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mpv/audio/out/ao_alsa.c

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/*
* ALSA 0.9.x-1.x audio output driver
*
* Copyright (C) 2004 Alex Beregszaszi
* Zsolt Barat <joy@streamminister.de>
*
* modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de>
* additional AC-3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org>
* 08/22/2002 iec958-init rewritten and merged with common init, zsolt
* 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
* 04/25/2004 printfs converted to mp_msg, Zsolt.
*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <errno.h>
#include <sys/time.h>
#include <stdlib.h>
#include <stdarg.h>
#include <limits.h>
#include <math.h>
#include <string.h>
#include "config.h"
#include "options/options.h"
#include "options/m_config.h"
#include "options/m_option.h"
#include "common/msg.h"
#include "osdep/endian.h"
#include <alsa/asoundlib.h>
#if defined(SND_CHMAP_API_VERSION) && SND_CHMAP_API_VERSION >= (1 << 16)
#define HAVE_CHMAP_API 1
#else
#define HAVE_CHMAP_API 0
#endif
#include "ao.h"
#include "internal.h"
#include "audio/format.h"
struct ao_alsa_opts {
char *mixer_device;
char *mixer_name;
int mixer_index;
int resample;
int ni;
int ignore_chmap;
int buffer_time;
int frags;
};
#define OPT_BASE_STRUCT struct ao_alsa_opts
static const struct m_sub_options ao_alsa_conf = {
.opts = (const struct m_option[]) {
OPT_FLAG("alsa-resample", resample, 0),
OPT_STRING("alsa-mixer-device", mixer_device, 0),
OPT_STRING("alsa-mixer-name", mixer_name, 0),
OPT_INTRANGE("alsa-mixer-index", mixer_index, 0, 0, 99),
OPT_FLAG("alsa-non-interleaved", ni, 0),
OPT_FLAG("alsa-ignore-chmap", ignore_chmap, 0),
OPT_INTRANGE("alsa-buffer-time", buffer_time, 0, 0, INT_MAX),
OPT_INTRANGE("alsa-periods", frags, 0, 0, INT_MAX),
{0}
},
.defaults = &(const struct ao_alsa_opts) {
.mixer_device = "default",
.mixer_name = "Master",
.mixer_index = 0,
.ni = 0,
.buffer_time = 100000,
.frags = 4,
},
.size = sizeof(struct ao_alsa_opts),
};
struct priv {
snd_pcm_t *alsa;
bool device_lost;
snd_pcm_format_t alsa_fmt;
bool can_pause;
bool paused;
snd_pcm_sframes_t prepause_frames;
double delay_before_pause;
snd_pcm_uframes_t buffersize;
snd_pcm_uframes_t outburst;
snd_output_t *output;
struct ao_convert_fmt convert;
struct ao_alsa_opts *opts;
};
#define CHECK_ALSA_ERROR(message) \
do { \
if (err < 0) { \
MP_ERR(ao, "%s: %s\n", (message), snd_strerror(err)); \
goto alsa_error; \
} \
} while (0)
#define CHECK_ALSA_WARN(message) \
do { \
if (err < 0) \
MP_WARN(ao, "%s: %s\n", (message), snd_strerror(err)); \
} while (0)
// Common code for handling ENODEV, which happens if a device gets "lost", and
// can't be used anymore. Returns true if alsa_err is not ENODEV.
static bool check_device_present(struct ao *ao, int alsa_err)
{
struct priv *p = ao->priv;
if (alsa_err != -ENODEV)
return true;
if (!p->device_lost) {
MP_WARN(ao, "Device lost, trying to recover...\n");
ao_request_reload(ao);
p->device_lost = true;
}
return false;
}
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct priv *p = ao->priv;
snd_mixer_t *handle = NULL;
switch (cmd) {
case AOCONTROL_GET_MUTE:
case AOCONTROL_SET_MUTE:
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME:
{
int err;
snd_mixer_elem_t *elem;
snd_mixer_selem_id_t *sid;
long pmin, pmax;
long get_vol, set_vol;
float f_multi;
if (!af_fmt_is_pcm(ao->format))
return CONTROL_FALSE;
snd_mixer_selem_id_alloca(&sid);
snd_mixer_selem_id_set_index(sid, p->opts->mixer_index);
snd_mixer_selem_id_set_name(sid, p->opts->mixer_name);
err = snd_mixer_open(&handle, 0);
CHECK_ALSA_ERROR("Mixer open error");
err = snd_mixer_attach(handle, p->opts->mixer_device);
CHECK_ALSA_ERROR("Mixer attach error");
err = snd_mixer_selem_register(handle, NULL, NULL);
CHECK_ALSA_ERROR("Mixer register error");
err = snd_mixer_load(handle);
CHECK_ALSA_ERROR("Mixer load error");
elem = snd_mixer_find_selem(handle, sid);
if (!elem) {
MP_VERBOSE(ao, "Unable to find simple control '%s',%i.\n",
snd_mixer_selem_id_get_name(sid),
snd_mixer_selem_id_get_index(sid));
goto alsa_error;
}
snd_mixer_selem_get_playback_volume_range(elem, &pmin, &pmax);
f_multi = (100 / (float)(pmax - pmin));
switch (cmd) {
case AOCONTROL_SET_VOLUME: {
ao_control_vol_t *vol = arg;
set_vol = vol->left / f_multi + pmin + 0.5;
