Wrote a test which exercises audio-only WebRTC calls.

Depends on https://codereview.chromium.org/260073013/ since it requires audio beeps generated even when the fake video device hasn't been acquired.

BUG=None

Review URL: https://codereview.chromium.org/252703003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@275790 0039d316-1c4b-4281-b951-d872f2087c98

content/browser/media/webrtc_browsertest.cc

@@ -59,7 +59,7 @@ class WebRtcBrowserTest : public WebRtcContentBrowserTest,
 #endif
   }
 
-  // Convenience function since most peerconnection-call.html tests just load
+  // Convenience method since most peerconnection-call.html tests just load
   // the page, kick off some javascript and wait for the title to change to OK.
   void MakeTypicalPeerConnectionCall(const std::string& javascript) {
     ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
@@ -71,6 +71,27 @@ class WebRtcBrowserTest : public WebRtcContentBrowserTest,
     ExecuteJavascriptAndWaitForOk(javascript);
   }
 
+  // Convenience method for making calls that detect if audio os playing (which
+  // has some special prerequisites, such that there needs to be an audio output
+  // device on the executing machine).
+  void MakeAudioDetectingPeerConnectionCall(const std::string& javascript) {
+    if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
+      // Bots with no output devices will force the audio code into a state
+      // where it doesn't manage to set either the low or high latency path.
+      // This test will compute useless values in that case, so skip running on
+      // such bots (see crbug.com/326338).
+      LOG(INFO) << "Missing output devices: skipping test...";
+      return;
+    }
+
+    ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
+        switches::kUseFakeDeviceForMediaStream))
+            << "Must run with fake devices since the test will explicitly look "
+            << "for the fake device signal.";
+
+    MakeTypicalPeerConnectionCall(javascript);
+  }
+
   void DisableOpusIfOnAndroid() {
 #if defined(OS_ANDROID)
     // Always force iSAC 16K on Android for now (Opus is broken).
@@ -327,39 +348,22 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, AddTwoMediaStreamsToOnePC) {
 }
 
 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
-                       EstablishAudioVideoCallAndMeasureOutputLevel) {
-  if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
-    // Bots with no output devices will force the audio code into a different
-    // path where it doesn't manage to set either the low or high latency path.
-    // This test will compute useless values in that case, so skip running on
-    // such bots (see crbug.com/326338).
-    LOG(INFO) << "Missing output devices: skipping test...";
-    return;
-  }
-
-  ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
-      switches::kUseFakeDeviceForMediaStream))
-          << "Must run with fake devices since the test will explicitly look "
-          << "for the fake device signal.";
+                       EstablishAudioVideoCallAndEnsureAudioIsPlaying) {
+  MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
+      "callAndEnsureAudioIsPlaying(%s, {audio:true, video:true});",
+      kUseLenientAudioChecking));
+}
 
-  MakeTypicalPeerConnectionCall(base::StringPrintf(
-      "callAndEnsureAudioIsPlaying(%s);", kUseLenientAudioChecking));
+IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
+                       EstablishAudioOnlyCallAndEnsureAudioIsPlaying) {
+  MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
+      "callAndEnsureAudioIsPlaying(%s, {audio:true});",
+      kUseLenientAudioChecking));
 }
 
 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
                        EstablishAudioVideoCallAndVerifyMutingWorks) {
-  if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
-    // See comment on EstablishAudioVideoCallAndMeasureOutputLevel.
-    LOG(INFO) << "Missing output devices: skipping test...";
-    return;
-  }
-
-  ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
-      switches::kUseFakeDeviceForMediaStream))
-          << "Must run with fake devices since the test will explicitly look "
-          << "for the fake device signal.";
-
-  MakeTypicalPeerConnectionCall(base::StringPrintf(
+  MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureAudioTrackMutingWorks(%s);", kUseLenientAudioChecking));
 }
 
