Keep the currently displayed subtitles even when the user cycles through
subtitle tracks, and the subtitle is decoded by libavcodec (such as
vobsubs). Do this by not clearing the subtitles on reset(). reset() is
also called on seek, so check the start PTS whether the subtitle should
really be displayed (there's already an end PTS). Note that sd_ass does
essentially something similar.
The existing code has checks for whether the PTS reported by the demuxer
is invalid (MP_NOPTS_VALUE). I don't know under what circumstances this
can happens, so fall back to the old behavior if the PTS is invalid.
Doesn't define AVPROBE_SCORE_RETRY for some reason. They use
AVPROBE_SCORE_MAX/4 directly internally. AV_DISPOSITION_ATTACHED_PIC
is not defined with the most recent Libav release.
AVIOContext.av_class exists in Libav, but is apparently disabled in
old releases. Disable it for now until people stop torturing me with
old crap releases.
Commit c02f25 switched the "http://" protocol to use ffmpeg's HTTP
implementation (stream_lavf.c), instead of the mplayer internal one
(http.c). Unfortunately, it turns out that there are some network
related options that are not respected by stream_lavf.c, and
consequently do not work anymore for "http://" URLs. This might be
fixed later. Mark them as deprecated for now, as it might take
arbitrarily long until this is taken care of.
This slightly improves display of the current playback time in files
with sparse video packets (like video tracks containing a slow MJPG
slideshows as in [1]), or audio files with cover art image attachments.
While the video PTS is always "stuck" at the last frame displayed or
the last seek, audio is usually continuous. Given sane samplerates and
working audio drivers (to query how much of the current audio buffer has
been played), the audio PTS should always be more reliable.
[1] http://www.podtrac.com/pts/redirect.mp3/traffic.libsyn.com/rtpodcast/Rooster_Teeth_Podcast_191.m4a
Improve EOF handling in ds_fill_buffer for the case where one stream ends
much earlier than the others, in particular make sure the "too many ..."
message is not printed over and over.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32823 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
libmpdemux/demuxer.c
Try to improve seeking in files with only few video packets,
in particular files with cover art.
This might cause issues with badly interleaved files, particularly
together with -audio-delay, even though I did not see issues
in my very limited testing.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35486 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
libmpdemux/demuxer.c
libmpdemux/demuxer.h
Fix code that detects streams temporarily lacking data to work
properly with e.g. DVDs.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35499 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
libmpdemux/demuxer.c
Make stream eof detection less sensitive.
Fixes bug #2111.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35543 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
libmpdemux/demuxer.c
ffmpeg pretends that image attachments (such as contained in ID3v2
metadata) are video streams. It injects the attached pictures as packets
into the packet stream received with av_read_frame().
Add the --audio-display option to allow configuring whether attached
pictures should be displayed. The default behavior doesn't change
(images are displayed).
Identify video streams, that are actually image attachments, with "[P]"
in the terminal output.
Modify the default stream selection such that real video streams are
preferred over attached pictures. (This is just for robustness; I do not
know of any samples where images are added before actual video streams
and could lead to bad default stream selection with the old code.)
ad_dvdpcm reads MPEG specific headers directly (passed through codecdata
by demux_mpg), so you couldn't use ffmpeg's "pcm_dvd" with demux_mpg.
Change demux_mpg to set the correct audio parameters directly. The code
for this is taken from ad_dvdpcm.
ad_dvdpcm is evil because it still does partial packet reads (with
demux_read_data()), and it's redundant to libavcodec anyway.
Since libavcodec doesn't have a "generic" PCM decoder, we have to go out
of out way to make it look like ad_lavc provides one: make it provide a
pseudo "pcm" decoder, which maps some format tags manually to the
individual libavcodec PCM decoders.
Format tags which uniquely map to one libavcodec could be mapped via
codecs.conf. Since defining these in tag_map[] is much shorter (one line
vs. a full codec entry in codecs.conf), and since we need tag_map[]
anyway, we don't use codecs.conf for these.
ad_pcm is evil because it still does partial packet reads (with
demux_read_data()), and it's redundant to libavcodec anyway.
This caused errors like:
core/mplayer.c:4308:5: error: implicit declaration of function 'pthread_win32_thread_detach_np' [-Werror=implicit-function-declaration]
It turns out a pthread.h include was missing. It's not clear why this
used to work (or rather, why it happens only sometimes). Possibly some
libraries or system headers recursively include pthread.h under certain
circumstances or configurations.
