Can be compiled without support for compressing the audio cache with FLAC
Supports more audio formats in matroska
RGB24 output no longer has swapped colors if the video is converted to it for output (there still seems to be some bugs lurking when conversion is done with libswscale)
Fixed an access violation on close when no audio is opened (introduced in 1.10)
Updated FFmpeg to rev 10423
1.10
The audio cache compression level is now ignored if the source isn't 16bit and the raw format is used instead
FLAC is now actually initialized properly so the cache actually works for files that aren't stereo (16bit limit still applies)
Now uses proper callbacks for FLAC so it works with larger than 2GB files
Doesn't (over)write the video cache with an empty one in certain cases when avformat is used for the source
1.9
Added the possibility to compress the audio cache with FLAC (currently only works with 16bit audio)
Added another planar YUV 4:2:0 format to the supported output formats (fixes certain mov files)
Less memory is now allocated on the stack which makes av_find_stream_info() work for all files (fixes certain mov files)
Updated FFmpeg to rev 10186
1.8
Updated FFmpeg to rev 10141
1.7
Updated FFmpeg
Fixed error with mkv for codecs without codec private data and the first packet doesn't belong to them
1.6
Fixed ac3 and other formats stored in mkv
Skip unnecessary seeking when index information already exists (gif file opening only 3/4 broken now)
Throws an error when the selected audio/video track has no frames/samples
1.5
Fixed a bug that made avformat opened files only return audio if only the audio cache needed to be created
Rejects more corrupt cache files
Fixed crash when a 0 byte audio cache file is present
Improved framerate guessing for avformat which now takes takes the duration of the first frame into account
Fixed a bug introduced in 1.4 that would make the number of reported frames too high for files opened with avformat
Fixed mpeg2 and probably some other formats stored in mkv
Fixed issues with large mkv files and large audio cache files
FFmpeg is now compiled with liba52 and faad2
1.4
Uses the average framerate for mkv files
Naming scheme of cache files changed to prevent confusion with the default names in files with multiple tracks of the same type
Use mmx optimizations in swscaler when possible
Now uses normal windows linebreaks in all files
Removed FFAudioSource
Merged FFVideoSource and FFAudioRefSource into FFmpegSource
Added postprocessing with libpostproc in FFmpegSource and separately in FFPP
1.3
Compiled against ffmpeg rev9620
Added FFAudioRefSource
Added FFAudioSource (has big issues)
Renamed FFmpegSource to FFVideoSource
Adjusted seeking in the forward direction to only be done if the requested frame is more than 10 frames away to reduce unnecessary seeking
Now outputs the last frame properly when there are decoding delays
1.2
Compiled against ffmpeg rev9451
Somewhat cleaner source code
Linear access in addition to a few other modes of seeking can now be forced
Can now save the index information to a file which makes subsequent file opening fast
No longer skips indexing for any format
1.1
Skip indexing for avi
Prefix all error messages with the plugin name
Can write v2 timecodes to a file
Fixed reported framerate
Compatibility - Video
AVI, MKV, MP4, FLV: Frame accurate
WMV: Frame accurate(?) but avformat seems to pick keyframes relatively far away
OGM: Messed up first frame and seeking produces smearing with seekmode=3, incredibly slow seeking without, remux to mkv or avi
VOB: No rff flags applied, frame accurate?
MPG: Seeking seems to be off by one or two frames now and then
TS: don't bother
Compatibility - Audio
Sample accurate in all containers
Usage
FFmpegSource(string source, int vtrack = -1, int atrack = -2, string timecodes, bool vcache = true, string vcachefile, string acachefile, int accompression = -1, string pp, int ppquality = 6, int seekmode = 1)
Note that the audio cache will always be created when opening files with audio and that it will be huge since it stores all audio as raw signed 16 bit pcm.
FFPP(clip, string pp, int ppquality = 6)
Separate postprocessing which also seems to include a few simple deinterlacers
source:
Source file.
atrack & vtrack:
Track number as seen by the relevant demuxer, starts from 0, -1 means it will pick the first suitable track and -2 means it's disabled.
timecodes:
File to output timecodes to, if the file exists it will be overwritten.
vcache:
Write video indexing information to a file for later use. This setting does not control if The video index is loaded which it always is if it exists.
vcachefile & acachefile:
Specifies the file to store the index information or raw audio in, if nothing is specified (source).ffv(tracknumber)cache is used for video and (source).ffa(tracknumber)cache for audio.
accompression
Audio cache compression, -1 means raw audio and 0-8 uses FLAC with that compression level.
pp:
See the table below for a full description, an empty string means no processing. It is recommended to avoid the autoq option since it's currently unknown what effect it will have on the processing.
ppquality:
The quality to use for the specified postprocessing. Valid values are 0-6 where 0 usually means that no actual processing is done.
seekmode:
Force how seeking is handled, has no effect on matroska files which always use the equivalent of seekmode=1 0: linear access, the definition of slow but should make some formats "usable" 1: safe normal, bases seeking decisions on the reported keyframe positions 2: unsafe normal, same as 1 but no error will be thrown if the exact destination has to be guessed 3: aggressive, seek in the forward direction even if no closer keyframe is known to exist, only useful for testing and containers where avformat doesn't report keyframes properly
PP string format
Available postprocessing filters:
Filters Options
short long name short long option Description
* * a autoq CPU power dependent enabler
c chrom chrominance filtering enabled
y nochrom chrominance filtering disabled
n noluma luma filtering disabled
hb hdeblock (2 threshold) horizontal deblocking filter
1. difference factor: default=32, higher -> more deblocking
2. flatness threshold: default=39, lower -> more deblocking
the h & v deblocking filters share these
so you can't set different thresholds for h / v
vb vdeblock (2 threshold) vertical deblocking filter
ha hadeblock (2 threshold) horizontal deblocking filter
va vadeblock (2 threshold) vertical deblocking filter
h1 x1hdeblock experimental h deblock filter 1
v1 x1vdeblock experimental v deblock filter 1
dr dering deringing filter
al autolevels automatic brightness / contrast
f fullyrange stretch luminance to (0..255)
lb linblenddeint linear blend deinterlacer
li linipoldeint linear interpolating deinterlace
ci cubicipoldeint cubic interpolating deinterlacer
md mediandeint median deinterlacer
fd ffmpegdeint ffmpeg deinterlacer
l5 lowpass5 FIR lowpass deinterlacer
de default hb:a,vb:a,dr:a
fa fast h1:a,v1:a,dr:a
ac ha:a:128:7,va:a,dr:a
tn tmpnoise (3 threshold) temporal noise reducer
1. <= 2. <= 3. larger -> stronger filtering
fq forceQuant force quantizer
Usage:
[: