| Commit message (Collapse) | Author | Age | Files | Lines |
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If demux_open_lavf fails between calling avformat_alloc_context() and
assigning the context to priv->avfc, it will never be properly freed.
Fixes #11793.
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Unfortunately Content-Type matching seems to be not enough as there are
mistagged streams. Try to match extension and pass-through URL.
Fixes #11700
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This fixes HLS playback. After FFmpeg@954d16f check is strict as per
RFC8216 requirement and demuxer need to have this information to work
properly.
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Fixes local HLS playback.
After FFmpeg@6b1f68c commit HLS is only processed if file extension
matches RFC8216 requirement.
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This reverts commit acababec208ec4f26c1462228a9ec1d4aac2c815.
Now it has been fixed upstream FFmpeg@6b1f68c.
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Pass "dummy.m3u8" filename to work around FFmpeg commit
6b1f68ccb04d791f0250e05687c346a99ff47ea1 which broke their HLS demuxer
and its ability to probe.
Since the above commit, libavformat will check the filename of the file
to be probed and reject it if it doesn't end with a valid HLS extension
i.e. m3u8,hls,m3u (never mind that .hls is not a valid HLS extension).
In addition to a bug with query strings, this also breaks mpv
functionality as mpv explicitly doesn't tell libavformat the filename
when probing, in order to properly detect the file based only on their
contents.
The [HLS specification](https://www.rfc-editor.org/rfc/rfc8216.txt) aka
RFC 8216, specifies in section 4 that "Each Playlist file MUST be
identifiable either by the path component of its URI or by HTTP
Content-Type." Notably, it does not require both, so this FFmpeg commit
is noncompliant. We work around this noncompliance by checking the MIME
type ourselves. If the mimetype matches one of the valid HLS mimetypes
(and also application/x-mpegurl, a legacy pre-standardization type),
then we pass "dummy.m3u8" to libavformat in order to work around its
overly strict checking of filenames.
Without this patch, we are unable to play many HLS streams, including a
few from the ytdl hook. This patch restores those ability to play those
streams when built against FFmpeg master. Do note that if the server
sends an invalid content-type header then we cannot implement this
workaround so those streams will still fail to play.
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`io_close2` was introduced as a superior replacement for `io_close` in
ffmpeg 5.0, and then deprecated in 6.0. The difference is that
`io_close2` can return errors. In our case, we're just calling through
to the original function anyway, so we don't need to do more than pass
the return value back.
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This can be useful in stream selection.
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c78482045444c488bb7948305d583a55d17cd236 introduced a bool option type
as a replacement for the flag type, but didn't actually transition and
remove the flag type because it would have been too much mundane work.
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In the previous change, I replaced the callsites that used
`av_init_packet`, but there are a handful of places that use stack
allocated packets with no initialisation.
In one case, I just switched to heap allocation as it's only done once
per stream at most.
In the other case, I removed our usage of the AVPackets as a
convenience mechanism to transfer data into a heap allocated packet.
Instead, I inlined the data copying.
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Now that 0.35 has been released, we can consider increasing our minimum
required ffmpeg version. Currently, we think 4.4 is the most recent
version we can move to (from the current requirement of 4.0).
This allows us to remove a few conditionals. There are more that we
won't be able to remove unless we move further up to 5.1.
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Detect avif files with 1 frame as images. This works because AV1 videos
and AVIF animations have nb_frames 0 or > 1.
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This define was always just a stopgap for that two month period (August
2021 - October 2021) where the bytes_read field in ffmpeg was completely
missing. Before that time, it was a private member in a struct (which
mpv used). Afterwards, it officially became public. Fortunately, the
lack of this field never actually made it into a release, so it could
have only possibly affected people building from the master branch.
Since ffmpeg 5.0 came out recently, and it's been plenty of months since
that two month window, we can go ahead and drop this check. This
finishes up the work done in 78cfeee2b93830f2988508a653b508336147b79d.
Sidenote: the cached ffmpeg version in the mingw ci were from that time
period when the bytes_read field was missing. The N in the workflow is
bumped to force a full rebuild and fresh clone of ffmpeg.
