similar to: Streaming WAV / Converting WAV to a streamable format on the fly

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey Gang, I figured this is a frequently asked question, however it is not in the faq. i am wondering the lowest possible bit-rate to stream ogg at. is this an ogg limitation or is this dependent on the streaming server. Thanks in advance, (please reply to me too as i am not on the list.) <p>- -------------------------------------- Jeff

Okay, before I submit my patch to make libao produce sorta-streamable wav files, I want to know what these partial wav files do to various players. I've posted two sample wav files on my webserver: http://volsung.dhcp.asu.edu/~stan/infinite.wav http://volsung.dhcp.asu.edu/~stan/zero.wav The first uses a riff and data length of 0xFFFFFFFF (approximately infinite) and second uses a riff and

I''m having a lot of trouble stubbing out an association extension for some view tests. Example rails code modeling a music album: class Album < ActiveRecord::Base has_many :songs do def streamable find(:all, :conditions => ''streamable = 1'') end end end So for a given Album instance (say @album), I need to be able to stub both

Hi there, first mail to this list, I hope to not disturb with my question. I'm trying to encode a VOB file (from a DVD) with ffmpeg2theora, but it have a lot of subtitles (about 26) and I can't find the correct --audiostream id to encode the file in my language. I have tried with ids from 20 to 40 and so on and all times the audio kbps it's 0. mplayer and VLC can play the .vob file

Hi. I'm trying to encode some high sample rate music (88.2kHz) into flac. When trying this with flac 1.1.4 I get the following error: ##### keithr@eng:~$ work/flac-1.1.4/src/flac/flac Serenissima02.wav <snip copyright> Serenissima02.wav: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 Serenissima02.wav: WARNING: skipping unknown sub-chunk 'fact'

Martin Kos wrote: > Hi > > i have seen that the dbPowerAmp ripping and encoding software supports a > new so-called "FLAC uncompressed" format, e.g. > > http://www.audiostream.com/content/dbpoweramps-flac-lossless-uncompressed-wish-come-true Wow, check this comment:

Le dimanche 30 janvier 2011 03:54:10, michel memeteau a ?crit : > Is there some evolution on Webm support, does it need a lot of changes > in Libshout for example ? or does the code structure of icecast makes > it easy to add new containers/codecs ? Well the interesting question is: "Is webm streamable?". I don't know much about it nut webm is wrapped into a matroska-based

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Hi i have seen that the dbPowerAmp ripping and encoding software supports a new so-called "FLAC uncompressed" format, e.g. http://www.audiostream.com/content/dbpoweramps-flac-lossless-uncompressed-wish-come-true i know only the normal flac compression levels from 0 to 8. have i missed an option on the flac comamnd line tool or how could i achieve that on the linux command line flac

Hi i have seen that the dbPowerAmp ripping and encoding software supports a new so-called "FLAC uncompressed" format, e.g. http://www.audiostream.com/content/dbpoweramps-flac-lossless-uncompressed-wish-come-true i know only the normal flac compression levels from 0 to 8. have i missed an option on the flac comamnd line tool or how could i achieve that on the linux command line

Hi, While building an ogg-vorbis stream encoder, I encountered some problems with silence in the audiostream. The bitrate drops to almost zero, and pages going out less then ones a minute what makes the stream to stop / buffer. In earlier postings I read that I shout use ogg_stream_flush as an alternative to ogg_stream_pageout, in case of silence. My question is this; Is there any

I have read FLAC's "--help", the man-page, and the HTML documentaion, but there are a few things that I don't understand. 1. I'll start with the thing I'm most confused about. The --best option is synonymous with -l 12 -b 4608 -m -e -r 6. Why is that? Is not -l 32 better that l- 12? And you can have -r 0,8 without using --lax, and -r 0,16 with --lax. 2. The --lax option

for the "best" compression: --super-secret-totally-impractical-compression-level takes forever on my system (athlon 2500+) and sometimes results in larger files. so i just stick with -8 -V --lax --no-padding. -V to ensure the encode was vail --no-padding to save a few more bytes, as "The encoder writes a PADDING block of 4096 bytes by default." though 4kb over 3000 tracks

Martin Leese wrote: > > At: > http://flac.sourceforge.net/documentation_tools_flac.html#flac_options_exhaustive_model_search > > it states, "If the max LPC order is high this > can significantly increase the encode time." > This suggest that if the max LPC order is low > then using -e will not be slow. > ---- That brings up another question then... What

Sorry I replied in private ... 2011/1/31 Quentin Drouet <kent1 at arscenic.info> > > > 2011/1/31 Romain Beauxis <toots at rastageeks.org> > > Le dimanche 30 janvier 2011 03:54:10, michel memeteau a ?crit : >> > Is there some evolution on Webm support, does it need a lot of changes >> > in Libshout for example ? or does the code structure of icecast

Hi ! 2011/1/31 Romain Beauxis <toots at rastageeks.org>: > Well the interesting question is: "Is webm streamable?". Well it seems to ma that Google made Webm especially for streaming and I've even read that it should be easier to stream webm than ogg for various reasons. > Alternatively, there seems to be a ogg mapping for vp8 floating around: >

Hi, I am trying to reencode a video from the European Parliament (WMV) which has multiple audio tracks for the different languages: ================================================================================================ Debian-50-lenny-32-minimal:/var/www/tmp# ffmpeg2theora --audiostream 0 VODChapter_20100323_09030000_12350000_Ch04.wmv [wmv3 @ 0xb7e21b50]Extra data: 8 bits left, value:

Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on

Josh explained, > search for "subset" on http://flac.sf.net/format.html > (I'll make a link in the documentation to it). That was over my head -- not the post, but the information at that URL. Does it mean that a non-Subset FLAC file isn't streamable, that a decoder can't read successive parts of the FLAC file and generate successive parts of the WAV?

Hey guys, Keep up the goodwork! I have a suggestion...why not make Ogg Vorbis streamable through QTSS or Darwin Streaming server. The people at Apple are willing to incorporate it I'm sure. Would be nice to be able to stream Ogg Vorbis from with QTSS or Darwin Server!! Lemme know what you think! Regards, Mark -- Mark de Bokx - Application Engineer Cistron Internet Services P.O. Box 297,