similar to: celt-dev Digest, Vol 30, Issue 7

Hello everyone, I was curious if anyone has performed any listening tests or done work with solo instrument performances with the CELT codec? I experience quite a bit of noise coming up with a heavily tonal stimulus, the worst offender being a trumpet. I understand some of this behavior is to be expected, this is a perceptual codec afterall, but was curious if anyone was able to find an elegant

Dear CELT Developers, I wrote a proposal on how to integrate CELT into Digital Radio Mondiale (DRM+). The CELT codec has been integrated in the Spark transmission software and SoDiRa receiver software. Please download the proposal here: http://www.drm-sender.de/celt_in_drm.pdf Suggestions and comments are welcome! A live on-air DRM+ transmission using CELT (v.0.7.1) for audio encoding was

Hi everyone, I'd like to announce CELT 0.7.1, which improves the quality of the packet loss concealment (PLC), but does not change the rest of the codec. For this reason it is the first release NOT to break bit-stream compatibility with the previous release (0.7.0). But I promise not to do it again, the next CELT release will likely break compatibility once again. Note that the PLC

Hi everyone, I'm pleased to announce version 0.6.0 of CELT, with many quality improvements. This includes better stereo coupling, better handling of transients, and better handling of highly tonal signals. In addition, packet loss robustness has been improved through the optional use of independent (intra) frames. CELT now supports a larger dynamic range, suitable for encoding 24-bit audio

The patch prevents KEIL MDK compile warnings, like:   warning:  #1035-d: single-precision operand implicitly converted to double-precision Actually ARM Cortex-M4F has only a *single precision* (float) FPU. It's suit for all platforms. See the comment at the begin of patch file. Sincerely Forrest Zhang -------------- next part -------------- Specify the floating point constant with single

-- Joshua Bowman Silverback Networks (559) 305-3770 silverbacknet at gmail.com www.silverbacknetworks.net -------------- next part -------------- src/analysis.c | 6 +++--- src/mlp.c | 3 +++ src/opus.vcxproj | 5 +++++ src/opus.vcxproj.filters | 15 +++++++++++++++ src/opus_demo.vcxproj | 4 ++++ src/opus_demo.vcxproj.filters |

Yes, WavPack is second in my list. FLAC doesn't preserve every chunk? I thought it did. I only gave a quick try but it seemed to have preserved even the most obscure chunks. Let me check: it even seems to preserve "MIDI note associated to marker", which is a very unknown metadata used by SoundForge (& even defined in a buggy way), so I assumed it was saving them

Hi Everyone, I just merged a lot of changes in the CELT master branch. This should solve the following issues: - tonal artefacts (birdies) at low bit-rate - tonal artefacts in stereo at all rates - low-frequency noise (roughness) on low bit-rate speech The code has been simplified a lot, but there's lots of changes, so please give this a try and see if anything got broken. Cheers,

Thanks Vamsi I have not been able to locate pwlib - 1.6.6 or openh323 1_13_5. I found the latest versions through sourceforge and I found some older versions on another site, but not these versions. This has been quite frustrating. Anyway, I think by using the asterisk-oh323 branch under channels in the asterisk source tree I will have more luck. At present it seems to compile successfully, but

I have successfully been using OH323 v0.6.5 with Asterisk 1.0.x. I now need to move to CVS HEAD in order to use some features that are not in v1.0.x, and am trying to compile OH323 to use with it. On the InaccessNetworks site, it ways that OH323 v0.7.1 is for HEAD. However, when I compile it, it appears that it hasn't been updated since the channel structures were revamped. I get many errors,

Has the requirement for FrameSize decreased? I cannot seem to instantiate a mode with a frame size smaller than 120. I have been using a framesize of 100 on previous versions. Thanks bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20110408/af13a1c6/attachment-0002.htm

Hi everyone. I did a rough recording of an instrumental (electric piano sound) & e-mailed it to a friend in Vorbis 11025 Hz / mono. I was seeking a bitrate in the range 8-16k/sec. The song is 2:55 in length. q=-1 happily achieved a 12.6k/sec bitrate. All file sizes I mention in this are for files without informational tags. And I hope this isn't interpreted as trying to plug my music or

Hello all, Please try this "patch". It changes the way the noise floor is used for quantization in a not-so-subtle way. At the very end of _vp_compute_mask, add the lines: for(i=0;i<n;i++) flr[i]=.01f*sqrt(flr[i]); The .01 is there to ensure the current codebooks will work. We will really need different, newly-trained codebooks with this change; then the

Hi all! Here is my problem: I have installed RedHat Linux version 6.0 on a computer and I want to share its resources via samba. Unfortunately, my computer - which is running WfWg 3.11 for Eastern Europe and the trumpet winsock TCP/IP stack is not able to see the samba server. Other computers, running Win95, Linux and WfWg with M$ TCP/IP stack perform OK. I have tried the following combinations:

Hello every body: I have RedHat fedora core 2 machine, using iptables and squid. I am having a lot of problems with peer2peer traffic. (bittorrent, kazaa, etc.) so I have installed ipp2p from rpm. Every thing was ok until I use iptables rules. I get this error. [root@router iptables]# iptables -A INPUT -p tcp -m ipp2p --ipp2p -j DROP iptables: No chain/target/match by that name sames

Hi -- I'm looking at adding Speex decoder and encoder support to the Miles Sound System, a commercial library used by game developers on various platforms including PCs, the Mac, and game consoles. As part of my initial evaluation, I've gone ahead and downloaded the 1.0.4 and 1.1.1 Win32 binary distributions and played around encoding and decoding various files with the speexenc.exe and

Hi I've been running asterisk for a year or so now, and recently upgraded to the lastest CVS-HEAD version along with oh323 v0.7.1. (previously oh323 version 0.6.3 used to crash about once per week during high call volumes). Since upgrading, however, asterisk seems unable to successfully receive h323 calls. Previously, an incoming h323 call would be answered by asterisk and show:

I'm doing testing on this at the moment. But to start with: >>Because music represents an analog signal, As I wrote, it would only apply to specific genres, not analog recordings. Electronic music quite often doesn't leave a computer these days. And it mainly consists of drums, synths & vocals/effects. Drums are often samples sequenced at sample (not sub-sample) accuracy,

Optimize opus encode (float only) usecase using ARM NE10 library. Mainly effects opus_fft and ctl_mdct_forward and related functions. This optimization can be used for ARM CPUs that have NEON VFP unit. This patch only enables optimizations for ARMv7. Official ARM NE10 library page available at http://projectne10.github.io/Ne10/ To enable this optimization, use --enable-intrinsics

Optimize opus encode (float only) usecase using ARM NE10 library. Mainly effects opus_fft and ctl_mdct_forward and related functions. This optimization can be used for ARM CPUs that have NEON VFP unit. This patch only enables optimizations for ARMv7. Official ARM NE10 library page available at http://projectne10.github.io/Ne10/ To enable this optimization, use --enable-intrinsics