1 files changed, 321 insertions, 0 deletions

diff --git a/audio/out/ao_sndio.c b/audio/out/ao_sndio.c
new file mode 100644
index 0000000..fce7139
--- /dev/null
+++ b/audio/out/ao_sndio.c
@@ -0,0 +1,321 @@
+/*
+ * Copyright (c) 2008 Alexandre Ratchov <alex@caoua.org>
+ * Copyright (c) 2013 Christian Neukirchen <chneukirchen@gmail.com>
+ * Copyright (c) 2020 Rozhuk Ivan <rozhuk.im@gmail.com>
+ * Copyright (c) 2021 Andrew Krasavin <noiseless-ak@yandex.ru>
+ *
+ * Permission to use, copy, modify, and distribute this software for any
+ * purpose with or without fee is hereby granted, provided that the above
+ * copyright notice and this permission notice appear in all copies.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
+ * WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
+ * ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
+ * WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
+ * ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
+ * OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
+ */
+
+#include <sys/types.h>
+#include <poll.h>
+#include <errno.h>
+#include <sndio.h>
+
+#include "options/m_option.h"
+#include "common/msg.h"
+
+#include "audio/format.h"
+#include "ao.h"
+#include "internal.h"
+
+struct priv {
+    struct sio_hdl *hdl;
+    struct sio_par par;
+    int delay;
+    bool playing;
+    int vol;
+    int havevol;
+    struct pollfd *pfd;
+};
+
+
+static const struct mp_chmap sndio_layouts[] = {
+    {0},                                        /* empty */
+    {1, {MP_SPEAKER_ID_FL}},                    /* mono */
+    MP_CHMAP2(FL, FR),                          /* stereo */
+    {0},                                        /* 2.1 */
+    MP_CHMAP4(FL, FR, BL, BR),                  /* 4.0 */
+    {0},                                        /* 5.0 */
+    MP_CHMAP6(FL, FR, BL, BR, FC, LFE),         /* 5.1 */
+    {0},                                        /* 6.1 */
+    MP_CHMAP8(FL, FR, BL, BR, FC, LFE, SL, SR), /* 7.1 */
+    /* Above is the fixed channel assignment for sndio, since we need to
+     * fill all channels and cannot insert silence, not all layouts are
+     * supported.
+     * NOTE: MP_SPEAKER_ID_NA could be used to add padding channels. */
+};
+
+static void uninit(struct ao *ao);
+
+
+/* Make libsndio call movecb(). */
+static void process_events(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    int n = sio_pollfd(p->hdl, p->pfd, POLLOUT);
+    while (poll(p->pfd, n, 0) < 0 && errno == EINTR);
+
+    sio_revents(p->hdl, p->pfd);
+}
+
+/* Call-back invoked to notify of the hardware position. */
+static void movecb(void *addr, int delta)
+{
+    struct ao *ao = addr;
+    struct priv *p = ao->priv;
+
+    p->delay -= delta;
+}
+
+/* Call-back invoked to notify about volume changes. */
+static void volcb(void *addr, unsigned newvol)
+{
+    struct ao *ao = addr;
+    struct priv *p = ao->priv;
+
+    p->vol = newvol;
+}
+
+static int init(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    struct mp_chmap_sel sel = {0};
+    size_t i;
+    struct af_to_par {
+        int format, bits, sig;
+    };
+    static const struct af_to_par af_to_par[] = {
+        {AF_FORMAT_U8,   8, 0},
+        {AF_FORMAT_S16, 16, 1},
+        {AF_FORMAT_S32, 32, 1},
+    };
+    const struct af_to_par *ap;
+    const char *device = ((ao->device) ? ao->device : SIO_DEVANY);
+
+    /* Opening device. */
+    MP_VERBOSE(ao, "Using '%s' audio device.\n", device);
+    p->hdl = sio_open(device, SIO_PLAY, 0);
+    if (p->hdl == NULL) {
+        MP_ERR(ao, "Can't open audio device %s.\n", device);
+        goto err_out;
+    }
+
+    sio_initpar(&p->par);
+
+    /* Selecting sound format. */
+    ao->format = af_fmt_from_planar(ao->format);
+
+    p->par.bits = 16;
+    p->par.sig = 1;
+    p->par.le = SIO_LE_NATIVE;
+    for (i = 0; i < MP_ARRAY_SIZE(af_to_par); i++) {
+        ap = &af_to_par[i];
+        if (ap->format == ao->format) {
+            p->par.bits = ap->bits;
+            p->par.sig = ap->sig;
+            break;
+        }
+    }
+
+    p->par.rate = ao->samplerate;
+
+    /* Channels count. */
+    for (i = 0; i < MP_ARRAY_SIZE(sndio_layouts); i++) {
+        mp_chmap_sel_add_map(&sel, &sndio_layouts[i]);
+    }
+    if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
+        goto err_out;
+
+    p->par.pchan = ao->channels.num;
+    p->par.appbufsz = p->par.rate * 250 / 1000;    /* 250ms buffer */
+    p->par.round = p->par.rate * 10 / 1000;    /*  10ms block size */
+
+    if (!sio_setpar(p->hdl, &p->par)) {
+        MP_ERR(ao, "couldn't set params\n");
+        goto err_out;
+    }
+
