Merge pull request #13897 from youennf/move-offerToReceive-tests-to-webrtc-legacy
Move some offerToReceive tests to webrtc/legacy
diff --git a/webrtc/RTCRtpReceiver-getStats.https.html b/webrtc/RTCRtpReceiver-getStats.https.html
index 4a2e6a0..05ca9f3 100644
--- a/webrtc/RTCRtpReceiver-getStats.https.html
+++ b/webrtc/RTCRtpReceiver-getStats.https.html
@@ -49,6 +49,12 @@
t.add_cleanup(() => caller.close());
const callee = new RTCPeerConnection();
t.add_cleanup(() => callee.close());
+
+ const stream = await getNoiseStream({audio:true});
+ t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+ const [track] = stream.getTracks();
+ callee.addTrack(track);
+
const { receiver } = caller.addTransceiver('audio');
await doSignalingHandshake(caller, callee);
diff --git a/webrtc/protocol/video-codecs.https.html b/webrtc/protocol/video-codecs.https.html
index 547856f..e728eff 100644
--- a/webrtc/protocol/video-codecs.https.html
+++ b/webrtc/protocol/video-codecs.https.html
@@ -21,7 +21,7 @@
// Section 5: Browsers MUST implement VP8 and H.264 Constrained Baseline
promise_test(async t => {
const pc = new RTCPeerConnection();
- const offer = await pc.createOffer({offerToReceiveVideo: true});
+ const offer = await generateVideoReceiveOnlyOffer(pc);
let video_section_found = false;
for (let section of offer.sdp.split(/\r\nm=/)) {
if (section.search('video') != 0) {