err = snd_mixer_selem_set_playback_volume(elem, 0, set_vol);
CHECK_ALSA_ERROR("Error setting left channel");
MP_DBG(ao, "left=%li, ", set_vol);
set_vol = vol->right / f_multi + pmin + 0.5;
err = snd_mixer_selem_set_playback_volume(elem, 1, set_vol);
CHECK_ALSA_ERROR("Error setting right channel");
MP_DBG(ao, "right=%li, pmin=%li, pmax=%li, mult=%f\n",
set_vol, pmin, pmax, f_multi);
break;
}
case AOCONTROL_GET_VOLUME: {
ao_control_vol_t *vol = arg;
snd_mixer_selem_get_playback_volume(elem, 0, &get_vol);
vol->left = (get_vol - pmin) * f_multi;
snd_mixer_selem_get_playback_volume(elem, 1, &get_vol);
vol->right = (get_vol - pmin) * f_multi;
MP_DBG(ao, "left=%f, right=%f\n", vol->left, vol->right);
break;
}
case AOCONTROL_SET_MUTE: {
bool *mute = arg;
if (!snd_mixer_selem_has_playback_switch(elem))
goto alsa_error;
if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
snd_mixer_selem_set_playback_switch(elem, 1, !*mute);
}
snd_mixer_selem_set_playback_switch(elem, 0, !*mute);
break;
}
case AOCONTROL_GET_MUTE: {
bool *mute = arg;
if (!snd_mixer_selem_has_playback_switch(elem))
goto alsa_error;
int tmp = 1;
snd_mixer_selem_get_playback_switch(elem, 0, &tmp);
*mute = !tmp;
if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
snd_mixer_selem_get_playback_switch(elem, 1, &tmp);
*mute &= !tmp;
}
break;
}
}
snd_mixer_close(handle);
return CONTROL_OK;
}
} //end switch
return CONTROL_UNKNOWN;
alsa_error:
if (handle)
snd_mixer_close(handle);
return CONTROL_ERROR;
}
struct alsa_fmt {
int mp_format;
int alsa_format;
int bits; // alsa format full sample size (optional)
int pad_msb; // how many MSB bits are 0 (optional)
};
// Entries that have the same mp_format must be:
// 1. consecutive
// 2. sorted by preferred format (worst comes last)
static const struct alsa_fmt mp_alsa_formats[] = {
{AF_FORMAT_U8, SND_PCM_FORMAT_U8},
{AF_FORMAT_S16, SND_PCM_FORMAT_S16},
{AF_FORMAT_S32, SND_PCM_FORMAT_S32},
{AF_FORMAT_S32, SND_PCM_FORMAT_S24, .bits = 32, .pad_msb = 8},
{AF_FORMAT_S32,
MP_SELECT_LE_BE(SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S24_3BE),
.bits = 24, .pad_msb = 0},
{AF_FORMAT_FLOAT, SND_PCM_FORMAT_FLOAT},
{AF_FORMAT_DOUBLE, SND_PCM_FORMAT_FLOAT64},
{0},
};
static const struct alsa_fmt *find_alsa_format(int mp_format)
{
for (int n = 0; mp_alsa_formats[n].mp_format; n++) {
if (mp_alsa_formats[n].mp_format == mp_format)
return &mp_alsa_formats[n];
}
return NULL;
}
#if HAVE_CHMAP_API
static const int alsa_to_mp_channels[][2] = {
{SND_CHMAP_FL, MP_SP(FL)},
{SND_CHMAP_FR, MP_SP(FR)},
{SND_CHMAP_RL, MP_SP(BL)},
{SND_CHMAP_RR, MP_SP(BR)},
{SND_CHMAP_FC, MP_SP(FC)},
{SND_CHMAP_LFE, MP_SP(LFE)},
{SND_CHMAP_SL, MP_SP(SL)},
{SND_CHMAP_SR, MP_SP(SR)},
{SND_CHMAP_RC, MP_SP(BC)},
{SND_CHMAP_FLC, MP_SP(FLC)},
{SND_CHMAP_FRC, MP_SP(FRC)},
{SND_CHMAP_FLW, MP_SP(WL)},
{SND_CHMAP_FRW, MP_SP(WR)},
{SND_CHMAP_TC, MP_SP(TC)},
{SND_CHMAP_TFL, MP_SP(TFL)},
{SND_CHMAP_TFR, MP_SP(TFR)},
{SND_CHMAP_TFC, MP_SP(TFC)},
{SND_CHMAP_TRL, MP_SP(TBL)},
{SND_CHMAP_TRR, MP_SP(TBR)},
{SND_CHMAP_TRC, MP_SP(TBC)},
{SND_CHMAP_RRC, MP_SP(SDR)},
{SND_CHMAP_RLC, MP_SP(SDL)},
{SND_CHMAP_MONO, MP_SP(FC)},
{SND_CHMAP_NA, MP_SPEAKER_ID_NA},
{SND_CHMAP_UNKNOWN, MP_SPEAKER_ID_NA},
{SND_CHMAP_LAST, MP_SPEAKER_ID_COUNT}
};
static int find_mp_channel(int alsa_channel)
{
for (int i = 0; alsa_to_mp_channels[i][1] != MP_SPEAKER_ID_COUNT; i++) {
if (alsa_to_mp_channels[i][0] == alsa_channel)
return alsa_to_mp_channels[i][1];
}
return MP_SPEAKER_ID_COUNT;
}
#define CHMAP(n, ...) &(struct mp_chmap) MP_CONCAT(MP_CHMAP, n) (__VA_ARGS__)
// Replace each channel in a with b (a->num == b->num)
static void replace_submap(struct mp_chmap *dst, struct mp_chmap *a,
struct mp_chmap *b)
{
struct mp_chmap t = *dst;
if (!mp_chmap_is_valid(&t) || mp_chmap_diffn(a, &t) != 0)
return;
assert(a->num == b->num);
for (int n = 0; n < t.num; n++) {
for (int i = 0; i < a->num; i++) {
if (t.speaker[n] == a->speaker[i]) {
t.speaker[n] = b->speaker[i];
break;
}
}
}
if (mp_chmap_is_valid(&t))
*dst = t;
}
static bool mp_chmap_from_alsa(struct mp_chmap *dst, snd_pcm_chmap_t *src)
{
*dst = (struct mp_chmap) {0};
if (src->channels > MP_NUM_CHANNELS)
return false;
dst->num = src->channels;
for (int c = 0; c < dst->num; c++)
dst->speaker[c] = find_mp_channel(src->pos[c]);
// Assume anything with 1 channel is mono.
