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Commit 93e6ab32 authored by Florent Castelli's avatar Florent Castelli Committed by Commit Bot
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Remove crash expectation on a RTCRtpSender.setParameters test 

The bug it was blocked on is now fixed, merging with the other tests.

Bug: webrtc/9141
Change-Id: If662ef9dfc4f58c0d8d43b41d32f745212ef462f
Reviewed-on: https://chromium-review.googlesource.com/1057668Reviewed-by: default avatarHenrik Boström <hbos@chromium.org>
Commit-Queue: Florent Castelli <orphis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#558319}
parent 574fe4e3
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Showing with 18 additions and 46 deletions
third_party/WebKit/LayoutTests/TestExpectations
View file @ 93e6ab32
...@@ -4329,8 +4329,6 @@ crbug.com/771540 external/wpt/webrtc/RTCPeerConnection-setLocalDescription-answe ...@@ -4329,8 +4329,6 @@ crbug.com/771540 external/wpt/webrtc/RTCPeerConnection-setLocalDescription-answe
# Timeout due to events not firing # Timeout due to events not firing
crbug.com/781935 fast/mediastream/MediaStreamTrack-observer-iterate-no-crash.html [ Timeout ] crbug.com/781935 fast/mediastream/MediaStreamTrack-observer-iterate-no-crash.html [ Timeout ]
crbug.com/781935 fast/mediastream/MediaStreamTrack.html [ Timeout ] crbug.com/781935 fast/mediastream/MediaStreamTrack.html [ Timeout ]
# DCHECK hit in WebRTC
crbug.com/webrtc/9141 fast/peerconnection/RTCRtpSender-setParameters-crash.html [ Crash Timeout Pass ]
crbug.com/807191 fast/media/mq-color-gamut-picture.html [ Pass Failure Timeout ] crbug.com/807191 fast/media/mq-color-gamut-picture.html [ Pass Failure Timeout ]
......
third_party/WebKit/LayoutTests/fast/peerconnection/RTCRtpSender-setParameters-crash.html deleted 100644 → 0
View file @ 574fe4e3
<!DOCTYPE html>
<html>
<head>
<title>RTCPeerConnection.getParameters</title>
<script src="../../resources/testharness.js"></script>
<script src="../../resources/testharnessreport.js"></script>
</head>
<body>
<script>
// TODO(orphis): Move test to main test file after https://crbug.com/webrtc/9141 is fixed
// WebRTC currently DCHECKs for a minimum maxBitrate value (currently 30000 bps).
// This can accidentally be hit when mixing units (Kbps vs B/s vs bps).
promise_test(async function() {
let pc1 = new RTCPeerConnection();
let pc2 = new RTCPeerConnection();
let stream = await navigator.mediaDevices.getUserMedia({video:true});
let videoSender = pc1.addTrack(stream.getVideoTracks()[0], stream);
await doSignalingHandshake(pc1, pc2);
let videoParameters = videoSender.getParameters();
videoParameters.encodings[0].maxBitrate = 50;
await videoSender.setParameters(videoParameters);
videoParameters = videoSender.getParameters();
assert_equals(videoParameters.encodings[0].maxBitrate, 50);
}, 'setParameters() set low maxBitrate value');
/**
* Helper functions to tests.
*/
async function doSignalingHandshake(localPc, remotePc) {
let offer = await localPc.createOffer();
await localPc.setLocalDescription(offer);
await remotePc.setRemoteDescription(offer);
let answer = await remotePc.createAnswer();
await remotePc.setLocalDescription(answer);
await localPc.setRemoteDescription(answer);
}
</script>
</body>
</html>
third_party/WebKit/LayoutTests/fast/peerconnection/RTCRtpSender-setParameters-expected.txt
View file @ 93e6ab32
...@@ -69,5 +69,6 @@ PASS audio setParameters() check for headerExtensions.encrypted removed ...@@ -69,5 +69,6 @@ PASS audio setParameters() check for headerExtensions.encrypted removed
PASS audio setParameters() check for rtcp removed PASS audio setParameters() check for rtcp removed
PASS audio setParameters() check for rtcp.cname modification PASS audio setParameters() check for rtcp.cname modification
PASS audio setParameters() check for rtcp.reducedSize modification PASS audio setParameters() check for rtcp.reducedSize modification
PASS setParameters() set low maxBitrate value
Harness: the test ran to completion. Harness: the test ran to completion.
third_party/WebKit/LayoutTests/fast/peerconnection/RTCRtpSender-setParameters.html
View file @ 93e6ab32
...@@ -314,6 +314,23 @@ for (testCase of readOnlyParameters) { ...@@ -314,6 +314,23 @@ for (testCase of readOnlyParameters) {
}).bind(undefined, testCase.name, testCase.transform, testCase.check), }).bind(undefined, testCase.name, testCase.transform, testCase.check),
'audio setParameters() check for ' + testCase.name); 'audio setParameters() check for ' + testCase.name);
} }
promise_test(async function() {
let pc1 = new RTCPeerConnection();
let pc2 = new RTCPeerConnection();
let stream = await navigator.mediaDevices.getUserMedia({video:true});
let videoSender = pc1.addTrack(stream.getVideoTracks()[0], stream);
await doSignalingHandshake(pc1, pc2);
let videoParameters = videoSender.getParameters();
videoParameters.encodings[0].maxBitrate = 50;
await videoSender.setParameters(videoParameters);
videoParameters = videoSender.getParameters();
assert_equals(videoParameters.encodings[0].maxBitrate, 50);
}, 'setParameters() set low maxBitrate value');
/** /**
* Helper functions to tests. * Helper functions to tests.
*/ */
......
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