This returned a stream error value directly to libavformat, which can't
make sense. For example STREAM_ERROR (0) means success in libavformat
error codes. (The meaning of the libavformat read_seek return value is
underdocumented too.)
Something like "char *s = ...; isdigit(s[0]);" triggers undefined
behavior, because char can be signed, and thus s[0] can be a negative
value. The is*() functions require unsigned char _or_ EOF. EOF is a
special value outside of unsigned char range, thus the argument to the
is*() functions can't be a char.
This undefined behavior can actually trigger crashes if the
implementation of these functions e.g. uses lookup tables, which are
then indexed with out-of-range values.
Replace all <ctype.h> uses with our own custom mp_is*() functions added
with misc/ctype.h. As a bonus, these functions are locale-independent.
(Although currently, we _require_ C locale for other reasons.)
Some of these might be security relevant.
The RealAudio code was especially bad. I'm not sure if all RealAudio
stuff still plays correctly; I didn't have that many samples for
testing. Some checks might be unnecessary or overcomplicated compared
to the (obfuscated) nature of the code.
CC: @mpv-player/stable
Also clarify the semantics.
It seems --idx didn't do anything. Possibly it used to change how the
now removed legacy demuxers like demux_avi used to behave. Or maybe
it was accidental.
--forceidx basically becomes --index=force. It's possible that new
index modes will be added in the future, so I'm keeping it
extensible, instead of e.g. creating --force-index.
Probably "needed" to get the correct alignment, although I'm not aware
of actual breakages or performance issues.
In fact we should probably always just allocate AVPackets, but for now
use the simple fix.
FFmpeg requires a bullshit padding after each input buffer, and they
just increased that padding without warning and without ABI or API bump.
We need this only in one file (although mp_image hardcodes something
similar, for which no FFmpeg API define is available), so drop our own
define.
While I'm not very fond of "const", it's important for declarations
(it decides whether a symbol is emitted in a read-only or read/write
section). Fix all these cases, so we have writeable global data only
when we really need.
Convert all these commands to properties. (Except tv_last_channel, not
sure what to do with this.) Also, internally, don't access stream
details directly, but dispatch commands with stream ctrls.
Many of the new properties are a bit strange, because they're write-
only. Also remove some OSD output these commands produced, because I
couldn't be bothered to port these.
In general, this makes everything much cleaner, and will also make it
easier to e.g. move the demuxer to its own thread.
Don't bother updating input.conf, but changes.rst documents how old
commands map to the new ones.
Mostly untested, due to lack of hardware.
In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.
The i_bps members of the sh_audio and dev_video structs are mostly used
for displaying the average audio and video bitrates. Keeping them in
bits-per-second avoids truncating them to bytes-per-second and changing
them back lateron.
Stop using it in most places, and prefer STREAM_CTRL_GET_SIZE. The
advantage is that always the correct size will be used. There can be no
doubt anymore whether the end_pos value is outdated (as it happens often
with files that are being downloaded).
Some streams still use end_pos. They don't change size, and it's easier
to emulate STREAM_CTRL_GET_SIZE using end_pos, instead of adding a
STREAM_CTRL_GET_SIZE implementation to these streams.
Make sure int64_t is always used for STREAM_CTRL_GET_SIZE (it was
uint64_t before).
Remove the seek flags mess, and replace them with a seekable flag. Every
stream must set it consistently now, and an assertion in stream.c checks
this. Don't distinguish between streams that can only be forward or
backwards seeked, since we have no such stream types.
stream.start_pos was needed for optical media only, and (apparently) not
for very good reasons. Just get rid of it.
For stream_dvd, we don't need to do anything. Byte seeking was already
removed from it earlier.
For stream_cdda and stream_vcd, emulate the start_pos by offsetting the
stream pos as seen by the rest of mpv.
The bits in discnav.c and loadfile.c were for dealing with the code
seeking back to the start in demux.c. Handle this differently by
assuming the demuxer is always initialized with the stream at start
position, and instead seek back if initializing the demuxer fails.
Remove the --sb option, which worked by modifying stream.start_pos. If
someone really wants this option, it could be added back by creating a
"slice" stream (actually ffmpeg already has such a thing).
Drop: sami, vplayer, rt, pjs, mpsub, aqt, jacosub. None of these seem
to be actually in use, except sami. Sami is very complex, and the
results subreader produces are not very useful.
