Somehow this was setup such that a BlockGroup can be incrementally
read (at least in theory). This makes no sense, as BlockGroup can
contain only one Block (despite its name). There's no need to read
this incrementally, and makes the code confusing for no gain.
Read all the BlockGroup sub-elements with a single function call,
without keeping global state for BlockGroup parsing.
The code for reading block data was duplicated. Move it into a function.
Instead of returning on error (possibly due to corrupt data) and
signalling EOF, continue by trying to find the next block. This makes
error handling slightly simpler too, because you don't have to care
about freeing the current block. We could still signal EOF in this case,
but trying to resync sounds better for dealing with corrupted files.
Matroska files prepared for streaming have clusters with unknown size.
These files are pretty rare, see e.g. test4.mkv from the official
Matroska test file collection.
The end positions of the current cluster and block were managed by
tracking their size and how much of them were read, instead of just
using the absolute end positions.
I'm not sure about the reasons why this code was originally written
this way. One obvious concern is reading from pipes and such, but the
stream layers hides this. stream_tell(s) works even when reading from
pipes. It's also a fast call, and doesn't involve the stream
implementation or syscalls. Keeping track of the cluster/block end is
simpler and there's no reason why this wouldn't work.
Incomplete files don't have a valid index, because the index is usually
located near the end of a file. In this case, an index is created on the
fly during demuxing, or when seeks are done.
This used a completely different code path, which leads to unnecessary
complications and code duplication. Use the normal index data structure
instead. The seeking code at the end of seek_creating_index() (in this
commit renamed to create_index_until()) is removed. The normal seek code
does the same thing instead.
This refactor makes the code easier to understand. Also corrects a bug that
caused the window to move to the left when the new size was bigger than the
visible frame.
I am aware this detection may occur too late, depending on other
settings, but at least it usually works and is portable.
Where the output fd can be changed, though, it'd be better to force a
similar behaviour via file descriptor use: use pipe:3 as output to FD 3,
and change the calling program to expect the stream on FD 3.
These require bleeding edge libass (latest git version), and will be
ignored otherwise.
I'm not sure about the blur factor and scaling. The ASS/VSFilter
semantics for blur scaling are a bad mess. Might require further
investigation.
Add dummy input and output filters to remove special cases in the format
negotiation code (af_fix_format_conversion() etc.). The output of the
filter chain is now negotiated in exactly the same way as normal
filters.
Negotiate setting the sample rate in the same way as other audio
parameters. As a side effect, the resampler is inserted at the start of
the filter chain instead of the end, but that shouldn't matter much,
especially since conversion and channel mixing are conflated into the
same filter (due to libavresample's API).
Anything this option did has been removed in the preceding 3 commits.
Note that even though these options sounded like a good idea (like
setting accuracy vs. speed tradeoffs), they were not really properly
implemented.
All this option did was deciding whether the resample filter was to be
insert at the beginning or end of the filter chain. Always do what the
option set for accuracy did. I doubt it makes much of a difference.
libavresample does most things in just one go anyway, so it won't
matter.
Dangerous and misleading. If it turns out that this is actually needed
to make certain setups work right, it should be added back in a better
way (in a way it doesn't cause random crashes).
The only thing this option did was changing the behavior of af_volume.
The option decided what sample format af_volume would use, but only if
the sample format was not already float. If the option was set, it would
default to float, otherwise to S16.
Remove use of the option and all associated code, and make af_volume
always use float (unless a af_volume specific sub-option is set).
Silence maximum value tracking. This message is printed when the filter
is destroyed, and it's slightly annoying. Was enabled due to enabling
float by default.
Switch the internal channel order to libavcodec's. If the channel number
mismatches at some point, use libavresample for up- or downmixing.
Remove the old af_pan automatic downmixing.
The libavcodec channel order should be equivalent to WAVEFORMATEX order,
at least nowadays. reorder_ch.h assumes that WAVEFORMATEX and libavcodec
might be different, but all defined channels have the same mappings.
Remove the downmixing with af_pan as well as the channel conversion with
af_channels from af.c, and prefer af_lavrresample for this. The
automatic downmixing behavior should be the same as before (if the
--channels option is set to 2, which is the default, the audio output
is forced to 2 channels, and libavresample does all downmixing).
