This is completely useless, and in this particular case, it broke the
fallback for MLP2 subtitles (stored as .txt files) to demux_subreader.
(Yes, libavformat should be fixed to handle this, but for now this will
_always_ break playback of subtitle files stored in .txt.)
You can still force this demuxer, but by default we will just pretend
that the "tty" demuxer does not exist.
Perhaps not very useful, but reserved for situations when a user reports
awful latency and experimentation/debugging might be required to find
out why or to fix it (happens often).
avio_alloc_context() is documented to require an av_malloc'ed buffer. It
appears libavformat can even reallocate the buffer while it is probing,
so passing a static buffer can in theory lead to crashes.
I couldn't reproduce such a crash, but apparently it happened to
mplayer-svn. This commit follows the mplayer fix in svn commit r36397.
Move the decoder parts from vo_vdpau.c to a new file vdpau_old.c. This
file is named so because because it's written against the "old"
libavcodec vdpau pseudo-decoder (e.g. "h264_vdpau").
Add support for the "new" libavcodec vdpau support. This was recently
added and replaces the "old" vdpau parts. (In fact, Libav is about to
deprecate and remove the "old" API without deprecation grace period,
so we have to support it now. Moreover, there will probably be no Libav
release which supports both, so the transition is even less smooth than
we could hope, and we have to support both the old and new API.)
Whether the old or new API is used is checked by a configure test: if
the new API is found, it is used, otherwise the old API is assumed.
Some details might be handled differently. Especially display preemption
is a bit problematic with the "new" libavcodec vdpau support: it wants
to keep a pointer to a specific vdpau API function (which can be driver
specific, because preemption might switch drivers). Also, surface IDs
are now directly stored in AVFrames (and mp_images), so they can't be
forced to VDP_INVALID_HANDLE on preemption. (This changes even with
older libavcodec versions, because mp_image always uses the newer
representation to make vo_vdpau.c simpler.)
Decoder initialization in the new code tries to deal with codec
profiles, while the old code always uses the highest profile per codec.
Surface allocation changes. Since the decoder won't call config() in
vo_vdpau.c on video size change anymore, we allow allocating surfaces
of arbitrary size instead of locking it to what the VO was configured.
The non-hwdec code also has slightly different allocation behavior now.
Enabling the old vdpau special decoders via e.g. --vd=lavc:h264_vdpau
doesn't work anymore (a warning suggesting the --hwdec option is
printed instead).
Remove the (now unused) code for determining correct-pts mode based on
the demuxer in use. Change its description in the manpage to reflect
what this option does now.
Gives really funky results with PNG attachments otherwise. The main
problem is that avcodec_flush_buffers() does not fully reset the
decoder, so passing multiple PNG packets without keyframe flags will
attempt to combine the new picture with the previously decoded
contents. (Makes no sense with proper PNG - maybe this codepath is
intended for MNG or APNG.)
In general, this warning can hint to actual bugs. We don't enable it
yet, because it would conflict with some unmerged code, and we should
check with clang too (this commit was done by testing with gcc).
This also affects --audiofile. The previous behavior wasn't really
useful. There are even separate switches for that: --audio-demuxer and
--sub-demuxer.
This fixes the sample RA_missing_timestamps.mkv. Pretty funny how this
code got it almost right, but not quite, so it was broken all these
years. And then, after everyone stopped caring, someone comes and fixes
it. (By the way, I know absolutely nothing about realaudio.)
This fixes playback of the sample linked by FFmpeg ticket 2508. The fix
follows ffmpeg commit 6158a3b (although it's not exactly the same).
The problem here is that the file contains an apparently non-sense
DefaultDuration value. DefaultDuration for audio tracks is used to
derive PTS values for packets with no timestamps, like they can happen
with frames inside a laced block. So the first packet of a SimpleBlock
will have a correct PTS, while the PTS values of the following packets
are calculated using DefaultDuration, and thus are broken.
This leads to seemingly ok playback, but broken A/V sync. Not using the
DefaultDuration value will leave the PTS values of these packets unset,
and the audio decoder can derive them from the output instead.
The fix more or less uses a heuristic to detect the broken case: if the
sample rate is 8 KHz (Matroska default, can assume unset), and the codec
is AC3 (as the broken file did), don't use it. I'm not sure why this
should be done only for AC3, maybe the muxing application (mkvmerge
v4.9.1) has known issues with AC3. AC3 also doesn't support 8 KHz as
sample rate natively.
(By the way, I'm not sure why we should honor the DefaultDuration at all
for audio. It doesn't seem to be needed. You can't seek to these frames,
and decoders should always be able to produce perfect PTS values by
adding the duration of the decoded audio to the first PTS.)
Matroska has an output sample rate (OutputSamplingFrequency), which in
theory should be forced instead of whatever the decoder outputs. But it
appears no software (other than mplayer2 and mpv until now) actually
respects this. Even worse, there were broken files around, which played
correctly with (in theory) broken software, but not mplayer2/mpv. Hacks
were added to our code to play these files correctly, but they didn't
catch all cases.
Simplify this by doing what everyone else does, and always use the
decoder's sample rate instead. In particular, we try to handle all
sample rate issues like libavformat's Matroska demuxer does.
Guess the colorspace directly in mpcodecs_reconfig_vo(), instead of in
set_video_colorspace(). The difference is that the latter function just
makes the video filter chain (and VOs) force the detected colorspace,
and then throws it away, while the former is a bit more general and
central. Not really a big difference and it doesn't matter much in
practice, but it guarantees that there is no internal disagreement about
the colorspace.
DVD playback had some trouble with PTS resets: libavformat's genpts
feature would try reading until EOF (worst case) to find a new usable
PTS in case a packet's PTS is not set correctly. Especially with slow
DVD access, this would make the player to appear frozen.