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This needs to be forwarded from the AVStream to the AVPacket itself, so
that it reaches the decoder. There exists an FFmpeg function for this,
we just need to call it. (Also add some logging)
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The bytes_read struct member in AVIOContext is now officially public,
so its usage no longer has to be specified as non-compliance with
FFmpeg's ABI/API rules.
That said, unfortunately there was a short period of time between
August 2021 and October 2021 where the struct member did not exist
in FFmpeg's git master, so keep a feature check for it alive for
now to enable building with those versions. Thankfully, no release
version of FFmpeg will be without this field, so it should be
possible to drop this check with time.
Finally, simplify the function in case the struct member is not
found. After all, there is zero reason to iterate through the AVIO
contexts if we cannot get the information we require.
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This exposes whether a video track is detected as an image, which is
useful for profile conditions, property expansion and lavfi-complex.
The lavf demuxer sets image to true when the existing check detects an
image.
When the lavf demuxer fails, the mf one guesses if the file is an image
by its extension, so sh->image is set to true when the mf demuxer
succeds and there's only one file.
The mkv demuxer just sets image to true for any attached picture.
The timeline demuxer just copies the value of image from source to
destination. This sets image to true for attached pictures, standalone
images and images added with !new_stream in EDL playlists, but it is
imperfect since you could concatenate multiple images in an EDL playlist
(which should be done with the mf demuxer anyway). This is good enough
anyway since the comment of the modified function already says it is
"Imperfect and arbitrary".
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This moves the image check to where the number of frames is available of
comparison, which allows not detecting jpg and png videos as images, and
detecting 1-frame gifs as images. This works with the mjpeg and png
videos in the FATE suite, though unfortunately the bmp video is still
detected as an image since it has nb_frames = 0.
aliaspix streams are also now considered images.
Attached pictures are now treated like standalone images, so audio with
attached pictures now has mf-fps as container-fps instead of
unavailable, which makes it consistent with external cover art, which
was already being assigned mf-fps.
Unfortunately images in a codec commonly used for videos are never
detected, and detection was inaccurate even using the now private
codec_info_nb_frames field in AVStream, and mediainfo gets them wrong
too, so I guess it's just a lost cause.
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Unfortunately, this functionality in large part based on a struct
member that was made private in FFmpeg/FFmpeg@7489f632815c98ad58c3db71d1a5239b5dae266c
in May. Unfortunately, this was not noticed during review.
This reverts commit 0862664ac952d21fef531a8923a58ae575268fc5.
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This exposes whether a video track is detected as an image. This is
useful for profile conditions, property expansion and lavfi-complex, and
is more accurate than any detection even Lua scripts can perform, since
they can't differentiate between images and videos without container-fps
and audio and with duration 1 (which is the duration set by the mf
demuxer with the default --mf-fps=1).
The lavf demuxer image check is moved to where the number of frames is
available for comparison, and is modified to check the number of frames
and duration instead of the video codec. This doesn't misdetect videos
in a codec commonly used for images (e.g. mjpeg) as images, and can
detect images in a codec commonly used for videos (e.g. 1-frame gifs).
pix files are also now detected as images, while before they weren't
since the condition was checking if the AVInputFormat name ends with
_pipe, and alias_pix doesn't.
Both nb_frames and codec_info_nb_frames are checked because nb_frames is
0 for some video codecs (hevc, av1, vc1, mpeg1video, vp9 if forcing
--demuxer=lavf), and codec_info_nb_frames is 1 for others (mpeg, mpeg4,
wmv3).
The duration is checked as well because for some uncommon codecs and
containers found in FFMpeg's FATE suite, libavformat returns nb_frames =
0 and codec_info_nb_frames = 1. For some of them it even returns
duration = 0, so they are blacklisted in order to never be considered
images.
