search for: 3khz

...n noise, use a small > codebook: large quantization noise) is *not* based on the actual > quantization noise for the frame being encoded, so the psycho > acoustics is not used to do MP3/AAC style noise shaping. If for some > reason there happens to be a lot of quantization noise in the 3khz > region, the codec is not aware of this and is unable to move bits > around to correct it. How important is this? I dont really know, but > we can find out when Vorbis if fully developed by comparing it with > AAC. Aleksandar -------------------------------------- Vorbis Xtreme: New...

...dication on that. Perhaps with people not very familiar with advanced audio tools know how to spot out a mp3, they just want to know if the flac they just bought or downloaded is good or not? Another test, if the FFT test is unclear, is to check the correlation between left and right channel below 3kHz, which should be just about nothing if the source is a mp3 (since mp3 encoders sum L and R below 3kHz at low bitrates). If also doing further testing, and knowing how the mp3 encoders work, it should be fairly easy to determine if a source file might have been an mp3. At least the program would be...

Thanks! Btw, have you tried using SBR-technology or similar with speech codecs? That might be a good idea I thought.. But I don't know if it produces as good quality with speech codecs as it does for music codecs. Do you know if there is any open source variant of SBR? /Pontus -----Ursprungligt meddelande----- Från: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]För Jean-Marc

...the lower pitched fundamentals which usually give rise to them. I don't know of any voice specific coder that even attempts to capture energy above 10kHz. SBR just isn't relevent. Most wideband speech coding captures only 7kHz to 8kHz bandwidth. The key improvement that gives over the 3kHz to 4kHz most mainstream voice coders capture is to clean up unvoiced sounds. fffff, sssss, and other unvoiced sounds appear almost the same at telephone bandwidth. At 7kHz bandwidth they have enough character to make them more distinguishable. The basic intelligibility improvement you get is us...

...as used (and at what settings) for uncompressed source audio, but I'd be really impressed if anyone could analyse an MP3 or other lossy file that had been transcoded more than once. > Another test, if the FFT test is unclear, is to check the correlation between left and right channel below 3kHz You need to be careful about stereo separation at lower frequencies. Every vinyl lab checks this, and almost every mastering house has warnings about it on their website. Due to the frequency response of vinyl as a medium, bass has to be cut when recording, and boosted at playback. The RIAA standa...

...h semi-stable weights, almost entirely in the (j==M-1) bin (0-20ms delay, which is quite natural). It's the same with normal speech as long as it's spoken reasonably fast. I see some "banding" of the output, it seems there's more output signal (and more to cancel) in the 1-3khz and 5-6 khz area, but I blame that on the headphones; they're cheap. However, when switching to AC DC: Big Gun, we see and hear a large residual echo from the opening el-guitar. This seems to be a result of a semi-stable sound that lasts more than 20 ms; the canceller finds a correlation i...

...mentals which usually give rise to them. > > I don't know of any voice specific coder that even attempts to capture > energy above 10kHz. SBR just isn't relevent. Most wideband speech coding > captures only 7kHz to 8kHz bandwidth. The key improvement that gives > over the 3kHz to 4kHz most mainstream voice coders capture is to clean > up unvoiced sounds. fffff, sssss, and other unvoiced sounds appear > almost the same at telephone bandwidth. At 7kHz bandwidth they have > enough character to make them more distinguishable. The basic > intelligibility impro...

...st entirely in the > (j==M-1) bin (0-20ms delay, which is quite natural). It's the same with > normal speech as long as it's spoken reasonably fast. Fine. > I see some "banding" of the output, it seems there's more output signal > (and more to cancel) in the 1-3khz and 5-6 khz area, but I blame that on > the headphones; they're cheap. Not sure what you mean but it doesn't seem to be a problem. > However, when switching to AC DC: Big Gun, we see and hear a large > residual echo from the opening el-guitar. This seems to be a result of a &g...

Hi, I'm still working on visualizing the echo canceller, but I discovered something that might be interresting. During testing, i did this: Generate a test signal (10+x sine waves per frame), where x increases by one for each iteration, and wraps around at 100. Set the speaker signal for the frame to the test signal. Add 0.5*test signal to the mic signal. When watching the power graph

First of all, I am not aware of any official source of FLAC files that provide MP3 sourced data. I meticulously check the music I purchase, especially when it is 24/48 or 24/96 material, because this is new technology, and sometimes people get it wrong. However, you should be aware that many modern producers use software to create their music, and when the software stores sound clips in