Reimplement it partially in demux_lavf.c, and use that code in the DVD
case. This is heavily "inspired" by the code in av_read_frame from
libavformat/utils.c. The difference is that we stop reading if no PTS
has been found after 50 packets (consider this a heuristic). Also, we
don't bother with the PTS wrapping and last-frame-before-EOF handling.
Even with normal PTS wraps, the player frontend will go to hell for the
duration of a frame anyway, and should recover quickly after that.
The terribleness of this commit is mostly that we duplicate libavformat
functionality, and that we suddenly need a packet queue.
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.
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.
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.
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.
Partial packet reads were needed because the video/audio parsers were
working on top of them. So it could happen that a parser read a part of
a packet, and returned that to the decoder. With libavformat/libavcodec,
packets are already parsed, and everything is much simpler.
Most of the simplifications in ad_spdif could have been done earlier.
Remove some other stuff as well, like the questionable slave mode start
time reporting (could be replaced by proper code, but we don't bother).
Remove the unused skip_audio_frame() functionality as well (it was used
by old demuxers). Some functions become private to demux.c, like
demux_fill_buffer(). Introduce new packet read functions, which have
simpler semantics. Packets returned from them are owned by the caller,
and all packets in the demux.c packet queue are considered unread.
Remove special code that dropped subtitle packets with size 0. This
used to be needed because it caused special cases in the old code.
The demux_open as well as demux_open_withparams calls don't use the
stream selection parameters anymore, so remove them everywhere.
Completes the previous commit.
These separate arrays were used by the old demuxers and are not needed
anymore. We can simplify track switching as well.
One interesting thing is that stream/tv.c (which is a demuxer) won't
respect --no-audio anymore. It will probably work as expected, but it
will still open an audio device etc. - this is because track selection
is now always done with the runtime track switching mechanism. Maybe
the TV code could be updated to do proper runtime switching, but I
can't test this stuff.
The audio parser was needed only by the "old" demuxers, and
demux_rawaudio. All other demuxers output already parsed packets.
demux_rawaudio is usually for raw audio, so using a parser with it
doesn't usually make sense. But you can also force it to read
compressed formats with fixed packet sizes, in which case the parser
would have been used. This use case is probably broken now, but you
will be able to do the same thing with libavformat demuxers.
Delete demux_avi, demux_asf, demux_mpg, demux_ts. libavformat does
better than them (except in rare corner cases), and the demuxers have
a bad influence on the rest of the code. Often they don't output
proper packets, and require additional audio and video parsing. Most
work only in --no-correct-pts mode.
Remove them to facilitate further cleanups.
STREAM_CTRL_GET_METADATA will be used to poll for streamcast metadata.
Also add DEMUXER_CTRL_UPDATE_INFO, which could in theory be used by
demux_lavf.c. (Unfortunately, libavformat is too crappy to read metadata
mid-stream for mp3 or ogg, so we don't implement it.)
The filter chain and the video ouputs have config() functions. They are
strictly limited to transfering the video size and format. Other
parameters (like color levels) have to be transferred separately.
Improve upon this by introducing a separate set of reconfig() functions,
which use mp_image_params to carry format parameters. This struct
contains all image format related parameters from config(), plus
additional parameters such as colorspace.
Change vf_rotate to use it, as well as vo_opengl. vf_rotate is just
an example/test case, but vo_opengl will need it later.
The intention is also to get rid of VOCTRL_SET_YUV_COLORSPACE. This
information is now handed to the VOs via reconfig(). The getter,
VOCTRL_GET_YUV_COLORSPACE, will still be needed though.
Old code used to use libass' recoding feature, which is a copy of the
old MPlayer code. We dropped that a few commits ago. Unfortunately,
this made it impossible to load some subtitle files, like UTF-16 files.
Make .ass loading respect -subcp again. We do this by recoding the
probe buffer to UTF-8, and then trying to load it normally. (Yep.)
Since UTF-16 in particular will effectively half the probe buffer size,
double the probe size.
demux_libass.c allows us to make subtitle format detection part of the
normal file loading process. libass has no probe function, but trying to
load the start of a file (the first 4 KB) is good enough. Hope that
libass can even handle random binary input gracefully without printing
stupid log messages, and that the libass parser doesn't accept too many
non-ASS files as input.
This doesn't handle the -subcp option correctly yet. This will be fixed
later.
The default correct-pts mode depended on which demuxer was opened last.
Often this is the subtitle demuxer. The correct-pts mode should be
decided on the demuxer for video instead.
subreader.c (before this commit renamed to demux_subreader.c) was
special cased to the -sub option. The plan is using the normal demuxer
codepath for all subtitle formats (so we can prefer libavformat demuxers
for most formats).
There are some subtle changes. The probe size is restricted to 32 KB
(instead of unlimitted + giving up after 100 lines of input). For
formats like MicroDVD, the video FPS isn't used anymore, because it's
not available on the subtitle demuxer level. Instead, hardcode it to
23.976 FPS (libavformat seems to do the same). The user can probably
still use -sub-fps to fix the timing. Checking the file extension for
".utf"/".utf8"/".utf-8" is simply removed (seems worthless, was in the
way, and I've never seen this anywhere).
Simpler, reduces the amount of copying.
We still have to malloc+memcpy the probe buffer though, because padding
with FF_INPUT_BUFFER_PADDING_SIZE is required by libavformat.
If a subtitle is external, read it completely and add all subtitle
events in advance when the subtitle track is selected. This is done
for text subtitles only. (Note that subreader.c and subtitles loaded
with libass are different and don't have anything to do with this
commit.)
Seems like a completely unnecessary complication. Instead, always add a
1 byte padding (could be extended if a caller needs it), and clear it.
Also add some documentation. There was some, but it was outdated and
incomplete.
