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-rw-r--r--DOCS/interface-changes.rst1
-rw-r--r--DOCS/man/ao.rst72
-rw-r--r--DOCS/man/options.rst54
-rw-r--r--audio/out/ao.c1
-rw-r--r--audio/out/ao_alsa.c82
-rw-r--r--audio/out/internal.h1
-rw-r--r--options/options.c4
-rw-r--r--options/options.h1
8 files changed, 117 insertions, 99 deletions
diff --git a/DOCS/interface-changes.rst b/DOCS/interface-changes.rst
index 493e3cefbc..ca7bec9b71 100644
--- a/DOCS/interface-changes.rst
+++ b/DOCS/interface-changes.rst
@@ -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
diff --git a/DOCS/man/ao.rst b/DOCS/man/ao.rst
index 0ab49e91fc..9e963d43f6 100644
--- a/DOCS/man/ao.rst
+++ b/DOCS/man/ao.rst
@@ -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
diff --git a/DOCS/man/options.rst b/DOCS/man/options.rst
index a4310208c5..8b7ffe02a5 100644
--- a/DOCS/man/options.rst
+++ b/DOCS/man/options.rst
@@ -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
-----------------------
diff --git a/audio/out/ao.c b/audio/out/ao.c
index 189434cf0b..df2fdd5416 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -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,
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index c6248d302b..e780d349c2 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -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}
},
};
diff --git a/audio/out/internal.h b/audio/out/internal.h
index 518661c2bd..bd8816366e 100644
--- a/audio/out/internal.h
+++ b/audio/out/internal.h
@@ -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
diff --git a/options/options.c b/options/options.c
index 1d840fd744..ddebceef36 100644
--- a/options/options.c
+++ b/options/options.c
@@ -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),
diff --git a/options/options.h b/options/options.h
index 59ed7b4a43..7d6701cd1e 100644
--- a/options/options.h
+++ b/options/options.h
@@ -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[];