| Commit message (Collapse) | Author | Age | Files | Lines |
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Stopping output implies that it can't be paused anymore.
This is consistent with the documented API in internal.h as well
as the behavior of other AOs.
resolves #13267
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c78482045444c488bb7948305d583a55d17cd236 introduced a bool option type
as a replacement for the flag type, but didn't actually transition and
remove the flag type because it would have been too much mundane work.
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Most sources don't need config.h.
The inclusion only leads to lots of unneeded recompilation if the
configuration is changed.
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It is now the AO's responsibility to handle period size alignment. The
ao->period_size alignment field is unused as of the recent audio
refactor commit. Remove it.
It turns out that ao_alsa shows extremely inefficient behavior as a
consequence of the removal of period size aligned writes in the
mentioned refactor commit. This is because it could get into a state
where it repeatedly wrote single samples (as small as 1 sample), and
starved the rest of the player as a consequence. Too bad. Explicitly
align the size in ao_alsa. Other AOs, which need this, should do the
same.
One reason why it broke so badly with ao_alsa was that it retried the
write() even if all reported space could be written. So stop doing that
too. Retry the write only if we somehow wrote less.
I'm not sure about ao_pulse.
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AOs which use the "push" API must set this field now. Actually, this was
sort of always required, but happened to work anyway. The future
intention is to use device_buffer as the pre-buffer amount, which has to
be available right before audio playback is started. "Pull" AOs really
need this too conceptually, just that the API is underspecified.
From what I can see, only ao_null did not do this yet.
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Instead of the relatively subtle underflow handling, simply signal
whether the stream is in a playing state. Should make it more robust.
Should affect ao_alsa and ao_pulse only (and ao_openal, but it's
broken).
For ao_pulse, I'm just guessing. How the hell do you query whether a
stream is playing? Who knows. Seems to work, judging from very
superficial testing.
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This affects "pull" AOs only: ao_alsa, ao_pulse, ao_openal, ao_pcm,
ao_lavc. There are changes to the other AOs too, but that's only about
renaming ao_driver.resume to ao_driver.start.
ao_openal is broken because I didn't manage to fix it, so it exits with
an error message. If you want it, why don't _you_ put effort into it? I
see no reason to waste my own precious lifetime over this (I realize the
irony).
ao_alsa loses the poll() mechanism, but it was mostly broken and didn't
really do what it was supposed to. There doesn't seem to be anything in
the ALSA API to watch the playback status without polling (unless you
want to use raw UNIX signals).
No idea if ao_pulse is correct, or whether it's subtly broken now. There
is no documentation, so I can't tell what is correct, without reverse
engineering the whole project. I recommend using ALSA.
This was supposed to be just a simple fix, but somehow it expanded scope
like a train wreck. Very high chance of regressions, but probably only
for the AOs listed above. The rest you can figure out from reading the
diff.
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Forgot in the previous commit to this file.
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The recent change to the common code removed all calls to ->drain. It's
currently emulated via a timed sleep and polling ao_eof_reached(). That
is actually fallback code for AOs which lacked draining. I could just
readd the drain call, but it was a bad idea anyway. My plan to handle
this better is to require the AO to signal a underrun, even if
AOPLAY_FINAL_CHUNK is not set. Also reinstate not possibly waiting for
ao_lavc.c. ao_pcm.c did not have anything to handle this; whatever.
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Change all OPT_* macros such that they don't define the entire m_option
initializer, and instead expand only to a part of it, which sets certain
fields. This requires changing almost every option declaration, because
they all use these macros. A declaration now always starts with
{"name", ...
followed by designated initializers only (possibly wrapped in macros).
The OPT_* macros now initialize the .offset and .type fields only,
sometimes also .priv and others.
I think this change makes the option macros less tricky. The old code
had to stuff everything into macro arguments (and attempted to allow
setting arbitrary fields by letting the user pass designated
initializers in the vararg parts). Some of this was made messy due to
C99 and C11 not allowing 0-sized varargs with ',' removal. It's also
possible that this change is pointless, other than cosmetic preferences.
Not too happy about some things. For example, the OPT_CHOICE()
indentation I applied looks a bit ugly.
Much of this change was done with regex search&replace, but some places
required manual editing. In particular, code in "obscure" areas (which I
didn't include in compilation) might be broken now.
In wayland_common.c the author of some option declarations confused the
flags parameter with the default value (though the default value was
also properly set below). I fixed this with this change.
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Helpful especially to test spdif fallback and so on.
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Before this change, AOs could have internal alignment, and play() would
not consume the trailing data if the size passed to it is not aligned.
Change this to require AOs to report their alignment (via period_size),
and make sure to always send aligned data.
The buffer reported by get_space() now always has to be correct and
reliable. If play() does not consume all data provided (which is bounded
by get_space()), an error is printed.
This is preparation for potential further AO changes.