The stream EOF flag should only be set when trying to read past the end
of the file (relatively similar to unix files). Always clear the EOF
flag on seeking. Trying to set it "properly" (depending whether data is
available at seek destination or not) might be an ok idea, but would
require attention to too many special cases. I suspect before this
commit (and in MPlayer etc. too), the EOF flag wasn't handled
consistently when the stream position was at the end of the file.
Fix one special case in ebml.c and stream_skip(): this function couldn't
distinguish between at-EOF and past-EOF either.
This function was called in various places. Most time, it was used
before a seek. In other cases, the purpose was apparently resetting
the EOF flag. As far as I can see, this makes no sense anymore. At
least the stream_reset() calls paired with stream_seek() are completely
pointless. A seek will either seek inside the buffer (and reset the
EOF flag), or do an actual seek and reset all state.
This branch heavily refactors the subtitle code (both loading and
rendering), and adds support for a few new formats through FFmpeg.
We don't remove any of the old code yet. There are still some subtleties
related to subreader.c to be resolved: code page detection & conversion,
timing post-processing, UTF-16 subtitle support, support for the -subfps
option. Also, SRT reading and loading ASS via libass should be turned
into proper demuxers. (SRT is needed because Libav's is gravely broken,
and we want ASS loading via libass to cover full libass format support.
Both should be demuxers which are probed _before_ libavformat, so that
all subtitles can be loaded through the demuxer infrastructure, and
libavformat subtitles don't need to be treated in a special way.)
Before this, subtitle packets were returned as data ptr/len pairs, and
mplayer.c got the rest (pts and duration) directly from the demuxer
data structures. Then mplayer.c reassembled the packet data structure
again.
Pass packets directly instead. The mplayer.c side stays a bit awkward,
because the (now by default unused) DVD path keeps getting in the way.
In demux.c there's lots of weird stuff (3 functions that read packets,
really?), but we want to keep the code equivalent for now to avoid
hitting weird issues and corner cases.
Makes WebVTT actually work.
Also simplify the logic for setting duration. Only the subtitle path
uses the packet duration, so the checks for STREAM_SUB as well as the
keyframe flag are redundant.
Apparently duration and convergence_duration are the same thing, but
convergence_duration was added as Matroska-specific hack to get a higher
value range (int vs. int64_t) with high resolution Matroska timebases.
For us it doesn't matter, because double floats are used for timestamps
and durations.
This means subassconvert.c is split in sd_srt.c and sd_microdvd.c. Now
this code is involved in the sub conversion chain like sd_movtext is.
The invocation of the converter in sd_ass.c is removed.
This requires some other changes to make the new sub converter code work
with loading external subtitles. Until now, subtitles loaded via
subreader.c was assumed to be in plaintext, or for some formats, in ASS
(except in -no-ass mode). Then these were added to an ASS_Track. Change
this so that subtitles are always in their original format (as far as
decoders/converters for them are available), and turn every sub event
read by subreader.c as packet to the dec_sub.c subtitle chain.
This removes differences between external/demuxed and -ass/-no-ass code
paths further.
Internally, stream_dvd.c returned DEMUXER_TYPE_MPEG_PS, and the same
value was hardcoded to enforced usage of demux_lavf in demux.c. But
"-demuxer mpegps" basically did the same, so that switch was broken
for this format. Undo this and don't request a demuxer in stream_dvd.c.
demux_lavf.c is (probably) good enough to probe correctly with DVD.
Otherwise, we'd actually have to do something completely different to
force the libavformat demuxer.
Make the sub decoder stuff independent from sh_sub (except for
initialization of course). Sub decoders now access a struct sd only,
instead of getting access to sh_sub. The glue code in dec_sub.c is
similarily independent from osd.
Some simplifications are made. For example, the switch_id stuff is
unneeded: the frontend code just has to make sure to call osd_changed()
any time subtitles are switched.
This is also preparation for introducing subtitle converters. It's much
cleaner to completely separate demuxer header/renderer glue/decoders
for this purpose, especially since sub converters might completely
change how demuxer headers have to be interpreted.
Also pass data as demux_packets. Currently, this doesn't help much, but
libavcodec converters might need scary stuff like packet side data, so
it's perhaps better to go with passing packets.
Subtitle files are opened in mplayer.c, not using the demuxer
infrastructure in general. Pretend that this is not the case (outside of
the loading code) by opening a pseudo demuxer that does nothing. One
advantage is that the initialization code is now the same, and there's
no confusion about what the difference between track->stream,
track->sh_sub and mpctx->sh_sub is supposed to be.
This is a bit stupid, and it would be much better if there were proper
subtitle demuxers (there are many in recent FFmpeg, but not Libav). So
for now this is just a transition to a more proper architecture. Look
at demux_sub like an artifical limb: it's ugly, but don't hate it - it
helps you to get on with your life.
This unifies the subtitle rendering path. Now all subtitle rendering
goes through sd_ass.c/sd_lavc.c/sd_spu.c.
Before that commit, the spudec.h functions were used directly in
mplayer.c, which introduced many special cases. Add sd_spu.c, which is
just a small wrapper connecting the new subtitle render API with the
dusty old vobsub decoder in spudec.c.
One detail that changes is that we always pass the palette as extra
data, instead of passing the libdvdread palette as pointer to spudec
directly. This is a bit roundabout, but actually makes the code simpler
and more elegant: the difference between DVD and non-DVD dvdsubs is
reduced.
Ideally, we would just delete spudec.c and use libavcodec's DVD sub
decoder. However, DVD playback with demux_mpg produces packets
incompatible to lavc. There are incompatibilities the other way around
as well: packets from libavformat's vobsub demuxer are incompatible to
spudec.c. So we define a new subtitle codec name for demux_mpg subs,
"dvd_subtitle_mpg", which only sd_spu can decode.
