It is wrong/incorrect in two aspects:
1. The pixel format is not enough to guarantee that the resulting file
will be any more compatible with media players.
2. Media players not supporting higher profiles are not necessarily
outdated (in fact this is simply an arrogant statement that
libavcodec can handle these particular features).
You could add that there are plenty of other ways to produce widely
incompatible files with ffmpeg, and these don't show any warnings.
What we really want to do here is defaulting to codec profiles that
have wide compatibility, such as main/high for h264. Also, if an
encoder does not accept certain pixfmts, we should automatically
convert them to a pixfmt the encoder can accept. But the existing
message certainly is not appropriate.
It also works for 2 specific encoders only. Extending it for other
cases would result in a lot of special cases, so this is not the
right place.
With the new decode API, you can't handle errors directly in the API
user - you only know that the hwaccel did not initialize at all.
Add some approximate logging.
This adds tons of code for no other benefit than making VideoToolbox
support conform with the new hwaccel API (using hw_device_ctx and
hw_frames_ctx).
Since VideoToolbox decoding does not actually require the user to
allocate frames, the new code does mostly nothing.
One benefit is that ffmpeg_videotoolbox.c can be dropped once generic
hwaccel support for ffmpeg.c is merged from Libav.
Does not consider VDA or VideoToolbox encoding.
Fun fact: the frame transfer functions are copied from vaapi, as the
mapping makes copying generic boilerplate. Mapping itself is not
exported by the VT code, because I don't know how to test.
This is a newer API that is intended for decoders like the cuvid
wrapper. Until now, the wrapper required to set an awkward
"incomplete" hw_frames_ctx to set the device. Now the device
can be set directly, and the user can get AV_PIX_FMT_CUDA output
for a specific device simply by setting hw_device_ctx.
This still does a dummy ff_get_format() call at init time, and should
be fully backward compatible.
Consider the following sequence of events:
- open a codec without AV_CODEC_CAP_DELAY
- decode call fails with an error
- ff_thread_flush() is called
- drain packet is sent
Then the last step would make ff_thread_decode_frame() return an error,
because p->result can still be set to an error value. This is because
submit_packet returns immediately if AV_CODEC_CAP_DELAY is not set, and
no worker thread gets the chance to reset p->result, yet its value is
trusted by ff_thread_decode_frame().
Fix this by clearing the error fields on flush.
Get rid of the "ret" variable, and always use err. Report the packet as
consumed if err is unset. This should be equivalent to the old code,
which obviously required err=0 for p->result>=0 (and otherwise,
p->result must have had the value err was last set to). The code block
added by commit 32a5b63126 is also not needed anymore, because the new
code strictly returns err if it's >=0.
Reviewed-by: "Ronald S. Bultje" <rsbultje@gmail.com>
Could lead to random behavior. This possibly happened due to commit
32a5b63126. This should/could probably be simplified, but for no apply
a minimal fix to quell the errors.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
async_mutex has is used in a very strange but intentional way: it is
locked by default, and unlocked only in regions that can be run
concurrently.
If the user was calling API functions to the same context from different
threads (in a safe way), this could unintentionally unlock the mutex on
a different thread than the previous lock operation. It's not allowed by
the pthread API.
Fix this by emulating a binary semaphore using a mutex and condition
variable. (Posix semaphores are not available on all platforms.)
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
The old "API" that signaled rotation as a metadata value has been
replaced by DISPLAYMATRIX side data quite a while ago.
There is no reason to make muxers/demuxers/API users support both. In
addition, the metadata API is dangerous, as user tags could "leak" into
it, creating unintended features or bugs.
ffmpeg CLI has to be updated to use the new API. In particular, we must
not allow to leak the "rotate" tag into the muxer. Some muxers will
catch this properly (like mov), but others (like mkv) can add it as
generic tag. Note applications, which use libavformat and assume the
old rotate API, will interpret such "rotate" user tags as rotate
metadata (which it is not), and incorrectly rotate the video.
