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.
Add the various decoders to codecs.conf and increase the maximum
number of buffered pts in stheader.h (apparently CrystalHD can have
very high decoder lag).
Patch by Philip Langdale, philipl overt org
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@33095 b3059339-0415-0410-9bf9-f77b7e298cf2
Add a framework for subtitle decoder modules that work more like
audio/video decoders do, and change libass rendering of demuxed
subtitles to use the new framework.
The old subtitle code is messy, with details specific to handling
particular subtitle types spread over high-level code. This should
make it easier to clean things up and fix some bugs/limitations.
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.
Allocates a new sh_sub and specifies language at the same time. I
don't think this is a particularly good API but add it for easier
compatibility with some svn changes.
The ad_functions structs are in rodata, mark some pointers to them
const.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@31606 b3059339-0415-0410-9bf9-f77b7e298cf2
via libavcodec.
Parsing can be done at the demuxer stage (currently disabled) or at the decoder
(ad_ffmpeg, enabled).
Should allow using the libavcodec AAC, DTS, ... decoders independent of container
format.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30130 b3059339-0415-0410-9bf9-f77b7e298cf2
and sub "stream headers".
One reason for this is to help avoid/make more obvious things like members with
the same function but different name (extradata vs. codecdata etc.), or members
with the same name but different semantics (pts for audio vs. pts for video).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@30128 b3059339-0415-0410-9bf9-f77b7e298cf2
Add a mode where libavcodec's reordered_opaque feature is used to
associate container packet timestamps with decoded frames. This should
improve behavior at least for MPEG files with interlaced h264; the
previous code does not cope well with the libavformat demuxer
producing two field packets with separate timestamps but the
libavcodec h264 decoder only producing a single output frame for those
two packets (so half the timestamps have no associated output frame).
The current libavformat mpeg demuxer seems to finally work with
interlaced h264 files and produce valid timestamps which are useful
with a mode like this.
By default MPlayer now selects between this new mode and the old one
automatically based on the number of timestamp problems they cause; by
default the new mode is used if both seem to work. The new option
-pts-association-mode can be used to force a particular mode. If
correct-pts mode is disabled this has no effect on timing.
Also remove the "EXPERIMENTAL" marker from the manpage description of
-correct-pts.
stheader.h used 'demuxer_t *' without including demuxer.h, causing
compilation to fail if demuxer.h had not been included first. Use
'struct demuxer *' instead so the type does not need to be defined.
Replace all USE_ prefixes by CONFIG_ prefixes to indicate
options which are configurable.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@27373 b3059339-0415-0410-9bf9-f77b7e298cf2
Remove the global variable mpvdec and move the video decoder pointer
to struct sh_video field vd_driver. Audio already had a similar
ad_driver field from before.
Remove some #include lines from headers, some of those removals made
possible by using incomplete struct types instead of typedefs. Include
mp_osd.h in mplayer.c and command.c after removing it from mp_core.h.
Remove "#ifdef USE_ASS" around some "struct ass_track_s *" fields
which will now compile even without ASS support.
Give sh_audio_t, sh_video_t and sh_sub_t which before had typedef
names only a matching struct name (without _t) too.
Change the a_streams, v_streams and s_streams demuxer fields from
void * to struct sh_audio *, struct sh_video * and struct sh_sub *.
Remove a now unnecessary cast from mplayer.c.
The first default track is chosen for playback if language-based selection
failes. Additionally, for audio tracks, the first one is chosen if there are
no default tracks at all.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@26301 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
by the user the video pipeline will use it as current aspect ratio
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24319 b3059339-0415-0410-9bf9-f77b7e298cf2
The value is related to overall a/v sync and is not used by audio
demuxers or decoders.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@22506 b3059339-0415-0410-9bf9-f77b7e298cf2
This mode has the following differences:
- Video timing is correct for streams with B frames, at least with some
demuxers.
- Video filters can modify frame timestamps and insert new frames, and
removing frames is handled better than before.
- Some things are known to break, it's not usable as the default yet.
Things should work as before when the -correct-pts option is not used.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@18922 b3059339-0415-0410-9bf9-f77b7e298cf2
(same as new_sh_audio()) instead of sh_audio_t *, use those to remove
the pointer from demuxer->a_streams[] before freeing it.
Some demuxers use free_sh_audio() to undo the creation of an
already-allocated audio stream in case of error. These uses were unsafe
since free_sh_audio() freed the data structure but left the pointer in
demuxer->a_streams[], leading to double free later in free_demuxer()
(and perhaps use of the freed stream before that, I didn't check).
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@18711 b3059339-0415-0410-9bf9-f77b7e298cf2
audio decoded with ad_libvorbis, ad_ffmpeg and ad_faad.
Patch by Uoti Urpala
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@18243 b3059339-0415-0410-9bf9-f77b7e298cf2