diff --git a/samples/js/demos/html/webaudio-and-webrtc.html b/samples/js/demos/html/webaudio-and-webrtc.html new file mode 100644 index 000000000..173a30845 --- /dev/null +++ b/samples/js/demos/html/webaudio-and-webrtc.html @@ -0,0 +1,254 @@ + + +
+ +The audio stream is:
+ o Recorded using live-audio
+ input.
+ o Filtered using an HP filter with fc=1500 Hz.
+ o Encoded using
+ Opus.
+ o Transmitted (in loopback) to remote peer using
+ RTCPeerConnection where it is decoded.
+ o Finally, the received remote stream is used as source to an <audio>
+ tag and played out locally.
+
Press any key to add an effect to the transmitted audio while talking.
+
Please note that:
+ o Linux is currently not supported.
+ o Sample rate and channel configuration must be the same for input and
+ output sides on Windows.
+ o Only the Default microphone device can be used for capturing.
+
For more information, see + WebRTC integration with the Web Audio API. +
+ +