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mpv/audio/out/ao_alsa.c
wm4 49df01323e ao_alsa: remove old multichannel method 
The "old" method (before the ALSA channel map API) used device aliases
like "surround51" to set the channel layout. The "interesting" part was
that these devices usually redirect to a hardware device. This means
playing stereo would lead you to the "default" device (dmix), while e.g.
5.1 to "surround51", which automatically takes care of the fact that
dmix can't do 5.1.

This is pretty much nonsense, though. It shouldn't depend on the damn
input media file whether the player is going to use shared access (dmix)
or exclusive access (direct hw device).

As a consequence, by default ao_alsa will do only what dmix can do. If
the user actually wants multichannel, he has to select a suitable hw
device with --audio-device. From there on, the correct speaker mapping
will be ensured via the channel mapping API.

The change is preparation for making multichannel output the default (as
far as supported by the audio output API). Of the common APIs, only ALSA
messes up beyond repair, so I feel like this change is needed.

On ancient alsa-lib versions, only stereo and mono can be played with
this branch.
2014-12-15 16:58:03 +01:00

921 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 MPlayer.
*
* MPlayer 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.
*
* MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#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_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 priv {
snd_pcm_t *alsa;
snd_pcm_format_t alsa_fmt;
int can_pause;
snd_pcm_sframes_t prepause_frames;
double delay_before_pause;
int buffersize; // in frames
int outburst; // in frames
char *cfg_device;
char *cfg_mixer_device;
char *cfg_mixer_name;
int cfg_mixer_index;
int cfg_resample;
int cfg_ni;
};
#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)
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_FORMAT_IS_SPECIAL(ao->format))
return CONTROL_FALSE;
snd_mixer_selem_id_alloca(&sid);
snd_mixer_selem_id_set_index(sid, p->cfg_mixer_index);
snd_mixer_selem_id_set_name(sid, p->cfg_mixer_name);
err = snd_mixer_open(&handle, 0);
CHECK_ALSA_ERROR("Mixer open error");
err = snd_mixer_attach(handle, p->cfg_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;
}
static const int mp_to_alsa_format[][2] = {
{AF_FORMAT_S8, SND_PCM_FORMAT_S8},
{AF_FORMAT_U8, SND_PCM_FORMAT_U8},
{AF_FORMAT_U16, SND_PCM_FORMAT_U16},
{AF_FORMAT_S16, SND_PCM_FORMAT_S16},
{AF_FORMAT_U32, SND_PCM_FORMAT_U32},
{AF_FORMAT_S32, SND_PCM_FORMAT_S32},
{AF_FORMAT_U24,
MP_SELECT_LE_BE(SND_PCM_FORMAT_U24_3LE, SND_PCM_FORMAT_U24_3BE)},
{AF_FORMAT_S24,
MP_SELECT_LE_BE(SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S24_3BE)},
{AF_FORMAT_FLOAT, SND_PCM_FORMAT_FLOAT},
{AF_FORMAT_UNKNOWN, SND_PCM_FORMAT_UNKNOWN},
};
static int find_alsa_format(int af_format)
{
af_format = af_fmt_from_planar(af_format);
for (int n = 0; mp_to_alsa_format[n][0] != AF_FORMAT_UNKNOWN; n++) {
if (mp_to_alsa_format[n][0] == af_format)
return mp_to_alsa_format[n][1];
}
return SND_PCM_FORMAT_UNKNOWN;
}
#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_MONO, MP_SP(FC)},
{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;
}
static int find_alsa_channel(int mp_channel)
{
for (int i = 0; alsa_to_mp_channels[i][1] != MP_SPEAKER_ID_COUNT; i++) {
if (alsa_to_mp_channels[i][1] == mp_channel)
return alsa_to_mp_channels[i][0];
}
return SND_CHMAP_UNKNOWN;
}
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)
return false;
for (int i = 0; maps[i] != NULL; i++) {
if (maps[i]->map.channels > MP_NUM_CHANNELS) {
