similar to: flac 1.3.0 NOT lossless

Okay, I was thinking it may have been something to do with the header, but I wasn't sure how to verify that. thanks guys. On Mon, Apr 15, 2013 at 3:00 PM, <flac-dev-request at xiph.org> wrote: > Send flac-dev mailing list submissions to > flac-dev at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

wow, thanks for the quick reply! > > My suggestion: try out the ffmpeg flac thingie since that speed > improvement is unreal in compare to the normal flac :) > > > Stefan Thanks for the suggestion - I'm reading the ffmpeg changelog and feature list right now, and it appears that it only supports flac decoding not encoding and decoding - is this because the documentation is

Tom Sparks wrote: > witch is the best lossless video codec that ffmpeg/ffmpeg2theora supports? FT doesn't support lossless at all. FFMPEG supports HuffYUV and YUV4MPEG (uncompressed). > Is it worth using dirac yet? It isn't available for now: format not frozen, no support in MP/ME, in FFMPEG it doesn't work :-( I doubt that Theora-lossless would be doable/useful ... maybe

I mean that the first statement [Subset streams must use one of 192/576/1152/2304/4608/256/512/1024/2048/4096 (and 8192/16384 if the sample rate is >48kHz).] published on https://www.xiph.org/flac/documentation_tools_flac.html#flac_options_blocksize page IS NOT EQUAL to second statement [The blocksize bits in the frame header must be 0001-1110. The blocksize must be <=16384; if the sample

Erik de Castro Lopo wrote: > > http://wootangent.net/~lsd/blah/snippet6.wav > > Great, thanks! Confirmed the problem here. Will look at it ASAP. Same problem with flac 1.2.1. Interesting! Erik -- ---------------------------------------------------------------------- Erik de Castro Lopo http://www.mega-nerd.com/

I'm misleading about FLAC subset constraints... Please help me understand exact FLAC subset limitation.

Yes, this is such a case. However, implementing this in a future encoder/decoder would break compatibility with most (likely all) existing decoders, and only in some very, very rare cases where the material is such that the encoder chooses to use negative shifts, which makes it even harder to troubleshoot. Furthermore, as this can only be used in very rare cases, there is no benefit from allowing

the audio was also 192,000khz sample rate, forgot to mention that, adn here are the audio files, the original, the flac, and the decoded from flac. the archive is 7zip, Idk where to upload it so I'll just send it to depositfiles. http://depositfiles.com/files/90anghniw -------------- next part -------------- An HTML attachment was scrubbed... URL:

Cristian Adam wrote: > On Wed, Feb 25, 2009 at 2:23 PM, Brian Willoughby <brianw at sounds.wa.com>wrote: > >> A better suggestion might be to start with libFLAC, optimize as >> needed, and then submit the optimizations back to the FLAC project >> where they will be more widely useful. >> >> But that's just my opinion. >> > > I agree. This

I wish to transcode an AVI from 29fps to 15fps theora. I have tried many combinations. The nearest I have so far is to: 1) ffmpeg -i input.avi -f avi -vcodec rawvideo -acodec pcm_s16le -r 15 -ar 22050 -ac 1 intermediate.avi then 2) ./ffmpeg2theora-0.15.linux.bin -f avi -c1 -H 22050 -o out.ogg intermediate.avi This gives me an output I want, however, the intermediate uncompressed stage

Hi, On 04/24/2010 01:26 PM, Silvia Pfeiffer wrote: > Did anybody ever reply to this, I wonder? in case nodbody did, > On Tue, Aug 11, 2009 at 7:30 AM, Frank Barchard <fbarchard at google.com> wrote: >> I've got a video that causes ffmpeg2theora (nightly) to crash. How/who do I >> go about getting you guys the source video. >> At times in the past this video

I've got a video that causes ffmpeg2theora (nightly) to crash. How/who do I go about getting you guys the source video. At times in the past this video encoded, but I had wanted about 2 mbit/s and it came out 200 kbit/s, and very low quality. It encodes okay with ffmpeg, but the reason I'm wanting to try ffmpeg2theora is that we see AV sync issues, and I want to know if its in the tool

I created some test samples and transcoded to FDK AAC and libopus at fairly low bitrates - I cannot recreate what bothered me about Opus & noisy music previously. It also seems I cannot tease ffmpeg into encoding FDK's AAC with VBR. As it stands, Opus clearly wins in this scenario.* Q: Is it possible to stream in variable bitrate? * ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac

Thanks for the info, Dave. Speed is a very important feature, but there might be some risk choosing the FFmpeg decoder. They've had trouble with their encoder in the past, which tells me it's possible your users might one day run into a valid FLAC that the FFmpeg decoder won't handle correctly. A better suggestion might be to start with libFLAC, optimize as needed, and

Many Thanks for your answer. Actually quite a few 'ifs', 'maybes' or 'shoulds'... ;-) I have tried most of them (except liquidsoap). However, even the mentioned apps like RadioDJ or Mixxx do NOT support meta data changes via the encoder (they all use the admin interface)! And as you correctly pointed out: none of the reference encoders support this. Not even the FLAC, OGG

?hel kenal p?eval (neljap?ev, 14. m?rts 2013 19:02:35) kirjutas Declan Kelly: > On Wed, Mar 13, 2013 at 10:06:51PM -0400, benski at winamp.com wrote: > > Flake is a completely independent codebase. When I used it years ago, I > > remember it being not only better compression but significantly faster as > > well. I believe some of the techniques used in libflake were added to

I don't think you guys should worry too much about messing up old decoders, but no matter what you choose to do FLAC MUST REMAIN LOSSLESS. On Thu, Mar 14, 2013 at 5:06 PM, <flac-dev-request at xiph.org> wrote: > Send flac-dev mailing list submissions to > flac-dev at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit >

Op 07-02-12 19:50, Ralph Giles schreef: > Basically the audio is chopped into a blocks and each block is coded > either uncompressed, as a constant value (good for silence), or with > linear predictive coding plus a rice-coded residual. I don't know how > the encoder decides where to put the block boundaries. AFAIK, FLAC uses a fixed block length so block boundaries are just put

CentOs zoneminder users, I have been trying to install zoneminder on Centos 5.3 (2.6.18-128.1.16.el5xen) and have hit a brick wall with ffmpeg which zoneminder has as a dependancy. There were no rpm's in centos or rpmforge so I have followed the instructions on the zoneminder website for a CentOs install. I have posted a note on the zoneminder list, but have not been able to get any takers

Hi, and thank to you all for this great codec ! I have this bug on Debian Lenny with compiled packages of last svn versions of ffmpeg and libtheora. This seems to append only on the amd64 arch. Here is a valgrind log : pre-barreau at augustins:~/video$ export LD_LIBRARY_PATH=/usr/local/lib pre-barreau at augustins:~/video$ valgrind /home/pub/apps/ffmpeg_dev/ffmpeg_svn/ffmpeg -i