Audio Quality - Opus Codec #208
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Hi! I think there's a discussion about this here: #177 I've not yet determined the best way to go about allowing adequate, useful codec configuration within PeerJS, but am actively thinking about it. What sort of interface would be most useful for you? How much would you need to customize this? |
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Michelle, Thanks for the reply! It would be good to have granular and explicit control over the codec and bitrate (quality) of the audio connection. If you like, an override variable that sticks the bitrate at a fixed position. The samplerate can remain at 48k, that's fine. I'd like to work on a high quality audio link between studios, so ideally we can choose increments up to 320kbps (if possible?), although for the Opus codec that may be quite high considering it's quality already. Anything higher than the default... I feel the current default Opus connection quality is too low, and audio is cut off at round 15kHz, and not completely wide band (20Hz - 20kHz). So if we can have access to these switches: codec: <-- (choose the codecs available with WebRTC) I think those simple switches will open up a lot of possibilities, especially for those developers who want to make broadcast applications (like me!) Let me know if this is possible. I've seen people achieve this, but not much info really. |
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Hi Michelle, Did you have any further thought regarding this? Do you have any beta builds that we can play around with yet =) Cheers |
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I think just codec set is enough. Webrtc js app code in https://bitbucket.org/webrtc/codelab and other examples, do have preferOpus() or preferISAC(). PeerJS just need provide the same thing. It impact the Android audio quality very much. I am writing the code and after test I will send a push. I just want to see how you think about this? Thanks! |
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Hi Will, Good to hear from you. Having a bandwidth control would be good as well, to Thanks On Thu, Nov 6, 2014 at 2:23 PM, Will Lee notifications@github.com wrote:
Marc Bakos _UK Land: +44 1932 80 80 20UK Mob: +44 7 909 514 505South Africa: +27 11 |
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this is ok, but different codec have.different bandwidth range. I will do the rangr check. --- 原始邮件 --- 发件人： "Marc5000" notifications@github.com Hi Will, Good to hear from you. Having a bandwidth control would be good as well, to Thanks On Thu, Nov 6, 2014 at 2:23 PM, Will Lee notifications@github.com wrote:
Marc Bakos _UK Land: +44 1932 80 80 20UK Mob: +44 7 909 514 505South Africa: +27 11 www.marcbakos.com Reply to this email directly or view it on GitHub: |
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Okay, or even if you just have the bandwidth control for Opus, as that is probably going to be the most popular case where users will need to change the bandwidth, manually. Thanks
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OK, I got it. --- 原始邮件 --- 发件人： "Marc5000" notifications@github.com Okay, or even if you just have the bandwidth control for Opus, as that is probably going to be the most popular case where users will need to change the bandwidth, manually. Thanks
Reply to this email directly or view it on GitHub: |
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Wow,thanks! This would be great :). Michelle On Thu, Nov 6, 2014 at 7:13 AM, Will Lee notifications@github.com wrote:
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What is the status of this? This is one of the most important features and it takes month to figure out how to config codec and bandwidth? I guess i have to change to another library as can't wait months to control the bandwidth LOL amazing.. Yeah peerjs is really cool i love it but when you totally blow off the ability to control bandwidth I don't understand this at all. |
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Hi, any have a fix on this? . Any way to set codec (Opus )for peerjs audio calls. |
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We added sdpTransform parameter to Connection's options, try use it. |
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@afrokick sdpTransform does not seem to work properly. For example, when I transform the sdp in peer.call to add a "b=AS:128" line to specify a 128kbps bitrate, the application executing peer.call sets that locally, but the peer.answer client does not have that line locally. In effect, bitrate never actually changes since only one of the two clients had their SDP transformed. This seems to also happen for a=fmtp parameters; sdptransform transforms the sdp for the client executing peer.call, but it is not negotiated with the client receiving the call. |
Hi,
I'd like to know if there is any granular control over the audio quality using PeerJS? I'd like to send audio in stereo at 128kb with the Opus codec to a connecting client. Is this possible to enter those parameters with PeerJS? Or any other way to achieve this?
Regards
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