MP_VERBOSE(ao, "skipping ALSA channel map with too many channels.\n");
continue;
}
struct mp_chmap entry = {.num = maps[i]->map.channels};
for (int c = 0; c < entry.num; c++)
entry.speaker[c] = find_mp_channel(maps[i]->map.pos[c]);
if (mp_chmap_is_valid(&entry)) {
MP_VERBOSE(ao, "Got supported channel map: %s (type %s)\n",
mp_chmap_to_str(&entry),
snd_pcm_chmap_type_name(maps[i]->type));
mp_chmap_sel_add_map(&chmap_sel, &entry);
} else {
char tmp[128];
if (snd_pcm_chmap_print(&maps[i]->map, sizeof(tmp), tmp) > 0)
MP_VERBOSE(ao, "skipping unknown ALSA channel map: %s\n", tmp);
}
}
snd_pcm_free_chmaps(maps);
return ao_chmap_sel_adjust(ao, &chmap_sel, chmap);
}
#else /* HAVE_CHMAP_API */
static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap)
{
return false;
}
#endif /* else HAVE_CHMAP_API */
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. Since 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)
{
struct priv *p = ao->priv;
if (AF_FORMAT_IS_IEC61937(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);
int err = snd_pcm_open
(&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, 0);
talloc_free(tmp);
if (!err)
return 0;
}
return snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, 0);
}
static void uninit(struct ao *ao)
{
struct priv *p = ao->priv;
if (p->alsa) {
int err;
err = snd_pcm_close(p->alsa);
CHECK_ALSA_ERROR("pcm close error");
}
alsa_error: ;
}
#define INIT_OK 0
#define INIT_ERROR -1
#define INIT_BRAINDEATH -2
static int init_device(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
if (!p->cfg_ni)
ao->format = af_fmt_from_planar(ao->format);
const char *device = "default";
if (AF_FORMAT_IS_IEC61937(ao->format)) {
device = "iec958";
MP_VERBOSE(ao, "playing AC3/iec61937/iec958, %i channels\n",
ao->channels.num);
}
const char *old_dev = device;
if (ao->device)
device = ao->device;
if (p->cfg_device && p->cfg_device[0])
device = p->cfg_device;
bool user_set_device = device != old_dev; // not strcmp()
MP_VERBOSE(ao, "using device: %s\n", device);
MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());
err = try_open_device(ao, device);
if (err == -EBUSY && !user_set_device && strcmp(device, "default") != 0) {
MP_WARN(ao, "Device '%s' busy, retrying default.\n", device);
err = try_open_device(ao, "default");
}
CHECK_ALSA_ERROR("Playback open error");
err = snd_pcm_nonblock(p->alsa, 0);
CHECK_ALSA_WARN("Unable to set blocking mode");
snd_pcm_hw_params_t *alsa_hwparams;
snd_pcm_sw_params_t *alsa_swparams;
snd_pcm_hw_params_alloca(&alsa_hwparams);
snd_pcm_sw_params_alloca(&alsa_swparams);
err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
CHECK_ALSA_ERROR("Unable to get initial parameters");
if (AF_FORMAT_IS_IEC61937(ao->format)) {
if (ao->format == AF_FORMAT_S_MP3) {
p->alsa_fmt = SND_PCM_FORMAT_MPEG;
} else {
p->alsa_fmt = SND_PCM_FORMAT_S16;
}
} else {
p->alsa_fmt = find_alsa_format(ao->format);
}
if (p->alsa_fmt == SND_PCM_FORMAT_UNKNOWN) {
p->alsa_fmt = SND_PCM_FORMAT_S16;
ao->format = AF_FORMAT_S16;
}
err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
if (err < 0) {
if (AF_FORMAT_IS_IEC61937(ao->format))
CHECK_ALSA_ERROR("Unable to set IEC61937 format");
MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n",
af_fmt_to_str(ao->format));
p->alsa_fmt = SND_PCM_FORMAT_S16;
ao->format = AF_FORMAT_S16;
}
err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
CHECK_ALSA_ERROR("Unable to set format");
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");
struct mp_chmap dev_chmap = ao->channels;
if (query_chmaps(ao, &dev_chmap)) {
ao->channels = dev_chmap;
} else {
// Assume only stereo and mono are supported.
