|
|
|
|
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
|
|
req.freq = as->freq;
|
|
req.nchannels = as->nchannels;
|
|
req.period_size = conf->period_size_out;
|
|
req.buffer_size = conf->buffer_size_out;
|
|
req.size_in_usec = conf->size_in_usec_out;
|
|
req.override_mask =
|
|
(conf->period_size_out_overridden ? 1 : 0) |
|
|
(conf->buffer_size_out_overridden ? 2 : 0);
|
|
|
|
if (alsa_open (0, &req, &obt, &handle, conf)) {
|
|
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->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
|
|
if (!alsa->pcm_buf) {
|
|
dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
|
|
hw->samples, 1 << hw->info.shift);
|
|
alsa_anal_close1 (&handle);
|
|
return -1;
|
|
}
|
|
|
|
alsa->handle = handle;
|
|
alsa->pollhlp.conf = conf;
|
|
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;
|
|
|
|
switch (cmd) {
|
|
case VOICE_ENABLE:
|
|
{
|
|
va_list ap;
|
|
int poll_mode;
|
|
|
|
va_start (ap, cmd);
|
|
poll_mode = va_arg (ap, int);
|
|
va_end (ap);
|
|
|
|
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;
|
|
ALSAConf *conf = drv_opaque;
|
|
|
|
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
|
|
req.freq = as->freq;
|
|
req.nchannels = as->nchannels;
|
|
req.period_size = conf->period_size_in;
|
|
req.buffer_size = conf->buffer_size_in;
|
|
req.size_in_usec = conf->size_in_usec_in;
|
|
req.override_mask =
|
|
(conf->period_size_in_overridden ? 1 : 0) |
|
|
(conf->buffer_size_in_overridden ? 2 : 0);
|
|
|
|
if (alsa_open (1, &req, &obt, &handle, conf)) {
|
|
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->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
|
if (!alsa->pcm_buf) {
|
|
dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
|
|
hw->samples, 1 << hw->info.shift);
|
|
alsa_anal_close1 (&handle);
|
|
return -1;
|
|
}
|
|
|
|
alsa->handle = handle;
|
|
alsa->pollhlp.conf = conf;
|
|
return 0;
|
|
}
|
|
|
|
static void alsa_fini_in (HWVoiceIn *hw)
|
|
{
|
|
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
|
|
|
|
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
|
|
|
|
g_free(alsa->pcm_buf);
|
|
alsa->pcm_buf = NULL;
|
|
}
|
|
|
|
static int alsa_run_in (HWVoiceIn *hw)
|
|
{
|
|
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
|
|
int hwshift = hw->info.shift;
|
|
int i;
|
|
int live = audio_pcm_hw_get_live_in (hw);
|
|
int dead = hw->samples - live;
|
|
int decr;
|
|
struct {
|
|
int add;
|
|
int len;
|
|
} bufs[2] = {
|
|
{ .add = hw->wpos, .len = 0 },
|
|
{ .add = 0, .len = 0 }
|
|
};
|
|
snd_pcm_sframes_t avail;
|
|
snd_pcm_uframes_t read_samples = 0;
|
|
|
|
if (!dead) {
|
|
return 0;
|
|
}
|
|
|
|
avail = alsa_get_avail (alsa->handle);
|
|
if (avail < 0) {
|
|
dolog ("Could not get number of captured frames\n");
|
|
return 0;
|
|
}
|
|
|
|
if (!avail) {
|
|
snd_pcm_state_t state;
|
|
|
|
state = snd_pcm_state (alsa->handle);
|
|
switch (state) {
|
|
case SND_PCM_STATE_PREPARED:
|
|
avail = hw->samples;
|
|
break;
|
|
case SND_PCM_STATE_SUSPENDED:
|
|
/* stream is suspended and waiting for an application recovery */
|
|
if (alsa_resume (alsa->handle)) {
|
|
dolog ("Failed to resume suspended input stream\n");
|
|
return 0;
|
|
}
|
|
trace_alsa_resume_in();
|
|
break;
|
|
default:
|
|
trace_alsa_no_frames(state);
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
decr = audio_MIN (dead, avail);
|
|
if (!decr) {
|
|
return 0;
|
|
}
|
|
|
|
if (hw->wpos + decr > hw->samples) {
|
|
bufs[0].len = (hw->samples - hw->wpos);
|
|
bufs[1].len = (decr - (hw->samples - hw->wpos));
|
|
}
|
|
else {
|
|
bufs[0].len = decr;
|
|
}
|
|
|
|
for (i = 0; i < 2; ++i) {
|
|
void *src;
|
|
struct st_sample *dst;
|
|
snd_pcm_sframes_t nread;
|
|
snd_pcm_uframes_t len;
|
|
|
|
len = bufs[i].len;
|
|
|
|
src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
|
|
dst = hw->conv_buf + bufs[i].add;
|
|
|
|
while (len) {
|
|
nread = snd_pcm_readi (alsa->handle, src, len);
