| <!DOCTYPE html> |
| <meta charset="utf-8"> |
| <title>Tests that RTCRtpReceiver is prepared to receive any negotiated video codec</title> |
| <script src="/resources/testharness.js"></script> |
| <script src="/resources/testharnessreport.js"></script> |
| <script src="/webrtc/RTCPeerConnection-helper.js"></script> |
| <script src="/webrtc/third_party/sdp/sdp.js"></script> |
| <body> |
| <script> |
| 'use strict' |
| |
| function filterOnlySecondaryCodec(description) { |
| const sections = SDPUtils.splitSections(description.sdp); |
| const dtls = SDPUtils.getDtlsParameters(sections[1], sections[0]); |
| const ice = SDPUtils.getIceParameters(sections[1], sections[0]); |
| const rtpParameters = SDPUtils.parseRtpParameters(sections[1]); |
| const setupValue = SDPUtils.matchPrefix(description.sdp, 'a=setup:')[0].substring(8); |
| const mline = SDPUtils.parseMLine(sections[1]); |
| |
| // Of all the codecs in the description, filter out one that has multiple |
| // payload types, and use only one of those payload types that is not the |
| // preferred codec. |
| // |
| // Ideally this would test all PTs through RTCRtpSender.setParameters, but |
| // Firefox does not at this time support RTCRtpEncodingParameters.codec: |
| // https://bugzilla.mozilla.org/show_bug.cgi?id=1894137 |
| const codecs = {}; |
| for (const codec of rtpParameters.codecs) { |
| if (["RED", "RTX", "ULPFEC"].includes(codec.name.toUpperCase())) { |
| continue; |
| } |
| codecs[codec.name] ??= []; |
| codecs[codec.name].push(codec); |
| } |
| |
| const multipleCodecs = []; |
| for (const name of Object.keys(codecs)) { |
| if (codecs[name].length > 1) { |
| multipleCodecs.push(codecs[name]); |
| } |
| } |
| |
| assert_implements_optional(multipleCodecs.length > 0, 'No codec with multiple payload types'); |
| |
| const multiplePtsCodecs = multipleCodecs[0]; |
| const nonPreferredCodec = multiplePtsCodecs[1]; |
| rtpParameters.codecs = [nonPreferredCodec]; |
| |
| return SDPUtils.writeSessionBoilerplate() + |
| SDPUtils.writeDtlsParameters(dtls, setupValue) + |
| SDPUtils.writeIceParameters(ice) + |
| SDPUtils.writeRtpDescription(mline.kind, rtpParameters); |
| } |
| |
| promise_test(async t => { |
| const pc1 = new RTCPeerConnection(); |
| t.add_cleanup(() => pc1.close()); |
| const pc2 = new RTCPeerConnection(); |
| t.add_cleanup(() => pc2.close()); |
| pc1.onicecandidate = ({candidate}) => pc2.addIceCandidate(candidate); |
| pc2.onicecandidate = ({candidate}) => pc1.addIceCandidate(candidate); |
| |
| const [track] = (await getNoiseStream({video: true})).getTracks(); |
| t.add_cleanup(() => track.stop()); |
| pc1.addTrack(track); |
| |
| await pc1.setLocalDescription(); |
| await pc2.setRemoteDescription(pc1.localDescription); |
| await pc2.setLocalDescription(); |
| |
| const nonPreferredAnswer = {type: 'answer', sdp: filterOnlySecondaryCodec(pc2.localDescription)}; |
| await pc1.setRemoteDescription(nonPreferredAnswer); |
| |
| // Verify that the right payloadType is sent, *and* received. |
| const start = performance.now(); |
| const timeoutThreshold = start + 5000; |
| while (true) { |
| const stats = [...(await pc1.getStats()).values()].find(({type}) => type === 'outbound-rtp'); |
| if (stats?.framesSent > 0) { |
| break; |
| } |
| if (performance.now() > timeoutThreshold) { |
| assert_unreached("Timed out waiting for sent frames"); |
| } |
| await new Promise(r => t.step_timeout(r, 100)); |
| } |
| |
| while (true) { |
| const stats = [...(await pc2.getStats()).values()].find(({type}) => type === 'inbound-rtp'); |
| if (stats?.framesReceived > 0) { |
| break; |
| } |
| if (performance.now() > timeoutThreshold) { |
| assert_unreached("Timed out waiting for received frames"); |
| } |
| await new Promise(r => t.step_timeout(r, 100)); |
| } |
| }, 'An RTCRtpReceiver is prepared to receive any negotiated codec.'); |
| |
| </script> |
| </body> |
