mirror of https://github.com/mpv-player/mpv
324 lines
8.7 KiB
C
324 lines
8.7 KiB
C
/*
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* Copyright (c) 2008 Alexandre Ratchov <alex@caoua.org>
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* Copyright (c) 2013 Christian Neukirchen <chneukirchen@gmail.com>
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* Copyright (c) 2020 Rozhuk Ivan <rozhuk.im@gmail.com>
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* Copyright (c) 2021 Andrew Krasavin <noiseless-ak@yandex.ru>
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*
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* Permission to use, copy, modify, and distribute this software for any
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* purpose with or without fee is hereby granted, provided that the above
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* copyright notice and this permission notice appear in all copies.
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*
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* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
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* WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
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* ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
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* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
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* ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
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* OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
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*/
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#include <sys/types.h>
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#include <poll.h>
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#include <errno.h>
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#include <sndio.h>
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#include "config.h"
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#include "options/m_option.h"
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#include "common/msg.h"
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#include "audio/format.h"
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#include "ao.h"
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#include "internal.h"
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struct priv {
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struct sio_hdl *hdl;
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struct sio_par par;
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int delay;
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bool playing;
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int vol;
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int havevol;
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struct pollfd *pfd;
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};
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static const struct mp_chmap sndio_layouts[] = {
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{0}, /* empty */
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{1, {MP_SPEAKER_ID_FL}}, /* mono */
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MP_CHMAP2(FL, FR), /* stereo */
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{0}, /* 2.1 */
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MP_CHMAP4(FL, FR, BL, BR), /* 4.0 */
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{0}, /* 5.0 */
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MP_CHMAP6(FL, FR, BL, BR, FC, LFE), /* 5.1 */
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{0}, /* 6.1 */
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MP_CHMAP8(FL, FR, BL, BR, FC, LFE, SL, SR), /* 7.1 */
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/* Above is the fixed channel assignment for sndio, since we need to
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* fill all channels and cannot insert silence, not all layouts are
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* supported.
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* NOTE: MP_SPEAKER_ID_NA could be used to add padding channels. */
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};
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static void uninit(struct ao *ao);
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/* Make libsndio call movecb(). */
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static void process_events(struct ao *ao)
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{
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struct priv *p = ao->priv;
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int n = sio_pollfd(p->hdl, p->pfd, POLLOUT);
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while (poll(p->pfd, n, 0) < 0 && errno == EINTR);
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sio_revents(p->hdl, p->pfd);
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}
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/* Call-back invoked to notify of the hardware position. */
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static void movecb(void *addr, int delta)
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{
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struct ao *ao = addr;
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struct priv *p = ao->priv;
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p->delay -= delta;
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}
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/* Call-back invoked to notify about volume changes. */
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static void volcb(void *addr, unsigned newvol)
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{
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struct ao *ao = addr;
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struct priv *p = ao->priv;
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p->vol = newvol;
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}
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static int init(struct ao *ao)
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{
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struct priv *p = ao->priv;
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struct mp_chmap_sel sel = {0};
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size_t i;
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struct af_to_par {
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int format, bits, sig;
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};
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static const struct af_to_par af_to_par[] = {
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{AF_FORMAT_U8, 8, 0},
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{AF_FORMAT_S16, 16, 1},
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{AF_FORMAT_S32, 32, 1},
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};
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const struct af_to_par *ap;
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const char *device = ((ao->device) ? ao->device : SIO_DEVANY);
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/* Opening device. */
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MP_VERBOSE(ao, "Using '%s' audio device.\n", device);
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p->hdl = sio_open(device, SIO_PLAY, 0);
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if (p->hdl == NULL) {
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MP_ERR(ao, "Can't open audio device %s.\n", device);
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goto err_out;
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}
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sio_initpar(&p->par);
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/* Selecting sound format. */
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ao->format = af_fmt_from_planar(ao->format);
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p->par.bits = 16;
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p->par.sig = 1;
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p->par.le = SIO_LE_NATIVE;
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for (i = 0; i < MP_ARRAY_SIZE(af_to_par); i++) {
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ap = &af_to_par[i];
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if (ap->format == ao->format) {
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p->par.bits = ap->bits;
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p->par.sig = ap->sig;
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break;
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}
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}
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p->par.rate = ao->samplerate;
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/* Channels count. */
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for (i = 0; i < MP_ARRAY_SIZE(sndio_layouts); i++) {
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mp_chmap_sel_add_map(&sel, &sndio_layouts[i]);
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}
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if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
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goto err_out;
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p->par.pchan = ao->channels.num;
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p->par.appbufsz = p->par.rate * 250 / 1000; /* 250ms buffer */
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p->par.round = p->par.rate * 10 / 1000; /* 10ms block size */
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if (!sio_setpar(p->hdl, &p->par)) {
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MP_ERR(ao, "couldn't set params\n");
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goto err_out;
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}
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/* Get current sound params. */
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if (!sio_getpar(p->hdl, &p->par)) {
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MP_ERR(ao, "couldn't get params\n");
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goto err_out;
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}
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if (p->par.bps > 1 && p->par.le != SIO_LE_NATIVE) {
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MP_ERR(ao, "swapped endian output not supported\n");
