FFmpeg's AAC decoder is much faster than libfaad2. The only known
exception is libfaad2 compiled in fixed-point mode on systems with
slow FPUs. Now that LATM support in FFmpeg is complete, FFmpeg's AAC
decoder has a similar feature set as libfaad2. This leaves no reason
not to use FFmpeg by default.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32605 b3059339-0415-0410-9bf9-f77b7e298cf2
Playing AVI files containing B-frames with demux_lavf printed two
"decreasing pts" info messages at the start of the file. We know the
timestamps from AVI won't be valid pts, so add a demuxer field to
convey that information to the timing code and make that not even try
to use the timestamps as valid pts.
lavf demuxers are mostly better and receive more maintenance,
therefore it makes sense to prefer them in most cases. Change the
"preferred" logic from listing all formats for which lavf is preferred
to listing exceptions for which it isn't. Currently there are 3
exceptions: Matroska, FLAC and RealMedia (.rm).
Add code to enforce matching pts with video when (re)starting the
audio stream, by either cutting away the first samples or inserting
silence at the beginning. New option -noinitial-audio-sync can be used
to disable this and return to old behavior.
demuxer_get_current_chapter() accessed sh_video/sh_audio pts fields to
determine playback position. demux layer shouldn't access those and
the values used weren't quite correct anyway. Give the playback
position as a parameter to the demux layer function instead. Also
change the top-level get_current_chapter() to use get_current_time()
in the timeline case where it didn't refer to demux layer.
If the option is enabled and all audio has been buffered to the AO,
then the player will move to the next file without waiting for the
buffered audio to drain, while leaving the AO initialized. If the
playback of the next file starts quickly enough (before the AO buffer
empties) then it should continue writing audio to the same AO with no
gap in between.
At least with PCM it's possible to get an audio stream that doesn't
end at a multiple of whole sample per channel. At least ao_alsa
refuses to accept that part of input, and so EOF detection in
fill_audio_out_buffers didn't trigger until the 0.04 second sanity
check (as there "was still audio not sent to AO left"). Change the
logic to detect EOF if there's less than one sample per channel of
unsent data left.
"libavformat file format detected" wasn't a very useful message due to
the many file formats supported to libavformat. Change the message so
that for demux_lavf it says something like
"Detected file format: QuickTime/MPEG-4/Motion JPEG 2000 format (libavformat)"
(using long name from FFmpeg), and for non-lavf something like
"Detected file format: Matroska".
When file format detection failed the output only said
"Exiting... (End of file)" after "Playing <file>." (or possibly error
messages triggered by format-specific check functions in between). Add
an explicit "Failed to recognize file format." error message.
The code choosing the demuxer to use only printed an error if given an
unknown demuxer name, then continued with default demuxer selection.
Change it to abort instead. This feels like more sensible behavior.
Also there's no fallback to autodetection in the case where the
demuxer name is recognized but the demuxer fails to open the file
either.
Seeking in MPEG files with pts resets could fail completely, as it was
always done by timestamps and those of course don't unambiguously
specify a file position in such files. Add basic functionality for
byte-based seeking and playback position reporting, and decide whether
to use that functionality based on a simple heuristic (could be
improved).
When -alang / -slang was specified the numerically first matching
track (if any) was always chosen. This meant that specifying "-alang
eng" could change the track choice even if all tracks were in English,
because now the default flag of tracks was ignored. Change the logic
to take the default flag into account as a secondary sorting key.
The code also accepted prefix matches, so that "-slang g" would match
track language "ger". I think that was not intentional. Change it to
require exact matches.
The Cue entries in typical Matroska files have information for the
video track only. This caused seeks to fail when playing with
-novideo, as demux_mkv tried to use audio track index entries then.
Add a fallback case that uses any index entries without caring what
track they're for if there are no entries specific to the track we're
interested in.
Relative seeks didn't add the current position as they should. Fix.
Note that this had no effect in normal playback case even if the file
had no index, because the "accurate_seek" logic at higher level would
convert all commands to absolute seeks before calling demuxer level.
Various code referred to "mpctx->demuxer" where it should really have
referred to the one used for audio/subtitles in case those differ. Fix
by using "mpctx->d_audio->demuxer" etc instead. Disable the copying of
streams in demux_demuxers; that was a partial workaround for things
referring to the main demuxer (and it wasn't enough anyway).
This fixes, among other things, switching audio tracks within the file
specified by -audiofile.
demux_demuxers doesn't run the normal demuxer.c initialization for new
demuxers. Initialize stream_pts separately (it won't ever be changed
with the current implementation). This at least avoids other code
assuming it was set properly.
Move functions to query current playback position, percentage position
and total video length from from the demuxer layer to top level. The
functions need access to playback state that doesn't belong on the
demuxing level. Make the new functions more capable and simplify some
code that can now rely on them. This fixes some errors in displayed in
OSD and slave mode information when using timeline (ordered chapters).
Use IMGFMT to compare instead of PIX_FMT to avoid issues with the
"JPEG" formats like PIX_FMT_YUVJ422P.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32594 b3059339-0415-0410-9bf9-f77b7e298cf2
For FFmpeg codecs YV12 should always be in the supported format
lists if I420 is.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32593 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix dvd:// fill_buffer function to actually write into the specified
buffer and check that the buffer is sufficiently large.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32591 b3059339-0415-0410-9bf9-f77b7e298cf2
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32592 b3059339-0415-0410-9bf9-f77b7e298cf2
Ensure we queue all subtitle packets before demuxing the next video
packet.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32587 b3059339-0415-0410-9bf9-f77b7e298cf2
fflatm seems to be working, whereas external faad does not work at all
with latm (internal does).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32585 b3059339-0415-0410-9bf9-f77b7e298cf2