if (dst->num == 1)
dst->speaker[0] = MP_SP(FC);
// Remap weird Intel HDA HDMI 7.1 layouts correctly.
replace_submap(dst, CHMAP(6, FL, FR, BL, BR, SDL, SDR),
CHMAP(6, FL, FR, SL, SR, BL, BR));
return mp_chmap_is_valid(dst);
}
static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
{
struct priv *p = ao->priv;
struct mp_chmap_sel chmap_sel = {.tmp = p};
snd_pcm_chmap_query_t **maps = snd_pcm_query_chmaps(p->alsa);
if (!maps) {
MP_VERBOSE(ao, "snd_pcm_query_chmaps() returned NULL\n");
return false;
}
for (int i = 0; maps[i] != NULL; i++) {
char aname[128];
if (snd_pcm_chmap_print(&maps[i]->map, sizeof(aname), aname) <= 0)
aname[0] = '\0';
struct mp_chmap entry;
if (mp_chmap_from_alsa(&entry, &maps[i]->map)) {
struct mp_chmap reorder = entry;
mp_chmap_reorder_norm(&reorder);
MP_DBG(ao, "got ALSA chmap: %s (%s) -> %s", aname,
snd_pcm_chmap_type_name(maps[i]->type),
mp_chmap_to_str(&entry));
if (!mp_chmap_equals(&entry, &reorder))
MP_DBG(ao, " -> %s", mp_chmap_to_str(&reorder));
MP_DBG(ao, "\n");
struct mp_chmap final =
maps[i]->type == SND_CHMAP_TYPE_VAR ? reorder : entry;
mp_chmap_sel_add_map(&chmap_sel, &final);
} else {
MP_VERBOSE(ao, "skipping unknown ALSA channel map: %s\n", aname);
}
}
snd_pcm_free_chmaps(maps);
return ao_chmap_sel_adjust2(ao, &chmap_sel, chmap, false);
}
// Map back our selected channel layout to an ALSA one. This is done this way so
// that our ALSA->mp_chmap mapping function only has to go one way.
// The return value is to be freed with free().
static snd_pcm_chmap_t *map_back_chmap(struct ao *ao, struct mp_chmap *chmap)
{
struct priv *p = ao->priv;
if (!mp_chmap_is_valid(chmap))
return NULL;
snd_pcm_chmap_query_t **maps = snd_pcm_query_chmaps(p->alsa);
if (!maps)
return NULL;
snd_pcm_chmap_t *alsa_chmap = NULL;
for (int i = 0; maps[i] != NULL; i++) {
struct mp_chmap entry;
if (!mp_chmap_from_alsa(&entry, &maps[i]->map))
continue;
if (mp_chmap_equals(chmap, &entry) ||
(mp_chmap_equals_reordered(chmap, &entry) &&
maps[i]->type == SND_CHMAP_TYPE_VAR))
{
alsa_chmap = calloc(1, sizeof(*alsa_chmap) +
sizeof(alsa_chmap->pos[0]) * entry.num);
if (!alsa_chmap)
break;
alsa_chmap->channels = entry.num;
// Undo if mp_chmap_reorder() was called on the result.
int reorder[MP_NUM_CHANNELS];
mp_chmap_get_reorder(reorder, chmap, &entry);
for (int n = 0; n < entry.num; n++)
alsa_chmap->pos[n] = maps[i]->map.pos[reorder[n]];
break;
}
}
snd_pcm_free_chmaps(maps);
return alsa_chmap;
}
static int set_chmap(struct ao *ao, struct mp_chmap *dev_chmap, int num_channels)
{
struct priv *p = ao->priv;
int err;
snd_pcm_chmap_t *alsa_chmap = map_back_chmap(ao, dev_chmap);
if (alsa_chmap) {
char tmp[128];
if (snd_pcm_chmap_print(alsa_chmap, sizeof(tmp), tmp) > 0)
MP_VERBOSE(ao, "trying to set ALSA channel map: %s\n", tmp);
err = snd_pcm_set_chmap(p->alsa, alsa_chmap);
if (err == -ENXIO) {
// A device my not be able to set any channel map, even channel maps
// that were reported as supported. This is either because the ALSA
// device is broken (dmix), or because the driver has only 1
// channel map per channel count, and setting the map is not needed.