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goto err_out;
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}
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/* Update sound params. */
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if (p->par.bits == 8 && p->par.bps == 1 && !p->par.sig) {
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ao->format = AF_FORMAT_U8;
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} else if (p->par.bits == 16 && p->par.bps == 2 && p->par.sig) {
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ao->format = AF_FORMAT_S16;
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} else if ((p->par.bits == 32 || p->par.msb) && p->par.bps == 4 && p->par.sig) {
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ao->format = AF_FORMAT_S32;
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} else {
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MP_ERR(ao, "couldn't set format\n");
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goto err_out;
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}
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p->havevol = sio_onvol(p->hdl, volcb, ao);
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sio_onmove(p->hdl, movecb, ao);
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p->pfd = talloc_array_ptrtype(p, p->pfd, sio_nfds(p->hdl));
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if (!p->pfd)
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goto err_out;
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ao->device_buffer = p->par.bufsz;
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MP_VERBOSE(ao, "bufsz = %i, appbufsz = %i, round = %i\n",
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p->par.bufsz, p->par.appbufsz, p->par.round);
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p->delay = 0;
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p->playing = false;
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if (!sio_start(p->hdl)) {
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MP_ERR(ao, "start: sio_start() fail.\n");
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goto err_out;
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}
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return 0;
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err_out:
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uninit(ao);
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return -1;
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}
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static void uninit(struct ao *ao)
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{
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struct priv *p = ao->priv;
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if (p->hdl) {
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sio_close(p->hdl);
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p->hdl = NULL;
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}
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p->pfd = NULL;
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p->playing = false;
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}
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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struct priv *p = ao->priv;
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float *vol = arg;
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switch (cmd) {
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case AOCONTROL_GET_VOLUME:
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if (!p->havevol)
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return CONTROL_FALSE;
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*vol = p->vol * 100 / SIO_MAXVOL;
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break;
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case AOCONTROL_SET_VOLUME:
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if (!p->havevol)
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return CONTROL_FALSE;
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sio_setvol(p->hdl, *vol * SIO_MAXVOL / 100);
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break;
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default:
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return CONTROL_UNKNOWN;
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}
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return CONTROL_OK;
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}
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static void reset(struct ao *ao)
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{
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struct priv *p = ao->priv;
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if (p->playing) {
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p->playing = false;
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#if HAVE_SNDIO_1_9
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if (!sio_flush(p->hdl)) {
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MP_ERR(ao, "reset: couldn't sio_flush()\n");
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#else
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if (!sio_stop(p->hdl)) {
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MP_ERR(ao, "reset: couldn't sio_stop()\n");
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#endif
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}
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p->delay = 0;
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if (!sio_start(p->hdl)) {
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MP_ERR(ao, "reset: sio_start() fail.\n");
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}
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}
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}
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static void start(struct ao *ao)
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{
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struct priv *p = ao->priv;
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p->playing = true;
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process_events(ao);
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}
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static bool audio_write(struct ao *ao, void **data, int samples)
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{
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struct priv *p = ao->priv;
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const size_t size = (samples * ao->sstride);
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size_t rc;
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rc = sio_write(p->hdl, data[0], size);
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if (rc != size) {
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MP_WARN(ao, "audio_write: unexpected partial write: required: %zu, written: %zu.\n",
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size, rc);
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reset(ao);
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p->playing = false;
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return false;
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}
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p->delay += samples;
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return true;
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}
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static void get_state(struct ao *ao, struct mp_pcm_state *state)
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{
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struct priv *p = ao->priv;
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process_events(ao);
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/* how many samples we can play without blocking */
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state->free_samples = ao->device_buffer - p->delay;
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state->free_samples = state->free_samples / p->par.round * p->par.round;
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/* how many samples are already in the buffer to be played */
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state->queued_samples = p->delay;
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/* delay in seconds between first and last sample in buffer */
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state->delay = p->delay / (double)p->par.rate;
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/* report unexpected EOF / underrun */
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if ((state->queued_samples &&
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(state->queued_samples < state->free_samples) &&
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p->playing) || sio_eof(p->hdl))
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{
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MP_VERBOSE(ao, "get_state: EOF/underrun detected.\n");
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MP_VERBOSE(ao, "get_state: free: %d, queued: %d, delay: %lf\n", \
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state->free_samples, state->queued_samples, state->delay);
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p->playing = false;
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state->playing = p->playing;
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ao_wakeup(ao);
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} else {
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state->playing = p->playing;
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}
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}
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const struct ao_driver audio_out_sndio = {
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.name = "sndio",
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.description = "sndio audio output",
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.init = init,
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.uninit = uninit,
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.control = control,
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.reset = reset,
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.start = start,
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.write = audio_write,
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.get_state = get_state,
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.priv_size = sizeof(struct priv),
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};
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