| Commit message (Collapse) | Author | Age | Files | Lines |
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Still uses termcap, but uses terminfo for loading the termcap database if
possible. Adds configure test to find terminfo; skips the termcap test
if terminfo is found since terminfo provides termcap.
Use termcap completely for special keys; if we can't get it from termcap
and it isn't one of the known fallbacks, we ignore its specialness and
treat as a sequence of UTF-8 codes.
Further hardcoded fallbacks can be added by calling keys_push_once in
load_termcap; there is no limit to the amount of keys pushed.
Uses the "ke" and "ks" capabilities to start / exit application mode, which
is necessary on vt100 emulators (including screen, xterm and all terminals
that emulate either of those) to correctly receive arrow keys.
It's now possible to compile getch2 even without termcap, though it won't
be of much use since it'll be unable to detect special keys.
Converted to 4 spaces per tab, prettified some statements.
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A cygwin mpv will usually run inside mintty, and even if it were to be run
in a regular console window, the cygwin dll emulates ANSI escapes.
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This was a memory leak: the ASS_Track was never deallocated.
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It appears the API requires you to cover all plane data with AVBuffers
(that is, one AVBuffer per plane in the most general case), because
certain code can make certain assumptions about this. (Insert rant
about how this is barely useful and increases complexity and potential
bugs.) I don't know any cases where the current code actually fails,
but we want to follow the API, so do it anyway.
Note that we don't really know whether or not planes are from a single
memory allocation, so we have to assume the most general case and create
an AVBuffer for each plane. We simply assume that the data is padded to
the full stride in the last image line. All these extra dummy references
are stupid, but the code might become much simpler once we only support
libavcodec versions with refcounting and can use AVFrame directly.
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The --profile top-level option is handled specially in m_config.c. But
this code also broke sub-options that happened to be named "profile"
(e.g. when trying to use "-af-add=bs2b=profile=cmoy"). Handle it
specially only if it's the top-level --profile option.
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This didn't setup the linked list of sub-commands for multi-part
commands ("a ; b ; c") correctly. Also, the new commands were attached
to the allocation of the _old_ command instead of the new one. (Wow,
whatever the hell I was (not) thinking when I wrote this code.)
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Gives really funky results with PNG attachments otherwise. The main
problem is that avcodec_flush_buffers() does not fully reset the
decoder, so passing multiple PNG packets without keyframe flags will
attempt to combine the new picture with the previously decoded
contents. (Makes no sense with proper PNG - maybe this codepath is
intended for MNG or APNG.)
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Attached pictures are now handled specially and on a separate codepath,
so this hack is not needed anymore. Remove it from the normal codepath.
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In general, this warning can hint to actual bugs. We don't enable it
yet, because it would conflict with some unmerged code, and we should
check with clang too (this commit was done by testing with gcc).
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This means that AOs/VOs with no options set do not take the legacy
option parsing path, but instead report that they have no options.
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Same as with VOs in the previous commit.
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All VOs use proper option parsing now, and compatibility hacks are not
needed.
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Finally not used by anything anymore. Farewell.
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The big endian case was not covered. Doesn't make much difference since mpv
runs on Macs with x86 only, but for the sake of correctness.
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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).
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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.
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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.
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* 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
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Previous code needlessly stored the input asbd before actually testing it's
support against the hardware.
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this is a wip
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The condition was checked wrongly on asbd which is the input format
description. This lead to the condition always being true, thus selecting lpcm
streams for digital input.
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kHALOutputParam_Volume is the linear gain so it should be at maximum 1 to
keep the audio quality good. No idea why it was more than that.
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Also extract this functionality inside a function in coreaudio_common
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Luckily they all were inside for loops so the functionality does not actually
change.
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The initialization is split more clearly between compressed and lpcm case.
For the compressed case, format selection is simplified a lot and negotiation
removed. The way it was written it just passed back to the core the original
requested format, not what was found available on hardware.
