Playing a .cue file directly will now parse the .cue file, and load and
play the file(s) referenced in the cue. If multiple files are referenced,
a timeline including all files will be created to create the impression
of a single, flat audio file containing all the tracks.
For each track, a chapter is created. The chapter navigation commands can
be used to jump between tracks. The chapter titles will use the string
provided by the track's TITLE cue command. (The -identify command can be
used to print all chapters in a not so user friendly way.)
Other than the chapter names, there is no attempt at displaying or exposing
any other meta data contained in the cue files yet.
The handling (or lack of thereof) of gaps (track pregaps and postgaps) is
probably not correct yet. In general, mplayer's mapping of tracks to the
source audio files can be verified by examining the timeline, which will
be printed when passing the -v switch.
Note that this has nothing to do with the old cue:// support. The old code
isn't touched, and is still only able to play .cue/.bin pairs. Prefixing a
.cue file with cue:// will always invoke the old code, while playing a .cue
file directly (i.e. "mplayer file.cue") will always use the new code.
Playing audio images (.cue/.bin pairs of files) doesn't work yet.
Update various code using Libav libraries to remove use of API
features that were deprecated at Libav release 0.7. I think this
removes them all with the exception of URLContext functions still used
in stream_ffmpeg.c (at least other uses that generated deprecation
warnings with libraries from 0.7 are removed).
Require versions of the Libav libraries corresponding to Libav release
0.7. These are:
libavutil 51.7.0
libavcodec 53.5.0
libavformat 53.2.0
libswscale 2.0.0
libpostproc 52.0.0
Also disable the fallback to simple header check if these libraries
could not be found with pkg-config; now compiling without pkg-config
support for these always requires explicitly setting --enable-libav
and any needed compiler/linker flags. The simple check would have let
compilation proceed even if a version mismatch was detected.
Information about individual chapters was printed during demuxer
opening phase, and total chapter count (ID_CHAPTERS) was printed
according to mpctx->demuxer->num_chapters. When playing a file with
ordered chapters, this meant that chapter information about every
source file was printed individually (even though only the chapters
from the first file would be used for playback) and the total chapter
count could be wrong. Remove the printing of chapter information from
the demuxer layer and print the chapter information and count actually
used for playback in core print_file_properties().
Also somewhat simplify the internal chapters API and remove possible
inconsistencies.
Demux_demuxers checked a pts value against 0 to see if it was unset,
but other code uses MP_NOPTS_VALUE for that now. As a result audio and
subtitle streams could seek to a large negative position (effectively
to 0) instead of the correct target position. This broke --audiofile;
the --initial-audio-sync code could compensate for wrong demuxer seek
up to 5 minutes from the start of the file, masking the bug, but
seeking further than that audio would seek to 0 instead.
Note that the current --audiofile implementation using the
demux_demuxers wrapper is a fundamentally unsound design and still not
expected to generally work properly even after fixing this particular
problem.
Codec selection for audio and video decoding had a "dynamic plugin"
feature that tried to load a shared library for any codec that had not
been enabled at compilation (disabled by default, but could be enabled
with --enable-dynamic-plugins configure switch; for unknown reasons
some distro packages have enabled it). The implementation was buggy
and could cause normal codec selection fallback to fail if the feature
was enabled. I'm not aware of any real uses of such dynamic plugins
and the feature seems questionable anyway (there are no ABI guarantees
that would make it safe to use). Remove the buggy feature.
Libav stopped automatically filling missing codec_tag field for raw
codecs based on pix_fmt in libav commit bb416bd68c ("lavf: do not set
codec_tag for rawvideo"). This broke demux_lavf for raw video in
formats like YUV4MPEG, as the video format was not exported from
demux_lavf in any form (the information only existed in the pix_fmt
field of the struct AVCodecContext from libavformat, and that is not
exported). Add an explicit call to avcodec_pix_fmt_to_codec_tag() to
set the codec tag again so that selecting the correct raw decoder
based on the tag works.
demux_mf allocated the "type" suboption of "--mf" with strdup if it
was not explicitly set. This caused a crash after playing an mf://
entry. Fix to use talloc instead.
