similar to: Opus vs Speex NB

For anything below 6 kb/s, I would actually recommend using codec2 rather than Speex or Opus. Cheers, Jean-Marc On 04/11/14 05:55 PM, Manpreet Singh wrote: > Hi, > > I noticed that speex.org <http://speex.org> has a banner that mentions > that Opus is better than Speex in all aspects. The supported bitrate > range for Speex seems to be as low as 2kbps though but Opus can

Hi Dragos, The issue is that not all packet loss maybe congestion related. Often, reducing bitrate seems to have no impact on improving packet loss. Thanks, Manpreet. On Tue, Dec 16, 2014 at 2:09 AM, Dragos Oancea <droancea at yahoo.com> wrote: > > Hi > > You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if > there is packet loss. You know if

Hi, Although this maybe considered out of scope here, but I'll ask anyway. Opus has remarkable flexibility for changing encoder bitrate during a call. Are there any suggestions about how bandwidth/capacity between the two endpoints can be measured/estimated during a call so that the outgoing bitrate can be adjusted accordingly? Thanks, Manpreet. -------------- next part -------------- An

The Opus RFC seems to recommend a frame size of 20ms for most applications. For wideband speech, the sweet spot range is recommended to be 16-20kbps. 20ms frames => 50 frames per second. For a VoIP application, the header overhead per frame (IP+UDP+RTP+SRTP) is 44bytes => 17.6kbps at 50 frames per second. So a 20ms frame size seems to cause a roughly 100% overhead of header data. Therefore,

Hello Daniele It would be worthwhile to attach an external serial flash or USB thumb drive, if the intent is store data. This allows for far more flexibility in storage Regards Amit On Fri, Aug 26, 2016 at 11:02 AM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 26/08/16 11:40 AM, Daniele Barzotti wrote: > > You're right! I forgot to say that I need only the encoder

Hi Jean-Marc, thanks a lot for your reply. > Well, the first question is whether you want encoding, decoding, or > both. If there's one you don't need then you can remove that > (unfortunately, there's no easy way right now). You're right! I forgot to say that I need only the encoder side (and only for voice). My application have to acquire a 16bit 8KHz PCM stream and

Hi You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if there is packet loss. ?You know if there's packet loss if you receive RTCP .?Linphone does this . Regards,Dragos Oancea From: Manpreet Singh <manpreets7 at gmail.com> To: opus at xiph.org Sent: Tuesday, December 16, 2014 7:54 AM Subject: [opus] Estimating bitrate during a real-time voip call Hi,

Hello. Voice quality when calling - this is one of the most important in the PBX. You need to record the quality parameters for each call to improve. Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11. In asterisk 13 I did not find a handler after the call, but before

Hi all, thanks a lot for your replies! Sorry for my typo, the flash size is 8MB (Byte) :-) Unfortunately I cannot use another flash because I'm working on a proprietary board. Jean-Marc, thanks for your suggestions. I thought to use fixed point for convenience, but I can work on floating point too, so I will take in account the codec2 (I didn't know it). Moreover, if you all have

Tried your fixed_generic.h change but that didn't help. Andy ----- Original Message ---- From: Andy Ngo <ndno72-speex@yahoo.com> To: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> Cc: speex-dev@xiph.org Sent: Wednesday, December 12, 2007 4:13:35 PM Subject: Re: [Speex-dev] 4kbps sounds robotic on TMS320C64 Jean-Marc, Yes, fixed-point is enabled (#define FIXED_POINT in

Hi Jean The problem is that for using this package I need to use all the libs like ogg, even Speex?, no, and also where i can find an easy example like opus_demo.c to create the .opus file Greetings Toni 2013/10/28 Jean-Marc Valin <jmvalin at jmvalin.ca> > Hi Toni, > > The package you want is opus-tools. You can get it from the download > section. For file distribution,

Hi everyone, We finally made it! Opus is now standardized by the IETF as RFC 6716 (http://tools.ietf.org/html/rfc6716). See the announcements at: https://hacks.mozilla.org/2012/09/its-opus-it-rocks-and-now-its-an-audio-codec-standard/ http://www.xiph.org/press/2012/rfc-6716/ Feel free to spread those around :-) We're also releasing both 1.0.0 (same code as the RFC) and 1.0.1, which is a

Hi everyone, We finally made it! Opus is now standardized by the IETF as RFC 6716 (http://tools.ietf.org/html/rfc6716). See the announcements at: https://hacks.mozilla.org/2012/09/its-opus-it-rocks-and-now-its-an-audio-codec-standard/ http://www.xiph.org/press/2012/rfc-6716/ Feel free to spread those around :-) We're also releasing both 1.0.0 (same code as the RFC) and 1.0.1, which is a

Thanks for all the amazing work with ambisonics Drew et al. We're looking forward to the 1.3 final release and have already been successfully using the ambisonic work in production code. Varun -- Engineering Manager Facebook Audio ---------------------------------------------------------------------- Message: 1 Date: Sun, 3 Jun 2018 13:02:18 +0100 From: Peter Robinson

On Thu, Feb 22, 2018 at 9:53 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 02/22/2018 09:34 PM, Stepan Salenikovich wrote: > > Its unexpected because the decoder continues to output all samples > > of -32768 even when the microphone input is silence or near silence, so > > I would expect the decoded values to be at or near 0. > > Oh, if the output is

Hi, I'm fairly new on codecs, I'm trying to implement a communication between two PCs. The data rate should be of approx. 6Kbits/s, this 6Kbits/s can just be the useful data. Any encapsulation (for example UDP , RTP, etc ...) can be present even if that means to rise the overall bit rate. Also if this rate can be achieved on Asterisk. For exemple I've tried to change de rate on

Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like <tftpserver-dir> <mac-address> firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _________________________________________________________________ Rethink your

I wonder if anybody try to compress a pure sine wave using OPUS codec. When I compressed the mono 1KHz, 16bits 48000 samples per sec. audio stream using the 'opus_demo' utility: opus_demo -e audio 48000 1 2min_1kHz_Sine_16bit_48kHz.wave 2min_1kHz_Sine_16bit_48kHz.opus_raw I had the output stream that is shown below. 00 00 00 01 00 00 00 00 78 00 00 00 01 00 00 00 00 78 00 00 00 01 00

Hi folks, I am totally new to audio streaming codecs, and just looking around. Trying to figure out what else I can put on top of my very long list of projects. So a few days ago I figured out that fldigi a digimode application for Amateur Radio supports a dual 1000 Baud PSK mode. I thought "that's fast" and started looking around which sort of Data I could put into that. A

On 26/08/16 11:40 AM, Daniele Barzotti wrote: > You're right! I forgot to say that I need only the encoder side (and > only for voice). Then you can remove all of the decoder. As for the encoder, it depends on the bitrate and sampling rate you want (more below). > My application have to acquire a 16bit 8KHz PCM stream and save a > compressed audio into a flash. Sounds like a job