MP_VERBOSE(ao, "device returned ENXIO when setting channel map %s\n",
mp_chmap_to_str(dev_chmap));
} else {
CHECK_ALSA_WARN("Channel map setup failed");
}
free(alsa_chmap);
}
alsa_chmap = snd_pcm_get_chmap(p->alsa);
if (alsa_chmap) {
char tmp[128];
if (snd_pcm_chmap_print(alsa_chmap, sizeof(tmp), tmp) > 0)
MP_VERBOSE(ao, "channel map reported by ALSA: %s\n", tmp);
struct mp_chmap chmap;
mp_chmap_from_alsa(&chmap, alsa_chmap);
MP_VERBOSE(ao, "which we understand as: %s\n", mp_chmap_to_str(&chmap));
if (p->opts->ignore_chmap) {
MP_VERBOSE(ao, "user set ignore-chmap; ignoring the channel map.\n");
} else if (af_fmt_is_spdif(ao->format)) {
MP_VERBOSE(ao, "using spdif passthrough; ignoring the channel map.\n");
} else if (!mp_chmap_is_valid(&chmap)) {
MP_WARN(ao, "Got unknown channel map from ALSA.\n");
} else if (chmap.num != num_channels) {
MP_WARN(ao, "ALSA channel map conflicts with channel count!\n");
} else {
if (mp_chmap_equals(&chmap, &ao->channels)) {
MP_VERBOSE(ao, "which is what we requested.\n");
} else if (!mp_chmap_is_valid(dev_chmap)) {
MP_VERBOSE(ao, "ignoring the ALSA channel map.\n");
} else {
MP_VERBOSE(ao, "using the ALSA channel map.\n");
ao->channels = chmap;
}
}
free(alsa_chmap);
}
return 0;
}
#else /* HAVE_CHMAP_API */
static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
{
return false;
}
static int set_chmap(struct ao *ao, struct mp_chmap *dev_chmap, int num_channels)
{
return 0;
}
#endif /* else HAVE_CHMAP_API */
static void dump_hw_params(struct ao *ao, const char *msg,
snd_pcm_hw_params_t *hw_params)
{
struct priv *p = ao->priv;
int err;
err = snd_pcm_hw_params_dump(hw_params, p->output);
CHECK_ALSA_WARN("Dump hwparams error");
char *tmp = NULL;
size_t tmp_s = snd_output_buffer_string(p->output, &tmp);
if (tmp)
mp_msg(ao->log, MSGL_DEBUG, "%s---\n%.*s---\n", msg, (int)tmp_s, tmp);
snd_output_flush(p->output);
}
static int map_iec958_srate(int srate)
{
switch (srate) {
case 44100: return IEC958_AES3_CON_FS_44100;
case 48000: return IEC958_AES3_CON_FS_48000;
case 32000: return IEC958_AES3_CON_FS_32000;
case 22050: return IEC958_AES3_CON_FS_22050;
case 24000: return IEC958_AES3_CON_FS_24000;
case 88200: return IEC958_AES3_CON_FS_88200;
case 768000: return IEC958_AES3_CON_FS_768000;
case 96000: return IEC958_AES3_CON_FS_96000;
case 176400: return IEC958_AES3_CON_FS_176400;
case 192000: return IEC958_AES3_CON_FS_192000;
default: return IEC958_AES3_CON_FS_NOTID;
}
}
// ALSA device strings can have parameters. They are usually appended to the
// device name. There can be various forms, and we (sometimes) want to append
// them to unknown device strings, which possibly already include params.
static char *append_params(void *ta_parent, const char *device, const char *p)
{
if (!p || !p[0])
return talloc_strdup(ta_parent, device);
int len = strlen(device);
char *end = strchr(device, ':');
if (!end) {
/* no existing parameters: add it behind device name */
return talloc_asprintf(ta_parent, "%s:%s", device, p);
} else if (end[1] == '\0') {
/* ":" but no parameters */
return talloc_asprintf(ta_parent, "%s%s", device, p);
} else if (end[1] == '{' && device[len - 1] == '}') {
/* parameters in config syntax: add it inside the { } block */
return talloc_asprintf(ta_parent, "%.*s %s}", len - 1, device, p);
} else {
/* a simple list of parameters: add it at the end of the list */
return talloc_asprintf(ta_parent, "%s,%s", device, p);
}
abort();
}
static int try_open_device(struct ao *ao, const char *device, int mode)
{
struct priv *p = ao->priv;
int err;
if (af_fmt_is_spdif(ao->format)) {
void *tmp = talloc_new(NULL);
char *params = talloc_asprintf(tmp,
"AES0=%d,AES1=%d,AES2=0,AES3=%d",
IEC958_AES0_NONAUDIO | IEC958_AES0_PRO_EMPHASIS_NONE,
IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
map_iec958_srate(ao->samplerate));
const char *ac3_device = append_params(tmp, device, params);
MP_VERBOSE(ao, "opening device '%s' => '%s'\n", device, ac3_device);
err = snd_pcm_open(&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, mode);
if (err < 0) {
// Some spdif-capable devices do not accept the AES0 parameter,
// and instead require the iec958 pseudo-device (they will play
// noise otherwise). Unfortunately, ALSA gives us no way to map
// these devices, so try it for the default device only.