The ffmpeg/ffplay tools drop the use of the old API for muxing and
demuxing, as all muxers/demuxers support the new API. This will mean
that the tools will not mistakenly interpret per-track "rotate" user
tags as rotate metadata. It will _not_ be treated as regression.
Unfortunately, hacks have been added, that allow the user to override
rotation by setting metadata explicitly, e.g. via
-metadata:s:v:0 rotate=0
See references to trac #4560. fate-filter-meta-4560-rotate0 tests this.
It's easier to adjust the hack for supporting it than arguing for its
removal, so ffmpeg CLI now explicitly catches this case, and essentially
replaces the "rotate" value with a display matrix side data. (It would
be easier for both user and implementation to create an explicit option
for rotation.)
When the code under FF_API_OLD_ROTATE_API is disabled, one FATE
reference file has to be updated (because "rotate" is not exported
anymore).
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
This supports retrieving the device from a provided hw_frames_ctx, and
automatically creating a hw_frames_ctx if hw_device_ctx is set.
The old API is not deprecated yet. The user can still use
av_vdpau_bind_context() (with or without setting hw_frames_ctx), or use
the API before that by allocating and setting hwaccel_context manually.
Cherry-picked from Libav commit 1a7ddba5.
(Adds missing APIchanges entry to the Libav version.)
Reviewed-by: Mark Thompson <sw@jkqxz.net>
This "reuses" the flags introduced for the av_vdpau_bind_context() API
function, and makes them available to all hwaccels. This does not affect
the current vdpau API, as av_vdpau_bind_context() should obviously
override the AVCodecContext.hwaccel_flags flags for the sake of
compatibility.
Cherry-picked from Libav commit 16a163b5.
Reviewed-by: Mark Thompson <sw@jkqxz.net>
libavcodec now automatically serializes decoding for hwaccels which
are not thread-safe. This means API users, which rely on the libavcodec
native software fallback mechanism, can now simply enable threading
without running into problems.
This patch deprecates anything that has to do with merging/splitting
side data. Automatic side data merging (and splitting), as well as all
API symbols involved in it, are removed completely.
Two FF_API_ defines are dedicated to deprecating API symbols related to
this: FF_API_MERGE_SD_API removes av_packet_split/merge_side_data in
libavcodec, and FF_API_LAVF_KEEPSIDE_FLAG deprecates
AVFMT_FLAG_KEEP_SIDE_DATA in libavformat.
Since it was claimed that changing the default from merging side data to
not doing it is an ABI change, there are two additional FF_API_ defines,
which stop using the side data merging/splitting by default (and remove
any code in avformat/avcodec doing this): FF_API_MERGE_SD in libavcodec,
and FF_API_LAVF_MERGE_SD in libavformat.
It is very much intended that FF_API_MERGE_SD and FF_API_LAVF_MERGE_SD
are quickly defined to 0 in the next ABI bump, while the API symbols are
retained for a longer time for the sake of compatibility.
AVFMT_FLAG_KEEP_SIDE_DATA will (very much intentionally) do nothing for
most of the time it will still be defined. Keep in mind that no code
exists that actually tries to unset this flag for any reason, nor does
such code need to exist. Code setting this flag explicitly will work as
before. Thus it's ok for AVFMT_FLAG_KEEP_SIDE_DATA to do nothing once
side data merging has been removed from libavformat.
In order to avoid that anyone in the future does this incorrectly, here
is a small guide how to update the internal code on bumps:
- next ABI bump (probably soon):
- define FF_API_LAVF_MERGE_SD to 0, and remove all code covered by it
- define FF_API_MERGE_SD to 0, and remove all code covered by it
- next API bump (typically two years in the future or so):
- define FF_API_LAVF_KEEPSIDE_FLAG to 0, and remove all code covered
by it
- define FF_API_MERGE_SD_API to 0, and remove all code covered by it
This forces anyone who actually wants packet side data to temporarily
use deprecated API to get it all. If you ask me, this is batshit fucked
up crazy, but it's how we roll. Making AVFMT_FLAG_KEEP_SIDE_DATA to be
set by default was rejected as an ABI change, so I'm going all the way
to get rid of this once and for all.
