Commit Graph

38 Commits

Author SHA1 Message Date
Marcin Kurczewski f43017bfe9 Update license headers
Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 12:10:01 +02:00
Stefano Pigozzi 70802d519f ao_coreaudio: add support for hotplug notifications
This commit adds notifications for hot plugging of devices. It also extends
the old behaviour of the `audio-out-detected-device` property which is now
backed by the hotplugging code. This allows clients to be notified when the
actual audio output device changes.

Maybe hotplugging should be supported for ao_coreaudio_exclusive too, but it's
device selection code is a bit fragile.
2015-02-14 12:51:15 +01:00
wm4 f061befb33 audio: add device change notification for hotplugging
Not very important for the command line player; but GUI applications
will want to know about this.

This only adds the internal API; support for specific audio outputs
comes later.

This reuses the ao struct as context for the hotplug event listener,
similar to how the "old" device listing API did. This is probably a bit
unclean and confusing. One argument got reusing it is that otherwise
rewriting parts of ao_pulse would be required (because the PulseAudio
API requires so damn much boilerplate). Another is that --ao-defaults is
applied to the hotplug dummy ao struct, which automatically applies such
defaults even to the hotplug context.

Notification works through the property observation mechanism in the
client API. The notification chain is a bit complicated: the AO notifies
the player, which in turn notifies the clients, which in turn will
actually retrieve the device list. (It still has the advantage that it's
slightly cleaner, since the AO stuff doesn't need to know about client
API issues.)

The weird handling of atomic flags in ao.c is because we still don't
require real atomics from the compiler. Otherwise we'd just use atomic
bitwise operations.
2015-02-12 17:17:41 +01:00
Stefano Pigozzi a3be14683a command: add property returning detected audio device
This can be useful to adjust some other audio related properties
at runtime depending on the audio device being used.
2015-02-03 00:40:02 +01:00
wm4 b021d038c2 audio/out: make ao_request_reload() idempotent
This is what you would expect. Before this commit, each
ao_request_reload() call would just queue a reload command, and then
recreate the AO for the number of times the function was called.

Instead of sending a command, introduce some sort of event retrieval
mechanism. At least for the reload case, use atomics, because we're too
lazy to setup an extra mutex.
2014-11-09 09:58:44 +01:00
wm4 edad4fc29b audio: change internal device listing API
Now we run ao_driver->list_devs on a dummy AO instance, which will
probably confuse everyone. This is done for the sake of PulseAudio.
2014-10-10 18:27:21 +02:00
wm4 35649a990a audio: add device selection & listing with --audio-device
Not sure how good of an idea this is.

This commit doesn't add support for this to any AO yet; the AO
implementations will follow later.
2014-10-09 21:21:31 +02:00
wm4 439a05d8c3 audio/out: remove old things
Remove the unnecessary indirection through ao fields.

Also fix the inverted result of AOCONTROL_HAS_TEMP_VOLUME. Hopefully the
change is equivalent. But actually, it looks like the old code did it
wrong.
2014-09-06 02:30:57 +02:00
wm4 68ff8a0484 Move compat/ and bstr/ directory contents somewhere else
bstr.c doesn't really deserve its own directory, and compat had just
a few files, most of which may as well be in osdep. There isn't really
any justification for these extra directories, so get rid of them.

The compat/libav.h was empty - just delete it. We changed our approach
to API compatibility, and will likely not need it anymore.
2014-08-29 12:31:52 +02:00
wm4 5059039c95 player: unrangle one aspect of audio EOF handling
For some reason, the buffered_audio variable was used to "cache" the
ao_get_delay() result. But I can't really see any reason why this should
be done, and it just seems to complicate everything.

One reason might be that the value should be checked only if the AO
buffers have been recently filled (as otherwise the delay could go low
and trigger an accidental EOF condition), but this didn't work anyway,
since buffered_audio is set from ao_get_delay() anyway at a later point
if it was unset. And in both cases, the value is used _after_ filling
the audio buffers anyway.

Simplify it. Also, move the audio EOF condition to a separate function.
(Note that ao_eof_reached() probably could/should whether the last
ao_play() call had AOPLAY_FINAL_CHUNK set to avoid accidental EOF on
underflows, but for now let's keep the code equivalent.)
2014-04-17 23:48:09 +02:00
wm4 fe298bc2a5 audio: explicitly document audio EOF condition
This should probably be an AO function, but since the playloop still has
some strange stuff (using the buffered_audio variable instead of calling
ao_get_delay() directly), just leave it and make it more explicit.
2014-04-17 22:45:49 +02:00
wm4 e16c91d07a audio/out: make draining a separate operation
Until now, this was always conflated with uninit. This was ugly, and
also many AOs emulated this manually (or just ignored it). Make draining
an explicit operation, so AOs which support it can provide it, and for
all others generic code will emulate it.

For ao_wasapi, we keep it simple and basically disable the internal
draining implementation (maybe it should be restored later).