bstr dev;
bstr_split_tok(bstr0(device), ":", &dev, &(bstr){0});
if (bstr_equals0(dev, "default")) {
const char *const fallbacks[] = {"hdmi", "iec958", NULL};
for (int n = 0; fallbacks[n]; n++) {
char *ndev = append_params(tmp, fallbacks[n], params);
MP_VERBOSE(ao, "got error '%s'; opening iec fallback "
"device '%s'\n", snd_strerror(err), ndev);
err = snd_pcm_open
(&p->alsa, ndev, SND_PCM_STREAM_PLAYBACK, mode);
if (err >= 0)
break;
}
}
}
talloc_free(tmp);
} else {
MP_VERBOSE(ao, "opening device '%s'\n", device);
err = snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, mode);
}
return err;
}
static void uninit(struct ao *ao)
{
struct priv *p = ao->priv;
if (p->output)
snd_output_close(p->output);
p->output = NULL;
if (p->alsa) {
int err;
err = snd_pcm_close(p->alsa);
p->alsa = NULL;
CHECK_ALSA_ERROR("pcm close error");
}
alsa_error: ;
}
#define INIT_DEVICE_ERR_GENERIC -1
#define INIT_DEVICE_ERR_HWPARAMS -2
static int init_device(struct ao *ao, int mode)
{
struct priv *p = ao->priv;
struct ao_alsa_opts *opts = p->opts;
int ret = INIT_DEVICE_ERR_GENERIC;
char *tmp;
size_t tmp_s;
int err;
p->alsa_fmt = SND_PCM_FORMAT_UNKNOWN;
err = snd_output_buffer_open(&p->output);
CHECK_ALSA_ERROR("Unable to create output buffer");
const char *device = "default";
if (ao->device)
device = ao->device;
err = try_open_device(ao, device, mode);
CHECK_ALSA_ERROR("Playback open error");
err = snd_pcm_dump(p->alsa, p->output);
CHECK_ALSA_WARN("Dump PCM error");
tmp_s = snd_output_buffer_string(p->output, &tmp);
if (tmp)
MP_DBG(ao, "PCM setup:\n---\n%.*s---\n", (int)tmp_s, tmp);
snd_output_flush(p->output);
err = snd_pcm_nonblock(p->alsa, 0);
CHECK_ALSA_WARN("Unable to set blocking mode");
snd_pcm_hw_params_t *alsa_hwparams;
snd_pcm_hw_params_alloca(&alsa_hwparams);
err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
CHECK_ALSA_ERROR("Unable to get initial parameters");
dump_hw_params(ao, "Start HW params:\n", alsa_hwparams);
// Some ALSA drivers have broken delay reporting, so disable the ALSA
// resampling plugin by default.
if (!p->opts->resample) {
err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
CHECK_ALSA_ERROR("Unable to disable resampling");
}
dump_hw_params(ao, "HW params after rate:\n", alsa_hwparams);
snd_pcm_access_t access = af_fmt_is_planar(ao->format)
? SND_PCM_ACCESS_RW_NONINTERLEAVED
: SND_PCM_ACCESS_RW_INTERLEAVED;
err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access);
if (err < 0 && af_fmt_is_planar(ao->format)) {
ao->format = af_fmt_from_planar(ao->format);
access = SND_PCM_ACCESS_RW_INTERLEAVED;
err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access);
}
CHECK_ALSA_ERROR("Unable to set access type");
dump_hw_params(ao, "HW params after access:\n", alsa_hwparams);
bool found_format = false;
int try_formats[AF_FORMAT_COUNT + 1];
af_get_best_sample_formats(ao->format, try_formats);
for (int n = 0; try_formats[n] && !found_format; n++) {
int mp_format = try_formats[n];
if (af_fmt_is_planar(ao->format) != af_fmt_is_planar(mp_format))
continue; // implied SND_PCM_ACCESS mismatches
int mp_pformat = af_fmt_from_planar(mp_format);
if (af_fmt_is_spdif(mp_pformat))
mp_pformat = AF_FORMAT_S16;
const struct alsa_fmt *fmt = find_alsa_format(mp_pformat);
if (!fmt)
continue;
for (; fmt->mp_format == mp_pformat; fmt++) {
p->alsa_fmt = fmt->alsa_format;
p->convert = (struct ao_convert_fmt){
.src_fmt = mp_format,
.dst_bits = fmt->bits ? fmt->bits : af_fmt_to_bytes(mp_format) * 8,
.pad_msb = fmt->pad_msb,
};
if (!ao_can_convert_inplace(&p->convert))
continue;
MP_VERBOSE(ao, "trying format %s/%d\n", af_fmt_to_str(mp_pformat),
p->alsa_fmt);
if (snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams,
p->alsa_fmt) >= 0)
{
ao->format = mp_format;
found_format = true;
break;
}
}
}
if (!found_format) {
MP_ERR(ao, "Can't find appropriate sample format.\n");
goto alsa_error;
}
err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
CHECK_ALSA_ERROR("Unable to set format");
dump_hw_params(ao, "HW params after format:\n", alsa_hwparams);
// Stereo, or mono if input is 1 channel.
