similar to: question on opus rtp

Hi Kelvin, The audio bandpass setting is only configurable when the encoder is instantiated (eg: start of a Voip call ) , but you can change the bitrate anytime.For example if you can read incoming RTCP packets , you can check if there's reported packet loss , and then lower the bitrate accordingly.Yes, the app has to be aware of the packetloss ?percentage. Cheers,Dragos From: Kelvin

Thanks Dragos, I assume I will be setting those parameters during initialization of encoder right? Question is, if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53

I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

Hi Kelvin, You can use something like :opus_encoder_ctl(enc,OPUS_SET_BITRATE(bitrate));opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(bandpass)); bandpass is the audio bandpass?, eg: OPUS_BANDWIDTH_WIDEBAND . You will need to calculate the codec bitrate from the available network bitrate (by taking into account the size of the IP+UDP+RTP headers which is 40 bytes ) . By default the audio bandwidth

Hi guys, I have been reading a lot about the "adaptiveness" of opus and i quote: ... can still change, e.g. to adapt to changing network conditions. useinbandfec ... can somebody please enlighten me on this "adaptiveness"? whatever way I do our tests, it sticks to the same sampling rate and the same average bitrate, it would go up, down a bit but that's it. When we get

On Fri, Nov 16, 2018 at 7:40 PM ongaku zettai <sergeinakamoto at gmail.com> wrote: > Hello. > i have over 30GB of Opus music and noticed that > solo instrumentals and solo vocals uses more bitrate > than full-mixes. > Here's example where Opus 1.3 used 190 kbps for > piano solo and 159 kbps for full-mix. > (--bitrate 160 --music) > Download example piano solo

On Nov 17 20:37:21, sergeinakamoto at gmail.com wrote: > Hello. Me again. > Have you tried to encode piano solo? > Noticed high bitrate Opus gave? > Download FLAC and Opus both files, > new link: > http://www.filedropper.com/example_3 > FLAC full: 1084 kbps; > FLAC solo: 465 kbps. > with --bitrate 160: > Opus full: 158 kbps; > Opus solo: 190 kbps. The two

Hello. Me again. Have you tried to encode piano solo? Noticed high bitrate Opus gave? And there's also artefact at 15kHz which wasn't in the original audio. Visible with Spek program. Download FLAC and Opus both files, new link: http://www.filedropper.com/example_3 FLAC full: 1084 kbps; FLAC solo: 465 kbps. with --bitrate 160: Opus full: 158 kbps; Opus solo: 190 kbps. Included also Spek

having a hard time communicating on IRC, thank you gmaxwell, very informative. anyway, we were discussing the proper implementation of FEC on the decoder side. well, encoder side is just a boolean thing so that's alright. i gave an example where the receiver lost 5 rtp packets, 1 2 3 4 and 5 during which, we call opus_decode with a null pointer and fec=0 for every packet lost. now, when it

Hi I am Rupesh from India. I have a huge directory of size 93.5 gb with 8500 mp3 files and 2000 sub directories. All these mp3s are speeches recorded by someone at 64 kbps. I want to compress these files recursively to opus using lame or another tool with 16 kbps bit rate and 11050 sample rate. I have compressed the above huge directory with above options using ffmpeg and the resulted

Hi, First, you can see a comparison of the codec features at http://www.speex.org/comparison.html As for quality/bitrate, the first thing is that Speex supports a lot more settings (from 4 to 42 kbps) and does wideband (16 kHz sampling), which iLBC doesn't do. I've only tested iLBC once, but I've found that Speex has a better quality for the same bit-rate (or lower bit-rate for the

> I am asking this because it is believed that Skype is using some iLBC and > iSAC since GlobalIPSound listed Skype as a partner. I think (from what I've heard) that's what Skype uses. I have no idea how iSac sounds because it's proprietary and I've never used Skype. Jean-Marc > > Thanks, > Joe > > -----Original Message----- > From: Jean-Marc Valin

Hello. i have over 30GB of Opus music and noticed that solo instrumentals and solo vocals uses more bitrate than full-mixes. Here's example where Opus 1.3 used 190 kbps for piano solo and 159 kbps for full-mix. (--bitrate 160 --music) Download example piano solo 15MB: https://mega.nz/#!wLBz3AZT!YmqQMkAGqc4kGHumNWZAfB7Cmcf4vFlHpT6IiiAVCNA FLAC uses 2 times less bitrate for solo than full-mix

comment inline. On Wed, 16 May 2007, Jean-Marc Valin wrote: >> Page 3: >> >> To be compliant with this specification, implementations MUST support >> 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. >> The sampling rate MUST be 8, 16 or 32 kHz. >> >> There is a type above after (narrowband), there is a " extra

hi guys, i'm using sept 30 cvs and oh323 5.5 i'm having 5 second latecy(on only 1 audio path) when a call is transferred.... the scenario is this: sip--------->asterisk----->h323:operator (who then transfers the call) ---------------->h323:destination ------------------audio path 5-second latency---------------->

Here my comments: Page 3: To be compliant with this specification, implementations MUST support 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. The sampling rate MUST be 8, 16 or 32 kHz. There is a type above after (narrowband), there is a " extra character. I don't understand what is the motivation to specify "SHOULD support 8 kbps

Jean-Marc, I am building a tool for producing the highest possible quality Internet interviews for "podcasting" applications. The goal is to produce a perfect recording of an interview or conference -- and giving the participants a glitch-free experience is secondary. My approach, therefore, is to build a Windows "wave" file asynchronously by using a streaming

After more than two years of development, we have released Opus 1.1. This includes: * new analysis code and tuning that significantly improves encoding quality, especially for variable-bitrate (VBR), * automatic detection of speech or music to decide which encoding mode to use, * surround with good quality at 128 kbps for 5.1 and usable down to 48 kbps, and * speed improvements on all

hi guys, have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing? ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030725/ae4b2f25/attachment.htm

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