Tested on Linux only.
2014-03-09 01:27:41 +01:00
wm4 41f2b26d11 audio/out: make ao struct opaque
We want to move the AO to its own thread. There's no technical reason
for making the ao struct opaque to do this. But it helps us sleep at
night, because we can control access to shared state better.
2014-03-09 00:19:31 +01:00
wm4 6b2a929ca7 ao: document some functions 2014-02-28 00:56:10 +01:00
wm4 0112143fda Split mpvcore/ into common/, misc/, bstr/ 2013-12-17 02:39:45 +01:00
wm4 347a86198b audio: switch output to mp_audio_buffer
Replace the code that used a single buffer with mp_audio_buffer. This
also enables non-interleaved output operation, although it's still
disabled, and no AO supports it yet.
2013-11-12 23:29:53 +01:00
wm4 380fc765e4 audio/out: prepare for non-interleaved audio
This comes with two internal AO API changes:

1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.

2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)

Change all AOs to use the new API.

The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
2013-11-12 23:27:51 +01:00
wm4 3cb4116243 ao: add ao_play_silence, use for ao_alsa and ao_oss
Also add a corresponding function to audio/format.c, which fills an
audio block with silence.
2013-11-10 23:05:59 +01:00
wm4 1a5c863a32 player: set PulseAudio stream title to window title
Set the PulseAudio stream title, just like the VO window title is set.
Refactor update_vo_window_title() so that we can use it for AOs too.

The ao_pulse.c bit is stolen from MPlayer.
2013-11-10 00:49:13 +01:00
wm4 8125252399 audio: don't let ao_lavc access frontend internals, change gapless audio
ao_lavc.c accesses ao->buffer, which I consider internal. The access was
done in ao_lavc.c/uninit(), which tried to get the left-over audio in
order to write the last (possibly partial) audio frame. The play()
function didn't accept partial frames, because the AOPLAY_FINAL_CHUNK
flag was not correctly set, and handling it otherwise would require an
internal FIFO.

Fix this by making sure that with gapless audio (used with encoding),
the AOPLAY_FINAL_CHUNK is set only once, instead when each file ends.
Basically, move the hack in ao_lavc's uninit to uninit_player.

One thing can not be entirely correctly handled: if gapless audio is
active, we don't know really whether the AO is closed because the file
ended playing (i.e. we want to send the buffered remainder of the audio
to the AO), or whether the user is quitting the player. (The stop_play
flag is overwritten, fixing that is perhaps not worth it.) Handle this
by adding additional code to drain the AO and the buffers when playback
is quit (see play_current_file() change).

Test case: mpv avdevice://lavfi:sine=441 avdevice://lavfi:sine=441 -length 0.2267  -gapless-audio
2013-11-08 20:00:58 +01:00
wm4 d58d4ec93c audio/out: remove useless info struct and redundant fields 2013-10-23 19:30:02 +02:00
wm4 cb54c2dda8 ao: remove some leftovers 2013-08-22 22:45:24 +02:00
Stefano Pigozzi 406241005e core: move contents to mpvcore (2/2)
Followup commit. Fixes all the files references.
2013-08-06 22:52:31 +02:00
Stefano Pigozzi 3449e893e1 audio/out: add support for new logging API 2013-08-01 20:32:49 +02:00
wm4 f32a90a839 audio/out: remove options argument from init()
Same as with VOs in the previous commit.
2013-07-22 22:58:09 +02:00
wm4 7eba27c125 options: use new option code for --ao
This requires completely refactoring the AO creation code too.
2013-07-21 23:27:31 +02:00
wm4 4d3a2c7e0d audio/out: remove ao->outburst/buffersize fields
The core didn't use these fields, and use of them was inconsistent
accross AOs. Some didn't use them at all. Some only set them; the values
were completely unused by the core. Some made full use of them.

Remove these fields. In places where they are still needed, make them
private AO state.

Remove the --abs option. It set the buffer size for ao_oss and ao_dsound
(being ignored by all other AOs), and was already marked as obsolete. If
it turns out that it's still needed for ao_oss or ao_dsound, their
default buffer sizes could be adjusted, and if even that doesn't help,
AO suboptions could be added in these cases.
2013-06-16 19:36:56 +02:00
wm4 b24bb7076d audio/out: remove wrapper for old AOs
It's unused now.
2013-06-16 18:33:19 +02:00
wm4 ab8f28a672 audio: add channel map selection function
The point is selecting a minimal fallback. The AOs will call this
through the AO API, so it will be possible to add options affecting
the general channel layout selection.