+    /* Get current sound params. */
+    if (!sio_getpar(p->hdl, &p->par)) {
+        MP_ERR(ao, "couldn't get params\n");
+        goto err_out;
+    }
+    if (p->par.bps > 1 && p->par.le != SIO_LE_NATIVE) {
+        MP_ERR(ao, "swapped endian output not supported\n");
+        goto err_out;
+    }
+
+    /* Update sound params. */
+    if (p->par.bits == 8 && p->par.bps == 1 && !p->par.sig) {
+        ao->format = AF_FORMAT_U8;
+    } else if (p->par.bits == 16 && p->par.bps == 2 && p->par.sig) {
+        ao->format = AF_FORMAT_S16;
+    } else if ((p->par.bits == 32 || p->par.msb) && p->par.bps == 4 && p->par.sig) {
+        ao->format = AF_FORMAT_S32;
+    } else {
+        MP_ERR(ao, "couldn't set format\n");
+        goto err_out;
+    }
+
+    p->havevol = sio_onvol(p->hdl, volcb, ao);
+    sio_onmove(p->hdl, movecb, ao);
+
+    p->pfd = talloc_array_ptrtype(p, p->pfd, sio_nfds(p->hdl));
+    if (!p->pfd)
+        goto err_out;
+
+    ao->device_buffer = p->par.bufsz;
+    MP_VERBOSE(ao, "bufsz = %i, appbufsz = %i, round = %i\n",
+        p->par.bufsz, p->par.appbufsz, p->par.round);
+
+    p->delay = 0;
+    p->playing = false;
+    if (!sio_start(p->hdl)) {
+        MP_ERR(ao, "start: sio_start() fail.\n");
+        goto err_out;
+    }
+
+    return 0;
+
+err_out:
+    uninit(ao);
+    return -1;
+}
+
+static void uninit(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    if (p->hdl) {
+        sio_close(p->hdl);
+        p->hdl = NULL;
+    }
+    p->pfd = NULL;
+    p->playing = false;
+}
+
+static int control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+    struct priv *p = ao->priv;
+    float *vol = arg;
+
+    switch (cmd) {
+    case AOCONTROL_GET_VOLUME:
+        if (!p->havevol)
+            return CONTROL_FALSE;
+        *vol = p->vol * 100 / SIO_MAXVOL;
+        break;
+    case AOCONTROL_SET_VOLUME:
+        if (!p->havevol)
+            return CONTROL_FALSE;
+        sio_setvol(p->hdl, *vol * SIO_MAXVOL / 100);
+        break;
+    default:
+        return CONTROL_UNKNOWN;
+    }
+    return CONTROL_OK;
+}
+
+static void reset(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    if (p->playing) {
+        p->playing = false;
+
+#if HAVE_SNDIO_1_9
+        if (!sio_flush(p->hdl)) {
+            MP_ERR(ao, "reset: couldn't sio_flush()\n");
+#else
+        if (!sio_stop(p->hdl)) {
+            MP_ERR(ao, "reset: couldn't sio_stop()\n");
+#endif
+        }
+        p->delay = 0;
+        if (!sio_start(p->hdl)) {
+            MP_ERR(ao, "reset: sio_start() fail.\n");
+        }
+    }
+}
+
+static void start(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+
+    p->playing = true;
+    process_events(ao);
+}
+
+static bool audio_write(struct ao *ao, void **data, int samples)
+{
+    struct priv *p = ao->priv;
+    const size_t size = (samples * ao->sstride);
+    size_t rc;
+
+    rc = sio_write(p->hdl, data[0], size);
+    if (rc != size) {
+        MP_WARN(ao, "audio_write: unexpected partial write: required: %zu, written: %zu.\n",
+            size, rc);
+        reset(ao);
+        p->playing = false;
+        return false;
+    }
+    p->delay += samples;
+
+    return true;
+}
+
+static void get_state(struct ao *ao, struct mp_pcm_state *state)
+{
+    struct priv *p = ao->priv;
+
+    process_events(ao);
+
+    /* how many samples we can play without blocking */
+    state->free_samples = ao->device_buffer - p->delay;
+    state->free_samples = state->free_samples / p->par.round * p->par.round;
+    /* how many samples are already in the buffer to be played */
+    state->queued_samples = p->delay;
+    /* delay in seconds between first and last sample in buffer */
+    state->delay = p->delay / (double)p->par.rate;
+
+    /* report unexpected EOF / underrun */
+    if ((state->queued_samples && state->queued_samples &&
+        (state->queued_samples < state->free_samples) &&
+        p->playing) || sio_eof(p->hdl))
+    {
+        MP_VERBOSE(ao, "get_state: EOF/underrun detected.\n");
+        MP_VERBOSE(ao, "get_state: free: %d, queued: %d, delay: %lf\n", \
+                state->free_samples, state->queued_samples, state->delay);
+        p->playing = false;
+        state->playing = p->playing;
+        ao_wakeup_playthread(ao);
+    } else {
+        state->playing = p->playing;
+    }
+}
+
+const struct ao_driver audio_out_sndio = {
+    .name      = "sndio",
+    .description = "sndio audio output",
+    .init      = init,
+    .uninit    = uninit,
+    .control   = control,
+    .reset     = reset,
+    .start     = start,
+    .write     = audio_write,
+    .get_state = get_state,
+    .priv_size = sizeof(struct priv),
+};