Since this is most likely useless for the compressed case, I didn't bother
with this. In the future I'd like to split this AO in two one that only uses
the AUHAL and the other with direct access to the hardware so that even
passthrough of lcpm can be possible. This would decrease the latency,
audiophiles would like that.
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Split out some utility functions that use the CoreAudio API but are not related
the main task of the AOs (which is to move data correctly to the ringbuffer).
These are mainly need for the verbosity of the CoreAudio API and are just
obscuring the 'real' code.
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property_address -> p_addr
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WIP
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Change the ca_msg macro to pass along MSGT_AO automatically. Also use it for
every output for consistency.
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It was reported that it also works by not setting the read size in the
AudioBuffer (now idea how, but I will discover it later).
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Read only the requested amount by the AUHAL (instead of all the buffered data).
No idea what the deal is with pausing the audio units if there is no audio to
play, maybe to avoid underruns of some sort. Anyway from my tests this
condition never occurred so I'm removing it all.
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Also get rid of the useless comment.
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Commit 9a83d03 accidentally removed this. (Overlooked "static"?)
The handling of this rather sucks. Maybe a better solution will be
possible once we clean up the mp_msg code.
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This also affects --audiofile. The previous behavior wasn't really
useful. There are even separate switches for that: --audio-demuxer and
--sub-demuxer.
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This makes it actually possible to use the filter with more complicated
filter graphs (such as graphs containing the "," character).
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Also add a "o" suboption, which should allow fine control over
libavresample.
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Make the VF/VO/AO option parser available to audio filters. No audio
filter uses this yet, but it's still a quite intrusive change.
In particular, the commands for manipulating filters at runtime
completely change. We delete the old code, and use the same
infrastructure as for video filters. (This forces complete
reinitialization of the filter chain, which hopefully isn't a problem
for any use cases. The old code forced reinitialization too, but it
could potentially allow a filter to cache things; e.g. consider loaded
ladspa plugins and such.)
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This is useful for debugging.
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This code is supposed to run if dynamic filter insertion (such as when
inserting a volume filter in mixer.c) fails. Then it removes all filters
and recreates the default list of filters. But the code just blew up and
entered an endless loop, because it removed even the sentinel in/out
filters. This could happen when trying to use softvol controls while
using spdif, but also other situations. Fix it by calling the correct
code.
Also remove these obnoxious yoda-conditions.
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Declare decoders directly, instead of using the LIBAD_EXTERN macro. This
is simpler (no weird magic) and more extensible.
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This was forgotten in commit b81f5e2.
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The mp_vcd_priv_t struct doesn't have a file descriptor but a file
handle on win32.
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Ahead of OSS because cygwin provides OSS.
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MSDN tells me to multiply the samplerates by 4 (for setting up the S/PDIF
signal frequency), but doesn't mention that I'm only supposed to do it
on the new, NT6.1+ IEC 61937 structs. Works on my Realtek Digital Output,
but as I can't connect any hardware to it I can't hear the result.
Also, always ask for little-endian AC3. I'm not sure if this is supposed
to be LE or NE, but Windows is LE on all platforms, so we go with LE.
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That's the sample format ad_spdif uses when the source is MP3.
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Entirely untested as this troper has no S/PDIF hardware.
Refuses trying any other format if we can't use passthrough, or we would
end up sending white noise at the user.
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Caused by incorrect conversion to the m_option API: since we don't allocate
the state ourselves, we also don't free it ourselves.
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Do an strstr match against the device description and, if we have only
a single match, take it. This works as long as the devices in the system
don't change, but it's not supposed to be reliable; if one wants
reliability, one uses the device ID string.
Formatting.
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This could turn valid parameters into syntax errors by the mere presence
or abscence of a device (e.g. USB audio devices), so don't do that.
We do validate that, if the parameter is an integer, it is not negative.
We also respond to the "help" parameter, which does the same as the "list"
suboption but exits after listing.
Demote the validation logging to MSGL_DBG2.
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