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
Replace memcpy with memmove since at least src==dst is possible.
Fixes another issue that is part of bug #1280.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32697 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace malloc+memset with calloc and use sizeof(*variable).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32694 b3059339-0415-0410-9bf9-f77b7e298cf2
Replace hard-coded number for loop limits for array index by
the define used in the array declaration.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32695 b3059339-0415-0410-9bf9-f77b7e298cf2
Add memset to avoid using uninitialized data with sample in bug 1280.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32693 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix mp_check_mp3_header: it checked for a byte-swapped MP3-header
on little-endian, and on big-endian it would only accept a MP3-header
that would be valid when read in both directions.
The latter was the reason for bug 905, causing the PS demuxer to
claim files far too agressively (the MP3 check avoiding misdetection
as DV is not exactly a sane approach, but it mostly works).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32692 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
Move setup of sh_audio->format to a more appropriate place (in asfheader.c).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32684 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove pointless assignments that are already handled in new_sh_audio.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32685 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove useless assignment already done in new_sh_video.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32686 b3059339-0415-0410-9bf9-f77b7e298cf2
Use FFMAX for slightly better readability.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32687 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix some unaligned writes and avoid some (incorrect due to alignment) casts.
Might also fix bug #371.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32683 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix TS files with MP4 ES AAC descriptor to be correctly recognized
as AAC and not AAC in LATM.
This fixes playback of http://samples.mplayerhq.hu/A-codecs/AAC/freetv_aac_latm.ts,
actual LATM samples seem unaffected.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32667 b3059339-0415-0410-9bf9-f77b7e298cf2
Add R_OVL_SHIFT to all R280 devices. Only actually confirmed for two.
Fixes bug #1826.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32663 b3059339-0415-0410-9bf9-f77b7e298cf2
Restore big-endian support removed in thoughtless upstream merge
at r23062.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32664 b3059339-0415-0410-9bf9-f77b7e298cf2
100l, fix vidix compilation on big-endian
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32665 b3059339-0415-0410-9bf9-f77b7e298cf2
Add horrible hack to make xvidix work on big-endian.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32666 b3059339-0415-0410-9bf9-f77b7e298cf2
Fixes:
ffmpeg://rtsp://stream.diffusion.ens.fr/2008_10_03_albarede.mov
and other X-SV3V-ES rtsp streams opened with ffmpeg://
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32660 b3059339-0415-0410-9bf9-f77b7e298cf2
Mark input-only buffers as const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32652 b3059339-0415-0410-9bf9-f77b7e298cf2
Use uint8_t type instead of unsigned char.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32653 b3059339-0415-0410-9bf9-f77b7e298cf2
Mark input buffer that is never modified as const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32654 b3059339-0415-0410-9bf9-f77b7e298cf2
Mark input-only buffer as const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32655 b3059339-0415-0410-9bf9-f77b7e298cf2
Add support for decoding Avid DNxHD through the QuickTime component.
This is needed for the 10-bit variant which the FFmpeg decoder does not
support (unfortunately both use the same FourCC).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32649 b3059339-0415-0410-9bf9-f77b7e298cf2
Bump codecs.conf version.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32650 b3059339-0415-0410-9bf9-f77b7e298cf2
change dnxhd to qtdnxhd. consistant with all other quicktime decoders
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32658 b3059339-0415-0410-9bf9-f77b7e298cf2
build_afilter_chain is not safe to use directly, thus make it
static and instead use reinit_audio_chain which should have
better error handling.
Fixes a crash with -af hrtf and changing speed, audio will
still stop playing though.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32648 b3059339-0415-0410-9bf9-f77b7e298cf2
Set the option value to disabled, not enabled, if the functionality is
not available at all. Without this, -font and -subfont do not work
when using -ass.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32636 b3059339-0415-0410-9bf9-f77b7e298cf2
Make it seek back to the stream->start_pos position instead of 0 in
that case.
Fixes bug 1790.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32635 b3059339-0415-0410-9bf9-f77b7e298cf2
"Authorization" header is for the destination server URL, even through
a proxy.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32633 b3059339-0415-0410-9bf9-f77b7e298cf2
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32630 b3059339-0415-0410-9bf9-f77b7e298cf2
Fix crash on path without directories.
Regression introduced in r32630. Patch by Yuriy Kaminskiy yumkam at mail ru.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32631 b3059339-0415-0410-9bf9-f77b7e298cf2
Handle correctly paths with mixed '/' and '\' in it.
Patch by Yuriy Kaminskiy (yumkam at mail ru)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32632 b3059339-0415-0410-9bf9-f77b7e298cf2
Handle ':' on systems with DOS paths: it allows paths like C:foo.avi.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@32642 b3059339-0415-0410-9bf9-f77b7e298cf2
The wrong variable was used as a function argument, and as a result
the code modified the usage_count field of non-refcounted mp_image
types. This error did not have any effect on visible behavior as no
code cares about the field value in the affected case.
-novideo is the right way to disable video, and should also work in
more cases now that lavf is used as the default demuxer for more formats
(like AVI; internal AVI demuxer fails with -novideo).
Also change the man page description of -novideo a bit to make it
sound less negative about the chances of the option working.
Some Matroska files have inaccurate ordered chapter endpoints, and so
parts where one chapter should end and the next begin at the same
timestamp were not merged. This resulted in an unnecessary seek over a
minimal distance. Add a heuristic to merge parts with a minimal gap or
overlap between them.
Based on patch by Hector Martin <hector@marcansoft.com>.
ogg/ogm demuxers can give first audio packets without timestamp after
a seek. Due to some backwards compatibility code this results in the
sync code getting audio timestamp 0. In this case a lot of audio was
dropped unnecessarily when seeking to a position later in the file, as
the code saw audio starting from 0, video from something larger.
Make the code more robust in two ways. First, add a special case to
not try syncing if we get audio timestamp <= 0 (hopefully there aren't
many files where we'd really get audio starting from 0 and video from
a later timestamp). Second, when throwing audio away, make the code
recalculate from scratch the amount of bytes that still need to be
thrown away after every decode call. This limits the amount of damage
initial too-small timestamps can do, as the code will see the better
timestamps after a while.
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 definitions for DisplayUnit, OutputSamplingFrequency and
FileDescription in matroska.py. Regenerate the C template files to
allow using all current definitions in code.