There is actually code in spudec.c to "assemble" fragments into complete
packets, but using the whole spudec.c is easier than trying to move this
code into demux_mpg to fix subtitle packets.
As additional complication, Libav 9.x can't decode DVD subs correctly,
so use sd_spu in that case as well.
The new wavpack packet format (see previous commit) doesn't work with
older libavcodec versions, so disable the new code in this case.
The version numbers are only approximate, since the libavcodec version
wasn't bumped with the wavpack change, but it's close enough.
Libav introduced a silent API breakage by changing what wavpack packets
the libavcodec decoder accepts. Originally the libavcodec codec accepted
Matroska-style wavpack packets. Libav commit 9b6f47c removed this
capability from the libavcodec code, and added code to libavformat's
Matroska demuxer to "rearrange" wavpack packets. Since demux_mkv still
sent Matroska-style packets, playback failed.
Fix this by "rearranging" packets in demux_mkv as well by copying
libavformat's code. (The best kind of fix.)
Tested with [CCCP]_Mega_Lossless_Audio_Test.mkv, as well as with a
sample generated by mkvmerge.
The core deselected all streams on initialization, and then selected the
streams it actually wanted. This was no problem for
demux_mkv/demux_lavf, but old demuxers (like demux_asf) could lose some
packets. The problem is that these demuxers can buffer some data on
initialization, which then is flushed on track switching. Fix this by
explicitly avoiding deselecting a wanted stream.
When AAC is streamed over HTTP, using libavformat defaults is
pathetically slow. One solution for that is skipping probing and using
the mimetype to identify that it's AAC instead. This is what we did
before this commit (and ffmpeg does it too, but their logic is too
"inaccessible" for mpv).
This is still pretty fragile though. Make it a bit more robust by
requiring minimal probing. A probescore of 25 is reached after feeding
2 KB to libavformat (instead of > 500 KB for the normal probescore), so
use that. This is done only when streaming AAC from HTTP to reduce the
possibility of weird breakages for other formats.
Also reduce analyzeduration. The default analyzeduration will make
libavformat read lots of data, which makes playback start slow. So we
set analyzeduration to a low value. On the other hand, doing that for
other formats is risky, because there are unspecified effects with
certain "strange" formats (like transport streams). So we do this only
if we're streaming AAC from HTTP as well.
tl;dr libavformat is shit for media players
This can control whether demux_lavf should use the HTTP mime type to
determine the format, instead of probing the data with the libavformat
API. Do this to allow easier debugging in case the mimetype is
incorrect. (This is done only for AAC streams right now.)
Playing Youtube videos often requires an additional seek to the end of
the file. This flushes the stream cache. The reason for the seek is
reading the cues (seek index). This poses the question why Google is
muxing its files in such a way, since nothing in Matroska mandates that
cues are located at the end of the file, but we want to handle this
situation better anyway.
The seek index is not needed for normal playback, only for seeking.
This commit changes header parsing such that the index is not read on
initialization in order to avoid the additional stream-level seek.
Instead, read the index on the first demuxer-level seek, when the seek
index is actually needed.
If the cues are at the beginning of the file, they are read immediately
as part of the normal header reading process. This commit changes
behavior only if cues are outside of the header (i.e. not in the area
between EBML header and clusters), and linked by a SeekHead. Other
level 1 elements linked by the SeekHead might still cause seeks to the
end of the file, although that seems to be rare.
Before this commit, the demuxer would in theory accept multiple cues
elements (and append its contents to the index in the order as
encountered during reading). According to the Matroska specification,
there can be only one cues element in the segment, so this seems like
an overcomplication.
Change it so that redundant elements are ignored, like with all other
unique header elements. This makes implementing deferred reading of the
cues element easier.
The sequence of avcodec_alloc_context3() / avcodec_copy_context() /
avcodec_close() / av_free() leaks some memory. So don't copy the context
and use it directly.
Originally avcodec_copy_context() was used to guarantee that libavformat
can't update the fields of the context during demuxing in order to make
things a little more robust, but it's not strictly needed, and
ffmpeg/ffplay don't do this anyway. Still might make the situation worse
should we move demuxing into a separate thread, though.
Using -demuxer mpegts -correct-pts triggered the assertion in
ds_get_packet2(). This is not surprising, because the correct-pts code
was changed to accept _complete_ packets, while all the old demuxers
(including the mpegts demuxer) require you to use "partial" packet
reads, together with the video_read_frame(). (That function actually
parses video frames, so fragments of the original "packets" can be fed
to the decoder.)
However, it returns out demux_ts packet's are mostly useable. demux_ts
still adds an offset (i.e. ds->buffer_pos != 0) to the packets when
calling internal parser functions, such as in parse_es.c. While this is
unclean design due to mplayer's old video demuxing/decoding path, it can
be easily be made work by modifying the packet as returned by
ds_get_packet2(). We also have to change the packet freeing code, as
demux_packet->buffer doesn't have to point to the start of the memory
allocation anymore.
MPlayer handles this "correctly" because it doesn't have a function that
reads a complete packet.
Nobody uses this, and this is an absolute waste of time. Even the user
who reported this turned out to have produced a sample manually.
Sample produced with:
wget http://diracvideo.org/download/test-streams/raw/vts/vts.LD-8Mb.drc
mkvmerge -o dirac.mkv vts.LD-8Mb.drc
mkvmerge writes a sort of broken aspect ratio. libavformat interprets it
as 1:1 PAR, while demux_mkv thinks this is a 1:1 DAR. Maybe libavformat
is more correct here.
This helps passing the channel layout correctly from decoder to audio
filter chain. (Because that part "reuses" the demuxer level codec
parameters, which is very disgusting.)
Note that ffmpeg stuff already passed the channel layout via
mp_copy_lav_codec_headers(). So other than easier dealing with the
demuxer/decoder parameters mess, there's no real advantage to doing
this.
