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.
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.
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.
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
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.
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.)
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
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.
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).
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.
Update endpos each time libavformat asks for it.
Fixes playback of still downloading files to not stop before we
really reached the end.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35107 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
libmpdemux/demux_lavf.c
Change STREAM_CTRL_GET_SIZE argument type from off_t to
uint64_t.
Also fix the incorrect type of the uint64_res variable.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35360 b3059339-0415-0410-9bf9-f77b7e298cf2
Conflicts:
libmpdemux/demux_lavf.c
libmpdemux/muxer_lavf.c
Note: also merges the "forgotten" cache support from r35107.
Make demux_lavf not error out if no video or audio track is present.
This allows opening subtitle files with the demuxer.
Improve the test whether subtitles read from demuxers must do explicit
packet reads. (I'm not sure whether always doing these reads could have
bad effects, such as reading too many audio and video packets at once,
so be conservative.)
Finish renaming directories and moving files. Adjust all include
statements to make the previous commit compile.
The two commits are separate, because git is bad at tracking renames
and content changes at the same time.
Also take this as an opportunity to remove the separation between
"common" and "mplayer" sources in the Makefile. ("common" used to be
shared between mplayer and mencoder.)
Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.
Renames the following directories:
libaf -> audio/filter
libao2 -> audio/out
libvo -> video/out
libmpdemux -> demux
Split libmpcodecs:
vf* -> video/filter
vd*, dec_video.* -> video/decode
mp_image*, img_format*, ... -> video/
ad*, dec_audio.* -> audio/decode
libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.
Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.
sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).
Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.
wm4
Rudolf Polzer
Uoti Urpala
reimar