must be zero when use a dvdsub_id greater than max sub id from demux.
To remove the implicit assumption, make it up here.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24962 b3059339-0415-0410-9bf9-f77b7e298cf2
which complains about 'unterminated string or character constant'.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24961 b3059339-0415-0410-9bf9-f77b7e298cf2
the most likely result is a NULL dereference which isn't much worse.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24946 b3059339-0415-0410-9bf9-f77b7e298cf2
r24897: Consistently use \- in option names.
r24909: Escape some more '-' where appropriate.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24944 b3059339-0415-0410-9bf9-f77b7e298cf2
VIDEO_MPEG12 and picture_coding_type==I_FRAME; fixes seeking in avi streams
with MPEG1/2 video
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24940 b3059339-0415-0410-9bf9-f77b7e298cf2
write the map also for ordinary mpeg audio and video streams (wasting space)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24936 b3059339-0415-0410-9bf9-f77b7e298cf2
Substract the delay caused by filter buffering when calculating
currently playing audio position. This matters for af_scaletempo which
buffers significant and varying amounts of data. For other current
filters the effect is normally insignificant.
Instead of the old time-based filter delay field (which was ignored)
this version stores the per-filter delay in units of bytes input read
without corresponding output. This allows the current scaletempo
behavior where other filters before and after it can see the same
nominal samplerate even though the real duration of the data varies;
in this case the other filters can not know the delay they're causing
in terms of real time.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24928 b3059339-0415-0410-9bf9-f77b7e298cf2
Make internal functions static and remove leading '_' from some
function names. Cast pointers to compatible 8-bit pointer types
instead of ints when calculating their difference.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24927 b3059339-0415-0410-9bf9-f77b7e298cf2
The value of the "speed" suboption was not initialized before calling
subopt_parse(). If the command line had suboptions but "speed" was not
one of them then the code accessed an uninitialized pointer and
possibly crashed. Fixed by initializing the option value.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24926 b3059339-0415-0410-9bf9-f77b7e298cf2
Because of a missing *num_channels factor the filter copied audio from
an incorrect position. This mixed up the channel order and hurt audio
quality even if the channels had identical content.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24925 b3059339-0415-0410-9bf9-f77b7e298cf2
The samplerate requested from the audio out depended on the initial
value of playback speed on startup. Changing speed later at runtime
does not affect output samplerate; audio is always resampled instead.
Change the init code so that speed does not affect the samplerate
requested and behavior matches what you'd get by starting the file
with speed 1 and then changing it.
The previous behavior could be desirable when using a constant speed
value. However it means that if you start with a low speed and later
switch to normal speed then audio will be resampled to a low output
frequency. You can still use -srate to explicitly select the frequency.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24923 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove the following arguments as redundant: in_channels, in_format,
out_minsize, out_maxsize. The first two always equal fields of the
sh_audio_t struct given as the first argument to the function. The
last two are unused after the allocation of sh_audio->a_out_buffer
was changed to be done on demand.
After the out_minsize and out_maxsize arguments are removed the
function preinit_audio_filters() is identical to init_audio_filters(),
so remove it and use the latter instead.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24922 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove the code allocating sh_audio->a_out_buffer from
init_audio_filters() and let the buffer be allocated by the new
dynamic allocation code.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24921 b3059339-0415-0410-9bf9-f77b7e298cf2
Rewrite decode_audio to better deal with filters that handle input in
large blocks. It now always places output in sh_audio->a_out_buffer
(which was always given as a parameter before) and reallocates the
buffer if needed. After the changes filters can return arbitrarily
large blocks of data without some of it being lost. The new version
also allows simplifying some code.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24920 b3059339-0415-0410-9bf9-f77b7e298cf2
sh_video_t: void *video_out: only assigned to, never read. Remove.
sh_video_t: void *vfilter: change type to struct vf_instance_s *
sh_audio_t: void *afilter: change type to struct af_stream_s *
The latter two never hold different types so there's no reason to use
void *. Maybe they were originally defined that way because the
option of using pointers to incomplete struct types was missed (the
typedefs vf_instance_t and af_stream_t would require extra headers)?
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24919 b3059339-0415-0410-9bf9-f77b7e298cf2
Most of these functions aren't even used in the same translation unit.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24918 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove the mul/cancel/gcd functions and some related code. Use ff_gcd
instead of the removed af_gcd in af_resample.c.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24917 b3059339-0415-0410-9bf9-f77b7e298cf2
Change the audio filters to use a double instead of rationals for the
ratio of output to input size. The rationals could overflow when
calculating the overall ratio of a filter chain and gave no real
advantage compared to doubles.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24916 b3059339-0415-0410-9bf9-f77b7e298cf2
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