similar to: Beginner's questions/suggestions

>>> Ralph Giles <giles at thaumas.net> schrieb am 21.12.2015 um 19:51 in Nachricht <CAEW_RkshUM55uwdvU6DsE17pLZki651Xvvu7d2Y6jObePXZwCQ at mail.gmail.com>: > On 21 December 2015 at 04:31, Ulrich Windl > <Ulrich.Windl at rz.uni-regensburg.de> wrote: > >> opusenc does not display the file name it processes. In Lunux when you use > some batch processing,

On 21 December 2015 at 04:31, Ulrich Windl <Ulrich.Windl at rz.uni-regensburg.de> wrote: > opusenc does not display the file name it processes. In Lunux when you use some batch processing, it might be interesting! Example output for > "for f in *.wav;do opusenc --bitrate 160 $f ${f%.wav}.opus; done": In situations like this, i usually insert an 'echo $f;' before he

>>> Amit Ashara <ashara.amit at gmail.com> schrieb am 11.05.2016 um 19:32 in Nachricht <CAEyg9sjvTWMBMMCJ8HQcYmbv1BtNt54CgpqWaGNm02MWrKcxaQ at mail.gmail.com>: > Hello Jean-Marc, > > So for the moment we can assume that this method is also OK to use? > > On Embedded Systems, both SRAM and Flash can be a restricting factor > besides the compute time. To

The overhead of Ogg (in file size) is pretty small and it's efficient enough for most applications (and uses far less CPU than the codec anyway). If anything, you might want to look at optimizing the existing Ogg implementation (e.g. like Tremor did in the context of Vorbis). Of course, you're always free to design a new container, but I doubt it's worth it and it's a lot of work

Hi! I have some MP3 audio material which is basically speech with some background noises, essentially > 120Hz and < 5kHz. I had the idea to reduce the file size by recoding the material to Opus at 56kbps. Unfortunately the result is a file sampled at 48kHz much larger than the original. I hope you agree that it does not make sense to create a file larger than the original (MP3). Of course

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

Hello Jean-Marc, As an example, I am using the output of opus encoder to store the file as the following format and read back the same during decode process, without having much overhead. (Thought it would be useful to put a picture rather than running text) [image: Inline image 2] Regards Amit On Thu, May 12, 2016 at 10:47 AM, Amit Ashara <ashara.amit at gmail.com> wrote: > Hello

On 05/12/2016 10:35 AM, Amit Ashara wrote: > For HMI panels, except for the capture pattern and a single page segment > entry, other fields are not important, and which results in almost 7% > overhead for a 20ms raw frame encoded with Opus. I'm not sure how you get a 7% overhead. In most uses I've seen, the overhead is more around 1%. > At the same time the > file

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

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

This patch adds a new option "channel-mapping" to opusenc which sets the channel mapping family used by the multistream encoder. Please let me know whether adding this option is worthwhile and whether the help string is okay. I tried to keep it short but accurate. The error message for an unimplemented channel mapping is "Error cannot create encoder: request not implemented".

On Mon, Nov 5, 2018 at 11:01 AM Jan Stary <hans at stare.cz> wrote: Attached I send the spectrogram (vic SoX) of the first 20 seconds > for the wav file and the opus file. Indeed, there is extra noise > for the low frequencies, but somewhere around -100 dB. > > Jan > That might be entirely due to SoX treating it as a 16-bit file, which it is not; -100dB is almost

>>> Agustín Dall'Alba <agustin at dallalba.com.ar> schrieb am 14.04.2017 um 22:53 in Nachricht <CAHBqS-w3v44WM5x+_4XdFMkD42A2iYTbEWKEBmvJc2P3Y-LJGA at mail.gmail.com>: > I halved the volume of the sample before encoding with > `sox -v 0.5 floex.wav quiet.wav` and now I can't ABX it succesfully anymore. > So the artifact I heard was just clipping when encoding

When I need to stop a running R script on Windows or Mac I just use the <esc> key which kills the current script and returns the control to R interpreter. But when I run R from JGR the <esc> is useless as well as the other available keyboard keys. Just recently not even clicking on the STOP-symbol (a big red X) placed on JGR top menu bar could terminate a process that had entered a

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

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

On Wed, 2014-02-12 at 14:29 +0000, Ian Malone wrote: > On 12 February 2014 13:02, Alice Wonder <alicewonder at shastaherps.org> wrote: > > Hello, > > > > I am new to using opus but so far I really am loving it. At 16kbps > > spoken content sounds fabulous. > > > > I am using opusenc on 64-bit linux - mostly with flac input. > > > > The one

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 am new to using opus but so far I really am loving it. At 16kbps spoken content sounds fabulous. I am using opusenc on 64-bit linux - mostly with flac input. The one thing I can't figure out how to do is add album art. I tried passing the vorbis way of adding a metadata_block_picture comment field to opusenc but it complained about the length. I can't seem to find any

I am interested in being able to encode a single Opus stream using several CPU cores. I get a raw audio input and "opusenc" can transcode it at 1200% speed (Raspberry PI 3B+). It saturates a single CPU core, but the other three are idle. Is out there any project to add multithreading options to "opusenc", or something in that line? Looking around, I have found this: