search for: 112kbps

Hello, I have problems with htb. The problem is that when I download any file via shaper with htb, the traffic is very dinamic, it jumps, for example: if i have set ceil = 128kbit the results that it jumps from 112kbps to 144kbps or smth like that maybe its not very bad, but when the traffic drops down to 40kbps or less and then after 1 or 2 seconds jumps to 144kbps, its bad :-( and it is often. Root class is 20Mbit There are about 7000 classes (on two interfaces) an example script: ==========================...

I've been reading through the source, and I can't understand where compression is occuring! It appears that it should be ouputing all of the LSP and residule data, but the output is smaller then the input (it's about 500Kb/s+, but it bzip2's to about 320Kb/s). The output sounds okay, but has signifantly worse S/N then mp3 at 320. --- >8 ---- List archives:

HI! So I’ve been at this for a few days with no luck. I am trying to pre-pend my stream with an intro file (mp3). I’ve saved the mp3 file (for the intro) the same as the stream > in my test right now it is set up stereo, 112kbps, 44100hz. I can play the file independently in a browser from the webfoot location. I can play the stream with no problems. I have modified my icecast.xml to include this <mount> <stream-name>/test.mp3</stream-name> <intro>/test-44100-stereo-112.mp3</...

...nd -vo yuv4mpeg -vop scale=640:480 SVCD_File.mpg Some 'data points': -a 1, for 44.1khz stereo .wav file, equates to ~64-72kbps -v 1, for 640x480 29.970fps video, equates to ~600kbps I've tested 'transcoding' a 13-minute-long, 368x480, 29.970 fps, 3750.0 kbps color mpeg2 video, 112kbps (?) audio file (Total file size ~315MB)from the Prelinger archives as described above, scaling to 640x480, with no problems. Synchronization seems fine throughout, and even at -v 1, the resulting video did not seem noticeably degraded from the original, though in fairness, the original in this cas...

hello, i have a question about the documentation: oggenc --help says: -b, --bitrate Choose a nominal bitrate to encode at. Attempt to encode at a bitrate averaging this. Takes an argument in kbps. This uses the bitrate management engine, and is not recommended for most users. See -q, --quality for a

...be better than in mp3, although it always sounds a little flat to me, I think the best description would be that the original (and mp3) have more "colours" (compared to <160 kbps vorbis). I was surprised that the 100kbps mode gave real bad quality at a higher real bitrate than the 112kbps-mode. This is the first mode that introduces a mp3-like "bubbling sound" that is not to ignore (112 kbps does this _a little_ as well at more silent passages). Modes higher than/equal to 128 kbps sound really fine. The encoder tripled in speed compared to beta3 on my duron 700. I'...

...be better than in mp3, although it always sounds a little flat to me, I think the best description would be that the original (and mp3) have more "colours" (compared to <160 kbps vorbis). I was surprised that the 100kbps mode gave real bad quality at a higher real bitrate than the 112kbps-mode. This is the first mode that introduces a mp3-like "bubbling sound" that is not to ignore (112 kbps does this _a little_ as well at more silent passages). Modes higher than/equal to 128 kbps sound really fine. The encoder tripled in speed compared to beta3 on my duron 700. I'...

...ot primarly care about quality but about filesizes. One could assume that such a user would have a collection of mp3's at 128kbps or higher bitrates, and uses an encoder like BladeEnc or Xing. He wants to take uses of ogg's supposed quality and transcode his 128-or-higher files into 96 or 112kbps oggs to save diskspace. Even if he realizes it will only lower quality, it will gain him/her diskspace. We know that transcoding will introduce errors, but it is interesting to check how bad they are. If the errors are small due to ogg's encoding quality, they may not be all that bad on medioc...

I started thinking about this after doing a little testing with AAC, MP3 and Ogg Vorbis. I was comparing the different formats at similiar bitrates. After a while I finally realized that they all sound more or less the same to me. >From 160 kbps on, I usually cannot detect any difference between a lossy encoding and the original source. If LAME is used, I have to strain to notice anything at

Hi all, For some reason VSAT or Satellite Internet services are not mentioned (or searchable) in this list's archives. I thought I'd let you know that I tested Asterisk using IAX (not IAX2) to make a phone call from an analog phone hooked up to an Asterisk system behind a Linksys router connected to a Gilat VSAT satmodem, and it worked. The "other end" (gateway) is a