Fix missing quoting in configure, which led to broken terminal output.
Closes#6.
This is a fix for web radio streams that send raw AAC [1]. libavformat's
AAC demuxer probe is picky enough to request hundreds of KBs data, which
makes for a slow startup. To speed up stream startup, try use the HTTP
MIME type to identify the format. The webstream in question sends an AAC
specific MIME type, for which demux_lavf will force the AAC demuxer,
without probing anything.
ffmpeg/ffplay do the same thing. Note that as of ffmpeg commit 76d851b,
av_probe_input_buffer() does the mapping from MIME type to demuxer. The
actual mapping is not publicly accessible, and can only be used by
calling that function. This will hopefully be rectified, and ideally
ffmpeg would provide a function like find_demuxer_from_mime_type().
[1] http://lr2mp0.latvijasradio.lv:8000
libavformat wants to read a full ~400KB of data to determine whether
it's really AAC. This causes slow startup with AAC web radio streams [1]
(possible due to a broken initial packet). There are similar issues
with other file formats.
Make the probe "score" (libavformat's mechanism for testing file
formats) configurable with the -lavfdtops:probescore option. This allows
lowering the amount of data read on probing. If the probe score is below
the probescore option value, demux_lavf will try to get a higher score
by feeding more data to libavformat, until the required score or the
max. probe size is reached.
Remove the lavf_preferred demuxer entry. This had a purpose in
mplayer-svn, but now there doesn't seem to be any good reason for it
to exist. Make sure that our native "good" demuxers are above
demux_lavf in demuxer_list[] instead (so that they are preferred).
[1] http://lr2mp0.latvijasradio.lv:8000
Use ffmpeg (stream_lavf) instead of internal mms support (asf_streaming.c)
for mms://, mmsh://, mmst:// URLs.
The old implementation is available under mp_mms:// etc.
There are some caveats with this:
- mms:// now always maps to mmsh://. It won't try mmst://. (I'm not sure
if mms:// URLs really can use the mmst protocol, though.)
- MMS streams under the http:// prefix are not handled. (ffmpeg ticket
#2001.) (Was already broken in mpv since c02f25.)
- It downloads all video streams now. MMS streams often have redundant
video streams, which encode the main stream at different quality. The
client is supposed to select one according to its bandwidth
requirements. (Explicit MMS stream selection has been broken in mpv
for a while, because MPOpts.vid maps to the stream number, not the
demuxer's stream ID - but the old logic doesn't work anyway when
using demuxer_lavf as opposed to demux_asf.)
ffmpeg recently added a demuxer that can read vobsubs (pairs of .sub and
.idx files). Get rid of the internal vobsub reader, and use the ffmpeg
demuxer instead.
Sneak in an unrelated manpage change (autosub default).
This affects streams loaded with -subfile and -audiofile. They could get
out of sync when they were deselected, and the main file was seeked. Add
code to seek external files when they are selected (see
init_demux_stream()).
Use avformat_seek_file() under certain circumstances. Both av_seek_frame()
("old" API) and avformat_seek_file() ("new" API) seem to be broken with
some formats. At least the vobsub demuxer doesn't implement the old API
(and the old API doesn't fallback to the new API), while the fallback
from new API to old API gives bad results. For example, seeking forward
with small step sizes seems to fail with the new API (tested with
Matroska by trying to seek 1 second forward relative to priv->last_pts).
Since only subtitle demuxers implement the new API anyway, checking
whether iformat->read_seek2 is set to test whether the old API is not
supported gives best results. This is a hack at best, but makes things
work.
Remove backwards seeking on seek failure. This was annoying, and only
was there to compensate for obscure corner cases (see 1ad332). In
particular, files with completely broken seeking that used to skip back
to the start on every seek request may now terminate playback.
Do this only if demux_lavf is used. Using demux_mpg and the ffmpeg DVD
subtitle decoder doesn't work. The problem is probably that demux_mpg
doesn't join split sub packets, while demux_lavf does. The internal
DVD sub decoder (spudec.c) can, while ffmpeg's dvdsub can't. I do not
know whether this is the actual problem.
