content/test/data/media/peerconnection-call.html - Issue 2167133002: Separates the WebRTC browser tests that deal with audio detection.

Unified Diff: content/test/data/media/peerconnection-call.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: Previous patch was moved to it's own cl. Created 4 years, 5 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 b0b6758d399312fd617bb60f3b16b10a52f3a2ab..9782ef90091d3d5dbb42aad0fbea734626aad7c8 100644

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

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

@@ -2,7 +2,6 @@

<head>

- <script type="text/javascript" src="webrtc_test_audio.js"></script>

$ = function(id) {

return document.getElementById(id);

@@ -22,7 +21,6 @@

var gRemoteStreams = {};

setAllEventsOccuredHandler(reportTestSuccess);

// Test that we can setup a call with an audio and video track (must request

@@ -99,28 +97,6 @@

});

}

- // 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);

- });

- }

// First makes a call between pc1 and pc2 where a stream is sent from pc1 to

// pc2. The stream sent from pc1 to pc2 is cloned from the stream received on

// pc2 to test that cloning of remote video tracks works as intended and is

@@ -270,33 +246,6 @@

offerOptions);

}

- 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;

@@ -307,78 +256,6 @@

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 callAndEnsureVideoTrackMutingWorks() {

createConnections(null);

navigator.webkitGetUserMedia({audio: true, video: true},

@@ -710,7 +587,6 @@

return sdp.replace(/a=group:BUNDLE .*\r\n/g, '');

}

function onRemoteStream(e, target) {

console.log("Receiving remote stream...");

if (gTestWithoutMsid && e.stream.id != "default") {