Use local pages for webrtc telemetry tests.

tools/perf/page_sets/webrtc_cases/audio.js

+/*
+ * Copyright 2017 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.
+ */
+
+'use strict';
+
+var audio2 = document.querySelector('audio#audio2');
+var callButton = document.querySelector('button#callButton');
+var hangupButton = document.querySelector('button#hangupButton');
+var codecSelector = document.querySelector('select#codec');
+hangupButton.disabled = true;
+callButton.onclick = call;
+hangupButton.onclick = hangup;
+
+var pc1;
+var pc2;
+var localStream;
+
+var bitrateGraph;
+var bitrateSeries;
+
+var packetGraph;
+var packetSeries;
+
+var lastResult;
+
+var offerOptions = {
+  offerToReceiveAudio: 1,
+  offerToReceiveVideo: 0,
+  voiceActivityDetection: false
+};
+
+function gotStream(stream) {
+  hangupButton.disabled = false;
+  trace('Received local stream');
+  localStream = stream;
+  var audioTracks = localStream.getAudioTracks();
+  if (audioTracks.length > 0) {
+    trace('Using Audio device: ' + audioTracks[0].label);
+  }
+  pc1.addStream(localStream);
+  trace('Adding Local Stream to peer connection');
+
+  pc1.createOffer(
+    offerOptions
+  ).then(
+    gotDescription1,
+    onCreateSessionDescriptionError
+  );
+
+  bitrateSeries = new TimelineDataSeries();
+  bitrateGraph = new TimelineGraphView('bitrateGraph', 'bitrateCanvas');
+  bitrateGraph.updateEndDate();
+
+  packetSeries = new TimelineDataSeries();
+  packetGraph = new TimelineGraphView('packetGraph', 'packetCanvas');
+  packetGraph.updateEndDate();
+}
+
+function onCreateSessionDescriptionError(error) {
+  trace('Failed to create session description: ' + error.toString());
+}
+
+function call() {
+  callButton.disabled = true;
+  codecSelector.disabled = true;
+  trace('Starting call');
+  var servers = null;
+  var pcConstraints = {
+    'optional': []
+  };
+  pc1 = new RTCPeerConnection(servers, pcConstraints);
+  trace('Created local peer connection object pc1');
+  pc1.onicecandidate = function(e) {
+    onIceCandidate(pc1, e);
+  };
+  pc2 = new RTCPeerConnection(servers, pcConstraints);
+  trace('Created remote peer connection object pc2');
+  pc2.onicecandidate = function(e) {
+    onIceCandidate(pc2, e);
+  };
+  pc2.onaddstream = gotRemoteStream;
+  trace('Requesting local stream');
+  navigator.mediaDevices.getUserMedia({
+    audio: true,
+    video: false
+  })
+  .then(gotStream)
+  .catch(function(e) {
+    alert('getUserMedia() error: ' + e.name);
+  });
+}
+
+function gotDescription1(desc) {
+  trace('Offer from pc1 \n' + desc.sdp);
+  pc1.setLocalDescription(desc).then(
+    function() {
+      desc.sdp = forceChosenAudioCodec(desc.sdp);
+      pc2.setRemoteDescription(desc).then(
+        function() {
+          pc2.createAnswer().then(
+            gotDescription2,
+            onCreateSessionDescriptionError
+          );
+        },
+        onSetSessionDescriptionError
+      );
+    },
+    onSetSessionDescriptionError
+  );
+}
+
+function gotDescription2(desc) {
+  trace('Answer from pc2 \n' + desc.sdp);
+  pc2.setLocalDescription(desc).then(
+    function() {
+      desc.sdp = forceChosenAudioCodec(desc.sdp);
+      pc1.setRemoteDescription(desc).then(
+        function() {
+        },
+        onSetSessionDescriptionError
+      );
+    },
+    onSetSessionDescriptionError
+  );
+}
+
+function hangup() {
+  trace('Ending call');
+  localStream.getTracks().forEach(function(track) {
+    track.stop();
+  });
+  pc1.close();
+  pc2.close();
+  pc1 = null;
+  pc2 = null;
+  hangupButton.disabled = true;
+  callButton.disabled = false;
+  codecSelector.disabled = false;
+}
+
+function gotRemoteStream(e) {
+  audio2.srcObject = e.stream;
+  trace('Received remote stream');
+}
+
+function getOtherPc(pc) {
+  return (pc === pc1) ? pc2 : pc1;
+}
+
+function getName(pc) {
+  return (pc === pc1) ? 'pc1' : 'pc2';
+}
+
+function onIceCandidate(pc, event) {
+  getOtherPc(pc).addIceCandidate(event.candidate)
+  .then(
+    function() {
+      onAddIceCandidateSuccess(pc);
+    },
+    function(err) {
+      onAddIceCandidateError(pc, err);
+    }
+  );
+  trace(getName(pc) + ' ICE candidate: \n' + (event.candidate ?
