Since Windows Vista, when running at 144 DPI or higher with composition
switched on, applications that don't declare themselves to be DPI aware
are stretched by the window manager, kind of like low resolution apps in
OSX.
To avoid this, declare DPI awareness in the manifest. Since mpv is
practically resolution independent this shouldn't cause any trouble. The
'True/PM' value declares per-monitor DPI awareness in Windows 8.1, so
that the mpv isn't shrunk when moved from a high DPI screen to one with
a lower DPI.
Also, avoid compatibility shims by declaring compatibility with all
Windows versions from Vista to 8.1 and add the missing uiAccess
attribute to the requestedExecutionLevel element.
Move the reading loop from read_all_fd_events to read_events. If
got_new_events is set when calling read_events, don't actually wait
and set the timeout to 0.
(Note that not waiting is sort of transparent to the caller: the caller
is just supposed to execute the event loop again, and then it will
actually wait. mplayer.c handles this correctly.)
This might reduce latency with some input sources.
Removes some code duplication. Also restructure the input waiting code
a bit: split the select() loop into a input_wait_read() function. On
systems which do not have POSIX select(), this function has an alternate
implementation, which waits unconditionally.
This was useless for anything but the raw demuxers. In most cases, this
would most likely lead to display of bogus duration values, because the
bitrates used are per-track, not the total file bitrate. There was
actually no case left where this code was helpful.
Note that demux_lavf has its own code for this using the total file
bitrate. Also, mplayer.c can calculate the playback percentage from
current file position / current file size. This is not removed.
All demuxers make a reasonable effort to set packet timestamps, and thus
support correct-pts mode. This commit also implicitly switches
demux_rawvideo to correct-pts mode.
We still allow demuxers to disable correct-pts mode in theory.
These were printed only with -v. Most streams had them set to useless
or redundant values, so it's just badly maintained bloat.
Since we remove the "author" field too, and since this may have
copyright implications, we add the contents of the author fields to
the file headers, except if the name is already part of the file header.
Stream implementations could set this to a unix file descriptor. The
generic stream code could use it as fallback for a few things. This
was confusing and insane. In most cases, the stream implementations
defined all callbacks, so setting the fd member didn't have any
advantages, other than avoiding defining a private struct to store it.
It appears that even if the stream implementation used close() on the
fd (or something equivalent), stream.c would close() it a second time
(and on windows, even would call closesocket()), which should be proof
for the insanity of this code.
For stream_file.c, additionally make sure we don't close stdin or
stdout if "-" is used as filename.
For stream_vcd.c, remove the control() code. This code most likely
didn't make the slightest sense, because it used a different type
for stream->priv. It also leaked memory. Maybe it worked, but it's
incorrect and insignificant anyway, so kill it. This code was added
with commit 9521c19 (svn commit 31019).
Untested for all protocols other than stream_file.c.
Get rid of the strange and messy reliance on DEMUXER_TYPE_ constants.
Instead of having two open functions for the demuxer callbacks (which
somehow are both optional, but you can also decide to implement both...),
just have one function. This function takes a parameter that tells the
demuxer how strictly it should check for the file headers. This is a
nice simplification and allows more flexibility.
Remove the file extension code. This literally did nothing (anymore).
Change demux_lavf so that we check our other builtin demuxers first
before libavformat tries to guess by file extension.
This removes the dependency on DEMUXER_TYPE_* and the file_format
parameter from the stream open functions.
Remove some of the playlist handling code. It looks like this was
needed only for loading linked mov files with demux_mov (which was
removed long ago).
Delete a minor bit of dead network-related code from stream.c as well.
The main problem is that this m_struct stuff uses pointers for offsets
(why...), so we mangle it by intptr_t. This stuff really should use ints
(or in theory ptrdiff_t) for offsets, but changing it would be too much
effort, and hopefully this m_struct stuff will go away and replaced by
the common option parser mechanism instead.
Shuts up warnings on Windows.
Patch suggested by jon_y and rossy on IRC.
Move codec_tags.h include to demux_mkv.c, because this is the only file
which still uses it.
Move new_sh_stream() to demux.h, because this is more proper.
Before this commit, we tried to play along with libavformat and tried
to pretend that attached pictures are video streams with a single
frame, and that the frame magically appeared at the seek position when
seeking. The playback core would then switch to a mode where the video
has ended, and the "remaining" audio is played.
This didn't work very well:
- we needed a hack in demux.c, because we tried to read more packets in
order to find the "next" video frame (libavformat doesn't tell us if
a stream has ended)
- switching the video stream didn't work, because we can't tell
libavformat to send the packet again
- seeking and resuming after was hacky (for some reason libavformat sets
the returned packet's PTS to that of the previously returned audio
packet in generic code not related to attached pictures, and this
happened to work)
- if the user did something stupid and e.g. inserted a deinterlacer by
default, a picture was never displayed, only an inactive VO window)
- same when using a command that reconfigured the VO (like switching
aspect or video filters)
- hr-seek didn't work
For this reason, handle attached pictures as separate case with a
separate video decoding function, which doesn't read packets. Also,
do not synchronize audio to video start in this case.
