similar to: Q: Bandwidth vs. bitrate

Hi! So here is what I got with different encoder settings; still not sure what the "best" setting is. 6kbps seems to add distortions, so I tried 12kbps. MP3-Original: LAME 3.99r, 120kbps, 44100Hz, Stereo, VBR V5 (22:23, 19.8MB) Opus (--raw-rate 44100 --bitrate 56 --vbr --comp 5): (44:45?, 23.4MB): Broken Opus (--bitrate 56 --vbr --comp 5 --ignorelength - %d): (22:23, 12.1 MB, 74kbps)

Hi! Useful advice, thanks! Actually I had been using foobar2000 to recode, because it just makes it so easy to convert multiple files while keeping the metadata (I confess, I'm a "tagger"). But it's easy to miss some encoder option when being presented some default suggestions in a dialog form... Apart form that I always had the impression that Opus could be quite smart

Erik de Castro Lopo wrote: > Err, thats a link to a post talking about flac's WAV reader being limited > to 4Gig files. Problem is, *all* WAV files greater than 4Gig are mal-formed. > Due to limitations in the way WAV files are specified, no valid WAV file > can ever be over 4Gig. And most don't work over 2 GB. The solution we (Xiph) have used in other projects (opusenc,

Hi Ulrich, I assume you've been using opusenc to encode the files. If that is the case you can try giving the encoder some more hints about your files: opusenc --speech --set-ctl-int 4008=1103 ... The latter should tell the encoder that the signal bandwidth is 8kHz (OPUS_SET_BANDWIDTH). See opus_defines.h for all valid numbers. You can also experiment with the complexity settings. 

This patch to opus-tools adds optional support to WavPack lossless format as input to opusenc. Like support to FLAC, it depends on an external library, libwavpack, and may be disabled on configure. Lucas Clemente Vella (1): Reading input from WavPack files. Makefile.am | 7 +- configure.ac | 37 ++++++++ src/audio-in.c | 71 ++++++++------- src/opusenc.c | 19 +++- src/opusenc.h

Hi all, I was reading the discussion about this 1.3.0pre1 release on HydrogenAudio and someone linked an old thread in which one patched FLAC 1.2.1 to support WAV-files larger then 2GB. It might be worth investigating: http://www.hydrogenaudio.org/forums/?showtopic=84014#entry725304

Hi! Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. With Opus I noticed that the file size for 48kHz and 48 kbps compared to 96kHz Vorbis at 31kbps is about double the size and it sounds even worse (than Vorbis) (there is a lot of noise in the lower frequencies when a low

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

Hello Thomas and Amit, Thanks for your notice and the detailed decode performance report. I describe the details of my encode/decode test on STM32F407ZG. A. opus version: latest 1.2.1 (TI: opus 1.1.2) B. KEIL 5.23 (TI: ARM compiler tool chain 5.2.7) C. setup the encoder as the below (fs is the sampling frequency) enc = opus_encoder_create(fs, chans, OPUS_APPLICATION_AUDIO, &opus_err);

Hi, I have found differences in quality between opus_demo and opusenc/opusdec. I used for both applications the same raw pcm file,16 bit,48khz,litle endian. i use libopus 1.1-beta and opus-tools-0.1.7. The command for opus_demo is: opus_demo audio 48000 1 64000 -cvbr -framesize 10 in.pcm out.pcm For opusenc/dec: opusenc --raw --raw-chan 1 bitrate 64 -cvbr --framesize 10 in.pcm in.opus

Hello!, i've just compared the 1.0.3 release with the master branch on a BeagleBone Black (AM335x 1GHz ARM Cortex-A8 with NEON floating-point accelerator) and Arch Linux ARM. At the moment I dont no why, but I see that 1.1 is much slower in encoding. Are there any default changes, that I missed and could explain this? Normaly I suggested a better performance with 1.1 and the ARM

Hello! I encoded a voice file (48kHz) with opusbin\opusenc.exe with the standard settings and decoded it. The output was amazing. I could not hear any loss at all. Then i encoded the same file with opus_demo.exe and standard settings and then decoded it. The output had a sizzling noise, even when I used full bandwidth. I think I have played around with any of the settings in opus_demo.exe,

Hi guys, as MP3 and Opus have very similar objectives, I think the original poster's question was a valid one: Why does Opus have more artefacts in the lower frequency ranges than MP3 has? The spontaneous suspect that lower frequency artefacts may be more noticeably than higher frequency artefacts seems plausible, also. Is it a matter of energy (which is higher for higher frequencies)? When

I had expected that the default bitrate for the encoder would be the same as setting it to OPUS_AUTO, but I'm getting difference results: >opusenc --comp 4 sample.wav sample.opus Encoding using libopus 1.3-rc2 (audio) ----------------------------------------------------- Input: 8 kHz, 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 25 kbit/s VBR Preskip: 312

Hi everyone, Some time ago, I sent a pull request <https://github.com/xiph/opus/pull/107> to the Opus github page. Jean-Marc asked me to post it to the mailing list so everyone can have a look at it. You can find the description and code changes below. Please let me know if you have any questions or concerns. Best regards Gustaf Ullberg In WebRTC, we would like to be able to

On 2007-12-30, Timothy B. Terriberry wrote: > In any format that is to be used on both, it is always better to pick > one and stick with it. Then recommend one single format. Nobody *has* to support all of the features present, yet it makes sense to *allow* common variances. Most of all, because: > Unless you can guarantee that you're writing streams that are only > going to

Hi list, I do not known, if this is really an issue but i noticed that when connecting to a remote ssh host with the standard linux openssh client using a private key, that there is no line of text indicating when the local key-passwd process was completed and the connection session was established. On a compromised host, the login shell could write the line 'Enter passphrase for key

On 2008-02-14, Conrad Parker wrote: > I tend to disagree with your sentiment. The specification of any > format or protocol has mandatory and recommended sections (not > "features"); MUST and SHOULD respectively for IETF and W3C stuff. Then why not make the common endianness MUST and the rest of it SHOULD? That was my sentiment, after all... -- Sampo Syreeni, aka decoy -

Ian Malone wrote: > This is all well and good but OggPCM is in an Ogg transport > stream, so that needs to be unpacked anyway. Fair enough. Since the ogg pages (which I beleive are 4k) need to be unpacked anyway, there is little harm in having to (possibly) do endswapping as well. Erik -- ----------------------------------------------------------------- Erik de Castro Lopo

List, A recent discussion over on XiphWiki is trying to decide if OggPCM should support only little-endianness or the usual combo of big and little. It started with the following statement by an user (Qqq): "Portable players are usually ARM, which is usually little-endian. The Macintosh is now little-endian. Obviously the PC is little-endian. Clearly there is a winner. It's long past