I have no idea what these do, but apparently they are needed to inform
ALSA about spdif configuration. First, replace the literal constant "6"
for the AES0 parameter with the symbolic constants from the ALSA
headers (the final value is the same). Second, copy paste some funky
looking parameter setup from VLC's alsa output for setting the AES1,
AES2, AES3 parameters. (The code is actually not literally copy-pasted,
but does exactly the same.)
My small but non-zero hope is that this could make DTS-HD work, or at
least work into that direction. I can't test spdif stuff though, and
for DTS-HD not even opening the ALSA device succeeds on my system.
Using spdif with alsa requires adding magic parameters to the device
name, and the existing code tried to deal with the situation when the
user wanted to add parameters too.
Rewrite this code, in particular remove the duplicated parameter string
as preparation for the next commit. The new code is a bit stricter, e.g.
it doesn't skip spaces before and after '{' and '}'. (Just don't add
spaces.)
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
The configure followed 5 different convetions of defines because the next guy
always wanted to introduce a new better way to uniform it[1]. For an
hypothetic feature 'hurr' you could have had:
* #define HAVE_HURR 1 / #undef HAVE_DURR
* #define HAVE_HURR / #undef HAVE_DURR
* #define CONFIG_HURR 1 / #undef CONFIG_DURR
* #define HAVE_HURR 1 / #define HAVE_DURR 0
* #define CONFIG_HURR 1 / #define CONFIG_DURR 0
All is now uniform and uses:
* #define HAVE_HURR 1
* #define HAVE_DURR 0
We like definining to 0 as opposed to `undef` bcause it can help spot typos
and is very helpful when doing big reorganizations in the code.
[1]: http://xkcd.com/927/ related
The code was selecting PA_CHANNEL_POSITION_MONO for MP_SPEAKER_ID_FC,
which is correct only with the "mono" channel layout, but not anything
else. Remove the mono entry, and handle mono separately.
See github issue #326.
It's true that ALSA uses alloca() in some of its API functions, but
since this is hidden behind macros in the ALSA headers, we have no
reason to include alloca.h ourselves.
Might help with portability (FreeBSD).
At least not with ffmpeg.
Honestly, I have no idea how little endian AC3 works at all, since
ao_pcm doesn't do anything special about it, and treats it like s16le.
Maybe it's broken and ffmpeg has special logic to detect it.
Was disabled by default, was never used, internal support was
inconsistent and poor, and there has been virtually no interest in
creating translations.
And I don't even think that a terminal program should be translated.
This is something for (hypothetical) GUIs.
Output silence to the output buffer during underruns. This removes small
occasional glitches that happen before the AUHAL is actually paused from the
`audio_pause` call.
Fixes#269
Trying to connect multiple mpv clients to JACK with the
JackUseExactName option would fail unless the user manually
specifies a unique client name. This changes the behavior
to automatically generate a unique name if the requested
one is already in use.
Use the new MP_ macros for some AOs instead of mp_msg.
Not all AOs are converted, and some only partially. In some cases, some
additional cosmetic changes are made.
Using the default output audio unit should provide a much better user
exeperience since it changes automatically the output device based on which
becomes the default one.
This was removed in d427b4fd. I now found a sample that causes underruns when
moving to a chapter and apparently this is also a problem when taking
screenshots.
Reverts one of the changes from 18777ecf. `kAudioObjectPropertyScopeOutput`
was introduced in the 10.8 SDK while `kAudioDevicePropertyScopeOutput` was
moved to `AudioHardwareDeprecated.h`. Since the deprecation is silent for now
(no warnings), just use the old constant.
Either way, they both evaluate to 'outp', and in the 10.8 SDK the deprecated
constant is defined in terms of the non-deprecated one.
Fixes#155
This is not done automatically by CoreAudio. I am told that it would a PITA
to have to switch back the format manually on the device (especially if the
same device is used for lpcm output).
b2f9e0610 introduced this functionality with code that was quite 'monolithic'.
Split the functionality over several functions and ose the new macros to get
array properties.
Introduce some macros to deal with properties. These allow to work around the
limitation of CoreAudio's API being `void **` based. The macros allow to keep
their client's code DRY, by not asking size and other details which can be
derived by the macro itself. I have no idea why Apple didn't design their API
like this in the first place.
* ao_coreaudio_utils: contains several utility function
* ao_coreaudio_properties: contains functions to set and get audio object
properties.
Conflicts:
audio/out/ao_coreaudio.c