It provides the following mechanism to AOs:
- forcing the correct channel order
- downmixing to stereo if no layout is available
- allow 5.1 <-> 5.1(side) fallback
- handling "unknown" channel layouts

This is quite weak and lots of code/complexity for little gain. All AOs
already made sure the channel order was correct, and the fallback is of
little value, and could perhaps be done in the frontend instead, like
stereo downmixing with --channels=2 is handled. But I'm not really sure
how this stuff should _really_ work, and the new code will hopefully
provides enough flexibility to make radical changes to channel layout
negotiation easier.
2013-05-12 21:24:57 +02:00
wm4 ce2515ddb8 ao: remove ao_driver.is_new field
Is unused, is completely pointless.
2013-05-12 21:24:56 +02:00
wm4 aea2328906 audio/out: switch to channel map
This actually breaks audio for 5/6/8 channels. There's no reordering
done yet. The actual reordering will be done inside of af_lavrresample
and has to be made part of the format negotiation.
2013-05-12 21:24:54 +02:00
wm4 f7a427676c audio: add some setters for mp_audio, and require filters to use them
mp_audio has some redundant fields. Setters like mp_audio_set_format()
initialize these properly.

Also move the mp_audio struct to a the file audio.c.

We can remove a mysterious line of code from af.c:

    in.format |= af_bits2fmt(in.bps * 8);

I'm not sure if this was ever actually needed, or if it was some kind of
"make it work" quick-fix that works against the way things were supposed
to work. All filters etc. now set the format correctly, so if there ever
was a need for this code, it's definitely gone.
2013-05-12 21:24:54 +02:00
wm4 4d016a92c8 core: redo how codecs are mapped, remove codecs.conf
Use codec names instead of FourCCs to identify codecs. Rewrite how
codecs are selected and initialized. Now each decoder exports a list
of decoders (and the codec it supports) via add_decoders(). The order
matters, and the first decoder for a given decoder is preferred over
the other decoders. E.g. all ad_mpg123 decoders are preferred over
ad_lavc, because it comes first in the mpcodecs_ad_drivers array.
Likewise, decoders within ad_lavc that are enumerated first by
libavcodec (using av_codec_next()) are preferred. (This is actually
critical to select h264 software decoding by default instead of vdpau.
libavcodec and ffmpeg/avconv use the same method to select decoders by
default, so we hope this is sane.)

The codec names follow libavcodec's codec names as defined by
AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders
have names different from the canonical codec name. The AVCodecDescriptor
API is relatively new, so we need a compatibility layer for older
libavcodec versions for codec names that are referenced internally,
and which are different from the decoder name. (Add a configure check
for that, because checking versions is getting way too messy.)

demux/codec_tags.c is generated from the former codecs.conf (minus
"special" decoders like vdpau, and excluding the mappings that are the
same as the mappings libavformat's exported RIFF tables). It contains
all the mappings from FourCCs to codec name. This is needed for
demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the
codec as determined by libavformat, while the other demuxers have to do
this on their own, using the mp_set_audio/video_codec_from_tag()
functions. Note that the sh_audio/video->format members don't uniquely
identify the codec anymore, and sh->codec takes over this role.

Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which
provide cover the functionality of the removed switched.

Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure
container/video combinations (e.g. the sample Film_200_zygo_pro.mov)
are played flipped. ffplay/avplay doesn't handle this properly either,
so we don't care and blame ffmeg/libav instead.
2013-02-10 17:25:56 +01:00
wm4 ae070a6f1e audio/out, video/out: hide encoding VO/AO
mpv -ao help and mpv -vo help shouldn't show the encoding outputs (named
"lavc" on both cases). Also make it impossible to select these manually
when not encoding.
2013-02-06 23:04:18 +01:00
wm4 7a6d26370c mixer: prefer AO softvol control over volume filter
This partially reverts earlier decisions, when I thought it would
always be better to prefer the audio volume filter over the AO's,
because the AO's relies on the underlying audio-API, which could
be broken or exhibit unusual behavior (like it happened with ao_dsound).

However, since the audio buffer can be quite large (500 ms), and we
don't attempt to flush & refilter the audio on volume changes, always
prefer AO volume control (as long as the AO mixer doesn't control the
system mixer).

Also document what the mixer.c related AO fields mean (hopefully not
too brief).
2013-02-06 23:04:18 +01:00
wm4 b0558e48b1 cleanup: remove ao.brokenpts
This field was used by ao_v4l2, and is now unused.
2012-12-12 23:05:57 +01:00
wm4 4873b32c59 Rename directories, move files (step 2 of 2)
Finish renaming directories and moving files. Adjust all include
statements to make the previous commit compile.

The two commits are separate, because git is bad at tracking renames
and content changes at the same time.

Also take this as an opportunity to remove the separation between
"common" and "mplayer" sources in the Makefile. ("common" used to be
shared between mplayer and mencoder.)
2012-11-12 20:08:18 +01:00
wm4 d4bdd0473d Rename directories, move files (step 1 of 2) (does not compile)
Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.

Renames the following directories:
    libaf -> audio/filter
    libao2 -> audio/out
    libvo -> video/out
    libmpdemux -> demux

Split libmpcodecs:
    vf* -> video/filter
    vd*, dec_video.* -> video/decode
    mp_image*, img_format*, ... -> video/
    ad*, dec_audio.* -> audio/decode

libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.

Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.

sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).

Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.
2012-11-12 20:06:14 +01:00