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

Issue 2167133002: Separates the WebRTC browser tests that deal with audio detection. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Typo DISABLE -> DISABLED Created 4 years, 5 months ago
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Index: content/test/data/media/peerconnection-call-audio.html
diff --git a/content/test/data/media/peerconnection-call-audio.html b/content/test/data/media/peerconnection-call-audio.html
new file mode 100644
index 0000000000000000000000000000000000000000..3c996887f97a42943df0824e3809df531db2f63b
--- /dev/null
+++ b/content/test/data/media/peerconnection-call-audio.html
@@ -0,0 +1,226 @@
+<html>
+<head>
+ <script type="text/javascript" src="webrtc_test_utilities.js"></script>
+ <script type="text/javascript" src="webrtc_test_common.js"></script>
+ <script type="text/javascript" src="webrtc_test_audio.js"></script>
+ <script type="text/javascript">
+ $ = function(id) {
+ return document.getElementById(id);
+ };
+
+ window.onerror = function(errorMsg, url, lineNumber, column, errorObj) {
+ failTest('Error: ' + errorMsg + '\nScript: ' + url +
+ '\nLine: ' + lineNumber + '\nColumn: ' + column +
+ '\nStackTrace: ' + errorObj);
+ }
+
+ var gFirstConnection = null;
+ var gSecondConnection = null;
+ var gLocalStream = null;
+ var gSentTones = '';
+
+ var gRemoteStreams = {};
+
+ setAllEventsOccuredHandler(reportTestSuccess);
+
+ // The second set of constraints should request audio (e.g. audio:true) since
+ // we expect audio to be playing after the second renegotiation.
+ function callAndRenegotiateToAudio(constraints, renegotiationConstraints) {
+ createConnections(null);
+ navigator.webkitGetUserMedia(constraints,
+ addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
+
+ waitForConnectionToStabilize(gFirstConnection, function() {
+ gFirstConnection.removeStream(gLocalStream);
+ gSecondConnection.removeStream(gLocalStream);
+
+ navigator.webkitGetUserMedia(renegotiationConstraints,
+ addStreamToTheFirstConnectionAndNegotiate, printGetUserMediaError);
+
+ var onCallEstablished = function() {
+ ensureAudioPlaying(gSecondConnection);
+ };
+
+ waitForConnectionToStabilize(gFirstConnection, onCallEstablished);
+ });
+ }
+
+ function callAndEnsureAudioIsPlaying(constraints) {
+ createConnections(null);
+
+ // Add the local stream to gFirstConnection to play one-way audio.
+ navigator.webkitGetUserMedia(constraints,
+ addStreamToTheFirstConnectionAndNegotiate, printGetUserMediaError);
+
+ var onCallEstablished = function() {
+ ensureAudioPlaying(gSecondConnection);
+ };
+
+ waitForConnectionToStabilize(gFirstConnection, onCallEstablished);
+ }
+
+ function callWithIsac16KAndEnsureAudioIsPlaying(constraints) {
+ setOfferSdpTransform(function(sdp) {
+ sdp = sdp.replace(/m=audio (\d+) RTP\/SAVPF.*\r\n/g,
+ 'm=audio $1 RTP/SAVPF 103 126\r\n');
+ sdp = sdp.replace('a=fmtp:111 minptime=10', 'a=fmtp:103 minptime=10');
+ if (sdp.search('a=rtpmap:103 ISAC/16000') == -1)
+ failTest('Missing iSAC 16K codec on Android; cannot force codec.');
+
+ return sdp;
+ });
+ callAndEnsureAudioIsPlaying(constraints);
+ }
+
+ function enableRemoteVideo(peerConnection, enabled) {
+ remoteStream = peerConnection.getRemoteStreams()[0];
+ remoteStream.getVideoTracks()[0].enabled = enabled;
+ }
+
+ function enableRemoteAudio(peerConnection, enabled) {
+ remoteStream = peerConnection.getRemoteStreams()[0];
+ remoteStream.getAudioTracks()[0].enabled = enabled;
+ }
+
+ function enableLocalVideo(peerConnection, enabled) {
+ localStream = peerConnection.getLocalStreams()[0];
+ localStream.getVideoTracks()[0].enabled = enabled;
+ }
+
+ function enableLocalAudio(peerConnection, enabled) {
+ localStream = peerConnection.getLocalStreams()[0];
+ localStream.getAudioTracks()[0].enabled = enabled;
+ }
+
+ function callAndEnsureRemoteAudioTrackMutingWorks() {
+ callAndEnsureAudioIsPlaying({audio: true, video: true});
+ setAllEventsOccuredHandler(function() {
+ setAllEventsOccuredHandler(reportTestSuccess);
+
+ // Call is up, now mute the remote track and check we stop playing out
+ // audio (after a small delay, we don't expect it to happen instantly).
