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Unified Diff: chrome/test/data/webrtc/message_handling.js

Issue 293123009: Cleaned up WebRTC browser test js, removed unneeded stuff. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Rebased Created 6 years, 7 months ago
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Index: chrome/test/data/webrtc/message_handling.js
diff --git a/chrome/test/data/webrtc/message_handling.js b/chrome/test/data/webrtc/message_handling.js
deleted file mode 100644
index 6329ed303480e5863dcfbecbe23664f2becfa87b..0000000000000000000000000000000000000000
--- a/chrome/test/data/webrtc/message_handling.js
+++ /dev/null
@@ -1,266 +0,0 @@
-/**
- * Copyright (c) 2012 The Chromium Authors. All rights reserved.
- * Use of this source code is governed by a BSD-style license that can be
- * found in the LICENSE file.
- */
-
-// This file requires these functions to be defined globally by someone else:
-// function createPeerConnection(stun_server, useRtpDataChannel)
-// function createOffer(peerConnection, constraints, callback)
-// function receiveOffer(peerConnection, offer, constraints, callback)
-// function receiveAnswer(peerConnection, answer, callback)
-
-// Currently these functions are supplied by jsep01_call.js.
-
-/**
- * We need a STUN server for some API calls.
- * @private
- */
-var STUN_SERVER = 'stun.l.google.com:19302';
-
-/**
- * This object represents the call.
- * @private
- */
-var gPeerConnection = null;
-
-/**
- * If true, any created peer connection will use RTP data
- * channels. Otherwise it will use SCTP data channels.
- */
-var gUseRtpDataChannels = true;
-
-/**
- * This stores ICE candidates generated on this side.
- * @private
- */
-var gIceCandidates = [];
-
-// Public interface to tests. These are expected to be called with
-// ExecuteJavascript invocations from the browser tests and will return answers
-// through the DOM automation controller.
-
-/**
- * Creates a peer connection. Must be called before most other public functions
- * in this file.
- */
-function preparePeerConnection() {
- if (gPeerConnection != null)
- throw failTest('creating peer connection, but we already have one.');
-
- gPeerConnection = createPeerConnection(STUN_SERVER, gUseRtpDataChannels);
- returnToTest('ok-peerconnection-created');
-}
-
-/**
- * Asks this page to create a local offer.
- *
- * Returns a string on the format ok-(JSON encoded session description).
- *
- * @param {!object} constraints Any createOffer constraints.
- */
-function createLocalOffer(constraints) {
- if (gPeerConnection == null)
- throw failTest('Negotiating call, but we have no peer connection.');
-
- // TODO(phoglund): move jsep01.call stuff into this file and remove need
- // of the createOffer method, etc.
- createOffer(gPeerConnection, constraints, function(localOffer) {
- returnToTest('ok-' + JSON.stringify(localOffer));
- });
-}
-
-/**
- * Asks this page to accept an offer and generate an answer.
- *
- * Returns a string on the format ok-(JSON encoded session description).
- *
- * @param {!string} sessionDescJson A JSON-encoded session description of type
- * 'offer'.
- * @param {!object} constraints Any createAnswer constraints.
- */
-function receiveOfferFromPeer(sessionDescJson, constraints) {
- if (gPeerConnection == null)
- throw failTest('Receiving offer, but we have no peer connection.');
-
- offer = parseJson_(sessionDescJson);
- if (!offer.type)
- failTest('Got invalid session description from peer: ' + sessionDescJson);
- if (offer.type != 'offer')
- failTest('Expected to receive offer from peer, got ' + offer.type);
-
- receiveOffer(gPeerConnection, offer , constraints, function(answer) {
- returnToTest('ok-' + JSON.stringify(answer));
- });
-}
-
-/**
- * Asks this page to accept an answer generated by the peer in response to a
- * previous offer by this page
- *
- * Returns a string ok-accepted-answer on success.
- *
- * @param {!string} sessionDescJson A JSON-encoded session description of type
- * 'answer'.
- */
-function receiveAnswerFromPeer(sessionDescJson) {
- if (gPeerConnection == null)
- throw failTest('Receiving offer, but we have no peer connection.');
-
- answer = parseJson_(sessionDescJson);
- if (!answer.type)
- failTest('Got invalid session description from peer: ' + sessionDescJson);
- if (answer.type != 'answer')
- failTest('Expected to receive answer from peer, got ' + answer.type);
-
- receiveAnswer(gPeerConnection, answer, function() {
- returnToTest('ok-accepted-answer');
- });
-}
-
-/**
- * Adds the local stream to the peer connection. You will have to re-negotiate
- * the call for this to take effect in the call.
