Commit Graph

31 Commits

Author SHA1 Message Date
wm4 1a2319f3e4 options: remove deprecated sub-option handling for --vo and --ao
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.)
2016-11-25 21:17:25 +01:00
wm4 69283bc0f8 options: deprecate suboptions for the remaining AO/VOs 2016-09-05 21:26:39 +02:00
wm4 8a9b64329c Relicense some non-MPlayer source files to LGPL 2.1 or later
This covers source files which were added in mplayer2 and mpv times
only, and where all code is covered by LGPL relicensing agreements.

There are probably more files to which this applies, but I'm being
conservative here.

A file named ao_sdl.c exists in MPlayer too, but the mpv one is a
complete rewrite, and was added some time after the original ao_sdl.c
was removed. The same applies to vo_sdl.c, for which the SDL2 API is
radically different in addition (MPlayer supports SDL 1.2 only).

common.c contains only code written by me. But common.h is a strange
case: although it originally was named mp_common.h and exists in MPlayer
too, by now it contains only definitions written by uau and me. The
exceptions are the CONTROL_ defines - thus not changing the license of
common.h yet.

codec_tags.c contained once large tables generated from MPlayer's
codecs.conf, but all of these tables were removed.

From demux_playlist.c I'm removing a code fragment from someone who was
not asked; this probably could be done later (see commit 15dccc37).

misc.c is a bit complicated to reason about (it was split off mplayer.c
and thus contains random functions out of this file), but actually all
functions have been added post-MPlayer. Except get_relative_time(),
which was written by uau, but looks similar to 3 different versions of
something similar in each of the Unix/win32/OSX timer source files. I'm
not sure what that means in regards to copyright, so I've just moved it
into another still-GPL source file for now.

screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but
they're all gone.
2016-01-19 18:36:06 +01:00
Dmitrij D. Czarkoff ea442fa047 mpv_talloc.h: rename from talloc.h
This change helps avoiding conflict with talloc.h from libtalloc.
2016-01-11 21:05:55 +01:00
wm4 831d7c3c40 audio: remove S8, U16, U24, U32 formats
They are useless. Not only are they actually rarely in use; but
libavcodec doesn't even output them, as libavcodec has no such sample
formats for decoded audio.

Even if it should happen that we actually still need them (e.g. if doing
direct hardware output), there are better solutions. Swapping the sign
is a fast and lossless operation and can be done inplace, so AO actually
needing it could do this directly.

If you wonder why we keep U8 instead of S8: because libavcodec does it.
2015-06-16 21:11:59 +02:00
Marcin Kurczewski f43017bfe9 Update license headers
Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 12:10:01 +02:00
wm4 26bc6b4831 Add some missing "const"s
The one in msg.c was mistakenly removed with commit e99a37f6.

I didn't actually test the change in ao_sndio.c (but obviously "ap"
shouldn't be static).
2014-10-10 13:44:08 +02:00
wm4 b745c2d005 audio: drop swapped-endian audio formats
Until now, the audio chain could handle both little endian and big
endian formats. This actually doesn't make much sense, since the audio
API and the HW will most likely prefer native formats. Or at the very
least, it should be trivial for audio drivers to do the byte swapping
themselves.

From now on, the audio chain contains native-endian formats only. All
AOs and some filters are adjusted. af_convertsignendian.c is now wrongly
named, but the filter name is adjusted. In some cases, the audio
infrastructure was reused on the demuxer side, but that is relatively
easy to rectify.

This is a quite intrusive and radical change. It's possible that it will
break some things (especially if they're obscure or not Linux), so watch
out for regressions. It's probably still better to do it the bulldozer
way, since slow transition and researching foreign platforms would take
a lot of time and effort.
2014-09-23 23:09:25 +02:00
wm4 c36faf8c49 audio/out/pull: remove race conditions
There were subtle and minor race conditions in the pull.c code, and AOs
using it (jack, portaudio, sdl, wasapi). Attempt to remove these.

There was at least a race condition in the ao_reset() implementation:
mp_ring_reset() was called concurrently to the audio callback. While the
ringbuffer uses atomics to allow concurrent access, the reset function
wasn't concurrency-safe (and can't easily be made to).

Fix this by stopping the audio callback before doing a reset. After
that, we can do anything without needing synchronization. The callback
is resumed when resuming playback at a later point.