struct mp_chmap reduced;
mp_chmap_from_channels(&reduced, MPMIN(2, ao->channels.num));
struct mp_chmap dev_chmap = {0};
if (!af_fmt_is_spdif(ao->format) && !p->opts->ignore_chmap &&
!mp_chmap_equals(&ao->channels, &reduced))
{
struct mp_chmap res = ao->channels;
if (query_chmaps(ao, &res))
dev_chmap = res;
// Whatever it is, we dumb it down to mono or stereo. Some drivers may
// return things like bl-br, but the user (probably) still wants stereo.
// This also handles the failure case (dev_chmap.num==0).
if (dev_chmap.num <= 2) {
dev_chmap.num = 0;
ao->channels = reduced;
} else if (dev_chmap.num) {
ao->channels = dev_chmap;
}
}
int num_channels = ao->channels.num;
err = snd_pcm_hw_params_set_channels_near
(p->alsa, alsa_hwparams, &num_channels);
CHECK_ALSA_ERROR("Unable to set channels");
dump_hw_params(ao, "HW params after channels:\n", alsa_hwparams);
if (num_channels > MP_NUM_CHANNELS) {
MP_FATAL(ao, "Too many audio channels (%d).\n", num_channels);
goto alsa_error;
}
err = snd_pcm_hw_params_set_rate_near
(p->alsa, alsa_hwparams, &ao->samplerate, NULL);
CHECK_ALSA_ERROR("Unable to set samplerate-2");
dump_hw_params(ao, "HW params after rate-2:\n", alsa_hwparams);
snd_pcm_hw_params_t *hwparams_backup;
snd_pcm_hw_params_alloca(&hwparams_backup);
snd_pcm_hw_params_copy(hwparams_backup, alsa_hwparams);
// Cargo-culted buffer settings; might still be useful for PulseAudio.
err = 0;
if (opts->buffer_time) {
err = snd_pcm_hw_params_set_buffer_time_near
(p->alsa, alsa_hwparams, &(unsigned int){opts->buffer_time}, NULL);
CHECK_ALSA_WARN("Unable to set buffer time near");
}
if (err >= 0 && opts->frags) {
err = snd_pcm_hw_params_set_periods_near
(p->alsa, alsa_hwparams, &(unsigned int){opts->frags}, NULL);
CHECK_ALSA_WARN("Unable to set periods");
}
if (err < 0)
snd_pcm_hw_params_copy(alsa_hwparams, hwparams_backup);
dump_hw_params(ao, "Going to set final HW params:\n", alsa_hwparams);
/* finally install hardware parameters */
err = snd_pcm_hw_params(p->alsa, alsa_hwparams);
ret = INIT_DEVICE_ERR_HWPARAMS;
CHECK_ALSA_ERROR("Unable to set hw-parameters");
ret = INIT_DEVICE_ERR_GENERIC;
dump_hw_params(ao, "Final HW params:\n", alsa_hwparams);
if (set_chmap(ao, &dev_chmap, num_channels) < 0)
goto alsa_error;
if (num_channels != ao->channels.num) {
int req = ao->channels.num;
mp_chmap_from_channels(&ao->channels, MPMIN(2, num_channels));
mp_chmap_fill_na(&ao->channels, num_channels);
MP_ERR(ao, "Asked for %d channels, got %d - fallback to %s.\n", req,
num_channels, mp_chmap_to_str(&ao->channels));
if (num_channels != ao->channels.num) {
MP_FATAL(ao, "mismatching channel counts.\n");
goto alsa_error;
}
}
err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &p->buffersize);
CHECK_ALSA_ERROR("Unable to get buffersize");
err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &p->outburst, NULL);
CHECK_ALSA_ERROR("Unable to get period size");
p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
snd_pcm_sw_params_t *alsa_swparams;
snd_pcm_sw_params_alloca(&alsa_swparams);
err = snd_pcm_sw_params_current(p->alsa, alsa_swparams);
CHECK_ALSA_ERROR("Unable to get sw-parameters");
snd_pcm_uframes_t boundary;
err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary);
CHECK_ALSA_ERROR("Unable to get boundary");
/* start playing when one period has been written */
err = snd_pcm_sw_params_set_start_threshold
(p->alsa, alsa_swparams, p->outburst);
CHECK_ALSA_ERROR("Unable to set start threshold");
/* play silence when there is an underrun */
err = snd_pcm_sw_params_set_silence_size
(p->alsa, alsa_swparams, boundary);
CHECK_ALSA_ERROR("Unable to set silence size");
err = snd_pcm_sw_params(p->alsa, alsa_swparams);
CHECK_ALSA_ERROR("Unable to set sw-parameters");
MP_VERBOSE(ao, "hw pausing supported: %s\n", p->can_pause ? "yes" : "no");
MP_VERBOSE(ao, "buffersize: %d samples\n", (int)p->buffersize);
MP_VERBOSE(ao, "period size: %d samples\n", (int)p->outburst);
ao->device_buffer = p->buffersize;
ao->period_size = p->outburst;
p->convert.channels = ao->channels.num;
return 0;
alsa_error:
uninit(ao);
return ret;
}
static int init(struct ao *ao)
{
struct priv *p = ao->priv;
p->opts = mp_get_config_group(ao, ao->global, &ao_alsa_conf);
if (!p->opts->ni)
ao->format = af_fmt_from_planar(ao->format);
MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());
int mode = 0;
int r = init_device(ao, mode);
if (r == INIT_DEVICE_ERR_HWPARAMS) {
// With some drivers, ALSA appears to be unable to set valid hwparams,
// but they work if at least SND_PCM_NO_AUTO_FORMAT is set. Also, it
// appears you can set this flag only on opening a device, thus there
// is the need to retry opening the device.