The code touched by this commit makes sure that DVD subtitle tracks
known by libdvdread but not known by demux_lavf can be selected and
displayed properly. These subtitle tracks have the first packet
some time late in the packet stream, so that libavformat won't
immediately recognize them, and will add the track as soon as the
first packet is seen during normal demuxing.
demux_mpg used to handle this elegantly: you just set the MPEG ID of
the stream you wanted. demux_lavf couldn't do this, so it was emulated
with a DEMUXER_CTRL. This commit changes it so that new streams are
selected by default (if autoselect is enabled), and the playloop
simply can take appropriate action before the lower layer throws away
the first packet.
This also changes the demux_lavf behavior that subtitle packets are
always demuxed, even if not needed. (They were immediately thrown away,
so there was no advantage to this.)
Further, this adds the ability to demux.c to deal with demuxing more
than one stream of a kind at once. (Though currently it's not useful.)
AVDISCARD_DEFAULT is probably a bit better for normal decoding.
AVDISCARD_NONE would (as by documentation) include "useless" packets
too, while DEFAULT filters these.
It turns out that some code that was removed earlier was still needed.
avcodec_decode_audio4() can decode packets "partially". In that case,
you have to "slice" the packet and call the decode function again.
Codecs which need this are obscure and in low numbers. One sample that
needs it is here:
rsync://fate-suite.ffmpeg.org/fate-suite/lossless-audio/luckynight-partial.shn
(This one decodes in rather small increments.)
The new code is much simpler than what has been removed earlier,
though. The fact that we own the packet returned by the demuxer helps
a lot.
Not sure what should happen if avcodec_decode_audio4() returns 0.
Currently, we throw away the packet in this case. We don't want to be
stuck in an endless loop (could happen if the decoder produces no
output either).
Generally remove all accesses to demux_stream from all the code, except
inside of demux.c. Make it completely private to demux.c.
This simplifies the code because it removes an extra concept. In demux.c
it is reduced to a simple packet queue. There were other uses of
demux_stream, but they were removed or are removed with this commit.
Remove the extra "ds" argument to demux fill_buffer callback. It was
used by demux_avi and the TV pseudo-demuxer only.
Remove usage of d_video->last_pts from the no-correct-pts code. This
field contains the last PTS retrieved after a packet that is not NOPTS.
We can easily get this value manually because we read the packets
ourselves. Reuse sh_video->last_pts to store the packet PTS values. It
was used only by the correct-pts code before, and like d_video->last_pts,
it is reset on seek. The behavior should be exactly the same.
Currently, all demuxer fill_buffer functions have a demux_stream
parameter. We want to remove that, but the TV code still depends on
it. Add a hack to remove that dependency.
The problem with the TV code is that reading video and audio frames
blocks, so in order to avoid a deadlock, you should read either of
them only if the decoder actually requests new data.
For now, we want to get rid of the demux->sub access, because this
field will become private to demux.c in a later commit. So replace the
current hack with another hack.
The need for the hack will be removed sooner or later. (Instead of
autoselecting a specific stream, all new streams will be enabled by
default, so that no packets can get lost. The frontend will then be
responsible to deselect unwanted streams.)
This is not directly related to the handling of format changes itself,
but playing audio normally after the change. This was broken: the output
byte rate was not recalculated, so audio-video sync was simply broken.
Fix this by calculating the byte rate on the fly, instead of storing it
in sh_audio.
Format changes are relatively common (switches between stereo and 5.1
in TV recordings), so this fixes a somewhat critical bug.
pts_bytes can't just be changed at the end. It must be offset to the pts
value, which is reset with each packet read from the demuxer. Make sure
the pts_byte field is always reset after receiving a new PTS, i.e.
increment it after actually writing to the output buffer.
Flush the AVFormatContext's write buffer, because otherwise the audio
PTS will jump around too much: the calculation doesn't use the exact
output buffer size if there's still data in the avio buffer.
Removing this code doesn't change anything. All remaining audio decoders
are well-behaved enough to not overwrite sh_audio->pts if they don't
know the PTS. And if they don't know the PTS, the d_audio->last_pts
field can't contain any usable value either, because both fields contain
theame value: the last known valid PTS found in an audio packet.
As the comment n the removed code says, this was once needed for
something subtitle related. This code has been cleaned up long ago,
so at least the original reason for it is gone.
wm4
James Ross-Gowan
Stefano Pigozzi
Jonathan Yong