dec_audio.c init_audio_codec() would in one case print
"ADecoder init failed :(\n" and return failure. Its only caller
init_best_audio_codec() printed exactly the same message if the
returned result was failure. Change the latter message to say
"Could not open audio decoder %s.\n" instead. Some of the
per-open-attempt messages are kind of value about their context; this
new message should make it more clear where the attempt to open one
specific codec ends.
The init() method in ad_ffmpeg tries to decode some audio data after
opening the libavcodec decoder; however the method returned success
even if this part failed. Change it to return failure instead,
indicating that the codec could not be successfully opened.
This improves behavior at least with some AAC files, for which the
libavcodec decoder can be successfully initialized but decoding
packets always fails. Before the audio would be decoded with
libavcodec, producing only a constant stream of errors; after this
commit audio decoder initialization falls back to FAAD (if available)
which works for these samples.
Setup default speex modes, allows decoding of speex in flv which does
not contain a header packet.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33327 b3059339-0415-0410-9bf9-f77b7e298cf2
Allow DR to work with reget_buffer when no buffer_hints are set.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33287 b3059339-0415-0410-9bf9-f77b7e298cf2
Allow reget_buffer to somewhat work after direct rendering failure.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33286 b3059339-0415-0410-9bf9-f77b7e298cf2
Do not try to set up the FS segment when using quicktime to decode,
it makes no sense to do that.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33259 b3059339-0415-0410-9bf9-f77b7e298cf2
Many video filters failed to calculate or even just pass through pts
values for their output frames. Fix this, and also make the two
remaining filters that called vf_next_put_image() twice for the same
input frame (vf_softpulldown, vf_telecine) use vf_queue_frame() so
that e.g. framestepping properly sees both frames.
Changed filters: vf_bmovl, vf_detc, vf_divtc, vf_filmdint, vf_ivtc,
vf_lavc, vf_phase, vf_pullup, vf_softpulldown, vf_telecine, vf_tile,
vf_tinterlace.
Update various code to use newer alternatives instead of deprecated
functions/fields that are being dropped at libav API bump. An
exception is avcodec_thread_init() which is being dropped even though
it's still _necessary_ with fairly recent libav versions, so there's
no good alternative which would work with both those recent versions
and latest libavcodec. I think there are grounds to consider the drop
premature and revert it for now; if that doesn't happen I'll add a
version-test #if check around it later.
Change MP_IMGFLAG_ACCEPT_STRIDE to MP_IMGFLAG_ACCEPT_ALIGNED_STRIDE in
get_buffer() as various FFmpeg assembly routines assume aligned input.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33097 b3059339-0415-0410-9bf9-f77b7e298cf2
Support 'lpcm' in mov files, has audible (clipping?) artefacts on some
systems.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33075 b3059339-0415-0410-9bf9-f77b7e298cf2
Support 32bit big endian float pcm in aiff.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33076 b3059339-0415-0410-9bf9-f77b7e298cf2
Audio with all codec tags other than 0x2000 was byte-swapped, while
only "dnet" should be.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33028 b3059339-0415-0410-9bf9-f77b7e298cf2
Delete mp3lib which has been the default mp3 decoder until now. In
addition to being an unnecessary embedded library it now fails to
compile correctly with the new gcc-4.6, producing noise.
After the deletion the default decoder priority for mp3 will be first
libmpg123 (a newer version of the code that mp3lib was based on) if
available, then ffmp3float which should be available in all normal
compiles. I think that some tweaking may be required as these decoder
alternatives get wider testing, but any problems should be solvable
and there should be no need for mp3lib.
Recent ffmpeg-mt versions changed the API for setting the number of
decoding threads to use (I'm not sure whether dropping backwards
compatibility was intentional or not). As a result only one thread was
used. Make the thread setting compatible with the new API to restore
proper multithreaded decoding.
Use color type getter instead of direct access to private member.
Using the getter is mandatory since recent libpng 1.5 release.
Patch by Gianluigi Tiesi (mplayer - netfarm it)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32840 b3059339-0415-0410-9bf9-f77b7e298cf2
af_lavcac3enc: use old SampleFormat names without AV_ prefix, the
latter were only added in 2010-11
vd_ffmpeg: add ifdef around CODEC_ID_LAGARITH use
demux_real: use ffmpeg_files/intreadwrite.h
stream/http.c, stream/realrtsp/real.c: define AV_BASE64_SIZE macro for
old libavutil versions lacking it
One of two alternative code parts passing codec extradata to
libavcodec didn't add the buffer padding that libavcodec requires,
resulting in invalid reads beoynd allocated memory area. Fix.
There were multiple files specific to Zoran support, and they also
depended on internal FFmpeg headers (so it would probably have been
hard to get them to compile now even if you tried). It's obsolete now,
so just drop the whole mess.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32741 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove AAC/FAAD2 installation instructions.
