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Bug 1250990 - Make RTCRtpEncodingParameters.scaleResolutionDownBy work with H.264 unicast. r=jesup
MozReview-Commit-ID: 2j8rRzZemql --HG-- extra : rebase_source : 1e039392219000830d869f0ad5432a297f6d6987
This commit is contained in:
Jan-Ivar Bruaroey
parent
a7a0605352
commit
2f66effab3
@ -12,33 +12,37 @@
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visible: true
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});
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var pc1 = new RTCPeerConnection();
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var pc2 = new RTCPeerConnection();
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var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
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pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
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pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
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pc1.onnegotiationneeded = e =>
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pc1.createOffer().then(d => pc1.setLocalDescription(d))
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.then(() => pc2.setRemoteDescription(pc1.localDescription))
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.then(() => pc2.createAnswer()).then(d => pc2.setLocalDescription(d))
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.then(() => pc1.setRemoteDescription(pc2.localDescription))
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.catch(generateErrorCallback());
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var mustRejectWith = (msg, reason, f) =>
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f().then(() => ok(false, msg),
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e => is(e.name, reason, msg));
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var v1, v2;
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runNetworkTest(function() {
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v1 = createMediaElement('video', 'v1');
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v2 = createMediaElement('video', 'v2');
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var removeVP8 = d => (d.sdp = d.sdp.replace("a=rtpmap:120 VP8/90000\r\n", ""), d);
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is(v2.currentTime, 0, "v2.currentTime is zero at outset");
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function testScale(codec) {
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var pc1 = new RTCPeerConnection();
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var pc2 = new RTCPeerConnection();
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navigator.mediaDevices.getUserMedia({ video: true })
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var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
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pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
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pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
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info("testing scaling with " + codec);
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pc1.onnegotiationneeded = e =>
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pc1.createOffer()
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.then(d => pc1.setLocalDescription(codec == "VP8" ? d : removeVP8(d)))
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.then(() => pc2.setRemoteDescription(pc1.localDescription))
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.then(() => pc2.createAnswer()).then(d => pc2.setLocalDescription(d))
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.then(() => pc1.setRemoteDescription(pc2.localDescription))
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.catch(generateErrorCallback());
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return navigator.mediaDevices.getUserMedia({ video: true })
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.then(stream => {
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var v1 = createMediaElement('video', 'v1');
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var v2 = createMediaElement('video', 'v2');
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is(v2.currentTime, 0, "v2.currentTime is zero at outset");
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v1.srcObject = stream;
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var sender = pc1.addTrack(stream.getVideoTracks()[0], stream);
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@ -46,23 +50,29 @@
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() => sender.setParameters({ encodings:
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[{ scaleResolutionDownBy: 0.5 } ] }))
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.then(() => sender.setParameters({ encodings: [{ maxBitrate: 60000,
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scaleResolutionDownBy: 2 }] }))
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scaleResolutionDownBy: 2 }] }))
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.then(() => new Promise(resolve => pc2.ontrack = e => resolve(e)))
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.then(e => v2.srcObject = e.streams[0])
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.then(() => new Promise(resolve => v2.onloadedmetadata = resolve))
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.then(() => waitUntil(() => v2.currentTime > 0 && v2.srcObject.currentTime > 0))
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.then(() => ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")"))
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.then(() => wait(1000)) // TODO: Bug 1248154
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.then(() => {
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ok(v1.videoWidth > 0, "source width is positive");
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ok(v1.videoHeight > 0, "source height is positive");
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is(v2.videoWidth, v1.videoWidth / 2, "sink is half the width of source");
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is(v2.videoHeight, v1.videoHeight / 2, "sink is half the height of source");
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})
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.then(() => {
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stream.getTracks().forEach(track => track.stop());
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v1.srcObject = v2.srcObject = null;
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})
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})
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.then(() => new Promise(resolve => pc2.ontrack = e => resolve(e)))
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.then(e => v2.srcObject = e.streams[0])
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.then(() => new Promise(resolve => v2.onloadedmetadata = resolve))