Make the --channels option accept a channel map. Since simple numbers
map to standard layouts with the given number of channels, this is
downwards compatible. Likewise for demux_rawaudio.
Allow the stream layer to report chapter times. Extend stream_dvd to do
this. I'm not 100% sure whether the re-used code is bug-free (because it
was used for slave-mode and/or debugging only).
MAke the frontend do time-based seeks when switching DVD chapters. I'm
not sure if there's a real reason STREAM_CTRL_SEEK_TO_CHAPTER exists
(maybe/hopefully not), but we will see.
Note that querying chapter times in demuxer_chapter_time() with the new
STREAM_CTRL_GET_CHAPTER_TIME could be excessively slow, especially with
the cache enabled. The frontend likes to query chapter times very often.
Additionally, stream_dvd uses some sort of quadratic algorithm to list
times for all chapters. For this reason, we try to query all chapters on
start (after the demuxer is opened), and add the chapters to the demuxer
chapter list. demuxer_chapter_time() will get the time from that list,
instead of asking the stream layer over and over again.
This assumes stream_dvd knows the list of chapters at the start, and
also that the list of chapters never changes during playback. This
seems to be true, and the only exception, switching DVD titles, is not
supported at runtime (and doesn't need to be supported).
The frontend doesn't use this.
Also use double for returning the chapter times. Everything uses double
for times, and there's no reason to use float here.
These were found by the cppcheck and scan-build static analyzers. Most
of these aren't interesting (the 2 previous commits fix some interesting
cases found by these analyzers), and they don't nearly fix all warnings.
(Most of the unfixed warnings are spam, things MPlayer never cared
about, or false positives.)
Also, mark demuxer as not capable if DVD playback is done. The problem
with DVD is that playback time (stream_pts) is not reported frame-exact,
and the time is a "guess" at best.
With the commit "demux_lavf: fix DEMUXER_CTRL_RESYNC", DVD playback
seems to work nicely with demux_lavf, and maybe works even better than
with demux_mpg.
The old demuxer can be forced with: --demuxer=mpegps
If no regressions surface, demux_mpg.c will be deleted later.
This used the libavformat current position, instead of the mp stream
(which reflects current DVD/Bluray read position). This was broken,
because libavformat won't update its position by calling the user's
stream callbacks, negating the whole point of DEMUXER_CTRL_RESYNC.
Now DVD playback with libavformat seems to work relatively well.
demux_mpg did the same, and doing this in demux_lavf fixes DVD playback
when using this demuxer.
Additionally this might make bluray work better in the future (but for
now, bluray playback doesn't change as it doesn't report stream PTS yet).
DVD playback uses a demuxer that signals to the frontend that timestamp
resets are possible. This made the frontend calculate the OSD playback
position based on the byte position and the total size of the stream.
This actually broke DVD playback position display. Since DVD reports a
a linear playback position, we don't have to rely on the demuxer
reported position, so disable this functionality in case of DVD
playback. This reverts the OSD behavior with DVD to the old behavior.
The stream ID handling as it was changed in commit 654c34f was still
a little bit insane, and caused a regression with the cover art hack
(the stream set in demux->video->sh was incorrect for demux_lavf).
Simplify by always using stream_index for demux_stream->id, and getting
rid of that tid thing. It turns out that the id for subtitles isn't
special either (maybe demux_ts.c was the only thing left that required
this).
This check was always false:
if (num == EBML_UINT_INVALID)
Fix it by using the proper type for the num variable.
This case actually doesn't really matter, and this is just for hiding
the warning and for being 100% correct.
When trying to seek before the start of the file, which usually happens
when using the arrow keys to seek to the start of the file, external
libavformat demuxed subtitles will be invisible. This is because seeking
in the external subtitle file fails, so the subtitle demuxer is left in
a random state.
This is actually similar to the normal seeking path, which has some
fallback code to handle this situation. Add such code to the subtitle
seeking path too.
(Normally, all demuxer support av_seek_frame(), except subtitles, which
support avformat_seek_file() only. The latter was meant to be the "new"
seeking API, but this never really took off, and using it normally seems
to cause worse seeking behavior. Or maybe we just use it incorrectly,
nobody really knows.)
Get rid of the 1-char subtitle type field. Use sh_stream->codec instead
just like audio and video do. Use codec names as defined by libavcodec
for simplicity, even if they're somewhat verbose and annoying.
Note that ffmpeg might switch to "ass" as codec name for ASS, so we
don't bother with the current silly "ssa" name.
MP_INPUT_BUFFER_PADDING_SIZE and FF_INPUT_BUFFER_PADDING_SIZE are both
16. The doxygen for FF_INPUT_BUFFER_PADDING_SIZE says only the first 23
bits must to be 0, but this is probably a lie.
This removes the stream handling mess by using a single list for all
stream types.
One consequence is that new streams are always set to AVDISCARD_ALL,
which could be an issue if packets are read before initializing other
streams. However, this doesn't seem to an issue for various reasons,
so we don't do anything about it.
The new code strictly assumes that libavformat never removes or
reorders streams once added to AVFormatContext->streams. Undefined
behavior will result if it does.
mkv_track_t now references sh_stream directly, instead of using an ID.
Also remove all accesses to demux_stream (demuxer->video etc.).
Remove some slave-mode things on the way, like "ID_SID_..." messages.
Some preparations to simplify demux_mkv and demux_lavf.
struct demux_stream manages state for each stream type that is being
demuxed (audio/video/sub). demux_stream is rather annoying, especially
the id and sh members, which are often used by the demuxers to determine
current stream and so on. Demuxers don't really have to access this,
except for testing whether a stream is selected and to add packets.
Add a new_sh_stream(), which allows creating streams without having the
caller specify any kind of stream ID. Demuxers should just use sh_stream
pointers, instead of multiple kinds of IDs and indexes.