- */
-function addLocalStream() {
- if (gPeerConnection == null)
- throw failTest('adding local stream, but we have no peer connection.');
-
- addLocalStreamToPeerConnection(gPeerConnection);
- returnToTest('ok-added');
-}
-
-/**
- * Loads a file with WebAudio and connects it to the peer connection.
- *
- * The loadAudioAndAddToPeerConnection will return ok-added to the test when
- * the sound is loaded and added to the peer connection. The sound will start
- * playing when you call playAudioFile.
- *
- * @param url URL pointing to the file to play. You can assume that you can
- * serve files from the repository's file system. For instance, to serve a
- * file from chrome/test/data/pyauto_private/webrtc/file.wav, pass in a path
- * relative to this directory (e.g. ../pyauto_private/webrtc/file.wav).
- */
-function addAudioFile(url) {
- if (gPeerConnection == null)
- throw failTest('adding audio file, but we have no peer connection.');
-
- loadAudioAndAddToPeerConnection(url, gPeerConnection);
-}
-
-/**
- * Mixes the local audio stream with an audio file through WebAudio.
- *
- * You must have successfully requested access to the user's microphone through
- * getUserMedia before calling this function (see getUserMedia.js).
- * Additionally, you must have loaded an audio file to mix with.
- *
- * When playAudioFile is called, WebAudio will effectively mix the user's
- * microphone input with the previously loaded file and feed that into the
- * peer connection.
- */
-function mixLocalStreamWithPreviouslyLoadedAudioFile() {
- if (gPeerConnection == null)
- throw failTest('trying to mix in stream, but we have no peer connection.');
- if (getLocalStream() == null)
- throw failTest('trying to mix in stream, but we have no stream to mix in.');
-
- mixLocalStreamIntoPeerConnection(gPeerConnection, getLocalStream());
-}
-
-/**
- * Must be called after addAudioFile.
- */
-function playAudioFile() {
- if (gPeerConnection == null)
- throw failTest('trying to play file, but we have no peer connection.');
-
- playPreviouslyLoadedAudioFile(gPeerConnection);
- returnToTest('ok-playing');
-}
-
-/**
- * Hangs up a started call. Returns ok-call-hung-up on success.
- */
-function hangUp() {
- if (gPeerConnection == null)
- throw failTest('hanging up, but has no peer connection');
- gPeerConnection.close();
- gPeerConnection = null;
- returnToTest('ok-call-hung-up');
-}
-
-/**
- * Retrieves all ICE candidates generated on this side. Must be called after
- * ICE candidate generation is triggered (for instance by running a call
- * negotiation). This function will wait if necessary if we're not done
- * generating ICE candidates on this side.
- *
- * Returns a JSON-encoded array of RTCIceCandidate instances to the test.
- */
-function getAllIceCandidates() {
- if (gPeerConnection == null)
- throw failTest('Trying to get ICE candidates, but has no peer connection.');
-
- if (gPeerConnection.iceGatheringState != 'complete') {
- console.log('Still ICE gathering - waiting...');
- setTimeout(getAllIceCandidates, 100);
- return;
- }
-
- returnToTest(JSON.stringify(gIceCandidates));
-}
-
-/**
- * Receives ICE candidates from the peer.
- *
- * Returns ok-received-candidates to the test on success.
- *
- * @param iceCandidatesJson a JSON-encoded array of RTCIceCandidate instances.
- */
-function receiveIceCandidates(iceCandidatesJson) {
- if (gPeerConnection == null)
- throw failTest('Received ICE candidate, but has no peer connection');
-
- var iceCandidates = parseJson_(iceCandidatesJson);
- if (!iceCandidates.length)
- throw failTest('Received invalid ICE candidate list from peer: ' +
- iceCandidatesJson);
-
- iceCandidates.forEach(function(iceCandidate) {
- if (!iceCandidate.candidate)
- failTest('Received invalid ICE candidate from peer: ' +
- iceCandidatesJson);
-
- gPeerConnection.addIceCandidate(new RTCIceCandidate(iceCandidate));
- });
-
- returnToTest('ok-received-candidates');
-}
-
-// Public interface to signaling implementations, such as JSEP.
-
-/**
- * Enqueues an ICE candidate for sending to the peer.
- *
- * @param {!RTCIceCandidate} The ICE candidate to send.
- */
-function sendIceCandidate(message) {
- gIceCandidates.push(message);
-}
-
-/**
- * Parses JSON-encoded session descriptions and ICE candidates.
- * @private
- */
-function parseJson_(json) {
- // Escape since the \r\n in the SDP tend to get unescaped.
- jsonWithEscapedLineBreaks = json.replace(/\r\n/g, '\\r\\n');
- try {
- return JSON.parse(jsonWithEscapedLineBreaks);
- } catch (exception) {
- failTest('Failed to parse JSON: ' + jsonWithEscapedLineBreaks + ', got ' +
- exception);
- }
-}
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