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mpv/audio/out/ao_alsa.c
Kevin Mitchell 8f33c65fe0 ao_alsa: add debug messages for format search
2015-10-06 02:24:36 -07:00

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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_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;
bool device_lost;
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;
int cfg_ignore_chmap;
};
#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->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_U8, SND_PCM_FORMAT_U8},
{AF_FORMAT_S16, SND_PCM_FORMAT_S16},
{AF_FORMAT_S32, SND_PCM_FORMAT_S32},
{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_DOUBLE, SND_PCM_FORMAT_FLOAT64},
{AF_FORMAT_UNKNOWN, SND_PCM_FORMAT_UNKNOWN},
};
static int find_alsa_format(int af_format)
{
if (af_fmt_is_spdif(af_format)) {
if (af_format == AF_FORMAT_S_MP3) {
return SND_PCM_FORMAT_MPEG;
} else {
return SND_PCM_FORMAT_S16;
}
}
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_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_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 int 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 -1;
dst->num = src->channels;
for (int c = 0; c < dst->num; c++)
dst->speaker[c] = find_mp_channel(src->pos[c]);
return 0;
}
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++) {
struct mp_chmap entry;
mp_chmap_from_alsa(&entry, &maps[i]->map);
if (mp_chmap_is_valid(&entry)) {
if (maps[i]->type == SND_CHMAP_TYPE_VAR)
mp_chmap_reorder_norm(&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;
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, 0);
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")) {
ac3_device = append_params(tmp, "iec958", params);
MP_VERBOSE(ao, "got error %d; opening iec fallback device '%s'\n",
err, ac3_device);
err = snd_pcm_open
(&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, 0);
}
}
talloc_free(tmp);
} else {
MP_VERBOSE(ao, "opening device '%s'\n", device);
err = snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, 0);
}
return err;
}
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, bool second_try)
{
struct priv *p = ao->priv;
int err;
const char *device = "default";
if (ao->device)
device = ao->device;
if (p->cfg_device && p->cfg_device[0])
device = p->cfg_device;
err = try_open_device(ao, device);
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");
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]; n++) {
p->alsa_fmt = find_alsa_format(try_formats[n]);
MP_DBG(ao, "Trying format %s\n", af_fmt_to_str(try_formats[n]));
if (snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt) >= 0) {
ao->format = try_formats[n];
found_format = true;
MP_DBG(ao, "Selectedformat %s\n", af_fmt_to_str(ao->format));
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");
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 (af_fmt_is_spdif(ao->format) || p->cfg_ignore_chmap) {
dev_chmap.num = 0; // disable chmap API
} else if (dev_chmap.num == 1 && dev_chmap.speaker[0] == MP_SPEAKER_ID_FC) {
// As yet another ALSA API inconsistency, mono is not reported correctly.
dev_chmap.num = 0;
} else 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) {
int req = ao->channels.num;
mp_chmap_from_channels_alsa(&ao->channels, num_channels);
if (!mp_chmap_is_valid(&ao->channels))
mp_chmap_from_channels(&ao->channels, 2);
MP_ERR(ao, "Asked for %d channels, got %d - fallback to %s.\n", req,
num_channels, mp_chmap_to_str(&ao->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]);
// mpv and ALSA use different conventions for mono
if (dev_chmap.num == 1 && dev_chmap.speaker[0] == MP_SP(FC))
alsa_chmap->pos[0] = SND_CHMAP_MONO;
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);
}
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;
mp_chmap_from_alsa(&chmap, alsa_chmap);
MP_VERBOSE(ao, "which we understand as: %s\n", mp_chmap_to_str(&chmap));
if (p->cfg_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)) {
// Is it one that contains NA channels?
struct mp_chmap without_na = chmap;
mp_chmap_remove_na(&without_na);
if (mp_chmap_is_valid(&without_na) &&
!mp_chmap_equals(&without_na, &chmap) &&
!mp_chmap_equals(&chmap, &ao->channels) &&
!second_try)
{
// 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,
// 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.
MP_VERBOSE(ao, "Working around braindead ALSA behavior.\n");
err = snd_pcm_close(p->alsa);
p->alsa = NULL;
CHECK_ALSA_ERROR("pcm close error");
ao->channels = without_na;
return INIT_BRAINDEATH;
}
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 {
MP_WARN(ao, "Got unknown channel map from ALSA.\n");
}
// mpv and ALSA use different conventions for mono
if (ao->channels.num == 1) {
MP_VERBOSE(ao, "assuming we actually got MONO from ALSA.\n");
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)
{
struct priv *p = ao->priv;
if (!p->cfg_ni)
ao->format = af_fmt_from_planar(ao->format);
MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());
int r = init_device(ao, false);
if (r == INIT_BRAINDEATH)
r = init_device(ao, true); // 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);
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 (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 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;
resume_device(ao);
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 (!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);
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) {
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);
}
#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),
OPT_FLAG("ignore-chmap", cfg_ignore_chmap, 0),
{0}
},
};
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