The extra codecs that would have to be blacklisted without checking the
duration are AV_CODEC_ID_4XM, AV_CODEC_ID_BINKVIDEO,
AV_CODEC_ID_DSICINVIDEO, AV_CODEC_ID_ESCAPE130, AV_CODEC_ID_MMVIDEO,
AV_CODEC_ID_NUV, AV_CODEC_ID_RL2, AV_CODEC_ID_SMACKVIDEO and
AV_CODEC_ID_XAN_WC3, while the containers are film-cpk, ivf and ogg.
The lower limit for duration is 10 because that's the duration of
1-frame gifs.
Streams with codec_info_nb_frames 0 are not considered images because
vp9 and av1 have nb_frames = 0 and codec_info_nb_frames = 0, and we
can't rely on just the duration to detect them because they could be
livestreams without an initial duration, and actually even if we could
for these codecs libavformat returns huge negative durations like
-9223372036854775808.
Some more images in the FATE suite that are really frames cut from a
video in an uncommon codec and container, like cine/bayer_gbrg8.cine,
could be detected by allowing codec_info_nb_frames = 0, but then any
present and future video codec with nb_frames = 0 and
codec_info_nb_frames = 0 would need to be added to the blacklist. Some
even have duration > 10, so to detect these images the duration check
would have to be removed, and all the previously mentioned extra codecs
and containers would have to be added added to the blacklists, which
means that images that use them (if they exist anywhere) will never be
detected. These FATE images aren't detected as such by mediainfo either
anyway, nor can a Lua script reliably detect them as images since they
have container-fps and duration > 0 and != 1, and you probably will
never see files like them anywhere else.
For attached pictures the lavf demuxer always set image to true, which
is necessary because they have duration > 10. There is a minor change in
behavior for which audio with attached pictures now has mf-fps as
container-fps instead of unavailable, but this makes it consistent with
external cover art, which was already being assigned mf-fps.
When the lavf demuxer fails, the mf one guesses if the file is an image
by its extension, so sh->image is set to true when the mf demuxer
succeds and there's only one file.
Even if you add a video's file type to --mf-type and open it with the mf
protocol, only the first frame is used, so setting image to true is
still accurate.
When converting an image to the extensions listed in demux/demux_mf.c,
tga and pam files are currently the only ones detected by the mf demuxer
rather than lavf. Actually they are detected with the image2 format, but
it is blacklisted; see d0fee0ac33a.
The mkv demuxer just sets image to true for any attached picture.
The timeline demuxer just copies the value of image from source to
destination. This sets image to true for attached pictures, standalone
images and images added with !new_stream in EDL playlists, but it is
imperfect since you could concatenate multiple images in an EDL playlist
(which should be done with the mf demuxer anyway). This is good enough
anyway since the comment of the modified function already says it is
"Imperfect and arbitrary".
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FFmpeg recently changed these to be const on their side.
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This was causing undefined behavior when playing streams without RG tags
but with RG enabled. Broken in 585f9ff42f3195c.
Thanks to uau for bisecting.
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This reverts commit 41243e7c4f98b410195397b6758f9796acd9de57.
This fixes image format detection. FFmpeg has an utter called "image2",
which is designed to read patterns in filenames (so you can play
something like "%*.jpg" for all jpg files in the current directory).
"image2" is not what we want; it's just broken with custom I/O like
mpv uses it, and we don't want to "accidentally" interpret filenames
as pattern. That's why mpv blacklists it.
Unfortunately, "image2" is sometimes the format that FFmpeg's probe API
returns as best match. Thus demux_lavf fails to detect the file type,
and after some more futile attempts, we end up at demux_mf, which uses
detection by file extension. (Not sure why. I guess MPlayer did that,
and foudn that sufficient.) If the file extension is wrong (which
happens a lot because apparently the world is full of idiots who don't
manage to get the most simple things right), the image "loads", but
decoding obviously fails.
There's no easy way around this. The FFmpeg API has no mechanism to
exclude a specific format from probing (like image2, which breaks stuff
for us). Out of the 5 probe functions the API provides, none can probe
a specific format or include or exclude specific formats. The main
problem is that AVInputFormat.read_probe is a private symbol.
