ao_alsa: change sub-options to global options

Same deal as with vo_opengl.

Also edit the outdated information about multichannel output a little.
This commit is contained in:
wm4 committed 2016-09-02 21:21:47 +02:00
1 parent eb14b18a33
commit a85eecfe40
8 files changed
+117 -99

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+1
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@@ -41,6 +41,7 @@ Interface changes
- remove all vo_opengl suboptions. Use global options with the same name
instead, e.g.: --vo=opengl:scale=nearest => --scale=nearest
Some options are prefixed with "opengl-", e.g. --opengl-pbo.
The same change is applied to ao_alsa.
- remove --vo=opengl-hq. Set --profile=opengl-hq instead. Note that this
profile does not force the VO. This means if you use the --vo option to
set another VO, it won't work. But this also means it can be used with
+4 -68
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@@ -34,76 +34,12 @@ Available audio output drivers are:
``alsa`` (Linux only)
ALSA audio output driver
``device=<device>``
Sets the device name. For ac3 output via S/PDIF, use an "iec958" or
"spdif" device, unless you really know how to set it correctly.
``resample=yes``
Enable ALSA resampling plugin. (This is disabled by default, because
some drivers report incorrect audio delay in some cases.)
``mixer-device=<device>``
Set the mixer device used with ``--no-softvol`` (default: ``default``).
``mixer-name=<name>``
Set the name of the mixer element (default: ``Master``). This is for
example ``PCM`` or ``Master``.
``mixer-index=<number>``
Set the index of the mixer channel (default: 0). Consider the output of
"``amixer scontrols``", then the index is the number that follows the
name of the element.
``non-interleaved``
Allow output of non-interleaved formats (if the audio decoder uses
this format). Currently disabled by default, because some popular
ALSA plugins are utterly broken with non-interleaved formats.
``ignore-chmap``
Don't read or set the channel map of the ALSA device - only request the
required number of channels, and then pass the audio as-is to it. This
option most likely should not be used. It can be useful for debugging,
or for static setups with a specially engineered ALSA configuration (in
this case you should always force the same layout with ``--audio-channels``,
or it will work only for files which use the layout implicit to your
ALSA device).
.. note::
MPlayer and mplayer2 required you to replace any ',' with '.' and
any ':' with '=' in the ALSA device name. mpv does not do this anymore.
Instead, quote the device name:
``--ao=alsa:device=[plug:surround50]``
Note that the ``[`` and ``]`` simply quote the device name. With some
shells (like zsh), you have to quote the option string to prevent the
shell from interpreting the brackets instead of passing them to mpv.
Actually, you should use the ``--audio-device`` option, instead of
setting the device directly.
.. warning::
Handling of multichannel/surround audio changed in mpv 0.8.0 from the
behavior in MPlayer/mplayer2 and older versions of mpv.
The old behavior is that the player always downmixed to stereo by
default. The ``--audio-channels`` (or ``--channels`` before that) option
had to be set to get multichannel audio. Then playing stereo would
use the ``default`` device (which typically allows multiple programs
to play audio at the same time via dmix), while playing anything with
more channels would open one of the hardware devices, e.g. via the
``surround51`` alias (typically with exclusive access). Whether the
player would use exclusive access or not would depend on the file
being played.
The new behavior since mpv 0.8.0 always enables multichannel audio,
i.e. ``--audio-channels=auto`` is the default. However, since ALSA
provides no good way to play multichannel audio in a non-exclusive
way (without blocking other applications from using audio), the player
is restricted to the capabilities of the ``default`` device by default,
which means it supports only stereo and mono (at least with current
typical ALSA configurations). But if a hardware device is selected,
then multichannel audio will typically work.
The short story is: if you want multichannel audio with ALSA, use
``--audio-device`` to select the device (use ``--audio-device=help``
to get a list of all devices and their mpv name).
To get multichannel/surround audio, use ``--audio-channels=auto``. The
default for this option is ``auto-safe``, which makes this audio otuput
explicitly reject multichannel output, as there is no way to detect
whether a certain channel layout is actually supported.
You can also try `using the upmix plugin <http://git.io/vfuAy>`_.
This setup enables multichannel audio on the ``default`` device
+54
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@@ -3597,6 +3597,60 @@ DVB
Default: ``no``
ALSA audio output options
-------------------------
``--alsa-device=<device>``
Sets the device name. For ac3 output via S/PDIF, use an "iec958" or
"spdif" device, unless you really know how to set it correctly.
.. note::
MPlayer and mplayer2 required you to replace any ',' with '.' and
any ':' with '=' in the ALSA device name. mpv does not do this anymore.
Instead, quote the device name:
``--ao=alsa:device=[plug:surround50]``
Note that the ``[`` and ``]`` simply quote the device name. With some
shells (like zsh), you have to quote the option string to prevent the
shell from interpreting the brackets instead of passing them to mpv.
Actually, you should use the ``--audio-device`` option, instead of
setting the device directly.
``--alsa-resample=yes``
Enable ALSA resampling plugin. (This is disabled by default, because
some drivers report incorrect audio delay in some cases.)
``--alsa-mixer-device=<device>``
Set the mixer device used with ``ao-volume`` (default: ``default``).
``--alsa-mixer-name=<name>``