Reviewed-by: James Almer <jamrial@gmail.com>
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Preparation for potentially disabling merged side data by default in the
libs. Do this in particular because it affects fate tests.
The changed tests either reflect added packet side data, or the changed
packet size due to merged side data removal reducing the packet size.
Make it clear that there is no timing-dependent behavior. In particular,
there is no state in which both input and output are denied, and where
you have to wait for a while yourself to make progress (apparently some
hardware decoders like to do this).
Avoid wording that makes references to time. It shouldn't be mistaken
for some kind of asynchronous API (like POSIX read() can return EAGAIN
if there is no new input yet). It's a state machine, so try to use
appropriate terms.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Merges Libav commit 8a60bba0ae.
Apparently the demuxer outputs the wrong padding for HE-AAC (based on
the raw sample rate, or so). aacdec contains a hack to adjust the muxer
padding accordingly before it's used to trim the decoder output. This
modified the packet side data, which in combination with the old
decoding API would change the packet the user passed to the decoder.
This is clearly not allowed, and it breaks running some gapless fate
tests with "-fflags +keepside" applied (without keepside, the packet
metadata is typically newly allocated, essentially making a copy and not
modifying the user's input packet).
This should probably be fixed in the demuxer (and consequently also the
muxer), but for now only fix the immediate problem.
Regression since 946ed78f5f (2012).
Ever since the codecpar changes, this has been always printed when
opening a flv file. This is because the codecpar changes made all
streams to be added lazily as read_packet is called.
There is no reason that draining couldn't return an error or two. But
some decoders don't handle this very well, and might always return an
error. This can lead to API users getting into an infinite loop and
burning CPU, because no progress is made and EOF is never returned.
In fact, ffmpeg.c contains a hack against such a case. It is made
unnecessary with this commit, and removed with the next one. (This
particular error case seems to have been fixed since the hack was
added, though.)
This might lose frames if decoding returns errors during draining.
The filter field is often used to check whether a filter is
configured. If configuring the filter actually fails somewhere in
the middle of it, these fields could still be set to non-NULL, which
lead to other code accessing the half-configured filter graph, which
in turn could lead to crashes within libavfilter.
Solve this by properly resetting all fields.
This was triggered by a fuzzed sample after the recent changes. It's
unknown whether this behavior could be triggered before that.
If a subtitle packet came before the first video frame could be fully
decoded, the subtitle packet would get discarded. This puts the subtitle
into a queue instead, and processes it once the attached filter graph is
initialized.
Some callers (like do_subtitle_out()) call this with an AVPacket that is
not refcounted. This can cause undefined behavior.
Calling av_packet_move_ref() does not make a packet refcounted if it
isn't yet. (And it can't be made to, because it always succeeds,
and can't return ENOMEM.)
Call av_packet_ref() instead to make sure it's refcounted.
I couldn't find a case that is fixed by this with the current code. But
it will fix the fate-pva-demux test with the later patches applied.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
If AVVideotoolboxContext.cv_pix_fmt_type is set to 0, don't set the
kCVPixelBufferPixelFormatTypeKey value on the VT decoder.
This makes VT output its native format, which can be much faster on
some hardware iterations (if the native format does not match with
the requested format, it will be converted, which is slow).
The default is still forcing nv12.
Public fields were added after the private fields (negating the entire
point of this). New private fields go into AVStreamInternal anyway.
The new marker was set by guessing which fields are supposed to be
private and wshich not. recommended_encoder_configuration is accessed by
ffserver_config.c directly, and is supposed to use the public API.
ffmpeg.c accesses AVStream.cur_dts, even though it's a private field,
but that seems to be an older error.
Allow all struct fields to be accessed directly, as long as they're
public.
Before this change, many fields were "public", but could be accessed via
AVOption only. This meant they were effectively not public, but were
present for documentation purposes, which was incredibly confusing at
best.
D3D9Ex uses different driver paths. This helps with "headless"
configurations when no user logs in. Plain D3D9 device creation will
fail if no user is logged in, while it works with D3D9Ex.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Merges Libav commit c2f97f0508.