FFmpeg itself has no problem opening such files. It turns out that it
works, because even though image2 by itself uses detection by file
extension, it uses private API to further probe the exact format. It
explicitly excludes itself to prevent recursion.
But fortunately, that also means that it's impossible to get the image2
format if no filename is passed to the prober. (No filename, no file
extension.) Apparently we pass it in because it helps in corner cases.
Until almost 3 years ago, we passed the filename only when normal
probing already failed. Restore this by this revert. It makes
incorrectly named files work. The revert also makes the (apparently
forgotten) comment above the touched line of code true again.
Yes, quite possible that this breaks some mp3s again. You can't win
with FFmpeg. Thanks FFmpeg for making us fail at opening simple image
files and/or the most widely used file format for audio.
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FFmpeg, being the pile of trash as usual, recently broke this. Add our
own trash hack to trashily workaround this shit.
Fixes: #7893
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Change all OPT_* macros such that they don't define the entire m_option
initializer, and instead expand only to a part of it, which sets certain
fields. This requires changing almost every option declaration, because
they all use these macros. A declaration now always starts with
{"name", ...
followed by designated initializers only (possibly wrapped in macros).
The OPT_* macros now initialize the .offset and .type fields only,
sometimes also .priv and others.
I think this change makes the option macros less tricky. The old code
had to stuff everything into macro arguments (and attempted to allow
setting arbitrary fields by letting the user pass designated
initializers in the vararg parts). Some of this was made messy due to
C99 and C11 not allowing 0-sized varargs with ',' removal. It's also
possible that this change is pointless, other than cosmetic preferences.
Not too happy about some things. For example, the OPT_CHOICE()
indentation I applied looks a bit ugly.
Much of this change was done with regex search&replace, but some places
required manual editing. In particular, code in "obscure" areas (which I
didn't include in compilation) might be broken now.
In wayland_common.c the author of some option declarations confused the
flags parameter with the default value (though the default value was
also properly set below). I fixed this with this change.
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It seems sporadic errors are possible, such as connection timeouts.
Before the recent demuxer change, the demuxer thread retried many times
even on EOF, so an error was only interpreted as EOF once the decoder
queues ran out.
Change it to use EOF only. Since this may actually lead to the demuxer
thread being "stuck" and retrying forever (depending on libavformat API
behavior), I'm also adding a heuristic to prevent this, using a random
retry counter. This should not be necessary, but libavformat cannot be
trusted. (This retrying forever could be stopped by the user, but
obviously it would still heat the CPU for a longer time if the user is
not present.)
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RTSP supports seeking, but at least the libavformat implementation makes
this dependent on runtime behavior. So you have to perform a seek, and
check if it fails. But even if you do this, the stream is interrupted
and restarted, and there seem to be other issues.
Assume that RTSP with unknown duration means it's a live stream, and
disable seeking in this case, as suggested by the issue reporter.
Fixes: #7472
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Aka hls-bitrate. In turn, remove the demux_lavf.c hack, which made the
track title use this.
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Libav seems rather dead: no release for 2 years, no new git commits in
master for almost a year (with one exception ~6 months ago). From what I
can tell, some developers resigned themselves to the horrifying idea to
post patches to ffmpeg-devel instead, while the rest of the developers
went on to greener pastures.
Libav was a better project than FFmpeg. Unfortunately, FFmpeg won,
because it managed to keep the name and website. Libav was pushed more
and more into obscurity: while there was initially a big push for Libav,
FFmpeg just remained "in place" and visible for most people. FFmpeg was
slowly draining all manpower and energy from Libav. A big part of this
was that FFmpeg stole code from Libav (regular merges of the entire
Libav git tree), making it some sort of Frankenstein mirror of Libav,
think decaying zombie with additional legs ("features") nailed to it.
"Stealing" surely is the wrong word; I'm just aping the language that
some of the FFmpeg members used to use. All that is in the past now, I'm
probably the only person left who is annoyed by this, and with this
commit I'm putting this decade long problem finally to an end. I just
thought I'd express my annoyance about this fucking shitshow one last
time.