Note that mpv still can't do channel layouts. It will pick the default
channel layout according to the channel count. This will be fixed later
by passing down the channel layout as well.
af_hrtf depends on the order of the input channels, so reorder to ALSA
(for which this code was written). This is better than changing the
filter code, which is more risky.
ao_pulse can accept waveext order directly, so set that as channel
mapping.
If format negotiation fails, and additional filters are inserted to fix
this, don't try to reinitialize the filter immediately. Instead, correct
the audio format, and let the caller retry.
Add a retry counter to af_reinit() to ensure that misbehaving filters
can't put the format negotiation into an endless loop.
Refactor to remove the duplicated format filter insertion code. Allow
other format converting filters to be inserted on format mismatches.
af_info.test_conversion checks whether conversion between two formats
would work with the given filter; do this to avoid having to insert
multiple conversion filters at once and such things. (Although this
isn't ideal: what if we want to avoid af_format for some conversions?
What if we want to split af_format in endian-swapping filters etc.?)
Prefer af_lavrresample for conversions that it supports natively,
otherwise let af_format handle the full conversion.
Make sure automatically inserted filters are removed on full reinit
(they are re-added later if they are really needed). Automatically
inserted filters were never explicitly removed, instead, it was
expected that redundant conversion filters detach themselves. This
didn't work if there were several chained format conversion filters,
e.g. s16le->floatle->s16le, which could result from repeated filter
insertion and removal. (format filters detach only if input format and
output format are the same.)
Further, the dummy filter (which exists only because af.c can't handle
an empty filter chain for some reason) could introduce bad conversions
due to how the format negotiation works. Change the code so that the
dummy filter never takes part on format negotiation. (It would be better
to fix format negotiation, but that would be much more complicated and
would involving fixing all filters.)
Simplify af_reinit() and remove the start audio filter parameter. This
means format negotiation and filter initialization is run more often,
but should be harmless.
The format was locked to s16. Extend it to accept all other ffmpeg
sample formats, and even allow different in- and output formats. The
generic filter code will still insert af_format on format mismatches,
though.
The change in af_scaletempo actually fixes a memory leak. af->data
contained a pointer to an allocated buffer, which was overwritten
during format negotiation. Set the format explicitly instead.
The condition that checked whether the chapters are out of order and
should be sorted was inverted. This likely wasn't noticed in testing,
because even if the chapters are unsorted, if the last two chapters
were sorted, the rest got sorted too.
Instead of doing this silly check, always sort the chapters after
demuxer initialization. Also make sure the sort order is stable in case
chapter start times are the same (original_index check).
This option can be used to selectively reset settings when playing the
next file in the playlist (i.e. restore mplayer and mplayer2 behavior).
Might remove this option again should it turn out that nobody uses it.
Consider:
mpv --volume 10 file1.mkv file2.mkv
Before this commit, the volume was reset to 10 when playing file2.mkv.
This was inconsistent to most other options. E.g. --brightness is a
rather similar case.
In general, settings should never be reset when playing the next file,
unless the option was explicitly marked file-local. This commit
corrects the behavior of the --volume and --mute options.
File local --volume still works as expected:
mpv --{ --volume 10 file1.mkv file2.mkv --}
This sets the volume always to 10 on playback start.
Move the m_config_leave_file_local() call down so that the mixer code
in uninit_player() can set the option volume and mute variables without
overwriting the global option values.
Another subtle issue is that we don't want to set volume if there's no
need to, which is why the user_set_volume/mute fields are introduced.
This is important because setting the volume might change the system
volume depending on other options.
At least libsdl adds -mwindows to the cflags, which marks the .exe
binary as GUI application. This means the program detaches from the
console when started in cmd.exe, instead of showing the playback
status, receiving console input, and so on.
Append -mconsole to the linker command line to disable -mwindows.
CC demux/demux.o
demux/demux.c: In function 'demuxer_switch_track':
demux/demux.c:1241:29: warning: array subscript is above array bounds [-Warray-bounds]
int new_id = demuxer->ds[type]->id;
^