Since demux_mkv queries the demuxer state when reading packets, track
switching is completely passive. Cycling etc. is done by the frontend.
As result, all track switching code can be removed.
Matroska files can contain multiple segments, which are literally
further Matroska files appended to the main file. They can be referenced
by segment linking.
While this is an extraordinarily useless and dumb feature, we support it
for the hell of it.
This is implemented by adding a further demuxer parameter for skipping
segments. When scanning for linked segments, each file is opened
multiple times, until there are no further segments found. Each segment
will have a separate demuxer instance (with a separate file handle
etc.).
It appears the Matroska spec. has an even worse feature for segments:
live streaming can completely reconfigure the stream by starting a new
segment. We won't add support for it, because there are 0 people on this
earth who think Matroska life streaming is a good idea. (As opposed to
serving Matroska/WebM files via HTTP.)
Matroska segment linking allows abusing Matroska files as playlists
without any actual video/audio/sub data, making files without any
clusters still useful for the frontend.
Relative seeks backwards didn't work too well with incomplete files, or
other files that are missing the seek index. The problem was that the
on-the-fly seek index generation simply added cluster positions as seek
entries. While this is perfectly fine, the seek code had no information
about the location of video key frames. For example, a 5 second long
cluster can have only 1 video key frame, which is located 4 seconds into
the cluster. Seeking backwards by one second while still located in the
same cluster would select this cluster as seek target again. Decoding
would resume with the key frame, giving the impression that seeking is
"stuck" at this frame.
Make the generated index aware of key frame and track information, so
that video can always be seeked in an idea way. This also uses the
normal block parsing code for indexing the clusters, instead of the
suspicious looking special code. (This code didn't parse the Matroska
elements correctly, but was fine for files with normal structure. Files
with corrupted clusters or clusters formatted for streaming were not
handled properly.)
Skipping is now quite a bit slower (takes about twice as long as
before), but it removes the special cased skipping code, and it's still
much faster (at least twice as fast) than libavformat. It needs to do
more I/O (no more skipping entire clusters, all data is read), and has
more CPU usage (more data needs to be parsed).
Move most code from demux_mkv_fill_buffer() to read_next_block(). The
former is supposed to read raw blocks, while ..fill_buffer() reads
blocks and turns them into packets.
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.
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).
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;
^
No subtitle selected was supposed to disable the preroll logic
completely. However, the packet skipping logic was not properly enabled,
so the demuxer would still return subtitle packets from before the seek
target timecode. This shouldn't matter at all in practice, but fixing
this makes the code clearer.
Makes sure that seeking to a given time position shows the subtitle at
that position. This can fail if the subtitle packet is not close enough
to the seek target. Always enabled for hr-seeks, and can be manually
enabled for normal seeks with --mkv-subtitle-preroll.
This helps displaying subtitles correctly with ordered chapters. When
switching ordered chapter segments, a seek is performed. If the subtitle
is timed slightly before the start of the segment, it normally won't be
demuxed. This is a problem with all seeks, but in this case normal
playback is affected. Since switching segments always uses hr-seeks,
the code added by this commit is always active in this situation.
If no subtitles are selected or the subtitles come from an external
file, the demuxer should behave exactly as before this commit.
Commit 546ae23 fixed aspect ratio if the DisplayWidth or DisplayHeight
elements were missing. However, some bogus files [1] can have these
elements present in the file, but set to 0. Use 1:1 pixel aspect for
such files.
[1] https://ffmpeg.org/trac/ffmpeg/ticket/2424
Commit ac1c5e6 (demux_mkv: improve robustness against broken files)
added code to skip to the next cluster on error conditions. However,
reaching normal EOF triggers this code as well, so explicitly check
for EOF before this happens. Note that the EOF flag is only set _after_
reading the last byte, so EOF needs to be checked after the fact. (Or
in other words, we must check for EOF after the ebml_read_id() call.)
(To answer the question why reading packets actually reaches EOF, even
if there's the seek index between the last packet and the end of the
file: the cluster reading code skips the seeking related EBML elements
as normal part of operation, so it hits EOF gracefully when trying to
find the next cluster.)
Fixes test7.mkv from the Matroska test file collection, as well as some
real broken files I've found in the wild. (Unfortunately, true recovery
requires resetting the decoders and playback state with a manual seek,
but it's still better than just exiting.)
If there are broken EBML elements, try harder to skip them correctly.
Do this by searching for the next cluster element. The cluster element
intentionally has a long ID, so it's a suitable element for
resynchronizing (mkvmerge does something similar).
We know that data is corrupt if the ID or length fields of an element
are malformed. Additionally, if skipping an unknown element goes past
the end of the file, we assume it's corrupt and undo the seek. Do this
because it often happens that corrupt data is interpreted as correct
EBML elements. Since these elements will have a ridiculous values in
their length fields due to the large value range that is possible
(0-2^56-2), they will go past the end of the file. So instead of
skipping them (which would result in playback termination), try to
find the next cluster instead. (We still skip unknown elements that
are within the file, as this is needed for correct operation. Also, we
first execute the seek, because we don't really know where the file
ends. Doing it this way is better for unseekable streams too, because
it will still work in the non-error case.)
This is done as special case in the packet reading function only. On
the other hand, that's the only part of the file that's read after
initialization is done.
Fixes test4.mkv from the Matroska test file collection.
demux_mkv_open() contains a loop that reads header elements. It starts
by reading the EBML element ID with ebml_read_id(). If there is broken
data in the header, ebml_read_id() might return EBML_ID_INVALID.
However, that is not handled specially, and the code for handling
unknown tags is invoked. This reads the EBML element length in order to
skip data, which, if the EBML ID is broken, is entirely random. This
caused a seek beyond the end of the file, making the demuxer fail.