MP_WARN(ao, "Attempting to work around even more ALSA bugs...\n");
mode |= SND_PCM_NO_AUTO_CHANNELS | SND_PCM_NO_AUTO_FORMAT |
SND_PCM_NO_AUTO_RESAMPLE;
r = init_device(ao, mode);
}
// Sometimes, ALSA will advertise certain chmaps, but it's not possible to
// set them. This can happen with dmix: as of alsa 1.0.29, dmix can do
// stereo only, but advertises the surround chmaps of the underlying device.
// In this case, e.g. setting 6 channels will succeed, but requesting 5.1
// afterwards will fail. Then it will return something like "FL FR NA NA NA NA"
// as channel map. This means we would have to pad stereo output to 6
// channels with silence, which would require lots of extra processing. You
// can't change the number of channels to 2 either, because the hw params
// are already set! So just fuck it and reopen the device with the chmap
// "cleaned out" of NA entries.
if (r >= 0) {
struct mp_chmap without_na = ao->channels;
mp_chmap_remove_na(&without_na);
if (mp_chmap_is_valid(&without_na) && without_na.num <= 2 &&
ao->channels.num > 2)
{
MP_VERBOSE(ao, "Working around braindead dmix multichannel behavior.\n");
uninit(ao);
ao->channels = without_na;
r = init_device(ao, mode);
}
}
return r;
}
static void drain(struct ao *ao)
{
struct priv *p = ao->priv;
snd_pcm_drain(p->alsa);
}
static int get_space(struct ao *ao)
{
struct priv *p = ao->priv;
snd_pcm_status_t *status;
int err;
snd_pcm_status_alloca(&status);
err = snd_pcm_status(p->alsa, status);
if (!check_device_present(ao, err))
goto alsa_error;
CHECK_ALSA_ERROR("cannot get pcm status");
unsigned space = snd_pcm_status_get_avail(status);
if (space > p->buffersize) // Buffer underrun?
space = p->buffersize;
return space / p->outburst * p->outburst;
alsa_error:
return 0;
}
/* delay in seconds between first and last sample in buffer */
static double get_delay(struct ao *ao)
{
struct priv *p = ao->priv;
snd_pcm_sframes_t delay;
if (p->paused)
return p->delay_before_pause;
if (snd_pcm_delay(p->alsa, &delay) < 0)
return 0;
if (delay < 0) {
/* underrun - move the application pointer forward to catch up */
snd_pcm_forward(p->alsa, -delay);
delay = 0;
}
return delay / (double)ao->samplerate;
}
// For stream-silence mode: replace remaining buffer with silence.
// Tries to cause an instant buffer underrun.
static void soft_reset(struct ao *ao)
{
struct priv *p = ao->priv;
snd_pcm_sframes_t frames = snd_pcm_rewindable(p->alsa);
if (frames > 0 && snd_pcm_state(p->alsa) == SND_PCM_STATE_RUNNING) {
frames = snd_pcm_rewind(p->alsa, frames);
if (frames < 0) {
int err = frames;
CHECK_ALSA_WARN("pcm rewind error");
}
}
}
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
if (p->paused)
return;
p->delay_before_pause = get_delay(ao);
p->prepause_frames = p->delay_before_pause * ao->samplerate;
if (ao->stream_silence) {
soft_reset(ao);
} else if (p->can_pause) {
if (snd_pcm_state(p->alsa) == SND_PCM_STATE_RUNNING) {
err = snd_pcm_pause(p->alsa, 1);
CHECK_ALSA_ERROR("pcm pause error");
p->prepause_frames = 0;
}
} else {
err = snd_pcm_drop(p->alsa);
CHECK_ALSA_ERROR("pcm drop error");
}
p->paused = true;
alsa_error: ;
}
static void resume_device(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
if (snd_pcm_state(p->alsa) == SND_PCM_STATE_SUSPENDED) {
MP_INFO(ao, "PCM in suspend mode, trying to resume.\n");
while ((err = snd_pcm_resume(p->alsa)) == -EAGAIN)
sleep(1);
}
}
static void audio_resume(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
if (!p->paused)
return;
resume_device(ao);
if (ao->stream_silence) {
p->paused = false;
get_delay(ao); // recovers from underrun (as a side-effect)
} else if (p->can_pause) {
if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED) {
err = snd_pcm_pause(p->alsa, 0);
CHECK_ALSA_ERROR("pcm resume error");
}
} else {
MP_VERBOSE(ao, "resume not supported by hardware\n");
err = snd_pcm_prepare(p->alsa);
CHECK_ALSA_ERROR("pcm prepare error");
}
if (p->prepause_frames)