For all these formats, there are still parsers in FFmpeg. We remove the
subreader implementation, because it might contain security relevant
bugs and such. (This is old, unmaintained C string parsing code, written
in times where absolutely nobody cared about security. The kind of
awesome code.)
We keep the other formats, because they're (mostly) commonly used and
relatively simple, for UTF16 support (still missing in FFmpeg), and for
the sake of Libav.
Caused failure to detect .pls files, because they were misdetected as
m3u. The problem is that "forcing playlist files" and "forcing a
specific playlist format" are not really treated separate, and in both
cases p->force is set to true. This made m3u detect all files as m3u
if --playlist was used. So correctly check whether the file format is
actually being probed or not.
Because the http playlist URL I had for testing claimed to be m3u by
file extension and mime type, but didn't have the header.
Note that this actually changes behavior only in the case the format is
detected by mime type. Then p->force will be set before calling the
parser, and the header becomes optional.
mp3 has a hack lowering the probescore for format detection. This is
because detecting mp3s is hard due to their nature, and the fact that
ID3v2 tags are sometimes several megabytes big.
When playing mp3 from network, the mime-type is usually set, and that
matches the format hack entry meant for webradios, overriding the normal
mp3 entry. This can lead to network mp3s not being detected. Lower the
network case to the same probescore as on-disk mp3s. The difference is
that for network mp3s, we don't load the full probe-buffer, and we lower
the amount of audio the demuxer will read to collect data on opening
(0.5 seconds instead of typically 5 seconds).
Also remove MSGL_SMODE and friends.
Note: The indent in options.rst was added to work around a bug in
ReportLab that causes the PDF manual build to fail.
VP9 packets can contain 2 frames in some video packets (from which 1
frame is invisible). Due to a design mismatch between libvpx and the
libavcodec vp9 decoder, libvpx can take the "full" packets, but lavc vp9
can not. The consequence is that we have to split the packets if we want
to feed them to the lavc codec.
This is not entirely correct yet: timestamp handling is missing.
--demuxer=lavf and ffmpeg native utilities have the same problem. We can
fix this only once the ffmpeg VP9 parser is fixed.
For some reason, some files appear to have broken mp3 packets, or at
least in a form that libavcodec can't deal with. The audio in the sample
file in question could not be decoded using libavcodec.
The problematic file had variable packet sizes, and the libavcodec
decoder kept printing "mp3: Header missing" for each packet it was fed.
Remuxing with mkvmerge fixes the problem. The mp3 data is probably not
VBR, and remuxing resulted in fixed-size mp3 frames. So I don't know why
the sample file was muxed this way - it might just be incorrect.
The sample file had "libmkv 0.6.4" as MuxingApp (although I could not
get mkvinfo to print this element, maybe the file uses an incorrect
element ID), and "HandBrake 0.9.4" as WritingApp.
Note that the libmpg123 decoder does not have any issues with it. It's
probably more robust, because libmpg123 was made to decode whole mp3
files, not just single frames.
Fixes issue #742.
glob() is mandated by POSIX. For the only non-POSIX platform we support,
Windows, we have our own replacement. So the ifdeffery is not needed.
Still leave the checks in the configure scripts, because they have to
decide whether to compile the replacement or not. (Although this could
be special cased to mingw-only, the wscript seems to make this hard.)
glob-win.c wasn't big, so it was easier to rewrite it. The new version
supports Unicode, handles directories properly, sorts the output and
puts all its allocations in the same talloc context to simplify the
implementation of globfree.
Notably, the old glob had error checking code, but didn't do anything
with the errors since the error reporting code was commented out. The
new glob doesn't copy this behaviour. It just treats errors as if there
were no more matching files, which shouldn't matter for mpv, since it
ignores glob errors too.
To match the other Windows I/O helper functions, the definition is moved
to osdep/io.h.
I hate tabs.
This replaces all tabs in all source files with spaces. The only
exception is old-makefile. The replacement was made by running the
GNU coreutils "expand" command on every file. Since the replacement was
automatic, it's possible that some formatting was destroyed (but perhaps
only if it was assuming that the end of a tab does not correspond to
aligning the end to multiples of 8 spaces).