So don't skip any data if the EBML ID was invalid, and simply try to
read the next element. ebml_read_id() reads at least one byte, so the
parsing loop won't get stuck.
All in all this is rather questionable, but since this affects error
situations only, makes behavior a bit more robust (no random seeks), and
actually fixes at least one sample, it's ok.
libavformat's demuxer handled this.
The code modified by this commit is supposed to prevent demuxing the
whole file when cover art is present. (The problem with cover art is
that the ffmpeg libavformat API doesn't signal video EOF correctly - so
we try to read more packets to find the next video frame, which results
in demuxing and queuing the whole audio stream.)
This caused regressions for files with extremely high audio offset (see
github issue #46). MY conclusion is that this cover art crap doesn't
work, and this is just another case of completely insane ffmpeg/libav
API.
Disable the hack in all cases, unless a cover art video track is
selected. Maybe I'll handle cover art directly in the frontend later, so
that we don't have to rely on whatever libavformat does.
Unfortunately, this also makes behavior with equally insane mp4 files
with sparse video tracks worse, but this issue takes priority.
Doing 'mpv mf://*' in a file with directories would crash, because even
though directories are skipped, the corresponding file entry is just
left at NULL, leading to a segfault on access. So explicitly skip NULL
entries.
FFmpeg recently changed how it writes Opus-in-Matroska to match
the A_OPUS/EXPERIMENTAL name that mkvmerge uses, with the caveat
that things will change and compatibility with old files can get
worked out when the spec is finalized.
This adds both A_OPUS and A_OPUS/EXPERIMENTAL so that *hopefully*
it can play both the newer files that use A_OPUS/EXPERIMENTAL, and
older ones muxed by FFmpeg that were simply A_OPUS, since this is
also what FFmpeg seems to be doing to handle the situation.
r_frame_rate was deprecated and was finally removed from Libav and
FFmpeg git.
Not sure what's the correct replacement. avg_frame_rate may or may not
be worse than the fallback of using the time_base as guess. The
framerate is mostly unused, but needed for frame-based subtitles and for
encoding. (It appears encoding guesses a timebase based on the FPS, and
I'm not sure why we don't just use the source timebase.)
The old names have been deprecated a while ago, but were needed for
supporting older ffmpeg/libav versions. The deprecated identifiers
have been removed from recent Libav and FFmpeg git.
This change breaks compatibility with Libav 0.8.x and equivalent
FFmpeg releases.
Emulate percentage-seeks (SEEK_FACTOR) as normal time-seeks if possible.
This fixes some issues with (let's call it) low quality implementations
of SEEK_FACTOR (e.g. demux_mkv basically interprets this as byte-seek,
and also seeking to 99.9% makes it seek back to the start).
For weird MPEG formats the demuxer level SEEK_FACTOR is still used.
These formats, which can have timestamp resets, are identified by
setting demuxer->ts_resets_possible to true.
Also, have get_current_pos_ratio() follow the same rules, and calculate
the percentage position with the file position if timestamp resets are
possible.
This actually fixes percentage-seeks in .ts files with demux_lavf.c.
This kind of seek is not really used now, but it will be more important
when we add a progress bar.
Note: seeking in chained ogg files is still completely broken. The main
issue is that ffmpeg doesn't provide a sane API for dealing with
timestamp resets, and trying to do byte seeks with ogg confuses demuxer
and decoder (or something like this) and just does random things.
(Tested with two concatenated flac-in-ogg files).
AVFormatContext.start_time is sometimes AV_NOPTS_VALUE, such as when
playing FLAC files. (For most other file formats it's set to 0, even if
the format doesn't support arbitrary start times.)
OPT_BASE_STRUCT defines which struct the OPT_ macros (like OPT_INT etc.)
reference implicitly, since these macros take struct member names but no
struct type. Normally, only cfg-mplayer.h should need this, and other
places shouldn't be bothered with having to #undef it.
(Some files, like demux_lavf.c, still store their options in MPOpts. In
the long term, this should be removed, and handled like e.g. with VO
suboptions instead.)
The percent position is used for the OSD, the status line, and for the
OSD bar (shown on seeks). By default, the PTS of the last demuxed packet
was used to calculate it. This led to a "jumpy" display when the
percentage value (casted to int) was changing. The reasons for this were
the presence of video frame reordering (packet PTS is not monotonic), or
getting PTS values from different streams (like audio/subs).
Since these rely on PTS values and correct file durations anyway,
simplify it by calculating it with the current playback position in
mplayer.c instead.
And support the PIX_FMT_MONOWHITE pixel format. (This is really weird:
unlike PIX_FMT_MONOBLACK, it uses white pixels. I have no idea why
libavcodec doesn't just convert the pixel format on the fly, instead of
bothering everyone with really special pixel formats.)
Seeking before the start of a .flac file (such as seeking backwards when
the file just started) generates a bunch of decoding errors and audible
artifacts. Also, the audio output doesn't match the reported playback
position. The errors printed to the terminal are:
[flac @ 0x8aca1c0]invalid sync code
[flac @ 0x8aca1c0]invalid frame header
[flac @ 0x8aca1c0]decode_frame() failed
This is most likely a problem with the libavformat API. When seeking
with av_seek_frame() fails, the demuxer can be left in an inconsistent
state. ffplay has the same issue [1].
Older versions of mpv somehow handled this fine. Bisection shows that
commit b3fb7c2 caused this regression by removing code that retried
failed seeks with an inverted AVSEEK_FLAG_BACKWARD flag. This code was
removed because it made it harder to stop playback of a file by seeking
past the end of the file (expecting this is rather natural when skipping
through multiple files by seeking, and the internal mplayer demuxers
also did this).
As a workaround, re-add the original code, but only for the backwards
seeking case.
