similar to: Change 48 khz sample rate limit

You asked in the Vorbis list, but your text only mentions OGG. The codec commonly used in OGG containers that is limited to 48 khz is Opus. Maybe you are trying to use the wrong codec (i.e. Opus instead of Vorbis)? Using a 44.1 khz wav file, I was able to encode a 192 khz ogg-vorbis file with the following command: $ oggenc --resample 192000 input.wav Of course, if your original material is

Hey. I just wondered,... which is the recommended bitrate and further settings of opusenc (like complexity and framesize) for general music (e.g. including classical music) to achieve more or less transparency? Talking about CD level audio (16bit; 44,1 kHz; Stereo) The only source I could find regarding that was: http://wiki.hydrogenaud.io/index.php?title=Opus#Music_encoding_quality But that

When experimenting with Opus encoder, I took a random FLAC file, encoded it, heard it, and heard the original file. When hearing the opus encoded file, I thought it was a flute solo, but when I heard the original, I nearly fell off my chain when I noticed the piano I couldn't hear from the Opus in the first time. Now I know there is a piano, I can hear it in the Opus file, but it is much

Hello, I am new to the list and I need to find a 32 or 64 bit windows dll to work with. I spent 2 hours on archive files and not easy to find if there is already one on windows. Can someone point me direction and where to start ? Thanks Charlie -------------- next part -------------- An HTML attachment was scrubbed... URL:

There is some time now that I reported this issue on the bugtracker: https://roundup.ffmpeg.org/roundup/ffmpeg/issue1325 And this guy, jbr, kindly created a patch that solved the problem. What is needed now for the patch to be included in the SVN? About his patch, there is a remap table that maps the channel order from SMPTE order to Vorbis order. What is this SMPTE order? Vorbis specification

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

Dear all: I'm a developer in china, and i really want to know how to calculate a ogg audio file's bitrate and then i can get the duration by file size. thanks very much for your answer. Best wishes. Aaron Jie. -------------- next part -------------- An HTML attachment was scrubbed...

That article is a bit too dismissive. I agree that one cannot hear the difference between 48KHz/16bit and 192KHz/24bit if you just transfer the data directly to the audio output device. As such, there is no good reason for Opus to support higher than 48KHz (especially since this is lossy compression, anyway). However, in general, that's not all you do with audio data. 192KHz is useful for

On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > Actually... no! 24-bit can indeed be useful as extra margin and Opus > can actually represent even more dynamic range than 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to

Greetings, In reviewing the changelogs it?s unclear in which release FLAC began supporting a sample rate of 192kHz. The reason for my question is that there are many forums and university studies that state that FLAC does not support a sample rate of 192kHz however the current documentation (assumed 1.2.1b) under FORMAT under FRAME_HEADER does note that it is supported. If it was not

hi,All, I want to know whether Opus has AEC features like Speex? Thanks 2012/10/17 <opus-request at xiph.org> > Send opus mailing list submissions to > opus at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.xiph.org/mailman/listinfo/opus > or, via email, send a message with subject or body 'help' to

One thing that has always bothered me about the ogg format is the distortion of high frequency sounds - even at data rates as high 128 and 160 kbps. I find the best way around this is to resample the wav file to 48 kHz (using SoundForge 6.0) before encoding (using CDex) to ogg. It takes a while, and adds a lot of extra wear and tear on my drive, but what a difference! The result is an 80k ogg file

Hi , I have read that it is posible to encode higher sample rates like 96 khz or 192khz? and the output is 48 khz, the resample is internally.? http://wiki.xiph.org/OpusFAQ But it is possible to encode? 44.1khz. It is resampled to 48khz or I have to make the resample by myself and then encode it with opus. thnx, arctor -------------- next part -------------- An HTML attachment was scrubbed...

Hi there - I am currently in the middle of a project where I need to configure an Icecast Server and I am using Ices as the encoder. I want to configure Ices to take in a stream with a sample rate of 128 kHz and output it to the Icecast server. My question is, what are the minimum and maximum sample rates that Ices will continue to work with? Thank you, Heidi Young Software

Hi, Does anyone know of a technical reason why FLAC cannot support 88.2 Khz files? I have a reason to uses this rate since it is easy to perform quality conversions from 24 bit 88.1 Khz master files (stored as flac files) to 16 bit 44.1 khz files for CD mastering purposes. I suppose I could Kludge the wav files so that they were half speed wav files at 44.1 khz and then hand the over to Flac, but

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

I found some posts regarding this issue dating of September 2003, but no real answer. The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help migration. Is there an existing format/codec for this? If not, can I make myself a shared object in /usr/lib/asterisk/modules? Is this easy??? :-( Thanks, Yves

hi, is it possible to encode 16 bit, 22 kHz, stereo/mono WAV files to FLAC files or could there be a problem with the low frequency 22 kHz (lower then CD quality)? PS: I'm a FLAC beginner thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac/attachments/20070512/63ed58f6/attachment.html

Good morning, On Mon, 2023-02-27 at 02:06 +0000, Coolvibes Reloaded wrote: > rite i've got the icecast2 page setup but i don't have any active > mounts? > why's that once you got your server running it will list those sources that are connected. So if it is freshly started and there are no sources connected nothing is shown as there is nothing to show. The next step would