|
|
|
|
if (nread <= 0) {
|
|
switch (nread) {
|
|
case 0:
|
|
trace_alsa_read_zero(len);
|
|
goto exit;
|
|
|
|
case -EPIPE:
|
|
if (alsa_recover (alsa->handle)) {
|
|
alsa_logerr (nread, "Failed to read %ld frames\n", len);
|
|
goto exit;
|
|
}
|
|
trace_alsa_xrun_in();
|
|
continue;
|
|
|
|
case -EAGAIN:
|
|
goto exit;
|
|
|
|
default:
|
|
alsa_logerr (
|
|
nread,
|
|
"Failed to read %ld frames from %p\n",
|
|
len,
|
|
src
|
|
);
|
|
goto exit;
|
|
}
|
|
}
|
|
|
|
hw->conv (dst, src, nread);
|
|
|
|
src = advance (src, nread << hwshift);
|
|
dst += nread;
|
|
|
|
read_samples += nread;
|
|
len -= nread;
|
|
}
|
|
}
|
|
|
|
exit:
|
|
hw->wpos = (hw->wpos + read_samples) % hw->samples;
|
|
return read_samples;
|
|
}
|
|
|
|
static int alsa_read (SWVoiceIn *sw, void *buf, int size)
|
|
{
|
|
return audio_pcm_sw_read (sw, buf, size);
|
|
}
|
|
|
|
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
|
|
{
|
|
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
|
|
|
|
switch (cmd) {
|
|
case VOICE_ENABLE:
|
|
{
|
|
va_list ap;
|
|
int poll_mode;
|
|
|
|
va_start (ap, cmd);
|
|
poll_mode = va_arg (ap, int);
|
|
va_end (ap);
|
|
|
|
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 ALSAConf glob_conf = {
|
|
.buffer_size_out = 4096,
|
|
.period_size_out = 1024,
|
|
.pcm_name_out = "default",
|
|
.pcm_name_in = "default",
|
|
};
|
|
|
|
static void *alsa_audio_init (void)
|
|
{
|
|
ALSAConf *conf = g_malloc(sizeof(ALSAConf));
|
|
*conf = glob_conf;
|
|
return conf;
|
|
}
|
|
|
|
static void alsa_audio_fini (void *opaque)
|
|
{
|
|
g_free(opaque);
|
|
}
|
|
|
|
static struct audio_option alsa_options[] = {
|
|
{
|
|
.name = "DAC_SIZE_IN_USEC",
|
|
.tag = AUD_OPT_BOOL,
|
|
.valp = &glob_conf.size_in_usec_out,
|
|
.descr = "DAC period/buffer size in microseconds (otherwise in frames)"
|
|
},
|
|
{
|
|
.name = "DAC_PERIOD_SIZE",
|
|
.tag = AUD_OPT_INT,
|
|
.valp = &glob_conf.period_size_out,
|
|
.descr = "DAC period size (0 to go with system default)",
|
|
.overriddenp = &glob_conf.period_size_out_overridden
|
|
},
|
|
{
|
|
.name = "DAC_BUFFER_SIZE",
|
|
.tag = AUD_OPT_INT,
|
|
.valp = &glob_conf.buffer_size_out,
|
|
.descr = "DAC buffer size (0 to go with system default)",
|
|
.overriddenp = &glob_conf.buffer_size_out_overridden
|
|
},
|
|
{
|
|
.name = "ADC_SIZE_IN_USEC",
|
|
.tag = AUD_OPT_BOOL,
|
|
.valp = &glob_conf.size_in_usec_in,
|
|
.descr =
|
|
"ADC period/buffer size in microseconds (otherwise in frames)"
|
|
},
|
|
{
|
|
.name = "ADC_PERIOD_SIZE",
|
|
.tag = AUD_OPT_INT,
|
|
.valp = &glob_conf.period_size_in,
|
|
.descr = "ADC period size (0 to go with system default)",
|
|
.overriddenp = &glob_conf.period_size_in_overridden
|
|
},
|
|
{
|
|
.name = "ADC_BUFFER_SIZE",
|
|
.tag = AUD_OPT_INT,
|
|
.valp = &glob_conf.buffer_size_in,
|
|
.descr = "ADC buffer size (0 to go with system default)",
|
|
.overriddenp = &glob_conf.buffer_size_in_overridden
|
|
},
|
|
{
|
|
.name = "THRESHOLD",
|
|
.tag = AUD_OPT_INT,
|
|
.valp = &glob_conf.threshold,
|
|
.descr = "(undocumented)"
|
|
},
|
|
{
|
|
.name = "DAC_DEV",
|
|
.tag = AUD_OPT_STR,
|
|
.valp = &glob_conf.pcm_name_out,
|
|
.descr = "DAC device name (for instance dmix)"
|
|
},
|
|
{
|
|
.name = "ADC_DEV",
|
|
.tag = AUD_OPT_STR,
|
|
.valp = &glob_conf.pcm_name_in,
|
|
.descr = "ADC device name"
|
|
},
|
|
{ /* End of list */ }
|
|
};
|
|
|
|
static struct audio_pcm_ops alsa_pcm_ops = {
|
|
.init_out = alsa_init_out,
|
|
.fini_out = alsa_fini_out,
|
|
.run_out = alsa_run_out,
|
|
.write = alsa_write,
|
|
.ctl_out = alsa_ctl_out,
|
|
|
|
.init_in = alsa_init_in,
|
|
.fini_in = alsa_fini_in,
|
|
.run_in = alsa_run_in,
|
|
.read = alsa_read,
|
|
.ctl_in = alsa_ctl_in,
|
|
};
|
|
|
|
struct audio_driver alsa_audio_driver = {
|
|
.name = "alsa",
|
|
.descr = "ALSA http://www.alsa-project.org",
|
|
.options = alsa_options,
|
|
.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)
|
|
};
|