this involved inverting the logic of find_formats, enumerate_devies
and wasapi_fill_VistaBlob. The latter two were trivial as their return
values were not actually checked (to be fixed in a later
commit).
Before these definitions were incorrectly guarded by and #ifdef
but since they aren't macros, this would never be true so that
if they were ever added to mingw headers we would have problems.
rename KSDATAFORMAT constants with the same mp prefix for consistency.
also use DEFINE_GUID rather than defining the bare structure
It was more complicated than necessary.
The behavior changes slightly. Now it might pass through extradata when
it didn't before (hopefully harmless), and doesn't fail with an error if
extradata is not available, even though it's needed (harmless, will fail
either way).
The if branch has a weak check to test whether the codec_id is the short
ID, and handles the long IDs in the else branch. The long IDs are all
longer than 12 bytes long, so hardcoding the string offset to get the
trailing part of the name makes sense. But the if condition checks for
another thing, which could get the else branch run even if the codec_id
is short.
Fix the bogus control flow and check if the codec_id is long enough. One
of these checks could be considered redundant, but include them both for
defensive coding.
I suspect this is what is happening in github issue #1265 (at least
partially).
If D3DFMT_A8 is not available, fall back to RGBA. This is less efficient
in general, so we normally want to avoid it.
Running "sub_add file.srt auto" during hook execution automatically
selected the first added track. This happened because all tracks added
with sub_add are marked as "external", and external subtitles are always
selected by default.
Fix this by negating the "external" flag when autoselecting subtitles
during loading. The no_default flag exists for this purpose; it was
probably added for libquvi originally, where we had the same issue.
This is a somewhat obscure situation, and happens only if audio starts
again after it has ended (in particular can happens with files where
audio starts later). It doesn't matter much whether audio starts
immediately or some milliseconds later, so simplify it.
When playing paused, the amount of decoded audio is limited to a small
amount (1 sample), because we don't write any audio to the AO when
paused. The small amount could trigger the case of the wanted audio
being too far in the future in the PTS sync code, which set the audio
status to STATUS_DRAINING, which in turn triggered the EOF code in the
next iteration. This was ok, but unfortunately, this triggered another
retry in order to check resuming from EOF by setting the status to
STATUS_SYNCING, which in turn lead to the busy loop by alternating
between the 2 states. So don't try resyncing while paused.
Since the PTS syncing code also calls ao_reset(), this could cause the
pulseaudio daemon to consume some CPU time as well.
This was caused by commit 33b57f55. Before that, the playloop was merely
run more often, but didn't cause any problems.
Fixes#1288.
...because everything is terrible.
strerror() is not documented as having to be thread-safe by POSIX and
C11. (Which is pretty much bullshit, because both mandate threads and
some form of thread-local storage - so there's no excuse why
implementation couldn't implement this in a thread-safe way. Especially
with C11 this is ridiculous, because there is no way to use threads and
convert error numbers to strings at the same time!)
Since we heavily use threads now, we should avoid unsafe functions like
strerror().
strerror_r() is in POSIX, but GNU/glibc deliberately fucks it up and
gives the function different semantics than the POSIX one. It's a bit of
work to convince this piece of shit to expose the POSIX standard
function, and not the messed up GNU one.
strerror_l() is also in POSIX, but only since the 2008 standard, and
thus is not widespread.
The solution is using avlibc (libavutil, by its official name), which
handles the unportable details for us, mostly. We avoid some pain.
Always create the context in mpgl_init(), instead of doing it when
mpgl_config_window() is called the first time. This is a small step
towards cleaning up the GL backend interface, and adding other things
like perhaps GLES support, or a callback-driven backend for libmpv.
this currently uses a sketchy but apparently working workaround,
which will be removed once the neccessary changes in youtube-dl
are implemented
Fixes#1277
This seems safer: otherwise, opening the AO could randomly fail if the
audio formats happens to be not float.
Unfortunately, this only works if the user does not select a device.
Since ALSA devices are arbitrary strings, including plugins with complex
parameters, it's not trivial or maybe even impossible to edit the string
in a way the "plug" plugin is added.
With --audio-device, it would be safe for users to select either
"default" or one of the "plughw" devices. Everything else seems
questionable.
Use the ALSA channel map API for querying and selecting supported
channel maps.
Since we (probably?) want to be compatible with ALSA versions before the
change, we still try to select the device name by channel map, and open
that device. There's no way to negotiate a channel map before opening,
so we're stuck with this approach. Fortunately, it seems these devices
allow selecting and setting any other supported channel layout, so maybe
this is not an issue at all. In particular, this avoids selecting the
default (dmix) device, which can only do stereo.
Most code is based on Martin Herkt <lachs0r@srsfckn.biz>'s alsa_ng
branch, with heavy modifications.
Don't crash if no fallback channel layout could be found (caller can't
handle NULL return from select_chmap()). Apparently this could never
actually happen, though.
Don't treat snd_pcm_hw_params_set_periods_near() failure as fatal error.
Same deal as with snd_pcm_hw_params_set_buffer_time_near().
Actually free channel maps returned by snd_pcm_get_chmap().
Adjust some messages.
Currently, --ytdl is off by default, but even if this is changed, never
enable it by default for the client API. It would be inappropriate to
start an intrusive external subprocess behind the host application's
back.
No functional changes.
ALSA_PCM_NEW_HW_PARAMS_API was a pre-ALSA 1.0.0 thing and does nothing
with modern ALSA. It stopped being necessary about 10 years ago.
3 functions are moved to avoid forward references.
