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Bug 1166832 - Add test to verify video (using capture stream) after renegotiation. r=bwc
--HG-- extra : transplant_source : I%AB%28%AA%D6%91%8F%CA5%8A%D9%0C%F2%8A%CD%B8m%DE%D7%05 extra : histedit_source : 1d30b92a3c6d128c24cfbb1f8778c665a6bb9eca
This commit is contained in:
Michael Froman
parent
6abe3c833c
commit
13a82c11bf
@ -209,10 +209,11 @@ skip-if = toolkit == 'gonk' || android_version == '18'
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skip-if = toolkit == 'gonk' # B2G emulator seems to be so slow that DTLS cannot establish properly
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[test_peerConnection_addDataChannelNoBundle.html]
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skip-if = toolkit == 'gonk' || (android_version == '18' && debug) # b2g(emulator seems to be so slow that DTLS cannot establish properly), android(bug 1240256, intermittent ICE failures starting w/bug 1232082, possibly from timeout)
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# b2g(Bug 960442, video support for WebRTC is disabled on b2g), android(Bug 1189784, timeouts on 4.3 emulator)
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[test_peerConnection_verifyVideoAfterRenegotiation.html]
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# B2G emulator is too slow to finish a renegotiation test in under 5 minutes, Bug 1180000 for Linux debug e10s, android(Bug 1189784, timeouts on 4.3 emulator)
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skip-if = toolkit == 'gonk' || android_version == '18'
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[test_peerConnection_webAudio.html]
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# b2g(Bug 960442, video support for WebRTC is disabled on b2g), android(Bug 1189784, timeouts on 4.3 emulator)
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tags = webaudio webrtc
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skip-if = toolkit == 'gonk' || buildapp == 'mulet' # b2g (Bug 1059867)
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[test_peerConnection_localRollback.html]
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@ -0,0 +1,100 @@
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<!DOCTYPE HTML>
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<html>
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<head>
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<script type="application/javascript" src="pc.js"></script>
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<script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
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</head>
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<body>
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<pre id="test">
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<script type="application/javascript">
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createHTML({
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bug: "1166832",
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title: "Renegotiation: verify video after renegotiation"
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});
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runNetworkTest(() => {
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var test = new PeerConnectionTest();
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var h1 = new CaptureStreamTestHelper2D(50, 50);
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var canvas1 = h1.createAndAppendElement('canvas', 'source_canvas1');
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var stream1;
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var vremote1;
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var h2 = new CaptureStreamTestHelper2D(50, 50);
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var canvas2;
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var stream2;
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var vremote2;
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test.setMediaConstraints([{video: true}], []);
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test.chain.replace("PC_LOCAL_GUM", [
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function DRAW_INITIAL_LOCAL_GREEN(test) {
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h1.drawColor(canvas1, h1.green);
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},
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function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
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stream1 = canvas1.captureStream(0);
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test.pcLocal.attachMedia(stream1, 'video', 'local');
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}
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]);
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test.chain.append([
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function FIND_REMOTE_VIDEO() {
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vremote1 = document.getElementById('pcRemote_remote1_video');
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ok(!!vremote1, "Should have remote video element for pcRemote");
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},
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function WAIT_FOR_REMOTE_GREEN() {
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return h1.waitForPixelColor(vremote1, h1.green, 128,
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"pcRemote's remote should become green");
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},
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function DRAW_LOCAL_RED() {
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// After requesting a frame it will be captured at the time of next render.
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// Next render will happen at next stable state, at the earliest,
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// i.e., this order of `requestFrame(); draw();` should work.
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stream1.requestFrame();
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h1.drawColor(canvas1, h1.red);
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},
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function WAIT_FOR_REMOTE_RED() {
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return h1.waitForPixelColor(vremote1, h1.red, 128,
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"pcRemote's remote should become red");
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}
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]);
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addRenegotiation(test.chain,
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[
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function PC_LOCAL_ADD_SECOND_STREAM(test) {
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canvas2 = h2.createAndAppendElement('canvas', 'source_canvas2');
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h2.drawColor(canvas2, h2.blue);
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stream2 = canvas2.captureStream(0);
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// can't use test.pcLocal.getAllUserMedia([{video: true}]);
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// because it doesn't let us substitute the capture stream
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return test.pcLocal.attachMedia(stream2, 'video', 'local');
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}
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]
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);
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test.chain.append([
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function FIND_REMOTE2_VIDEO() {
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vremote2 = document.getElementById('pcRemote_remote2_video');
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ok(!!vremote2, "Should have remote2 video element for pcRemote");
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},
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function WAIT_FOR_REMOTE2_BLUE() {
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return h2.waitForPixelColor(vremote2, h2.blue, 128,
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"pcRemote's remote2 should become blue");
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},
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function DRAW_NEW_LOCAL_GREEN(test) {
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stream1.requestFrame();
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h1.drawColor(canvas1, h1.green);
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},
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function WAIT_FOR_REMOTE1_GREEN() {
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return h1.waitForPixelColor(vremote1, h1.green, 128,
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"pcRemote's remote1 should become green");
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}
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]);
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test.run();
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});
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</script>
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</pre>
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</body>
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</html>
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@ -1144,6 +1144,8 @@ WebrtcVideoConduit::SelectSendResolution(unsigned short width,
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bool changed = false;
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if (mSendingWidth != width || mSendingHeight != height)
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{
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CSFLogDebug(logTag, "%s: resolution changing to %ux%u (from %ux%u)",
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__FUNCTION__, width, height, mSendingWidth, mSendingHeight);
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// This will avoid us continually retrying this operation if it fails.
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// If the resolution changes, we'll try again. In the meantime, we'll
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// keep using the old size in the encoder.
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@ -1155,6 +1157,8 @@ WebrtcVideoConduit::SelectSendResolution(unsigned short width,
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// uses mSendingWidth/Height
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unsigned int framerate = SelectSendFrameRate(mSendingFramerate);
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if (mSendingFramerate != framerate) {
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CSFLogDebug(logTag, "%s: framerate changing to %u (from %u)",
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__FUNCTION__, framerate, mSendingFramerate);
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mSendingFramerate = framerate;
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changed = true;
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}
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@ -1221,6 +1225,10 @@ WebrtcVideoConduit::ReconfigureSendCodec(unsigned short width,
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CSFLogError(logTag, "%s: GetSendCodec failed, err %d", __FUNCTION__, err);
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return NS_ERROR_FAILURE;
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}
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CSFLogDebug(logTag,
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"%s: Requesting resolution change to %ux%u (from %ux%u)",
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__FUNCTION__, width, height, vie_codec.width, vie_codec.height);
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// Likely spurious unless there was some error, but rarely checked
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if (vie_codec.width != width || vie_codec.height != height ||
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vie_codec.maxFramerate != mSendingFramerate)
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@ -1399,6 +1407,8 @@ WebrtcVideoConduit::SendVideoFrame(webrtc::I420VideoFrame& frame)
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return kMediaConduitNoError;
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}
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if (frame.width() != mLastWidth || frame.height() != mLastHeight) {
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CSFLogDebug(logTag, "%s: call SelectSendResolution with %ux%u",
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__FUNCTION__, frame.width(), frame.height());
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if (SelectSendResolution(frame.width(), frame.height(), &frame)) {
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// SelectSendResolution took ownership of the data in i420_frame.
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// Submit the frame after reconfig is done
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