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Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best

switch between speech and music encoding rather quickly. I wanted to try that for a CD track that has a vocal (talk) introduction = followed by music (just two instruments). Using foobar2000 with some custom setup for Opus, I tried VBR encoding = with a 256kbps setting (opus-tools-0.1.9-win32). My expectation was that = the bitrate would be significantly lower than 256kbps while speech is =

It might be good to use the (uncompressed) samples on the opus page, as a common starting point? http://www.opus-codec.org/examples/ On Dec 21, 2013, at 9:43 AMEST, Stuart Marsden wrote: > I have run a few more test at different bitrates and 1.1 is looking even worse in terms of speed compared to previous versions. > > I have shared a google sheet which has the raw data and charts for

An idea would be like for WMA 9 lower bitrates (32-42-48Kbps) to use a 'crystallizer'; which is basically a harmonics booster focussed at transposing sharp tones some octaves higher. Eg: A file has been recorded @ 20khz computer (or 10khz real life) to preserve space. While playing back the file sounds a bit mushy, almost as if someone was speaking through a cardboard wall. The higher

Thanks Jim, that's understood. When I say AMBE isn't working well, I only mean from the audience acceptance point of view. Technically it is fine. It is exactly doing the job we had expected. It's the long standing wish that everyone wants... More for less. We are just seeking a bit of magic that just may not be there. Ideally finding a codec that can perform

On 7 Nov 2017 13:36, Lucas Clemente Vella <lvella at gmail.com> wrote: 2017-11-07 11:10 GMT-02:00 encrupted anonymous <sergeinakamoto at gmail.com<mailto:sergeinakamoto at gmail.com>>: did another test of many. NeroAAC q=1 @400kbps and Vorbis q=10 @412kbps shared 2nd place. OPUS @330 kbps - 3rd place. LAME MP3 q=0 @320 kbps - 1st place. ---JPEG file attached--- Please disable

I have to admit that I am impressed by your results -- making 1.1 look slower than 1.0 is by no means an easy task. On the other hand, it's a great tutorial on how not to use Opus, so for the benefit of everyone, this is a summary of what we learned in this exercise: 1) When running on ARM, the fixed-point build is usually faster than floating point. This is true on the majority of ARM archs

did another test of many. NeroAAC q=1 @400kbps and Vorbis q=10 @412kbps shared 2nd place. OPUS @330 kbps - 3rd place. LAME MP3 q=0 @320 kbps - 1st place. ---JPEG file attached--- Please disable speech synthezation in OPUS for 96 kbps and up. I don't want my music sound like from a phone speaker! Or what is the problem? Modern codec at high bitrates should produce nearly bit-exact sound, not

Hi again, I was just looking at reviews on vorbis and reading some listening test results when I came across several web sites that were saying something about Vorbis quality -2. From what I read a few programs have Vorbis going down to quality -2 and I even think hydrogen audio forums mentions it in a speech compression topic. Does this quality really exist and has anyone even heard of it? I

OK. Different mailing lists are set up differently. This list is unusual because your answers only go to the person who replied to you. So if you want the other people on the listserv to see your answer, you should make sure that Speex-dev at xiph.org<mailto:Speex-dev at xiph.org> is added to the TO: field of your outgoing message. Hopefully someone else will also attempt to answer your

I have some speech encoded at 140 kbps that I would like to have encoded at 40-60 kbps instead. I could just convert the file to wav and re-encode the file, but I seem to recall that ogg-files very easily can be converted to lower bitrates (something with cutting off the last bits). I would assume that this also will preserve more of the original data. Is there a utility for this? /Ole --

On 2017-11-01, Jean-Marc Valin wrote: > I'm not sure, but my best guess would be "because MP3's window is very > leaky and MP3 has to waste a lot of bits in the LF because of that". > It could also be just the MP3 encoder being silly, or other things. Was the original poster speaking about the SILK or the CELT derived mode? Because at least wrt SILK (and the rest of

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

Cliff, Yes it would be good, but very hard to get a figure for the quality. At 6kbps I assume it does not bother trying to figure what mode to use as at that rate it can only use SILK. When I run some other bitrates it may get a bit slower trying to decide whether it is voice or music. I started with low bit rate because I am only really interested in Voice and very low bit rate. I think there

Message: 2 Date: 29 Dec 2004 15:30:25 +0000 From: Karl Heyes Subject: Re: [Icecast] icecast2.2 and aac? To: qiang Bao Cc: icecast Message-ID: <1104334224.30220.23.camel@bogus.hackers.club> Content-Type: text/plain On Wed, 2004-12-29 at 08:21, qiang Bao wrote: > it seems that icecast 2.2 can only stream aac at 128kbps! > > mp3 and ogg are ok at all bitrates. > > is it a

Hi again, shortly after beta 4 came out i noticed that rumbling artifact oggenc puts into a certain tune of mine. Back then, it was even audible at an ABR of 350kbps. Now with RC2 that bug is completely gone at bitrates of 128kbps and higher. I can hear it again at 96kbps and it becomes more apparent with lower bitrates. At 64kbps, I assume, everybody should be able to hear it. This is not a

Hi All, Got today the Orban Opticodec-PC encoder LE and it sounds great at 32Kbps and 44.1Khz and even lower it sounds great!! Now even people with dial-up can listen to FM quality streams! Got a test stream up, if you're intrested, with some rights free (RIAA free) music (dutch language made by a friend of mine). http://dir.xiph.org/index.php?sgenre=&stype=&search=doosfm there

Jean-Marc, I am building a tool for producing the highest possible quality Internet interviews for "podcasting" applications. The goal is to produce a perfect recording of an interview or conference -- and giving the participants a glitch-free experience is secondary. My approach, therefore, is to build a Windows "wave" file asynchronously by using a streaming

Well, I don't know what SBR is, but there's something in the wideband mode that may be similar: It's possible to encode the whole 4-8 kHz band with just ~1-2 kbps by only encoding the (LPC) shape of the spectrum and then just filling that band with "something that makes sense". Quality is quite reasonable... Jean-Marc Le dim 13/10/2002 à 06:18, Steve Underwood a

Hello, We're streaming radio programs at both 128kbsp and 32kbps, but only archiving the 128kbps stream to save storage space. I'd like to give users a similar choice of bitrates when they request an archived stream (served through icecast's /file/ functionality). Is there a way to change the bitrate on the fly, or do I really need to save archive both bitrates? Thanks for the