This is an extended version of the AVFrame.opaque field, which can be
used to attach arbitrary user information to an AVFrame.
The usefulness of the opaque field is rather limited, because it can
store only up to 32 bits of information (or 64 bit on 64 bit systems).
It's not possible to set this field to a memory allocation, because
there is no way to deallocate it correctly.
The opaque_ref field circumvents this by letting the user set an
AVBuffer, which makes the user data refcounted.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Merges Libav commit 04f3bd3496.
This is a bit messy, mainly due to timestamp handling.
decode_video() relied on the fact that it could set dts on a flush/drain
packet. This is not possible with the old API, and won't be. (I think
doing this was very questionable with the old API. Flush packets should
not contain any information; they just cause a FIFO to be emptied.) This
is replaced with checking the best_effort_timestamp for AV_NOPTS_VALUE,
and using the suggested DTS in the drain case.
The modified tests (fate-cavs and others) still fails due to dropping
the last frame. This happens because the timestamp of the last frame
goes backwards (ffprobe -show_frames shows the same thing). I suspect
that this "worked" due to the best effort timestamp logic picking the
DTS over the decreasing PTS. Since this logic is in libavcodec (where
it probably shouldn't be), this can't be easily fixed. The timestamps
of the cavs samples are weird anyway, so I chose not to fix it.
Another strange thing is the timestamp handling in the video path of
process_input_packet (after the decode_video() call). It looks like
the code to increase next_dts and next_pts should be run every time
a frame is decoded - but it's needed even if output is skipped.
With the new decode API, doing this in ffmpeg.c is impractical. There
was resistance against removing the warning, so put it into libavcodec.
Not bothering with reducing the warning to verbose log level for
subsequent wanrings. The warning should be rare, and only happen when
developing new codecs for the old API.
Includes a change suggested by Michael Niedermayer.
This uses a new MMAL feature, which limits the number of extra frames
that can be buffered within the decoder. VIDEO_MAX_NUM_CALLBACKS can
be defined as positive or negative number. Positive numbers are
absolute, and can lead to deadlocks if the user underestimates the
number of required buffers. Negative numbers specify the number of extra
buffers, e.g. -1 means no extra buffer, (-1-N) means N extra buffers.
Set a gratuitous default of -11 (N=10). This is much lower than the
firmware default, which appears to be 96.
This is backwards compatible, but needs a symbol only present in newer
firmware headers. (It's an enum item, so it requires a check in
configure.)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Slight simplification. The result is the same. Also, change the
wording of the message as requested in patch review.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Fixes apparent mmal_port_disable() freezes in ffmmal_stop_decoder() when
calling ffmmal_decode() with flush semantics a large number of times in
a row.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The assert in ffmmal_stop_decoder() could trigger sometimes. The
packets_buffered counter was indeed not correctly maintained, and
packets were not subtracted from it if they were still in the waiting
queue.
For some reason, this happened especially with VC-1.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The deprecated avcodec_decode_video2() and avcodec_decode_audio4()
functions called av_packet_split_side_data() on the input packets. This
is required for packets produced by libavformat with the
AVFMT_FLAG_KEEP_SIDE_DATA flag unset (which is unfortunately the
default).
The new API didn't do this yet, although it didn't matter as no decoder
supports the new API yet. The emulation layer for the old API calls the
old API functions, which took care of the splitting. Add this code to
the new API codec entrypoints too, because we shouldn't send essentially
corrupted data to decoders.
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
From Libav commit 05f66706d1.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This uses a new MMAL feature, which limits the number of extra frames
that can be buffered within the decoder. VIDEO_MAX_NUM_CALLBACKS can
be defined as positive or negative number. Positive numbers are
absolute, and can lead to deadlocks if the user underestimates the
number of required buffers. Negative numbers specify the number of extra
buffers, e.g. -1 means no extra buffer, (-1-N) means N extra buffers.
Set a gratuitous default of -11 (N=10). This is much lower than the
firmware default, which appears to be 96.