Set the name of the mixer element (default: ``Master``). This is for
example ``PCM`` or ``Master``.
``--alsa-mixer-index=<number>``
Set the index of the mixer channel (default: 0). Consider the output of
"``amixer scontrols``", then the index is the number that follows the
name of the element.
``--alsa-non-interleaved``
Allow output of non-interleaved formats (if the audio decoder uses
this format). Currently disabled by default, because some popular
ALSA plugins are utterly broken with non-interleaved formats.
``--alsa-ignore-chmap``
Don't read or set the channel map of the ALSA device - only request the
required number of channels, and then pass the audio as-is to it. This
option most likely should not be used. It can be useful for debugging,
or for static setups with a specially engineered ALSA configuration (in
this case you should always force the same layout with ``--audio-channels``,
or it will work only for files which use the layout implicit to your
ALSA device).
OpenGL renderer options
-----------------------
+1
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@@ -140,6 +140,7 @@ static struct ao *ao_alloc(bool probing, struct mpv_global *global,
*ao = (struct ao) {
.driver = desc.p,
.probing = probing,
.global = global,
.input_ctx = input_ctx,
.log = mp_log_new(ao, log, name),
.def_buffer = opts->audio_buffer,
+51 -31
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@@ -35,6 +35,7 @@
#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"
@@ -48,6 +49,37 @@
#include "internal.h"
#include "audio/format.h"
struct ao_alsa_opts {
char *device;
char *mixer_device;
char *mixer_name;
int mixer_index;
int resample;
int ni;
int ignore_chmap;
};
#define OPT_BASE_STRUCT struct ao_alsa_opts
const struct m_sub_options ao_alsa_conf = {
.opts = (const struct m_option[]) {
OPT_STRING("alsa-device", device, 0),
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;
@@ -61,13 +93,7 @@ struct priv {
snd_output_t *output;
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;
struct ao_alsa_opts *opts;
};
#define BUFFER_TIME 250000 // 250ms
@@ -125,13 +151,13 @@ static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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);
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->cfg_mixer_device);
err = snd_mixer_attach(handle, p->opts->mixer_device);
CHECK_ALSA_ERROR("Mixer attach error");
err = snd_mixer_selem_register(handle, NULL, NULL);
@@ -447,7 +473,7 @@ static int set_chmap(struct ao *ao, struct mp_chmap *dev_chmap, int num_channels
MP_VERBOSE(ao, "which we understand as: %s\n", mp_chmap_to_str(&chmap));
if (p->cfg_ignore_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");
@@ -615,8 +641,8 @@ static int init_device(struct ao *ao, int mode)
const char *device = "default";
if (ao->device)
device = ao->device;
if (p->cfg_device && p->cfg_device[0])
device = p->cfg_device;
if (p->opts->device && p->opts->device[0])
device = p->opts->device;
err = try_open_device(ao, device, mode);
CHECK_ALSA_ERROR("Playback open error");
@@ -641,7 +667,7 @@ static int init_device(struct ao *ao, int mode)
// Some ALSA drivers have broken delay reporting, so disable the ALSA
// resampling plugin by default.
if (!p->cfg_resample) {
if (!p->opts->resample) {
err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
CHECK_ALSA_ERROR("Unable to disable resampling");
}
@@ -684,7 +710,7 @@ static int init_device(struct ao *ao, int mode)
dump_hw_params(ao, MSGL_DEBUG, "HW params after format:\n", alsa_hwparams);
struct mp_chmap dev_chmap = ao->channels;
if (af_fmt_is_spdif(ao->format) || p->cfg_ignore_chmap) {
if (af_fmt_is_spdif(ao->format) || p->opts->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.
@@ -806,7 +832,9 @@ alsa_error:
static int init(struct ao *ao)
{
struct priv *p = ao->priv;
if (!p->cfg_ni)
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());
@@ -1125,8 +1153,6 @@ static void list_devs(struct ao *ao, struct ao_device_list *list)
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",
@@ -1144,20 +1170,14 @@ const struct ao_driver audio_out_alsa = {
.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),
OPT_SUBOPT_LEGACY("device", "alsa-device"),
OPT_SUBOPT_LEGACY("resample", "alsa-resample"),
OPT_SUBOPT_LEGACY("mixer-device", "alsa-mixer-device"),
OPT_SUBOPT_LEGACY("mixer-name", "alsa-mixer-name"),
OPT_SUBOPT_LEGACY("mixer-index", "alsa-mixer-index"),
OPT_SUBOPT_LEGACY("non-interleaved", "alsa-non-interleaved"),
OPT_SUBOPT_LEGACY("ignore-chmap", "alsa-ignore-chmap"),
{0}
},
};
+1
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@@ -40,6 +40,7 @@ struct ao {
const struct ao_driver *api; // entrypoints to the wrapper (push.c/pull.c)
const struct ao_driver *driver;
void *priv;
struct mpv_global *global;
struct encode_lavc_context *encode_lavc_ctx;
struct input_ctx *input_ctx;
struct mp_log *log; // Using e.g. "[ao/coreaudio]" as prefix
+4
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@@ -75,6 +75,7 @@ extern const struct m_sub_options encode_config;
extern const struct m_sub_options image_writer_conf;
extern const struct m_sub_options gl_video_conf;
extern const struct m_sub_options vo_opengl_conf;
extern const struct m_sub_options ao_alsa_conf;
extern const struct m_obj_list vf_obj_list;
extern const struct m_obj_list af_obj_list;
@@ -648,6 +649,9 @@ const m_option_t mp_opts[] = {
OPT_SUBSTRUCT("", gl_video_opts, gl_video_conf, 0),
OPT_SUBSTRUCT("", vo_opengl_opts, vo_opengl_conf, 0),
#endif
#if HAVE_ALSA
OPT_SUBSTRUCT("", ao_alsa_opts, ao_alsa_conf, 0),
#endif
#if HAVE_ENCODING
OPT_SUBSTRUCT("", encode_opts, encode_config, 0),
+1
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@@ -334,6 +334,7 @@ typedef struct MPOpts {
struct gl_video_opts *gl_video_opts;
struct vo_opengl_opts *vo_opengl_opts;
struct ao_alsa_opts *ao_alsa_opts;
} MPOpts;
extern const m_option_t mp_opts[];