+ enableRemoteAudio(gSecondConnection, false);
+ ensureSilence(gSecondConnection);
+ });
+ }
+
+ function callAndEnsureLocalAudioTrackMutingWorks() {
+ callAndEnsureAudioIsPlaying({audio: true, video: true});
+ setAllEventsOccuredHandler(function() {
+ setAllEventsOccuredHandler(reportTestSuccess);
+
+ // Call is up, now mute the local track of the sending side and ensure
+ // the receiving side stops receiving audio.
+ enableLocalAudio(gFirstConnection, false);
+ ensureSilence(gSecondConnection);
+ });
+ }
+
+ function callAndEnsureAudioTrackUnmutingWorks() {
+ callAndEnsureAudioIsPlaying({audio: true, video: true});
+ setAllEventsOccuredHandler(function() {
+ setAllEventsOccuredHandler(reportTestSuccess);
+
+ // 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);
+ }, 500);
+
+ setTimeout(function() {
+ ensureAudioPlaying(gSecondConnection);
+ }, 1500);
+ });
+ }
+
+ function callAndEnsureLocalVideoMutingDoesntMuteAudio() {
+ callAndEnsureAudioIsPlaying({audio: true, video: true});
+ setAllEventsOccuredHandler(function() {
+ setAllEventsOccuredHandler(reportTestSuccess);
+ enableLocalVideo(gFirstConnection, false);
+ ensureAudioPlaying(gSecondConnection);
+ });
+ }
+
+ function callAndEnsureRemoteVideoMutingDoesntMuteAudio() {
+ callAndEnsureAudioIsPlaying({audio: true, video: true});
+ setAllEventsOccuredHandler(function() {
+ setAllEventsOccuredHandler(reportTestSuccess);
+ enableRemoteVideo(gSecondConnection, false);
+ ensureAudioPlaying(gSecondConnection);
+ });
+ }
+
+ function createConnections(constraints) {
+ gFirstConnection = createConnection(constraints, 'remote-view-1');
+ assertEquals('stable', gFirstConnection.signalingState);
+
+ gSecondConnection = createConnection(constraints, 'remote-view-2');
+ assertEquals('stable', gSecondConnection.signalingState);
+ }
+
+ function createConnection(constraints, remoteView) {
+ var pc = new webkitRTCPeerConnection(null, constraints);
+ pc.onaddstream = function(event) {
+ onRemoteStream(event, remoteView);
+ }
+ return pc;
+ }
+
+ function displayAndRemember(localStream) {
+ var localStreamUrl = URL.createObjectURL(localStream);
+ $('local-view').src = localStreamUrl;
+
+ gLocalStream = localStream;
+ }
+
+ // Called if getUserMedia succeeds and we want to send from both connections.
+ function addStreamToBothConnectionsAndNegotiate(localStream) {
+ displayAndRemember(localStream);
+ gFirstConnection.addStream(localStream);
+ gSecondConnection.addStream(localStream);
+ negotiate();
+ }
+
+ // Called if getUserMedia succeeds when we want to send from one connection.
+ function addStreamToTheFirstConnectionAndNegotiate(localStream) {
+ displayAndRemember(localStream);
+ gFirstConnection.addStream(localStream);
+ negotiate();
+ }
+
+ function negotiate() {
+ negotiateBetween(gFirstConnection, gSecondConnection);
+ }
+
+ function onRemoteStream(e, target) {
+ console.log("Receiving remote stream...");
+ gRemoteStreams[target] = e.stream;
+ var remoteStreamUrl = URL.createObjectURL(e.stream);
+ var remoteVideo = $(target);
+ remoteVideo.src = remoteStreamUrl;
+ }
+
+ </script>
+</head>
+<body>
+ <table border="0">
+ <tr>
+ <td><video width="320" height="240" id="local-view" style="display:none"
+ autoplay muted></video></td>
+ <td><video width="320" height="240" id="remote-view-1"
+ style="display:none" autoplay></video></td>
+ <td><video width="320" height="240" id="remote-view-2"
+ style="display:none" autoplay></video></td>
+ <!-- Canvases are named after their corresponding video elements. -->
+ <td><canvas width="320" height="240" id="remote-view-1-canvas"
+ style="display:none"></canvas></td>
+ <td><canvas width="320" height="240" id="remote-view-2-canvas"
+ style="display:none"></canvas></td>
+ </tr>
+ </table>
+</body>
+</html>
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