webrtc-sendrecv.py: improve debug and documentation

diff --git a/webrtc/README.md b/webrtc/README.md
index 5456c90..20e3488 100644 (file)
--- a/webrtc/README.md
+++ b/webrtc/README.md
@@ -74,6 +74,12 @@ $ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstrea
 * python3 -m pip install --user websockets
 * run `python3 sendrecv/gst/webrtc-sendrecv.py ID` with the `id` from the browser. You will see state changes and an SDP exchange.
 
+> The python version currently requires the master branches from `gst-plugins-bad` and `gst-plugins-base`.
+
+<!---
+TODO: replace the note above when 1.16 is released
+-->
+
 With all versions, you will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app.
 
 You can pass a --server argument to all versions, for example `--server=wss://127.0.0.1:8443`.

diff --git a/webrtc/sendrecv/gst/webrtc-sendrecv.py b/webrtc/sendrecv/gst/webrtc-sendrecv.py
index dee4800..83b49ae 100644 (file)
--- a/webrtc/sendrecv/gst/webrtc-sendrecv.py
+++ b/webrtc/sendrecv/gst/webrtc-sendrecv.py
@@ -42,7 +42,7 @@ class WebRTCClient:
 
     def send_sdp_offer(self, offer):
         text = offer.sdp.as_text()
-        print ('Sending offer:\n%s', text)
+        print ('Sending offer:\n%s' % text)
         msg = json.dumps({'sdp': {'type': 'offer', 'sdp': text}})
         loop = asyncio.new_event_loop()
         loop.run_until_complete(self.conn.send(msg))
@@ -120,7 +120,7 @@ class WebRTCClient:
             sdp = msg['sdp']
             assert(sdp['type'] == 'answer')
             sdp = sdp['sdp']
-            print ('Received answer:\n%s', sdp)
+            print ('Received answer:\n%s' % sdp)
             res, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp)
             answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
             promise = Gst.Promise.new()