| <!DOCTYPE html> |
| <html> |
| <head> |
| <meta charset="utf-8"> |
| <title>Testing replacing an audio track during a WebRTC call</title> |
| <script src="../resources/testharness.js"></script> |
| <script src="../resources/testharnessreport.js"></script> |
| <script src ="routines.js"></script> |
| </head> |
| <body> |
| <video id=video autoplay controls></video> |
| <script> |
| var usesLowSampleRate = false; |
| promise_test(async (test) => { |
| if (window.testRunner) |
| testRunner.setUserMediaPermission(true); |
| |
| const localStream = await navigator.mediaDevices.getUserMedia({ audio: true }); |
| setInterval(() => { |
| usesLowSampleRate = !usesLowSampleRate; |
| const rate = usesLowSampleRate ? 44100 : 48000; |
| localStream.getAudioTracks()[0].applyConstraints({sampleRate: rate}); |
| }, 200); |
| const remoteStream = await new Promise((resolve, reject) => { |
| createConnections((firstConnection) => { |
| sender = firstConnection.addTrack(localStream.getAudioTracks()[0], localStream); |
| }, (secondConnection) => { |
| receivingConnection = secondConnection; |
| secondConnection.ontrack = (trackEvent) => { resolve(trackEvent.streams[0]); }; |
| }); |
| setTimeout(() => reject("Test timed out"), 5000); |
| }); |
| video.srcObject = remoteStream; |
| |
| await new Promise(resolve => setTimeout(resolve, 1000)); |
| localStream.getAudioTracks()[0].stop(); |
| }, "Audio connection with track changing sample rate"); |
| </script> |
| </body> |
| </html> |
