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Unified Diff: content/test/data/media/peerconnection-call.html

Issue 216523003: Tighten up webrtc audio tests. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Created 6 years, 9 months ago
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Index: content/test/data/media/peerconnection-call.html
diff --git a/content/test/data/media/peerconnection-call.html b/content/test/data/media/peerconnection-call.html
index 29a23b8f9151bde9b034f3b21a6c4eb347144d21..85b0ea636a551da6c3ee9555854db2aa1a3fed50 100644
--- a/content/test/data/media/peerconnection-call.html
+++ b/content/test/data/media/peerconnection-call.html
@@ -291,8 +291,8 @@
// Wait until we have gathered samples and can conclude if audio is playing.
addExpectedEvent();
var onCallEstablished = function() {
- gatherAudioLevelSamples(gSecondConnection, 300, 100,
- function(samples) {
+ // Gather 10 samples per second for 4 seconds.
+ gatherAudioLevelSamples(gSecondConnection, 40, 10, function(samples) {
phoglund_chromium 2014/03/28 13:55:26 Ok, I doubled the beep frequency and halved the ga
verifyAudioIsPlaying(samples);
eventOccured();
});
@@ -312,7 +312,7 @@
enableRemoteAudio(gSecondConnection, false);
setTimeout(function() {
- gatherAudioLevelSamples(gSecondConnection, 200, 100, function(samples) {
+ gatherAudioLevelSamples(gSecondConnection, 20, 10, function(samples) {
verifyIsSilent(samples);
reportTestSuccess();
});
@@ -331,7 +331,7 @@
}, 500);
setTimeout(function() {
- gatherAudioLevelSamples(gSecondConnection, 200, 100, function(samples) {
+ gatherAudioLevelSamples(gSecondConnection, 40, 10, function(samples) {
perkj_chrome 2014/03/28 13:29:17 why is this 40 samples to check for playing while
phoglund_chromium 2014/03/28 13:55:26 We need more of a safety margin when determining i
verifyAudioIsPlaying(samples);
reportTestSuccess();
});
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