This new vo is heavily based on vo_gl.c. It provides better scale
filters, dithering, and optional color management with LittleCMS2.
It requires OpenGL 3.
Many features are enabled by default, so it will be slower than vo_gl.
However, it can be tuned to behave almost as vo_gl.
The code used OpenGL 3 specific functions for querying the extension
string when the actual GL 3 context wasn't created yet. This appears to
work fine on nVidia, but could break otherwise. Remove the offending
getFunctions call and retrieve the needed function pointer manually.
(This way the wglCreateContextAttribsARB function pointer can be removed
from struct GL too.)
(Amusingly exposes a wine bug; they made the same mistake.)
Explicitly check the extension string whether the function is available,
although this probably doesn't matter in practice.
Also retrieve bit depth information on win32.
Also include GL/glext.h on windows:
Mingw's (and cygwin's) GL/gl.h has GL/glext.h's inclusion commented
out for some reason. Their glext.h is also ancient, so do yourself
a favor and replace your GL/glext.h with the one from
http://www.opengl.org/registry/api/glext.h .
A workaround is needed for NVidia's broken wglCreateContextAtrribsARB:
It'll return an error if the requested OpenGL version is previous to
3.2 *and* you request a profile... which is exactly *not* what the
wgl_create_context spec says should happen.
Handle it by removing the profile request from attribs[] and retrying
the context creation once more if the first try fails.
And after my first foray into OpenGL I already find a driver quirk.
Oh well.
Also add a bunch of GL functions to the function loader, which will be
needed by vo_gl3. Remove some unused legacy GL functions from the
loader.
Use the proper name for glGetProgramivARB. glGetProgramiv is a different
and incompatible function. The ARB variant is used for ARB shaders,
while the proper one is for GLSL.
The function mp_get_yuv2rgb_coeffs() expects valid values for
input_bits.
When using RGB formats, input_bits is outside the range of what
mp_get_yuv2rgb_coeffs() expects. This doesn't matter since we don't
use the result of that function in the RGB case, but it triggered an
assertion.
This is a regression from commit a816810266,
"vo_gl: improve 10-bit YUV->RGB conversion accuracy slightly".
pa_stream_flush() seems to work pretty badly in general. The visible
symptoms included at least old audio continuing for a significant time
after the call, and bogus latency reporting causing temporary video
freezes after a seek. Add some hacks to work around these problems.
The result seems to work most of the time on my machine at least...
For ao_pulse, the current latency is not a good indicator of how soon
the AO requires new data to avoid underflow. Add an internal pipe that
can be used to wake up the input loop from select(), and make the
pulseaudio main loop (which runs in a separate thread) use this
mechanism when pulse requests more data. The wakeup signal currently
contains no information about the reason for the wakup, but audio
buffers are always filled when the event loop wakes up.
Also, request a latency of 1 second from the Pulseaudio server. The
default is normally significantly higher. We don't need low latency,
while higher latency helps prevent underflows reduces need for
wakeups.
Compared to converting to Y444 this should be faster and lossless.
Based on patch by Hans-Kristian Arntzen [maister archlinux us]
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@34317 b3059339-0415-0410-9bf9-f77b7e298cf2
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@34245 b3059339-0415-0410-9bf9-f77b7e298cf2
Note: ffmpeg first introduced PIX_FMT_GBR24P, which was used in this
commit. Later, it was renamed to PIX_FMT_GBRP in ffmpeg and libav. This
was updated in revision 34492 in mplayer, but the mplayer specific
names (such as IMGFMT_GBR24) were left unchanged.
If the user moved the window to another screen, fullscreen mode would
still use the original screen. Fix to use the screen the window is
currently on (unless overridden by --xineramascreen).
Change the macosx_finder_args function so that when mplayer2 is
invoked from the Finder in a Mac application bundle, it redirects the
output to ~/Library/Logs/mplayer2.log instead of cluttering the global
system.log.
This doesn't affect terminal use which keeps writing to stdout and
stderr.
The gl video output is faster and has more features than corevideo, so
it should be preferred on mac osx.
This doesn't affect GUI compatibility because they specify the
corevideo video output along with the suboptions for the shared buffer
name to mmap in.
The recommended way to get function pointers to the functions in the
OpenGL library is through dlopen/dlsym/dlclose. This causes problems
in the Cocoa OpenGL backend when -lGL (X11's OpenGL headers) is linked
to the binary together with -framework OpenGL.