This is backwards compatible, but needs a symbol only present in newer
firmware headers. (It's an enum item, so it requires a check in
configure.)
Fixes a specific srt sample, which has an event with negative duration.
libavcodec will convert an event with negative duration to an ASS event
which will be displayed forever, which is not wanted here.
Treat negative duration always as unknown duration instead, and show it
until the next subtitle event.
If videotoolbox_common_end_frame failed, then the AVFrame was returned
to the API user with the dummy buffer (in AVFrame.buf[0]) still set, and
the decode call indicating success.
These "half-set" AVFrames with dummy buffer are a videotoolbox specific
hack, because the decoder requires an allocated AVFrame for its internal
logic. Videotoolbox on the other hand allocates its frame itself
internally, and outputs it only on end_frame. At this point, the dummy
buffer is replaced with the real frame (unless decoding fails).
Fixes apparent mmal_port_disable() freezes in ffmmal_stop_decoder() when
calling ffmmal_decode() with flush semantics a large number of times in
a row.
The assert in ffmmal_stop_decoder() could trigger sometimes. The
packets_buffered counter was indeed not correctly maintained, and
packets were not subtracted from it if they were still in the waiting
queue.
For some reason, this happened especially with VC-1.
I've got some m4a samples that had jpeg cover art marked as png. Since
these files were supposedly written by iTunes, and other software can
read it (e.g. clementine does), this should be worked around.
Since png has a very simple to detect header, while it's apparently a
real pain to detect jpeg in the general case, try to detect png and
assume jpeg otherwise. Not bothering with bmp, as I have no test case.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Commit 2b3e9bbfb5 caused problems for a
certain API user:
https://code.google.com/p/chromium/issues/detail?id=537725https://code.google.com/p/chromium/issues/detail?id=542032
The problem seems rather arbitrary, because if there's junk, anything
can happen. In this case, the imperfect junk skipping just caused it to
read different junk, from what I can see.
We can improve the accuracy of junk detection by a lot by checking if 2
consecutive frames use the same configuration. While in theory it might
be completely fine for the 1st frame to have a different format than the
2nd frame, it's exceedingly unlikely, and I can't think of a legitimate
use-case.
This is approximately the same mpg123 does for junk skipping. The
set of compared header bits is the same as the libavcodec mp3 parser
uses for similar purposes.
The AVCodecContext.get_format callback is not only used for pixel format
negotiation with the API user, but also for hwaccel init. For the
latter, it's required that some codec parameters, in particular the
codec profile, are set when the callback is invoked.
This patch removes a get_format invocation where this is not guaranteed.
The codec parameters, including the profile, are really set further
below. (The same code path that sets the profile also calls get_format
properly too.)
This just happened to work by coincidence in most cases. For example, if
the API user just copied or reused the AVStream's AVCodecContext when
decoding, the profile would be set properly. But in some cases it
fails., such as with the sample WolfensteinTwitch.mp4 on the samples
server.
Remove the redundant get_format call. Apparently it serves no purpose
anymore, although it is possible that this was different at the time it
was added in commit ffd77f94a2.
This fixes hwaccel usage for API users which do not set the profile
when setting up the AVCodecContext (which is allowed).
Work on the AVFrame references directly.
Instead of setting up a flipped/swapped "view" on the pictures,
flip/swap them when returning decoded frames to the API user.
In some situations, MMAL won't return a decoded frame for certain input
frames. This can happen if a frame fails to decode, or if a packet does
not actually contain a complete frame. In these situations, we would
deadlock (or actually timeout) waiting for an expected output frame,
which is not ideal. On the other hand, there are situations where we
definitely have to block to avoid deadlocks. (This mess is a
consequence of trying to map MMAL's asynchronous and flexible
dataflow to libavcodec, which is more static and rigid.)
Solve this by doing a blocking wait only if the amount of buffered data
is too big. The whole purpose of the blocking wait is to avoid excessive
buffering of input data, so we can skip it if it appears to be low. The
consequence is that libavcodec can gracefully return no frame to the
API user.
We want to track the number of full packets to make our heuristic work.
