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b016760a28
mpv/audio/out/ao_alsa.c
wm4 4cb5e53ada ao_alsa: drop use of AF_FORMAT_S24 
Instead of the infrastructure added in the previous commit to do the
conversion within the AO.

If this is used, and snd_pcm_status_get_avail() returns more frames than
snd_pcm_write*() actually accepts, you will get some nice audio
corruption.

Also, this mutates the data passed via play(), which is rather fishy,
but sort of doesn't matter for now. Surely this will cause unintended
bugs and WTFs.
2017-07-07 17:56:18 +02:00

1239 lines
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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 General Public License as published by
* the Free Software Foundation; either version 2 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 General Public License for more details.
*
* You should have received a copy of the GNU 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 <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>
#define HAVE_CHMAP_API \
(defined(SND_CHMAP_API_VERSION) && SND_CHMAP_API_VERSION >= (1 << 16))
#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;
};
#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),
{0}
},
.defaults = &(const struct ao_alsa_opts) {
.mixer_device = "default",
.mixer_name = "Master",
.mixer_index = 0,
.ni = 0,
},
.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 BUFFER_TIME 250000 // 250ms
#define FRAGCOUNT 16
#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, SND_MIXER_SCHN_FRONT_LEFT, 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, SND_MIXER_SCHN_FRONT_RIGHT, 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, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
vol->left = (get_vol - pmin) * f_multi;
snd_mixer_selem_get_playback_volume
(elem, SND_MIXER_SCHN_FRONT_RIGHT, &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, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
}
snd_mixer_selem_set_playback_switch
(elem, SND_MIXER_SCHN_FRONT_LEFT, !*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, SND_MIXER_SCHN_FRONT_LEFT, &tmp);
*mute = !tmp;
if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
snd_mixer_selem_get_playback_switch
(elem, SND_MIXER_SCHN_FRONT_RIGHT, &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, int msglevel, 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, msglevel, "%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;
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, MSGL_DEBUG, "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, MSGL_DEBUG, "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, MSGL_DEBUG, "HW params after access:\n", alsa_hwparams);
bool found_format = false;
int try_formats[AF_FORMAT_COUNT];
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, MSGL_DEBUG, "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, MSGL_DEBUG, "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, MSGL_DEBUG, "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 = snd_pcm_hw_params_set_buffer_time_near
(p->alsa, alsa_hwparams, &(unsigned int){BUFFER_TIME}, NULL);
CHECK_ALSA_WARN("Unable to set buffer time near");
if (err >= 0) {
err = snd_pcm_hw_params_set_periods_near
(p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
CHECK_ALSA_WARN("Unable to set periods");
}
if (err < 0)
snd_pcm_hw_params_copy(alsa_hwparams, hwparams_backup);
dump_hw_params(ao, MSGL_V, "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, MSGL_DEBUG, "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");
/* disable underrun reporting */
err = snd_pcm_sw_params_set_stop_threshold
(p->alsa, alsa_swparams, boundary);
CHECK_ALSA_ERROR("Unable to set stop 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;
do {
ao_convert_inplace(&p->convert, data, samples);
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 {
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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