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Opus 1.2 Audio Codec Officially Released

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  • Opus 1.2 Audio Codec Officially Released

    Phoronix: Opus 1.2 Audio Codec Officially Released

    Following the earlier development releases, Opus 1.2 is now official...

    http://www.phoronix.com/scan.php?pag...s-1.2-Released

  • #2
    I just read that Safari supports Opus now for WebRTC: https://webkit.org/blog/7726/announc...media-capture/

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    • #3
      And look at what you can do with 700 b/s: http://www.rowetel.com/?p=5373

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      • #4
        Opus is turning into an outlier codec.
        Fascinating.

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        • #5
          You should check the link in the article, they have audio samples, it is awesome

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          • #6
            Originally posted by ellisgl View Post
            And look at what you can do with 700 b/s: http://www.rowetel.com/?p=5373
            The problem with very low bitrates is that on the internet you wouldn't be sending completely filled up packets and would waste a portion of bits that you could use for speech instead. This is why Opus (being a codec for the internet) doesn't even try to go very low. Codec2 on the other hand is a codec created for HAM radio where every bit counts.

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            • #7
              Originally posted by quikee View Post

              The problem with very low bitrates is that on the internet you wouldn't be sending completely filled up packets and would waste a portion of bits that you could use for speech instead. This is why Opus (being a codec for the internet) doesn't even try to go very low. Codec2 on the other hand is a codec created for HAM radio where every bit counts.
              What if (in an online game) player position could be sent in that empty space?

              Edit: Wait, what you said doesn't matter if we're talking about low latency because you can send the data at a higher frequency in smaller packets.
              Last edited by profoundWHALE; 06-21-2017, 04:27 AM.

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              • #8
                Originally posted by profoundWHALE View Post
                Edit: Wait, what you said doesn't matter if we're talking about low latency because you can send the data at a higher frequency in smaller packets.
                The TCP/IP overhead is 64 bytes per packet (UDP 52 bytes). At 700 bits/s the TCP overhead is so huge (when sending more then 1 packet per second) that it makes no real sense to use a so low bandwidth codec to begin with. 10 UDP packets per second are already over 4kbit/s. So at ~5kbit you need an audio-ping of ~230ms which is pretty bad. Assuming 30ms internet ping, if you want to get below 100ms total audio-ping, each packet can only be 35ms long. So you have to send around 30 packets per second, that's around 12kbit/s just for the UDP overhead. At that rate it doesn't make a huge difference, if you have a codec that uses 0.7 or 4 kbit/s. The lowest Opus sample they provide starts at 12kbit/s, that means each packet is half data, half overhead.

                Sure if you fill the packet with other data (player position and stuff) and can fit the complete game state including audio into one packet (~1500 bytes max) then it may make a difference to save a few byte on the audio.

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                • #9
                  Well I was trying to compare audio bit rate to actual audio quality. And if you to have it as a Server Client, you could push from the server to the client a multi channel stream (1 channel per user/voice - so you could record pod casts that are editable.)

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                  • #10
                    I still believe it would be useful for the odd time that one is extremely bandwidth limited or with a very poor connection as codec2 is incredible when dealing with packet loss.

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