The linked OpenGL symbols are always from -lGL, causing all the
function pointers to point to null when getFunctions is called against
a Cocoa OpenGL context.
For this reason change the configure autodetection code to disable
the vo_gl X11 backend when cocoa is active.
This video output is not useful anymore. It is based on Carbon to draw
the mplayer window and this has been deprecated by Apple in 10.5.
The upcoming 10.8 OSX release should deprecate most of Carbon, so it
doesn't make sense to keep vo_quartz in the codebase when there are
modern and better alternatives (vo_gl and vo_corevideo).
macosx_finder_args was using Carbon and wasn't usable any longer on
modern versions of MacOSX. This is very useful to embed mplayer in a
mac application bundle.
When using application bundles, the operating system will call the
main function with only one argument that identifies the process
serial number (this is some additional process identifier in osx other
than the pid). File open events are then dispatched to the application
through events that must be handled accordingly.
The Cocoa framework generates only a NS*MouseDown event when handling
the second click of a double click (no NS*MouseUp). If that's the case
put mouse up key in mplayer2's fifo when dealing with the MouseDown
Cocoa event.
Change the window to accept mouse drag events not only on the title
bar, but also on the rest of the window surface; this includes the
video area.
It looks like the changing of the window mask resets the behaviour
specified in the delegate method, probably due to some strange
interaction with NSBorderlessWindow. For this reason call
-setPresentationOptions in the -fullscreen method to remind cocoa the
behaviour we want.
Add option --cursor-autohide-delay to control the number of milliseconds
with no user interaction before the mouse cursor is hidden.
There are two negative values with useful special meanings:
* A value of -1 prevents the cursor from hiding (useful for users
with multiple displays).
* A value of -2 prevents the cursor from showing upon activity.
The default is 1 second to keep the behaviour consistent with the
past X11 backend implementation.
Remove the vo_mouse_autohide field as it was always true.
There were some slight differences between what input.conf mapped, and
what was in input.c def_cmd_binds[]. Make them match.
Add some minor documentation improvements in input.cfg.
Also remove double comments ('##'), because they were confusing.
At least on some keyboards, the key between '0' and 'Enter' on the
key pad is mapped to KP_Separator. Since X11 VOs accept unicode
input, the mplayer keycode this key generates depended on the numlock
state, and with numlock enabled this mapped to an ASCII character.
This is probably not what the user wanted, since two physical keys
will always map to the same key code.
Map it to KP_DEC.
This change allows using non-ASCII keys with X11. These keys were ingored
before.
Technically, this creates an invisible, non-interactive input method
context. If creation fails, the code falls back to the old method, which
allows a subset of ASCII only.
This assumes the terminal uses UTF-8. If invalid UTF-8 is encountered (for
example because the terminal uses a legacy encoding), the code falls back
to the old method and feeds each byte as key code to the input code.
In theory, UTF-8 input could randomly fail, because the code in getch2.c
doesn't try to fill the input buffer correctly with input sequences
longer than a byte. This is a problem with the design of the existing
code.
This moves all key codes above the highest valid unicode code point
(which is 0x10FFFF). All key codes below MP_KEY_BASE now directly map
to unicode (KEY_ENTER is 13, carriage return). Configuration files
(input.conf) can contain unicode characters in UTF-8 to map non-ASCII
characters/keys.
This shouldn't change anything user visible, except that "direct key
codes" (as used in input.conf) will change their meaning.
Parts of the bstr functions taken from libavutil's GET_UTF8 and
slightly modified.
Setting the WM_NAME/WM_ICON_NAME window properties didn't always work:
apparently there are some characters that can't be represented in the X
STRING or COMPOUND_TEXT encodings, such as U+2013 EN DASH. The function
Xutf8TextListToTextProperty partially converts the string, and returns
a value different from 'Success'. This means vo_x11_set_property_string
didn't set these window properties.
On most modern window managers, this is not a problem, since these use
the _NET_WM_NAME/_NET_ICON_NAME and the UTF8_STRING encoding. Some older
WMs like IceWM don't read these, and the window title remains blank.
It's not clear what exactly we should do in this situation, but fix it
by setting set the WM_NAME/WM_ICON_NAME properties as UTF8_TEXT. This
violates the ICCCM, but at least IceWM seems to handle this well.
See also:
http://lists.freedesktop.org/archives/xorg/2004-September/003391.htmlhttp://lists.freedesktop.org/archives/xorg/2004-September/003395.html
Timeline handling converted the pts values from demuxed subtitles to
timeline scale. Change the code to do most subtitle handling in
original subtitle source pts, and instead convert current playback
timeline pts to those units when deciding which subtitle to show.