mp_chmap_from_channels(&ao->channels, MPMIN(2, dev_chmap.num));
dev_chmap.num = 0;
}
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");
if (num_channels > MP_NUM_CHANNELS) {
MP_FATAL(ao, "Too many audio channels (%d).\n", num_channels);
goto alsa_error;
}
if (num_channels != ao->channels.num) {
MP_ERR(ao, "Couldn't get requested number of channels.\n");
mp_chmap_from_channels_alsa(&ao->channels, num_channels);
}
// Some ALSA drivers have broken delay reporting, so disable the ALSA
// resampling plugin by default.
if (!p->cfg_resample) {
err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
CHECK_ALSA_ERROR("Unable to disable resampling");
}
err = snd_pcm_hw_params_set_rate_near
(p->alsa, alsa_hwparams, &ao->samplerate, NULL);
CHECK_ALSA_ERROR("Unable to set samplerate-2");
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");
err = snd_pcm_hw_params_set_periods_near
(p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
CHECK_ALSA_WARN("Unable to set periods");
/* finally install hardware parameters */
err = snd_pcm_hw_params(p->alsa, alsa_hwparams);
CHECK_ALSA_ERROR("Unable to set hw-parameters");
/* end setting hw-params */
#if HAVE_CHMAP_API
if (mp_chmap_is_valid(&dev_chmap)) {
snd_pcm_chmap_t *alsa_chmap =
calloc(1, sizeof(*alsa_chmap) +
sizeof(alsa_chmap->pos[0]) * dev_chmap.num);
if (!alsa_chmap)
goto alsa_error;
alsa_chmap->channels = dev_chmap.num;
for (int c = 0; c < dev_chmap.num; c++)
alsa_chmap->pos[c] = find_alsa_channel(dev_chmap.speaker[c]);
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) {
MP_WARN(ao, "Device does not support requested channel map\n");
} else {
CHECK_ALSA_WARN("Channel map setup failed");
}
}
snd_pcm_chmap_t *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 = {.num = alsa_chmap->channels};
for (int c = 0; c < chmap.num; c++)
chmap.speaker[c] = find_mp_channel(alsa_chmap->pos[c]);
MP_VERBOSE(ao, "which we understand as: %s\n", mp_chmap_to_str(&chmap));
if (mp_chmap_is_valid(&chmap)) {
if (mp_chmap_equals(&chmap, &ao->channels)) {
MP_VERBOSE(ao, "which is what we requested.\n");
} else if (chmap.num == ao->channels.num) {
MP_VERBOSE(ao, "using the ALSA channel map.\n");
ao->channels = chmap;
} else {
MP_WARN(ao, "ALSA channel map conflicts with channel count!\n");
}
} else {
// Is it one that contains NA channels?
struct mp_chmap chmap2 = {0};
for (int c = 0; c < alsa_chmap->channels; c++) {
int alsa_ch = alsa_chmap->pos[c];
if (alsa_ch != SND_CHMAP_NA)
chmap2.speaker[chmap2.num++] = find_mp_channel(alsa_ch);
}
if (mp_chmap_is_valid(&chmap2)) {
// Sometimes, ALSA will advertise certain chmaps, but it's not
// possible to set them. This can happen with dmix: as of
// alsa 1.0.28, dmix can do stereo only, but advertises the
// surround chmaps of the underlying device. In this case,
// requesting e.g. 5.1 will fail, but it will still allow
// setting 6 channels. 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
// is way too complicated in the general case. 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.