@@ -373,18 +377,7 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 #endif
 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
                        MAYBE_EstablishAudioVideoCallAndVerifyUnmutingWorks) {
-  if (!media::AudioManager::Get()->HasAudioOutputDevices()) {
-    // See comment on EstablishAudioVideoCallAndMeasureOutputLevel.
-    LOG(INFO) << "Missing output devices: skipping test...";
-    return;
-  }
-
-  ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
-      switches::kUseFakeDeviceForMediaStream))
-          << "Must run with fake devices since the test will explicitly look "
-          << "for the fake device signal.";
-
-  MakeTypicalPeerConnectionCall(base::StringPrintf(
+  MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureAudioTrackUnmutingWorks(%s);", kUseLenientAudioChecking));
 }
 

content/test/data/media/peerconnection-call.html

@@ -301,9 +301,9 @@
     remoteAudioTrack.enabled = enabled;
   }
 
-  function callAndEnsureAudioIsPlaying(beLenient) {
+  function callAndEnsureAudioIsPlaying(beLenient, constraints) {
     createConnections(null);
-    navigator.webkitGetUserMedia({audio: true, video: true},
+    navigator.webkitGetUserMedia(constraints,
       addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 
     // Wait until we have gathered samples and can conclude if audio is playing.
@@ -314,16 +314,13 @@
         verifyAudioIsPlaying(samples, beLenient);
         eventOccured();
       });
-
-      // (Also, ensure video muting doesn't affect audio).
-      enableRemoteVideo(gSecondConnection, false);
     };
 
-    detectVideoPlaying('remote-view-2', onCallEstablished);
+    waitForConnectionToStabilize(gFirstConnection, onCallEstablished);
   }
 
   function callAndEnsureAudioTrackMutingWorks(beLenient) {
-    callAndEnsureAudioIsPlaying(beLenient);
+    callAndEnsureAudioIsPlaying(beLenient, {audio: true, video: true});
     setAllEventsOccuredHandler(function() {
       // Call is up, now mute the track and check everything goes silent (give
       // it a small delay though, we don't expect it to happen instantly).
@@ -339,10 +336,12 @@
   }
 
   function callAndEnsureAudioTrackUnmutingWorks(beLenient) {
-    callAndEnsureAudioIsPlaying(beLenient);
+    callAndEnsureAudioIsPlaying(beLenient, {audio: true, video: true});
     setAllEventsOccuredHandler(function() {
       // Mute, wait a while, unmute, verify audio gets back up.
+      // (Also, ensure video muting doesn't affect audio).
       enableRemoteAudio(gSecondConnection, false);
+      enableRemoteVideo(gSecondConnection, false);
 
       setTimeout(function() {
         enableRemoteAudio(gSecondConnection, true);

content/test/data/media/webrtc_test_audio.js

@@ -15,7 +15,16 @@ function gatherAudioLevelSamples(peerConnection, numSamples, frequency,
   var audioLevelSamples = []
   var gatherSamples = setInterval(function() {
     peerConnection.getStats(function(response) {
-      audioLevelSamples.push(getAudioLevelFromStats_(response));
+      audioOutputLevels = getAudioLevelFromStats_(response);
+      if (audioOutputLevels.length == 0) {
+        // The call probably isn't up yet.
+        return;
+      }
+
+      // If more than one audio level is reported we get confused.
+      assertEquals(1, audioOutputLevels.length);
+      audioLevelSamples.push(audioOutputLevels[0]);
+
       if (audioLevelSamples.length == numSamples) {
         console.log('Gathered all samples.');
         clearInterval(gatherSamples);
@@ -83,8 +92,5 @@ function getAudioLevelFromStats_(response) {
       audioOutputLevels.push(report.stat('audioOutputLevel'));
     }
   }
-  // Should only be one audio level reported, otherwise we get confused.
-  assertEquals(1, audioOutputLevels.length);
-
-  return audioOutputLevels[0];
+  return audioOutputLevels;
 }