The most intrusive change in this commit is the resample filter, which
originally used libavresample. Since the FFmpeg developer refused to
enable libavresample by default for drama reasons, and the API was
slightly different, so the filter used some big preprocessor mess to
make it compatible to libswresample. All that falls away now. The
simplification to the build system is also significant.
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Demuxers can call demux_close_stream() to close the underlying stream if
it's not needed anymore. (Useful to release "heavy" resources like FDs
and sockets. Plus merely keeping a file open can have visible side
effects such as inability to unmount a filesystem or, on Windows, to do
anything with the file.)
Until now, this set demuxer->stream to a dummy stream, because most code
used to assume that the stream field is non-NULL. But this requirement
disappeared (in some cases, the stream field is already NULL), so stop
doing that. demux_lavf.c, one of the demuxers which calls this function,
still had some of this, though.
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Fucking gross that you need this in almost-2020.
Fixes: #7255
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See previous commit. libavformat exports this information as AVStream.id
field.
The big problem is that the libavformat field is simply 0 if it's
unknown (i.e. the demuxer never sets it). So it needs to remain a
whitelist. Just add more formats which are known to have a meaningful
ID.
I considered exporting IDs for all formats, and then either leaving the
values as they are, or filtering duplicate values (and choosing
arbitrary but unique different IDs). But then again, I think it's sort
of mpv's job to filter FFmpeg's absurd bullshit API, and it should make
an effort to hide it rather than to reflect it.
See: #7211
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Signed-off-by: Aman Gupta <aman@tmm1.net>
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It sometimes happens that HLS streams freeze because the HTTP server is
not responding for a fragment (or something similar, the exact
circumstances are unknown). The --timeout option didn't affect this,
because it's never set on HLS recursive connections (these download the
fragments, while the main connection likely nothing and just wastes a
TCP socket).
Apply an elaborate hack on top of an existing elaborate hack to somehow
get these options set. Of course this could still break easily, but hey,
it's ffmpeg, it can't not try to fuck you over. I'm so fucking sick of
ffmpeg's API bullshit, especially wrt. HLS.
Of course the change is sort of pointless. For HLS, GET requests should
just aggressively retried (because they're not "streamed", they're just
actual files on a CDN), while normal HTTP connections should probably
not be made this fragile (they could be streamed, i.e. they are backed
by some sort of real time encoder, and block if there is no data yet).
The 1 minute default timeout is too high to save playback if this
happens with HLS.
Vaguely related to #5793.
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Until now, we've made FFmpeg use the default network timeout - which is
apparently infinite. I don't know if this was changed at some point,
although it seems likely, as I was sure there was a more useful default.
For most use cases, a smaller timeout is more useful (for example
recording something in the background), so force a timeout of 1 minute.
See: #5793
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In some corner cases (see #6802), it can be beneficial to use a larger
stream buffer size. Use this as argument to rewrite everything for no
reason.
Turn stream.c itself into a ring buffer, with configurable size. The
latter would have been easily achievable with minimal changes, and the
ring buffer is the hard part. There is no reason to have a ring buffer
at all, except possibly if ffmpeg don't fix their awful mp4 demuxer, and
some subtle issues with demux_mkv.c wanting to seek back by small
offsets (the latter was handled with small stream_peek() calls, which
are unneeded now).
In addition, this turns small forward seeks into reads (where data is
simply skipped). Before this commit, only stream_skip() did this (which
also mean that stream_skip() simply calls stream_seek() now).
Replace all stream_peek() calls with something else (usually
stream_read_peek()). The function was a problem, because it returned a
pointer to the internal buffer, which is now a ring buffer with
wrapping. The new function just copies the data into a buffer, and in
some cases requires callers to dynamically allocate memory. (The most
common case, demux_lavf.c, required a separate buffer allocation anyway
due to FFmpeg "idiosyncrasies".) This is the bulk of the demuxer_*
changes.
I'm not happy with this. There still isn't a good reason why there
should be a ring buffer, that is complex, and most of the time just
wastes half of the available memory. Maybe another rewrite soon.
It also co |