There is nothing special about building and installing FAAD2, so there is
no longer a need to keep maintaining instructions for it.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32742 b3059339-0415-0410-9bf9-f77b7e298cf2
* sub:
sub/OSD: move some related files to sub/
subtitles: options: enable -ass by default
subtitles: change default libass rendering style
demux_mkv, chapters: change millisecond arithmetic to ns
cleanup: rename ass_* functions to mp_ass_*
subs: use correct font aspect ratio for libass + converted subs
cleanup: some random minor code simplification and cleanup
vf_vo: fix EOSD change detection bug
sd_ass: remove subreader use, support plaintext markup
subtitles: style support for common SubRip tags and MicroDVD
core: ordered chapters: fix bad subtitle parameter
subs/demux: don't try to enable sub track when creating it
subtitles/demux: store duration instead of endpts in demux packets
subtitles: add framework for subtitle decoders
options: add special -leak-report option
subtitles: remove code trying to handle text subs with libavcodec
cleanup: move MP_NOPTS_VALUE definition to mpcommon.h
subtitles: move global ass_track to struct osd_state
core: move most mpcommon.c contents to mplayer.c
core: move global "subdata" and "vo_sub_last" to mpctx
subtitles: remove sub_last_pts hack
options: move -noconfig to option struct, simplify
The various ass_* functions were created when libass was part of the
MPlayer tree and the distinction between MPlayer-specific and other
functions was less clear. Now that libass is a clearly separate
library, using the same ass_* namespace for player functions is ugly.
Rename the functions to use mp_ass_ prefix instead.
Rendering of ASS subtitles tries to be bug compatible with VSFilter
and stretches fonts when the video is anamorphic (some scripts try to
compensate for this VSFilter behavior, so trying to render them
"correctly" would give the wrong result). However this behavior is not
appropriate for subtitles we converted to ASS format ourselves for
libass rendering, as they certainly don't have VSFilter bug
workarounds. Change the code to use different behavior for "native"
ASS tracks and converted ones. It's questionable whether the
VSFilter-compatible behavior is appropriate for external .ass files
either, as there could be anamorphic and non-anamorphic versions of
the same video and the bug-compatible behavior can only be correct for
one alternative at most. However it's probably better to keep it as a
default at least, so that extracting a muxed subtitle track and using
that does not give behavior different from the original muxed one.
The aspect ratio setting is per ASS_Renderer, and changing it resets
libass caches. For that reason this commit adds separate renderer
instances to use for the "correct" and "VSFilter bug compatible"
cases.
vf_vo had code setting its prev_visibility variable correctly, then a
line that overrode the value just set with an incorrect one. Remove
the wrong extra line. As a result of the bug the "contents changed"
indicator wasn't forced to true when switching from a subtitle track
to "no track" and then back. A visible effect was at least that a
currently visible static subtitle disappeared when doing that switch
back and forth.
Make "-lavdopts threads=0" mean an autodetected number of threads, and
make that the default value of the option. Also increase the upper
limit of the option from 8 to 16. Add new file osdep/numcores.c which
tries to determine the number of cores available on the machine.
numcores.c is based (heavily modified) on public domain numcpus.c by
Philip Willoughby <pgw99@doc.ic.ac.uk>, downloaded from
http://csgsoft.doc.ic.ac.uk/numcpus/
Improve speex codec pts handling, make audio timestamps work reasonably
even with the native demuxer as long as seeking is not done.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32704 b3059339-0415-0410-9bf9-f77b7e298cf2
Change direct rendering buffer allocation code to treat non-ref frames
like B-frames even if has_b_frames is not set and they are indeed not
B-frames (no reordering). Treating it as an I/P frame would violate
the assumptions of MPlayer's buffering system, which thinks only the
latest previous I/P frame is needed (in addition to one possibly being
decoded). In this case the previous I/P frame will still be needed in
the future, not the non-ref frame being decoded now.
This happens with flv files, as in bug #1079, and this change fixes that
corruption.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32700 b3059339-0415-0410-9bf9-f77b7e298cf2
Otherwise we might think the filter chain/vo is ready when it
actually is not, leading to a crash.
Fixes crash part of bug 1156.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32690 b3059339-0415-0410-9bf9-f77b7e298cf2
Use the value of the OutputSamplingFrequency element instead of the
SamplingFrequency element as the "container samplerate". In most cases
this only removes a warning, as those typically differ for SBR AAC
files and there was already a special case detecting this in
ad_ffmpeg.
The implementation adds a new "container_out_samplerate" field to the
sh_audio struct. Reusing the existing "samplerate" field and the
equivalent inside the 'wf' struct and just setting those to the new
value instead would probably work (at least I'm not aware of any codec
that would need the original SamplingFrequency for initialization).
However using a separate field also avoids some ugliness: the 'wf'
struct may not exist (though most demuxers create it), and the
'samplerate' field is overwritten to reflect the final value decided
by codec when decoding is first initialized.
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.
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