FFmpeg has increased FF_INPUT_BUFFER_PADDING_SIZE to 16 (unlike Libav
which still has it at 8). Raise MP_INPUT_BUFFER_PADDING_SIZE to 16 to
allow compilation against FFmpeg too (demuxer.c checks the padding
size for packets is at least as much as libavcodec wants for its
decoders, and this check failed with the previous value of 8).
Commit 6e8d420a41 ("demux: avoid a copy of demux packets with lavf,
reduce padding") was missing an av_dup_packet() line. As a result at
least formats that use parsing on the lavf side could fail (with
parsing the packet may contain pointers to temporary fields that
will be overwritten/freed when reading further packets, and
av_dup_packet() is required to allocate permanent storage).
After 0ece360eea ("demux_mkv: skip files faster in ordered chapter
file search") some Matroska files failed to open. The problem was that
demux_mkv_read_info() returned 0 on success, but the opening code
interpreted this as a value to stop parsing further headers. Fix this
and also modify some of the other return value handling.
Pass the libavformat packet side_data field from demux_lavf to
vd_ffmpeg. Libavcodec/libavformat use this field for palette data, and
passing it is required for the playback of some paletted video codecs.
The implementation works by giving vd_ffmpeg a copy of the struct
demux_packet used to store the video packet (from which it can access
the avpacket field). The definition of struct demux_packet is moved to
new file demux_packet.h so that vd_ffmpeg.c can use it without
including all of demuxer.h.
Export the codec private data field for WavPack and TrueHD audio
tracks. At least for WavPack this is necessary to make some samples
work.
Also change some other cases to use the same data-copying code.
When switching audio or video tracks, demux_mkv only checked that the
new index fell in the range corresponding to tracks existing in the
file being played. However, if the demuxer can not recognize the
format of a track or detects an error, some of those tracks in the
file may not be exported from the demuxer and are not visible to the
rest of the player. Selecting such a track would cause a crash. Add
checks skip such tracks when cycling to next track and switch to
nosound instead if given an explicit track number corresponding to
such a track.
demuxer.c calls demuxer->close() even if opening failed. Thus the
mkv_free() call added in 0ece360eea ("demux_mkv: skip files faster
in ordered chapter file search") was wrong, and could cause a crash
from a double free if some data structures were allocated before the
opening attempt was aborted.
Drop the unnecessary include and add a missing direct include in some
files. This also revealed that demux_rtp_internal.h was missing a
config.h include, fix that too.
Remove unnecessary demuxer.h include from aviheader.h. Through
stheader.h aviheader.h is included in a lot of files. Add missing
mp_msg.h includes to av_sub.c and sd_ass.c (previously hidden by
indirect inclusion through demuxer.h and stream.h).
When demux_lavf read a new packet it used to copy the data from
libavformat's struct AVPacket to struct demux_packet and then free the
lavf packet. Change it to instead keep the AVPacket allocated and
point demux_packet fields to the buffer in that.
Also change MP_INPUT_BUFFER_PADDING_SIZE to 8 which matches
FF_INPUT_BUFFER_PADDING SIZE; demux_lavf packets won't have more
padding now anyway (it was increased from 8 earlier when
FF_INPUT_BUFFER_PADDING_SIZE was increased in libavcodec, but that
change was reverted).
Don't interpret native MPEG codec tags using our generic
format-agnostic codec tag tables. MPEG may use tag 3 for MP3, whereas
the generic tables map 3 to uncompressed PCM. Make the code ignore the
codec_tag field for the "mpeg" and "mpegts" libavformat demuxers and
rely on the codec_id value provided by lavf only.
Ordered chapter code tries opening files to find those matching the
SegmentUID values specified in the timeline. Previously this scan did
a full initialization of the Matroska demuxer for each file, then
checked whether the UID value in the demuxer was a match. Make the
scan code instead provide a list of searched-for UIDs to the demuxer
open code, and make that do a comparison against the list as soon as
it sees the UID in the file, aborting if there is no match.