If DVD playback is used, create "fake" vobsub-style text extradata
(like .idx files) to pass resolution and palette information to the
ffmpeg decoder. We could use the "palette" AVOpt and avcodec_set_dimensions()
instead, but it's actually simpler this way. Note that the decoder
doesn't parse any other fields. Also note that DVD playback still uses
demux_mpg by default, so this code is inactive unless -demuxer lavf is
specified. This is mainly preparation for the case when we manage to get
rid of demux_mpg for DVD playback.
The option sub-forced-only was accidentally renamed back to
forcedsubsonly in commit 72205635ab, causing a segfault in
mp_property_generic_option due to missing option.
Button 3 and 4 are mapped to the mouse wheel. Double click events for
them annoying and not useful at all.
I don't know about buttons 5-19; for all I know these could be mapped
to wheels as well. Even if not, double click events are probably not
very important for these. Disable double clicks for these as well.
libavformat demuxes ass subtitles in a broken way, that forces the
player to throw away all subtitle events received so far. See mplayer
svn commit 31293.
Do not fall back to 0 for samplerate when parser is not initialized.
Might fix some issues with using -ac spdifenc with audio in MKV
or MP4.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35517 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace outdated list of unsupported formats by list of supported formats.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35534 b3059339-0415-0410-9bf9-f77b7e298cf2
Do not call af_fmt2str on the same data over and over.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35535 b3059339-0415-0410-9bf9-f77b7e298cf2
ad_spdif: use the more specific AF_FORMAT_AC3_LE when
we handle AC3.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35536 b3059339-0415-0410-9bf9-f77b7e298cf2
Make AF_FORMAT_IS_IEC61937 include AF_FORMAT_IS_AC3.
Our AC3 "sample format" is also iec61937.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35537 b3059339-0415-0410-9bf9-f77b7e298cf2
af_format: support endianness conversion also for iec61937
formats in general, not just AC3.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35538 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
audio/filter/af_format.c
af_format: Fix check_format, non-special formats are of course supported.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35545 b3059339-0415-0410-9bf9-f77b7e298cf2
Note: see mplayer bug #2110
stream ftp: Pass full buffer size to snprintf
Previously the buffer size was always passed as one less than
the underlying buffer's size. This is not using the underlying
buffer to its full potential according to the C99 standard. The
last byte of the buffers were never used.
No vulnerabilities should have been caused by this mistake because
the strings stored in the buffers were zero terminated at all
times. Neither were out-of-array writes nor reads possible.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35488 b3059339-0415-0410-9bf9-f77b7e298cf2
stream ftp: open_f: Mark parameter file_format unused
We have nothing to say about it, so we do not set *file_format.
No need for compilers to emit a warning about it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35489 b3059339-0415-0410-9bf9-f77b7e298cf2
stream ftp: Set type to STREAMTYPE_STREAM
Previously this was not set at all from within the stream_ftp module.
This caused the run-time warning message "Streams need a type!".
The actual behaviour should not be affected by this change.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35490 b3059339-0415-0410-9bf9-f77b7e298cf2
stream ftp: Use C99 designated initializers
Simplify the initialization of the stream private struct's defaults.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35491 b3059339-0415-0410-9bf9-f77b7e298cf2
stream ftp: Remove unneeded cast
At worst these kind of casts can hide real errors. As it is, it is
just not needed at all, thus remove it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35492 b3059339-0415-0410-9bf9-f77b7e298cf2
Reinitialize sh_audio->samplesize and sample_format before falling back
to another audio decoder (some decoders rely on default values). Remove
code setting these fields from demux_mkv and demux_lavf (no decoder
should depend on demuxer-set values for these fields).
Conflicts:
audio/decode/ad_lavc.c
Merged from mplayer2 commit 6b9567. The changes to ad_lavc.c are not
merged, as they are very specific to the mplayer2 libavresample hack;
we deplanarize manually, so we can't get unsupported sample formats
yet (except on raw audio with "pcm_f64le", as we don't support
AV_SAMPLE_FMT_DBL in the audio chain).
The subreader.c ASS parser (used when playing an external ASS subtitle
file with --no-ass for "plaintext" display) hardcodes dialogue line
syntax instead of correctly reading it from the "Format: " line in the
file, but tried to support a varying amount of fields by guessing
where the text field (which should be last) begins. This guessing code
was buggy in many ways. Remove it and hardcode skipping 9 commas
before the text field, which should work for most files. I don't
consider the --no-ass case important enough to implement correct
parsing now. Also fix the code removing formatting tags, which failed
to remove the second in a pair of two consecutive tags.