I casually checked alsa/lavc/null/pcm, the other AOs might or might not
work.
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All contributors of the current code have agreed. ao.c requires a
"driver" entry for each audio output - we assume that if someone who
didn't agree to LGPL added a line, it's fine for ao.c to be LGPL
anyway. If the affected audio output is not disabled at compilation
time, the resulting binary will be GPL anyway, and ootherwise the
code is not included.
The audio output code itself was inspired or partially copied from
libao in 7a2eec4b59f4 (thus why MPlayer's audio code is named libao2).
Just to be sure we got permission from Aaron Holtzman, Jack Moffitt, and
Stan Seibert, who according to libao's SVN history and README are the
initial author. (Something similar was done for libvo, although the
commit relicensing it forgot to mention it.)
242aa6ebd40: anders mostly disagreed with the LGPL relicensing, but we
got permission for this particular commit.
0ef8e555735: nick could not be reached, but the include statement was
removed again anyway.
879e05a7c17: iive agreed to LGPL v3+ only, but this line of code was
removed anyway, so ao_null.c can be LGPL v2.1+.
9dd8f241ac2: patch author could not be reached, but the corresponding
code (old slave mode interface) was completely removed later.
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Long planned. Leads to some sanity.
There still are some rather gross things. Especially g_groups is ugly,
and a hack that can hopefully be removed. (There is a plan for it, but
whether it's implemented depends on how much energy is left.)
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This change helps avoiding conflict with talloc.h from libtalloc.
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Signed-off-by: wm4 <wm4@nowhere>
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I'm not sure how common this behavior possibly is; well whatever. This
option will allow reproducing such behavior, and help debugging it.
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ao_get_delay() returns double, but the get_delay callback still
returned float.
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Round get_space() results in the same way play() rounds the input size.
Some audio APIs do this for various reasons.
This affects only "push" based AOs. Some of these need no change,
because they either do it already right (like ao_openal), or they seem
not to have any such requirements (like ao_pulse).
Needed for the following commit.
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ao_null is used to stop autoprobing (if all AOs before fail to init).
After it come things like ao_pcm, which should never be automatically
selected.
Remove a certain theoretically possible failure case, and force "some"
fallback.
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Doesn't work quite right, and will pause for the latency duration after
seeking. Some users use --ao=null to disable audio (even though they
should probably use --no-audio), and this use-case is broken by this
issue too.
CC: @mpv-player/stable
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It's in seconds, but the code used it as sample count.
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This EOF problems happen at least with PulseAudio, but since it's hard
to reproduce, let ao_null optionally simulate it.
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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.
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Helps with testing and debugging.
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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.
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Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.
Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
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The size accidentally defaulted to 200 seconds instead of 200
milliseconds, which had fatal consequences when trying to use it.
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Simulate proper handling of AOPLAY_FINAL_CHUNK. Print when underruns
occur (i.e. running out of data). Add some options that control
simulated buffer and outburst sizes.
All this is useful for debugging and self-documentation. (Note that
ao_null always was supposed to simulate an ideal AO, which is the reason
why it fools people who try to use it for benchmarking video.)
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ao_null should simulate a "perfect" AO, but framestepping behaved quite
badly with it. Framstepping usually exposes problems with AOs dropping
their buffers on pause, and that's what happened here.
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Simply change internals from using byte counts to sample counts.
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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.
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No AO can handle these, so it would be a problem if they get added
later, and non-interleaved formats get accepted erroneously. Let them
gracefully fall back to other formats.
Most AOs actually would fall back, but to an unrelated formats. This is
covered by this commit too, and if possible they should pick the
interleaved variant if a non-interleaved format is requested.
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Same as with VOs in the previous commit.
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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.
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Some still do, because they use the value in other places of the init
function. ao_portaudio is tricky and reads ao->bps in the stream
thread, which might be started on initialization (not sure about that,
but better safe than sorry).
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GetTimer() is generally replaced with mp_time_us(). Both calls return
microseconds, but the latter uses int64_t, us defined to never wrap,
and never returns 0 or negative values.
GetTimerMS() has no direct replacement. Instead the other functions are
used.
For some code, switch to mp_time_sec(), which returns the time as double
float value in seconds. The returned time is offset to program start
time, so there is enough precision left to deliver microsecond
resolution for at least 100 years. Unless it's casted to a float
(or the CPU reduces precision), which is why we still use mp_time_us()
out of paranoia in places where precision is clearly needed.
Always switch to the correct time. The whole point of the new timer
calls is that they don't wrap, and storing microseconds in unsigned int
variables would negate this.
In some cases, remove wrap-around handling for time values.
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Make all AOs use what has been introduced in the previous commit.
Note that even AOs which can handle all possible layouts (like ao_null)
use the new functions. This might be important if in the future
ao_select_champ() possibly honors global user options about downmixing
and so on.
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Is unused, is completely pointless.
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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.
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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.)
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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.
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