Don't call driver->pause, and make driver->resume and driver->reset
start/stop the audio callback. In the initial state, the audio callback
must be disabled.

JackAudio of course is different. Maybe there is no way to suspend the
audio callback without "disconnecting" it (what jack_deactivate() would
do), so I'm not trying my luck, and implemented a really bad hack doing
active waiting until we get the audio callback into a state where it
won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we
can be sure that the callback doesn't access the ringbuffer or anything
else anymore. Since both sched_yield() and pthread_yield() apparently
are not always available, use mp_sleep_us(1) to avoid burning CPU during
active waiting.

The ao_jack.c change also removes a race condition: apparently we didn't
initialize _all_ ao fields before starting the audio callback.

In ao_wasapi.c, I'm not sure whether reset really waits for the audio
callback to return. Kovensky says it's not guaranteed, so disable the
reset callback - for now the behavior of ao_wasapi.c is like with
ao_jack.c, and active waiting is used to deal with the audio callback.
2014-05-29 02:24:17 +02:00
wm4 7221d96ba3 ao_sdl: make sure our buffer is always larger than what SDL requests
Assume obtained.samples contains the number of samples the SDL audio
callback will request at once. Then make sure ao.c will set the buffer
size at least to 3 times that value (or more).

Might help with bad SDL audio backends like ESD, which supposedly uses a
500ms buffer.
2014-03-10 22:56:23 +01:00
wm4 346c687d5a ao_sdl: use new pull API helpers
One strange issue is that we apparently can't stop the audio API on
audio reset (ao_driver.reset). We could use SDL_PauseAudio, but that
doesn't specify whether remaining audio is dropped. We also could use
SDL_LockAudio, but holding that over a long time will probably be bad,
and it probably doesn't drop audio. This means we simply play silence
after a reset, instead of stopping the callback completely. (The
existing code ran into an underrun in this situation.)

The delay estimation works about the same. We simply assume that the
callback is locked to audio timing (like ao_jack), and that 1 callback
corresponds to 1 period. It seems this (removed) code fragment assumes
there 1 one period size delay:

// delay subcomponent: remaining audio from the next played buffer, as
// provided by the callback
buffer_interval += callback_interval;

so we explicitly do that too.
2014-03-09 19:08:47 +01:00
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 0112143fda Split mpvcore/ into common/, misc/, bstr/ 2013-12-17 02:39:45 +01:00
wm4 eb15151705 Move options/config related files from mpvcore/ to options/
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.
2013-12-17 02:07:57 +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 bf60281ffb audio/out: reject non-interleaved formats
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.
2013-11-12 23:16:31 +01:00
wm4 d58d4ec93c audio/out: remove useless info struct and redundant fields 2013-10-23 19:30:02 +02:00
wm4 edd36a3afc audio/out: do some mp_msg conversions
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.
2013-08-22 23:12:35 +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
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 12e645fc24 ao_sdl: use new option API 2013-07-21 23:27:32 +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 f88193091b audio/out: don't require AOs to set ao->bps
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).
2013-06-16 19:32:18 +02:00
wm4 e56d8a200d Replace all calls to GetTimer()/GetTimerMS()
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.
2013-05-26 16:44:20 +02:00
wm4 ecc6e379b2 audio/out: channel map selection
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.
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 fe8d3e70b2 ao_sdl: fix compilation with Libav
On Libav, <libavutil/fifo.h> doesn't recursively include common.h, but
the code in ao_sdl.c uses some macros defined by this header.
2013-01-06 16:04:17 +01:00
Rudolf Polzer 6be50fa773 sdl, encode_lavc: fix copyright headers
Some of them had changes in 2012; extend their header.

Fix project name.
2012-12-28 11:41:30 +01:00
Rudolf Polzer c3cc38e4c4 vo/ao: SDL 1.2+ audio driver, SDL 2.0+ accelerated video driver
This mainly serves as a fallback for platforms where nothing better is
available; also as a debugging help. Both the audio and video driver are
not first class - the audio driver lacks delay detection, and the video
driver only supports a single YUV color space.

Configure options: --disable-sdl2 to disable SDL 2.0+ detection,
--disable-sdl to disable SDL 1.2+ detection. Both options need to be
specified to turn off SDL support entirely.
2012-12-28 08:40:28 +01:00