Conflicts:
sub/subreader.c
Merged from mplayer2 commit 91f516. Essentially reverts d5b964.
Remove the options --utf8 and --unicode which had no effect any more
(what they once did should be doable with --subcp). The only use of
corresponding variables left in code was subreader.c code using
sub_utf8 as a flag indicating whether iconv conversion was active.
Change the code to test the existence of iconv context instead.
Conflicts:
DOCS/man/en/options.rst
core/cfg-mplayer.h
sub/sub.c
sub/sub.h
sub/subreader.c
Merged from mplayer2 commit ea7311.
Note: --unicode was already removed
When the cache fill status goes below a certain threshold, automatically
pause the player. When the cache is filled again, unpause again.
This is intended to help with streaming from http. It's better to pause
a while, rather than exposing extremely crappy behavior when packet
reads during decoding block the entire player.
In theory, we should try to increase the cache if underruns happen too
often. Unfortunately, changing the cache implementation would be very
hard, because it's insane code (forks, uses shared memory and "volatile"
etc.). So for now, this just reduces the frequency of the stuttering if
the network is absolutely too slow to play the stream in realtime.
The internal mplayer http implementation is terrible and old. Try to use
ffmpeg instead. Enable ffmpeg https as well (we don't have native
support for it and never will).
Enable the cache for ffmpeg streams. In theory, we want to enable the
cache for network streams only (or at least, not enable it for files),
but there doesn't seem to be any way to query this. ffmpeg has its own
read-cache, but apparently it doesn't get filled asynchronously (only
to the degree internal socket/kernel buffers do).
Should it turn out that the mplayer implementation is better than
ffmpeg's in some way (unlikely), this commit might be reverted until
ffmpeg is ready. So far, the ffmpeg implementation seems to work very
well though.
The original mplayer stream implementation is made available under
"mp_http://" and "mp_http_proxy://". This is for testing, until the
http implementation is completely removed. Note that this is unlikely
to happen soon: ffmpeg doesn't support Microsoft streaming, while
mplayer has some support. This code has dependencies on http.
Note: "http_proxy://" doesn't work anymore. Use "httpproxy://" to get
the ffmpeg implementation of it.
Using https requires enabling OpenSSL in ffmpeg by adding
the --enable-openssl switch to ffmpeg's configure.
This commit is separate from the previous one to separate our own
changes from changes merged from mplayer2 (as far as that was possible).
Make it easier for stream implementations to request being cached. Set
a default cache size in stream.c, and remove them from various stream
implementations. Only MS streaming support sets a meaningful cache size.
Make querying cache size saner. This reduces the amount of #ifdefs
needed.
Code enabling the cache by default for network streams did that by
modifying the value of the "cache" option. This wasn't sane, as
multiple streams may be created and all share the same options. Change
the code to not modify options but store data in the stream instance
instead.
Conflicts:
core/mplayer.c
demux/demux.c
stream/cache2.c
stream/network.c
stream/network.h
stream/pnm.c
stream/stream.c
stream/stream_rtp.c
Merged from mplayer2 commit e26070. Note that this doesn't solve any
actual bug, as the playlist crashing bug has been fixed before.
Since the global cache size option value is not overwritten anymore, the
option doesn't need to be restored on end of playback (M_OPT_LOCAL).
Based on a patch by qyot27. Add the missing parts in mp_get_chroma_shift(),
which allow allocation of such images, and which make vo_opengl
automatically accept the new formats. Change the IMGFMT_IS_YUVP16_LE/BE
macros to properly report IMGFMT_444P14 as supported: this pixel format
has the highest numerical bit width identifier (0x55), which is not
covered by the mask ~0xfc. Remove 1 bit from the mask (makes it 0xf8) so
that IMGFMT_IS_YUVP16(IMGFMT_444P14) is 1. This is slightly risky, as
the organization of the image format IDs (actually FourCCs + mplayer
internal IDs) is messy at best, but it should be ok.
Based on a patch by qyot27. Add export LC_ALL=C on top of version.sh to
make the output locale independent.