Also fix units used in "Merging timeline part" verbose message.
Change written_audio_pts() and playing_audio_pts() to return
MP_NOPTS_VALUE if no reasonable pts estimate is available. Before they
returned some incorrect value typically around zero (but not
necessarily exactly that).
Recent commit 5d5ca22a6d ("options: commandline: accept --foo=xyz
style options") left some bad code under "#ifdef MP_DEBUG" in
playtree.c, which caused a compilation failure if configured with
"--enable-debug". Fix this. Having the "#ifdef MP_DEBUG" there was
completely unnecessary; it only increased the risk for this kind of
problems for no real benefit - executing the asserts under it would
have no noticeable performance or other penalty in default builds
either. Remove several cases of such harmful "#ifdef MP_DEBUG".
Rename the BSTR() function to bstr(). The former caused a conflict
with some Windows OS name, and it's no longer a macro so uppercase
naming is less appropriate.
Do the global initialization of libavcodec and libavformat
(avcodec_register_all(), av_register_all()) immediately on program
startup and remove the initialization calls from various individual
modules that use libavcodec/libavformat functionality.
Some versions of lavf abuse codec_tag for passing Bink version
information to the decoder, which broke detection based on codec tag
(though this has already stopped again in latest Libav). Move bink
audio codec IDs from mp_wav_tags to mp_codecid_override_tags so that
codec tags are completely ignored for them.
Setting AVIOContext for AVFMT_NOFILE formats now triggers a warning
from libavformat (and triggered an error for a while), so add a check
to avoid setting AVIOContext when not necessary.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33695 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix printing of subtitle type, the wrong index was used to look up the
type.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33664 b3059339-0415-0410-9bf9-f77b7e298cf2
Acording to the ASF documentation, the play duration is zero
if the preroll value is greater than the play duration.
The new way of determining it (suggested by reimar) prevents
overflows as well.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33492 b3059339-0415-0410-9bf9-f77b7e298cf2
According to the ASF documentation,
MF_PD_ASF_FILEPROPERTIES_PREROLL (preroll) is UINT64. Fix type
mentioned in comment.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33484 b3059339-0415-0410-9bf9-f77b7e298cf2
If the played file has per-track titles for audio and subtitles show
those on the OSD when switching tracks. This changes the OSD message
from 'Audio: (2) eng' to 'Audio: (2) eng ("Director's commentary")'.
Move the buffer storing audio data ready to be fed to the audio output
driver from the audio decoder object to the AO object. This will help
encoding code deal with end of input, and may also be useful to
improve other general gapless audio behavior (as AOs which do not
accept chunks smaller than a certain size may keep them in the buffer
while the decoder changes).
Less data may be dropped now when changing audio filters or switching
timeline parts.
Selecting the colorspace to output from a decoder is done in the
function mpcodecs_config_vo(). Add a new version of this function,
mpcodecs_config_vo2(), that allows the decoder to specify a list of
candidate colorspaces instead of always using a hardcoded list
specified in the codecs.conf entry. If the codecs.conf entry has any
"out" lines then those still take priority and the decoder-provided
list (if any) is ignored. Make vd_ffmpeg provide a list of the
colorspaces it's willing to output. Remove "out" lines from most
entries for libavcodec video decoders in codecs.conf, so that the
automatic values are now used instead.
sd_ass relies on there being a zero byte after packet data. However
the packet allocation routines special-cased data length 0 and left
the data pointer as NULL in that case. This could cause a crash in
sd_ass if there was an empty subtitle packet. Change the allocation
routines to stop special-casing empty data and always allocate
padding. Empty packets are not so common that special casing them
would be a worthwhile optimization.
Also fix resize_demux_packet() to use MP_INPUT_BUFFER_PADDING SIZE as
the padding size, instead of a hardcoded value of 8.