But MMAL buffers are fixed-size, requiring splitting large packets. This
is why the previous commit is needed. We use the ..._FRAME_END flag to
remember packet boundaries, but MMAL does not preserve these buffer
flags when returning buffers to the user.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The next commit needs 1 bit of additional information per MMAL buffer
sent to the MMAL input port. This information will be needed when the
buffer is recycled (i.e. returned by the input port's callback).
Normally, we could use MMAL_BUFFER_HEADER_FLAG_USER0, but that is
unexpectedly not preserved.
Do this by storing a pointer to FFBufferEntry in the MMAL buffer's
user data, instead of an AVBufferRef. This also changes the lifetime
of FFBufferEntry.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This affects Annex B streams (such as demuxed from .ts and others). It
also handles the format change in reinit-large_420_8-to-small_420_8.h264
correctly.
Instead of passing through the extradata, create it on the fly it from
the currently active SPS and PPS. Since reconstructing the PPS and SPS
NALs would be very complicated and verbose, we use the NALs as they
originally appeared in the bitstream.
The code for writing the extradata is somewhat derived from
libavformat/avc.c, but it's small and different enough that sharing it
is not really worth it.
We assume an upper bound of 4096 bytes for each raw SPS/PPS. It's hard
to determine an exact maximum size, but this value was was considered
high enough and safe.
Needed for the following VideotoolBox commit.
Commits 43bc5cf9 and c5371f77 add code for skipping initial zeros in mp3
packets. This code forgot to report to the user that data was skipped at
all.
Since audio codecs allow partial packet decoding, the user application
has to rely on the return value. It will remove the data reported as
consumed by the decoder, and feed it to the decoder again. This resulted
in the mp3 frame after the zero region to be decoded over and over
again, until the zero region was finally skipped by the application.
Fix this by including the amount of skipped bytes to the number of
consumed bytes returned by the decode call.
Fixes trac ticket #4890.
Note that convergence_duration had another meaning, one which was in
practice never used. The only real use for it was a 64 bit replacement
for the duration field. It's better just to make duration 64 bits, and
to get rid of it.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
It appears vdpau drivers can return constrained baseline as unsupported,
even if libvdpau knows about the symbol, and the main profile is
supported.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Note that this slightly changes behavior: it sets AVMEDIA_TYPE_UNKNOWN
if the codec type is unknown. This should be ok.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
If cmd_pos is broken, this would just keep accumulating packets in the
reassembly buffer, until it fails and flushes the buffer on overflow.
Since packets are usually rather small, this will take a lot of subtitle
packets. The perceived effect is that subtitles are not displayed
anymore after the faulty packet was passed to the decoder.
I'm not terribly sure about this, but on the other hand this code is
active only when fragmented packets need to be reassembled.
Fixes sample file in trac issue #4872.
Assuming the first and second packets are partial, this would append the
reassembly buffer (ctx->buf) to itself with the second
append_to_cached_buf() call, because buf is set to ctx->buf.
I do not know a valid sample file which triggers this, and do not know
if packets can be split into more than 2 sub-packets, but it triggered
with a (differently) broken sample file in trac issue #4872.
Broken by commit ba12ba859a. This only
happens with HLS streams which use encryption and require preserving
cookies sent by the server.
Fixes trac issue #4846.
Some .idx files actually contain duplicate subtitle events:
timestamp: 00:07:52:600, filepos: 00004e800
timestamp: 00:07:52:600, filepos: 00004f800
The second will be dropped, because it has same pts, duration, and text
(the text is just a dummy empty string; the real data is retrieved when
actually reading vobsub subtitle packets).
Dropping this is probably not intended/safe, so avoid it.
See trac issue #4872 for a sample. This patch doesn't fix decoding of
the sample, though.
The stream ID is essentially an arbitrary number defined by the .idx
file headers. They have to match the IDs in the .sub stream. The vobsub
demuxer assumed the IDs would just start from 0, increassing by 1 for
each stream. This is not correct. In the sample I had, the IDs were
starting from 1, leading to no subtitles being displayed at all.
Fix this by using the correct stream ID.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
wm4