ao_play_silence(ao, p->prepause_frames);
alsa_error: ;
p->paused = false;
}
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
p->paused = false;
p->prepause_frames = 0;
p->delay_before_pause = 0;
if (ao->stream_silence) {
soft_reset(ao);
} else {
err = snd_pcm_drop(p->alsa);
CHECK_ALSA_ERROR("pcm prepare error");
err = snd_pcm_prepare(p->alsa);
CHECK_ALSA_ERROR("pcm prepare error");
}
alsa_error: ;
}
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *p = ao->priv;
snd_pcm_sframes_t res = 0;
if (!(flags & AOPLAY_FINAL_CHUNK))
samples = samples / p->outburst * p->outburst;
if (samples == 0)
return 0;
ao_convert_inplace(&p->convert, data, samples);
do {
if (af_fmt_is_planar(ao->format)) {
res = snd_pcm_writen(p->alsa, data, samples);
} else {
res = snd_pcm_writei(p->alsa, data[0], samples);
}
if (res == -EINTR || res == -EAGAIN) { /* retry */
res = 0;
} else if (!check_device_present(ao, res)) {
goto alsa_error;
} else if (res < 0) {
if (res == -ESTRPIPE) { /* suspend */
resume_device(ao);
} else if (res == -EPIPE) {
// For some reason, writing a smaller fragment at the end
// immediately underruns.
if (!(flags & AOPLAY_FINAL_CHUNK))
MP_WARN(ao, "Device underrun detected.\n");
} else {
MP_ERR(ao, "Write error: %s\n", snd_strerror(res));
}
res = snd_pcm_prepare(p->alsa);
int err = res;
CHECK_ALSA_ERROR("pcm prepare error");
res = 0;
}
} while (res == 0);
p->paused = false;
return res < 0 ? -1 : res;
alsa_error:
return -1;
}
#define MAX_POLL_FDS 20
static int audio_wait(struct ao *ao, pthread_mutex_t *lock)
{
struct priv *p = ao->priv;
int err;
int num_fds = snd_pcm_poll_descriptors_count(p->alsa);
if (num_fds <= 0 || num_fds >= MAX_POLL_FDS)
goto alsa_error;
struct pollfd fds[MAX_POLL_FDS];
err = snd_pcm_poll_descriptors(p->alsa, fds, num_fds);
CHECK_ALSA_ERROR("cannot get pollfds");
while (1) {
int r = ao_wait_poll(ao, fds, num_fds, lock);
if (r)
return r;
unsigned short revents;
err = snd_pcm_poll_descriptors_revents(p->alsa, fds, num_fds, &revents);
CHECK_ALSA_ERROR("cannot read poll events");
if (revents & POLLERR) {
snd_pcm_status_t *status;
snd_pcm_status_alloca(&status);
err = snd_pcm_status(p->alsa, status);
check_device_present(ao, err);
return -1;
}
if (revents & POLLOUT)
return 0;
}
return 0;
alsa_error:
return -1;
}
static bool is_useless_device(char *name)
{
char *crap[] = {"rear", "center_lfe", "side", "pulse", "null", "dsnoop", "hw"};
for (int i = 0; i < MP_ARRAY_SIZE(crap); i++) {
int l = strlen(crap[i]);
if (name && strncmp(name, crap[i], l) == 0 &&
(!name[l] || name[l] == ':'))
return true;
}
// The standard default entry will achieve exactly the same.
if (name && strcmp(name, "default") == 0)
return true;
return false;
}
static void list_devs(struct ao *ao, struct ao_device_list *list)
{
void **hints;
if (snd_device_name_hint(-1, "pcm", &hints) < 0)
return;
ao_device_list_add(list, ao, &(struct ao_device_desc){"", ""});
for (int n = 0; hints[n]; n++) {
char *name = snd_device_name_get_hint(hints[n], "NAME");
char *desc = snd_device_name_get_hint(hints[n], "DESC");
char *io = snd_device_name_get_hint(hints[n], "IOID");
if (!is_useless_device(name) && (!io || strcmp(io, "Output") == 0)) {
char desc2[1024];
snprintf(desc2, sizeof(desc2), "%s", desc ? desc : "");
for (int i = 0; desc2[i]; i++) {
if (desc2[i] == '\n')
desc2[i] = '/';
}
ao_device_list_add(list, ao, &(struct ao_device_desc){name, desc2});
}
free(name);
free(desc);
free(io);
}
snd_device_name_free_hint(hints);
}
const struct ao_driver audio_out_alsa = {
.description = "ALSA audio output",
.name = "alsa",
.init = init,
.uninit = uninit,
.control = control,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = audio_pause,
.resume = audio_resume,
.reset = reset,
.drain = drain,
.wait = audio_wait,
.wakeup = ao_wakeup_poll,
.list_devs = list_devs,
.priv_size = sizeof(struct priv),
.global_opts = &ao_alsa_conf,
};
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