+      event.candidate.candidate : '(null)'));
+}
+
+function onAddIceCandidateSuccess() {
+  trace('AddIceCandidate success.');
+}
+
+function onAddIceCandidateError(error) {
+  trace('Failed to add ICE Candidate: ' + error.toString());
+}
+
+function onSetSessionDescriptionError(error) {
+  trace('Failed to set session description: ' + error.toString());
+}
+
+function forceChosenAudioCodec(sdp) {
+  return maybePreferCodec(sdp, 'audio', 'send', codecSelector.value);
+}
+
+// Copied from AppRTC's sdputils.js:
+
+// Sets |codec| as the default |type| codec if it's present.
+// The format of |codec| is 'NAME/RATE', e.g. 'opus/48000'.
+function maybePreferCodec(sdp, type, dir, codec) {
+  var str = type + ' ' + dir + ' codec';
+  if (codec === '') {
+    trace('No preference on ' + str + '.');
+    return sdp;
+  }
+
+  trace('Prefer ' + str + ': ' + codec);
+
+  var sdpLines = sdp.split('\r\n');
+
+  // Search for m line.
+  var mLineIndex = findLine(sdpLines, 'm=', type);
+  if (mLineIndex === null) {
+    return sdp;
+  }
+
+  // If the codec is available, set it as the default in m line.
+  var codecIndex = findLine(sdpLines, 'a=rtpmap', codec);
+  console.log('codecIndex', codecIndex);
+  if (codecIndex) {
+    var payload = getCodecPayloadType(sdpLines[codecIndex]);
+    if (payload) {
+      sdpLines[mLineIndex] = setDefaultCodec(sdpLines[mLineIndex], payload);
+    }
+  }
+
+  sdp = sdpLines.join('\r\n');
+  return sdp;
+}
+
+// Find the line in sdpLines that starts with |prefix|, and, if specified,
+// contains |substr| (case-insensitive search).
+function findLine(sdpLines, prefix, substr) {
+  return findLineInRange(sdpLines, 0, -1, prefix, substr);
+}
+
+// Find the line in sdpLines[startLine...endLine - 1] that starts with |prefix|
+// and, if specified, contains |substr| (case-insensitive search).
+function findLineInRange(sdpLines, startLine, endLine, prefix, substr) {
+  var realEndLine = endLine !== -1 ? endLine : sdpLines.length;
+  for (var i = startLine; i < realEndLine; ++i) {
+    if (sdpLines[i].indexOf(prefix) === 0) {
+      if (!substr ||
+          sdpLines[i].toLowerCase().indexOf(substr.toLowerCase()) !== -1) {
+        return i;
+      }
+    }
+  }
+  return null;
+}
+
+// Gets the codec payload type from an a=rtpmap:X line.
+function getCodecPayloadType(sdpLine) {
+  var pattern = new RegExp('a=rtpmap:(\\d+) \\w+\\/\\d+');
+  var result = sdpLine.match(pattern);
+  return (result && result.length === 2) ? result[1] : null;
+}
+
+// Returns a new m= line with the specified codec as the first one.
+function setDefaultCodec(mLine, payload) {
+  var elements = mLine.split(' ');
+
+  // Just copy the first three parameters; codec order starts on fourth.
+  var newLine = elements.slice(0, 3);
+
+  // Put target payload first and copy in the rest.
+  newLine.push(payload);
+  for (var i = 3; i < elements.length; i++) {
+    if (elements[i] !== payload) {
+      newLine.push(elements[i]);
+    }
+  }
+  return newLine.join(' ');
+}
+
+// query getStats every second
+window.setInterval(function() {
+  if (!window.pc1) {
+    return;
+  }
+  window.pc1.getStats(null).then(function(res) {
+    res.forEach(function(report) {
+      var bytes;
+      var packets;
+      var now = report.timestamp;
+      if ((report.type === 'outboundrtp') ||
+          (report.type === 'outbound-rtp') ||
+          (report.type === 'ssrc' && report.bytesSent)) {
+        bytes = report.bytesSent;
+        packets = report.packetsSent;
+        if (lastResult && lastResult.get(report.id)) {
+          // calculate bitrate
+          var bitrate = 8 * (bytes - lastResult.get(report.id).bytesSent) /
+              (now - lastResult.get(report.id).timestamp);
+
+          // append to chart
+          bitrateSeries.addPoint(now, bitrate);
+          bitrateGraph.setDataSeries([bitrateSeries]);
+          bitrateGraph.updateEndDate();
+
+          // calculate number of packets and append to chart
+          packetSeries.addPoint(now, packets -
+              lastResult.get(report.id).packetsSent);
+          packetGraph.setDataSeries([packetSeries]);
+          packetGraph.updateEndDate();
+        }
+      }
+    });
+    lastResult = res;
+  });
+}, 1000);