err = snd_pcm_close(p->alsa);
p->alsa = NULL;
CHECK_ALSA_ERROR("pcm close error");
ao->channels = chmap2;
return INIT_BRAINDEATH;
}
MP_WARN(ao, "Got unknown channel map from ALSA.\n");
}
if (ao->channels.num == 1)
ao->channels.speaker[0] = MP_SP(FC);
free(alsa_chmap);
}
#endif
snd_pcm_uframes_t bufsize;
err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize);
CHECK_ALSA_ERROR("Unable to get buffersize");
p->buffersize = bufsize;
MP_VERBOSE(ao, "got buffersize=%i samples\n", p->buffersize);
snd_pcm_uframes_t chunk_size;
err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL);
CHECK_ALSA_ERROR("Unable to get period size");
MP_VERBOSE(ao, "got period size %li\n", chunk_size);
p->outburst = chunk_size;
/* setting software parameters */
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, chunk_size);
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");
/* end setting sw-params */
p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
return INIT_OK;
alsa_error:
uninit(ao);
return INIT_ERROR;
}
static int init(struct ao *ao)
{
int r = init_device(ao);
if (r == INIT_BRAINDEATH)
r = init_device(ao); // retry with normalized channel layout
return r == INIT_OK ? 0 : -1;
}
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);
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 (snd_pcm_state(p->alsa) == SND_PCM_STATE_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;
}
static void audio_pause(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
if (p->can_pause) {
if (snd_pcm_state(p->alsa) == SND_PCM_STATE_RUNNING) {
p->delay_before_pause = get_delay(ao);
err = snd_pcm_pause(p->alsa, 1);
CHECK_ALSA_ERROR("pcm pause error");
}
} else {
MP_VERBOSE(ao, "pause not supported by hardware\n");
if (snd_pcm_delay(p->alsa, &p->prepause_frames) < 0
|| p->prepause_frames < 0)
p->prepause_frames = 0;
p->delay_before_pause = p->prepause_frames / (double)ao->samplerate;
err = snd_pcm_drop(p->alsa);
CHECK_ALSA_ERROR("pcm drop error");
}
alsa_error: ;
}
static void audio_resume(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);
}
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: ;
}
static void reset(struct ao *ao)
{
struct priv *p = ao->priv;
int err;
p->prepause_frames = 0;
p->delay_before_pause = 0;
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 {
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 (res == -ESTRPIPE) { /* suspend */
audio_resume(ao);
} else if (res < 0) {
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);
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;
snd_pcm_poll_descriptors_revents(p->alsa, fds, num_fds, &revents);
CHECK_ALSA_ERROR("cannot read poll events");
if (revents & POLLERR)
return -1;
if (revents & POLLOUT)
return 0;
}
return 0;
alsa_error:
return -1;
}
static void list_devs(struct ao *ao, struct ao_device_list *list)
{
void **hints;
if (snd_device_name_hint(-1, "pcm", &hints) < 0)
return;
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 (io && strcmp(io, "Output") != 0)
continue;
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});
}
snd_device_name_free_hint(hints);
}
#define OPT_BASE_STRUCT struct priv
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),
.priv_defaults = &(const struct priv) {
.cfg_mixer_device = "default",
.cfg_mixer_name = "Master",
.cfg_mixer_index = 0,
.cfg_ni = 0,
},
.options = (const struct m_option[]) {
OPT_STRING("device", cfg_device, 0),
OPT_FLAG("resample", cfg_resample, 0),
OPT_STRING("mixer-device", cfg_mixer_device, 0),
OPT_STRING("mixer-name", cfg_mixer_name, 0),
OPT_INTRANGE("mixer-index", cfg_mixer_index, 0, 0, 99),
OPT_FLAG("non-interleaved", cfg_ni, 0),
{0}
},
};
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