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Cleaned up WebRTC browser test js, removed unneeded stuff.
This patch merges message_handling.js and jsep01_call.js. The
distinction isn't useful anymore since we used to support multiple
signaling bases like ROAP, we don't anymore. Besides, message_handling
had evolved to practically be the public interface to the test C++,
so this patch makes this role more explicit.
Also the code used to be very flexible to support all kinds of
permutations that we need to do in the manual test page, but since that
code has moved off on its own we can eliminate the flexibility and
thereby greatly simplify the code.
Also improves the error reporting by including the js stack trace
when there's a test failure.
BUG= 375240
Committed: https://src.chromium.org/viewvc/chrome?view=rev&revision=272994
Total comments: 1
Total comments: 2
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chrome/browser/media/chrome_webrtc_browsertest.cc
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chrome/browser/media/webrtc_browsertest_base.cc
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chrome/test/data/webrtc/getusermedia.js
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chrome/test/data/webrtc/jsep01_call.js
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chrome/test/data/webrtc/manual/peerconnection.html
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chrome/test/data/webrtc/manual/peerconnection.js
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chrome/test/data/webrtc/manual/peerconnection_manual.js
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chrome/test/data/webrtc/message_handling.js
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chrome/test/data/webrtc/peerconnection.js
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chrome/test/data/webrtc/test_functions.js
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chrome/test/data/webrtc/webrtc_audio_quality_test.html
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chrome/test/data/webrtc/webrtc_jsep01_test.html
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chrome/test/data/webrtc/webrtc_video_quality_test.html
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Total messages: 14 (0 generated)
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