This was broken at some unknown point (even before the recent cache
changes). There are several problems:
- stream_dvd returning a random stream position, confusing the cache
layer (cached data and stream data lost their 1:1 corrospondence by
position)
- this also confused the mechanism added with commit a9671524, which
basically triggered random seeking (although this was not the only
problem)
- demux_lavf requesting seeks in the stream layer, which resulted in
seeks in the cache or the real stream
Fix this by completely removing byte-based seeking from stream_dvd. This
already works fine for stream_dvdnav and stream_bluray. Now all these
streams do time-based seeks, and pretend to be infinite streams of data,
and the rest of the player simply doesn't care about the stream byte
positions.
Instead, always use the mpctx->chapters array. Before this commit, this
array was used only for ordered chapters and such, but now it's always
populated if there are chapters.
Stream-level chapters (like DVD etc.) did potentially not have
timestamps for each chapter, so STREAM_CTRL_SEEK_TO_CHAPTER and
STREAM_CTRL_GET_CURRENT_CHAPTER were needed to navigate chapters. We've
switched everything to use timestamps and that seems to work, so we can
simplify the code and remove this old mechanism.
Instead of parsing the ASS file in demux_libass.c and trying to pass the
ASS_Track to the subtitle renderer, just read all file data in
demux_libass.c, and let the subtitle renderer pass the file contents to
ass_process_codec_private(). (This happens to parse full files too.)
Makes the code simpler, though it also relies harder on the (messy)
probe logic in demux_libass.c.
The mplayer decoder (spudec.c) actually handled this. There was explicit
code for binary palettes (16 32 bit values), and the subtitle resolution
was handled by video resolution coincidentally matching the subtitle
resolution.
Whoever puts vobsub into mp4 should be punished.
Fixes the sample gundam_sample.mp4, closes github issue #547.
This skipped all audio packets before the first video key frame was
found. I'm not really sure why this would be needed; most likely it
isn't. So get rid of it. Even if audio packets are returned to the
player too soon, the player will sync the audio start to the video
start by decoding and discarding audio data.
Note that although the removed code was just added in the previous
commit, it merely kept the old keeping semantics which demux_mkv
always followed. This commit removes these special semantics.
v_skip_to_keyframe is set to true while non-keyframe video packets are
skipped. Until now, audio packets were also skipped when doing this. I
can't see any good reason why this would be done, but for now I want to
keep the old logic when audio+video seeks are done.
However, for audio-only mode, do proper seeking, which also fixes
behavior when trying to seek past the end of the file: playback is
terminated properly, instead of starting playback on the start of the
last cluster.
Note that a_no_timecode_check is used only for audio+video seek. I'm
not sure what this is needed for, but it might influence A/V sync after
seeking.
Trying to set a non-existent flag (like +keepside on Libav) causes
libavutil print an incomprehensible warning (something about eval;
probably the overengineered libavutil option parser tripping over the
'+' normally used for flags, and trying to interpret it as formula).
There's apparently no easy way to check for the existence of a flag,
so add some more ifdeffery to shut it up.
There is some logic to discard packets from streams that are not
selected. Run the metadata update code before this, just to make 100%
sure that no metadata updates can be lost when streams are deselected.
(I'm not sure why this logic would be needed, since both libavformat and
the generic demuxer code do this already. But a quick test shows that
av_read_frame() can return a packet from a stream even if the stream has
AVStream.discard set to AVDISCARD_ALL. This happened after stream
switching. Maybe libavformat doesn't discard already queued packets.)
Instead of printing lines like:
Demuxer info GENRE changed to Alternative Rock
Just output all tags once they change. The assumption is that individual
tags rarely change, while all tags change in the common case.
This changes tag updates to use polling. This could be fixed later,
although the ICY stuff makes it a bit painful, so maybe it will remain
this way.
Also remove DEMUXER_CTRL_UPDATE_INFO. This was intended to check for tag
updates, but now we use a different approach.
Use an arbitrary constant instead, which is as good as PATH_MAX.
This helps us to avoid having to think about pull request #523.
Also fix a case where a potentially signed char was passed to isspace().
This used to work; I'm not sure when or why it regressed. When setting
AVProbeData.filename to NULL, libavformat will crash in rtp_probe() by
unconditionally accessing the string.
We used to set the filename to NULL to prevent probing by file extension
when we don't deem it as necessary. Using an empty string also works for
this purpose.
This generally affects mp3 files that don't have any (or many) mp3
frames in the first 2 MB. 2 MB is the maximum probe size, and
libavformat returns a low probescore even if we give it the full 2 MB.
Trying to probe a larger buffer (or even the full file) doesn't work for
mysterious reasons.