Note that the build time will not be updated on every "make" invocation,
but only when the git revision is updated. This is a good thing, as
repeated make invocations should not rebuild the binary. (This would
break "sudo make install" too.)
libavdevice supports various "special" video and audio inputs, such
as screen-capture or libavfilter filter graphs.
libavdevice inputs are implemented as demuxers. They don't use the
custom stream callbacks (in AVFormatContext.pb). Instead, input
parameters are passed as filename. This means the mpv stream layer has
to be disabled. Do this by adding the pseudo stream handler avdevice://,
whose only purpose is passing the filename to demux_lavf, without
actually doing anything.
Change the logic how the filename is passed to libavformat. Remove
handling of the filename from demux_open_lavf() and move it to
lavf_check_file(). (This also fixes a possible bug when skipping the
"lavf://" prefix.)
libavdevice now can be invoked by specifying demuxer and args as in:
mpv avdevice://demuxer:args
The args are passed as filename to libavformat. When using libavdevice
demuxers, their actual meaning is highly implementation specific. They
don't refer to actual filenames.
Note:
libavdevice is disabled by default. There is one problem: libavdevice
pulls in libavfilter, which in turn causes symbol clashes with mpv
internals. The problem is that libavfilter includes a mplayer filter
bridge, which is used to interface with a set of nearly unmodified
mplayer filters copied into libavfilter. This filter bridge uses the
same symbol names as mplayer/mpv's filter chain, which results in symbol
clashes at link-time.
This can be prevented by building ffmpeg with --disable-filter=mp, but
unfortunately this is not the default.
This means linking to libavdevice (which in turn forces linking with
libavfilter by default) must be disabled. We try doing this by compiling
a test file that defines one of the clashing symbols (vf_mpi_clear).
To enable libavdevice input, ffmpeg should be built with the options:
--disable-filter=mp
and mpv with:
--enable-libavdevice
Originally, I tried to auto-detect it. But the resulting complications
in configure did't seem worth the trouble.
Opening files with the libavformat AVISynth demuxer ("avs"/avisynth.c)
fails, because the filename we pass to avformat_open_input() is prefixed
with "mp:". Normally, this doesn't matter, because data is read with the
stream interface. The AVISynth demuxer can't use this, because the
Avisynth API (apparently) can't read scripts from memory, and requires
a filename.
The "mp:" prefix used to be required when mplayer's stream layer was
made available as protocol to ffmpeg. This was replaced by setting
custom stream callbacks in de4908 (svn commit 25499), but the prefix
wasn't removed. Since this prefix doesn't have any purpose anymore and
prevents AVS playback from functioning, remove it. Fixes#5.
To get guaranteed alignment for the chroma planes with typical YV12
playback, we have to double the alignment on the image width, as the
chroma planes have half the image width.
Clear the image with black instead of green to hide scaling artifacts
on the right border of the screen.
(It might be possible to create the image layout ourselves by not
calling XvShmCreateImage(), and filling in our own image width and exact
strides, but that's probably too risky: the Xv client library sends an
X protocol request to query the real image dimension and strides. It is
unknown to me whether X servers or drivers would generally accept an
image with mismatching parameters, even if the image is conceptually
valid.)
Allocate the image with av_malloc() in the non-SHM case. I suspect the
non-SHM case doesn't matter much, though.
The presence of inttypes.h is guaranteed by POSIX. We don't need to
check for it. We don't need to provide a compatibility header either.
Apparently libc5 systems didn't provide inttypes.h. libc5 is ancient,
unmaintained, and not used by modern Linux systems.
The option is -no-video. Remove the deprecated "fast" suboption, which
did nothing and instructed the user to use "-novideo" instead.
Fix a reference to -novideo in encoding.rst.
Add a "generic" entry about -no-* to the list of renamed options. The
change is already explicitly mentioned in the text above the table, but
even if it's redundant, it makes it harder to overlook.
This fixes operation with current ffmpeg releases.
Note that this planarization is slow and should be reverted once proper
planar audio support is there in mpv.
Gestalt is deprecated since 10.8. Change the code to read the OS version from
a system plist file.
As mentioned http://stackoverflow.com/a/11072974/499456 Apple engineers are
suggesting this plist reading approach.
This is consistent with the demuxer/decoder info output mpv already has,
and is also generally useful to know, especially if using
--ao=codec1,codec2,... syntax.
The following HEX formats are now parsed correctly: `#RRGGBB`, `RRGGBB`.
Moreover this implementation doesn't show the HTML on screen if the input
color is not recognized. A warning is still displayed in the terminal.