Also note that the original intention of the code removed in b3fb7c2 was
not dealing with this case, but something else. It also had to do with
working around weird libavformat situations, though. It's not perfectly
clear what exactly. See commit 1ad332f.
[1] https://ffmpeg.org/trac/ffmpeg/ticket/2282
Apparently this cuases trouble for legacy demuxers. demux_mpg stopped
doing PCM audio. (The problem was probably that it read a bunch of
video packets on detection, and then the sparse video hack prevented
audio packets from being read, because it looked like there were no
more audio packets. With sparse video, this normally helps not reading
too many audio packets.)
Since the legacy demuxers do not need this hack, enable it for
demux_lavf and demux_mkv only.
Some additional hacks that were needed to handle legacy demuxers can be
removed, making the code simpler.
Also see commit 4a40eed.
There should be no functional changes, except that way how avoiding
spamming the terminal with the overflow warning is handled changes a
bit.
The removed check for ds->eof looks suspicious, but it should be
redundant now.
Commit 4d016a9 changed how demuxers report the codec of each stream.
Some of that was missed in video.c, which is important for legacy
demuxers (demux_mpg was broken by this, which is needed for DVD
playback).
Not sure about the ASF/AVI related change, but this is also a legacy
demuxers only codepath.
Instead of putting codec header data into WAVEFORMATEX and
BITMAPINFOHEADER, pass it directly via AVCodecContext. To do this, we
add mp_copy_lav_codec_headers(), which copies the codec header data
from one AVCodecContext to another (originally, the plan was to use
avcodec_copy_context() for this, but it looks like this would turn
decoder initialization into an even worse mess).
Get rid of the silly CodecID <-> codec_tag mapping. This was originally
needed for codecs.conf: codec tags were used to identify codecs, but
libavformat didn't always return useful codec tags (different file
formats can have different, overlapping tag numbers). Since we don't
go through WAVEFORMATEX etc. and pass all header data directly via
AVCodecContext, we can be absolutely sure that the codec tag mapping is
not needed anymore.
Note that this also destroys the "standard" MPlayer method of exporting
codec header data. WAVEFORMATEX and BITMAPINFOHEADER made sure that
other non-libavcodec decoders could be initialized. However, all these
decoders have been removed, so this is just cruft full of old hacks that
are not needed anymore. There's still ad_spdif and ad_mpg123, bu neither
of these need codec header data. Should we ever add non-libavcodec
decoders, better data structures without the past hacks could be added
to export the headers.
Rearrange some code to make it easier readable. Remove some dead code,
and stop printing AVI headers in demux_lavf. (These are not actual AVI
headers, just for internal use.)
There should be no functional changes, other than reducing output in
verbose mode.
Also move the lang field into the general stream header. (SH_COMMON is
an old hack to "share" code between audio/video/sub headers.)
There should be no functional changes, other than not printing stream
info in verbose mode or with slave mode. (The frontend already prints
stream info, and this is just a leftover when individual demuxers did
this, and slave mode remains broken.)
Use codec names instead of FourCCs to identify codecs. Rewrite how
codecs are selected and initialized. Now each decoder exports a list
of decoders (and the codec it supports) via add_decoders(). The order
matters, and the first decoder for a given decoder is preferred over
the other decoders. E.g. all ad_mpg123 decoders are preferred over
ad_lavc, because it comes first in the mpcodecs_ad_drivers array.
Likewise, decoders within ad_lavc that are enumerated first by
libavcodec (using av_codec_next()) are preferred. (This is actually
critical to select h264 software decoding by default instead of vdpau.
libavcodec and ffmpeg/avconv use the same method to select decoders by
default, so we hope this is sane.)
The codec names follow libavcodec's codec names as defined by
AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders
have names different from the canonical codec name. The AVCodecDescriptor
API is relatively new, so we need a compatibility layer for older
libavcodec versions for codec names that are referenced internally,
and which are different from the decoder name. (Add a configure check
for that, because checking versions is getting way too messy.)
demux/codec_tags.c is generated from the former codecs.conf (minus
"special" decoders like vdpau, and excluding the mappings that are the
same as the mappings libavformat's exported RIFF tables). It contains
all the mappings from FourCCs to codec name. This is needed for
demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the
codec as determined by libavformat, while the other demuxers have to do
this on their own, using the mp_set_audio/video_codec_from_tag()
functions. Note that the sh_audio/video->format members don't uniquely
identify the codec anymore, and sh->codec takes over this role.
Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which
provide cover the functionality of the removed switched.
Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure
container/video combinations (e.g. the sample Film_200_zygo_pro.mov)
are played flipped. ffplay/avplay doesn't handle this properly either,
so we don't care and blame ffmeg/libav instead.
The ffmpeg/libav attached picture hack usually set the PTS of video
packets to AV_NOPTS_VALUE. Set it to 0 to avoid printing a warning by
the filter code.
Should be dead code. Stream selection is handled either during
demuxer initialization, or via DEMUXER_CTRL_SWITCH_*.
(If there were actually situations where this code did something, it
was probably broken anyway.)
It appears this is not needed anymore. ffmpeg can handle "chained" ogg
files fine. These can be created with "cat file1.ogg file2.ogg > chained.ogg",
and are similar (or equal) to some internet radio streams. Apparently
ffmpeg used to add new tracks when crossing boundaries in chained files,
and the hack in demux_lavf.c handled this. At some later point, ffmpeg's
ogg demuxer was improved, and stopped adding new tracks as long as the
codec doesn't change.
Since the hack in demux_lavf.c was hardcoded to Vorbis (i.e. only active
if the new and old track were both Vorbis), it's dead code, and we can
remove it. I couldn't find any stream that triggered this hack, or fails
without it.
Firefox had a similar issue, and its bug tracker makes a good reference:
https://bugzilla.mozilla.org/show_bug.cgi?id=455165
NOTE: this doesn't update metadata on track changes anymore.