The workaround consists in accepting a very weak probescore if the
format is detected as mp3 and we probed already 2 MB.
Restructure it a bit, so we can use the format hack list even if no mime
type applies. Shouldn't change anything functionally yet. Preparation
for the next commit.
If there's more than one edition, print the list of editions, including
the edition name, whether the edition is selected, whether the edition
is default, and the command line option to select the edition. (Similar
to stream list.)
Move reading the tags to a separate function process_tags(), which is
called when all other state is parsed. Otherwise, that tags will be lost
if chapters are read after the tags.
Pretty worthless. This is called from the seek code, which will
reinitialize these anyway. Even if seeking somehow decides to fail, the
new values are still valid.
One could say a failed seek (if that happens) should jump back to the
original position, and thus it would be better to make sure the state
is restored. But then demux_mkv_seek needs to do this correctly,
including not setting up skipping to the target timestamp. But not
bothering with this.
Extremely obscure corner case with concatenated segments, in which EOF
wasn't recognized correctly, and it tried to demux clusters from the
next segment.
See [MKV]_Editions,_Linked_Segments,_&_Tracksets.mkv from the CCCP test
file collection.
This basically used to be part of the user interface, before mpv moved
printing the track list to the frontend, and this code was raised to
verbose output level.
For some reason, if an error happened when reading headers, it merely
stopped reading the headers, and then continued normally. (It looks like
the case to exit hard (-2) was mainly used for skipping unwanted ordered
chapter segments.)
I can't comprehend this. Always exit on error when reading headers.
(Maybe some more error tolerance would be good, but I have no test case,
and there's some danger of entering endless loops.)
This makes everything more robust, and also somewhat simpler (even if
the diffstat isn't very impressive).
Instead of recursively following SeekHeads while reading headers, just
read the headers until the first cluster, and then possibly use
SeekHeads to read the remaining missing headers.
As of this commit, stream_read_line() can't actually error (except in
the case the passed in buffer is 0, which never happens here). This
commit is preparation for the following commit, which checks harder
whether the read data is actually text. Before this commit, an error
was treated as end-of-file, but the data read so far was considered
valid.
Many ebml_read_* functions have a length int pointer parameter, which
returns the number of bytes skipped. Nothing actually needed this
(anymore), and code using it was rather hard to understand, so get rid
of them.
Matroska makes it pretty hard to resync correctly on broken files:
random data returns "valid" EBML IDs with a high probability, and when
trying to skip them it's likely that you skip a random amount of data
(instead of considering the element length invalid).
Improve upon this by skipping known level 1 elements only. Consider
everything else invalid and call the resync code. This might result in
annoying behavior when Matroska adds new level 1 elements, although it
won't be particularly harmful. Matroska doesn't really allow us to do
better (even mkvtoolnix explicitly checks for known level 1 elements).
Since we now don't always want to combine EBML element skipping and
resyncing, remove ebml_read_skip_or_resync_cluster(), and make
ebml_read_skip() more tolerant against skipping broken elements.
Also, don't resync when reading sub-elements, and instead do resyncing
when reading them results in an error.
Until now, corrupted files were detected if the size of an element (that
should be skipped) was larger than the remaining file. This still could
skip larger regions of the file itself if the broken size happened to be
within the file.
Change it so that it's never allowed to skip outside the parent's
element.
MicroDVD files _can_ contain real timestamps instead of frame timestamps
if they declare a FPS. But this seems to be rare, so ignore that if the
FPS happens to match with the libavformat microdvd parser's default FPS.
This might actually break files that declare 23.976 FPS, but the video
file is not 23.976 FPS, but the chance that this happens is probably
very low, and the commit fixes the more common breakage with 25 FPS
video.
demux_subreader.c contains the old MPlayer subtitle parser, and I have
absolutely no confidence in this (very crappy) code. There might be
one or two security risks associated with running that code on
arbitrary input.
Apparently, Matroska packs TrueHD packets in a way lavc doesn't expect.
This broke decoding of some files [1] completely. A short look at the
libavcodec parser shows that parsing this ourselves would probably be
too much work, so make use of the libavcodec parser API.
[1] http://www.cccp-project.net/beta/test_files/mzero_truehd_sample.mkv
The TV code pretends to be part of stream/, but it's actually demuxer
code too. The audio_in code is shared between the TV code and
stream_radio.c, so stream_radio.c needs a small hack until stream.c is
converted.