The main functionality changes are that now demuxed subtitles which
overlap chapter boundaries are handled correctly (at least for libass
subtitles), and external subtitles are assumed to use same pts scale
as current source (this needs improvements later).
Before, a video subtitle that had a duration continuing past the end
of the chapter would continue to be shown for the original duration,
even if the chapter ended and playback switched to a position in the
source where the subtitle shouldn't exist. Now, the subtitle will
correctly end.
Before, external subtitle files were interpreted as specifying pts
values in timeline scale. Now, they're interpreted as specifying pts
values in source file time scale, for _every_ source file. This is
probably more likely to be what the user wants for the "main" source
file in case there is one, but almost certainly not quite right for
multiple source files where the same subs could be shown over
different scenes. If the user wants them to match some main source
file, it's probably still better to have incorrect extra subs for
video from some files than to have every subtitle appearing at the
wrong time. The new code makes it easier to change the interpretation
of the subtitle times, and some configurability should be added in
the future.
Direct rendering support in vo_xv (used with --dr) had at least two
problems. First, OSD drawing modified the buffers; this meant that
if the buffers were used for reference frames there would be video
corruption. I don't think "performance optimization" with this level
of drawbacks is appropriate with today's machines any more. Direct
rendering could still be used for non-reference frames, but there's a
second problem: with direct rendering enabled the same buffer is used
for every frame, and with the XShm extension that is used by default
there's no checking that the previous frame has been completely
uploaded to the graphics card before it's overwritten by the next one.
This could be fixed, but as Xv is becoming obsolete I don't see it as
a priority to improve it. Thus I'm simply removing the parts of
functionality that were more likely to break things than improve
playback.
When switching to a timeline part from another file, decoders were
reinitialized after doing the demuxer-level seek. This is necessary
for audio because some decoders read from the demuxer stream during
initialization and the previous stream position before seek could have
been at EOF. However, this initialization sequence could lose first
subtitles or first part of audio.
The problem for subtitles was that the seek itself or audio
initialization could already have buffered subtitle packets from the
new position, and the way subtitles are reinitialized flushes packet
buffers. Thus early subtitles could be lost (even if they were demuxed
- unfortunately demuxers may not know about still active subtitles
earlier in the file, but that's another issue). Fix this by moving
subtitle and video reinitialization before the demuxer seek; they
don't have the problems which prevent that for audio.
Audio initialization can already decode and buffer some output.
However, the seek_reset() call done last would then throw away this
buffered output. Work around this by adding an extra flag to
seek_reset().
Restructure parts of the code in the main play loop. The main
functionality difference is that if a video track ends first, now
audio will continue to be played until it ends too.
Now the process also wakes up less often if there's no need to update
video or audio. This will reduce unnecessary wakeups especially when
paused, but may make handling of input events laggier when fd-based
notifications are not supported (like most input on Windows).
Now "-ao openal:device=<subdevice>" will pass <subdevice> as device to
OpenAL. This allows selecting both the OpenAL backend (OS-level audio
API) and the physical output device.
The available devices can be listed with "-ao openal:device=help".
The recent changes in mixer.c require the AO to return a volume of
exactly 0 when audio has been muted. Rather than adding just another
special case to mixer.c, fix ao_dsound.c to return previously set
volumes exactly. Because DirectSound volume control is not connected
with the system mixer, which could change the volume without mplayer
knowing, reading the volume back from DirectSound is pointless.
Also, the code tried to calculate log10(0). Clip the volume to 1,
which results in -10000, DirectSound's definition of silence.
When layback of a file ends, the audio output doesn't receive new audio
data, but the rest of the data must be played properly. ao_dsound.c
doesn't handle this properly: DirectSound will continue to play the
ringbuffer, even if mplayer doesn't write any data. There's no explicit
way to prevent such a buffer underrun. Try to detect it and stop
playback.
Add the flag D3DCREATE_FPU_PRESERVE, which tells Direct3D not to switch
the FPU to single precision mode. Single precision mode would mean that
all floating point calculations are done in float precision, even if
using double variables.
The MSDN documentation seems to discourage use of this flag with scary
warnings about bad performance and stability, but I suspect in practice
switching off this completely unreasonable behavior is fine.
wm4
Stefano Pigozzi
Kovensky
Uoti Urpala
reimar
cehoyos