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.then(() => waitUntil(() => v2.currentTime > 0 && v2.srcObject.currentTime > 0))
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.then(() => ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")"))
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.then(() => wait(1000)) // TODO: Bug 1248154
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.then(() => {
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ok(v1.videoWidth > 0, "source width is positive");
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ok(v1.videoHeight > 0, "source height is positive");
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is(v2.videoWidth, v1.videoWidth / 2, "sink is half the width of source");
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is(v2.videoHeight, v1.videoHeight / 2, "sink is half the height of source");
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})
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.catch(generateErrorCallback())
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.then(networkTestFinished);
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});
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.catch(generateErrorCallback());
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}
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runNetworkTest(() => testScale("VP8").then(() => testScale("H264"))
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.then(networkTestFinished));
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</script>
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</pre>
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</body>
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@ -1277,80 +1277,76 @@ WebrtcVideoConduit::ReconfigureSendCodec(unsigned short width,
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CSFLogDebug(logTag,
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"%s: Requesting resolution change to %ux%u (from %ux%u)",
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__FUNCTION__, width, height, vie_codec.width, vie_codec.height);
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// Likely spurious unless there was some error, but rarely checked
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if (vie_codec.width != width || vie_codec.height != height ||
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vie_codec.maxFramerate != mSendingFramerate)
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{
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vie_codec.width = width;
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vie_codec.height = height;
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vie_codec.maxFramerate = mSendingFramerate;
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SelectBitrates(vie_codec.width, vie_codec.height, 0,
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vie_codec.width = width;
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vie_codec.height = height;
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vie_codec.maxFramerate = mSendingFramerate;
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SelectBitrates(vie_codec.width, vie_codec.height, 0,
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mLastFramerateTenths,
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vie_codec.minBitrate,
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vie_codec.startBitrate,
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vie_codec.maxBitrate);
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for (size_t i = vie_codec.numberOfSimulcastStreams; i > 0; --i) {
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webrtc::SimulcastStream& stream(vie_codec.simulcastStream[i - 1]);
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stream.width = width;
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stream.height = height;
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MOZ_ASSERT(stream.jsScaleDownBy >= 1.0);
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uint32_t new_width = uint32_t(width / stream.jsScaleDownBy);
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uint32_t new_height = uint32_t(height / stream.jsScaleDownBy);
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// TODO: If two layers are similar, only alloc bits to one (Bug 1249859)
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if (new_width != width || new_height != height) {
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if (vie_codec.numberOfSimulcastStreams == 1) {
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// Use less strict scaling in unicast. That way 320x240 / 3 = 106x79.
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ConstrainPreservingAspectRatio(new_width, new_height,
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&stream.width, &stream.height);
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} else {
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// webrtc.org supposedly won't tolerate simulcast unless every stream
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// is exactly the same aspect ratio. 320x240 / 3 = 80x60.
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ConstrainPreservingAspectRatioExact(new_width*new_height,
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&stream.width, &stream.height);
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}
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}
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// Give each layer default appropriate bandwidth limits based on the
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// resolution/framerate of that layer
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SelectBitrates(stream.width, stream.height, stream.jsMaxBitrate,
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mLastFramerateTenths,
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vie_codec.minBitrate,
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vie_codec.startBitrate,
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vie_codec.maxBitrate);
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stream.minBitrate,
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stream.targetBitrate,
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stream.maxBitrate);
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for (size_t i = vie_codec.numberOfSimulcastStreams; i > 0; --i) {
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webrtc::SimulcastStream& stream(vie_codec.simulcastStream[i - 1]);
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stream.width = width;
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stream.height = height;
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MOZ_ASSERT(stream.jsScaleDownBy >= 1.0);
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uint32_t new_width = uint32_t(width / stream.jsScaleDownBy);
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uint32_t new_height = uint32_t(height / stream.jsScaleDownBy);
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// TODO: If two layers are similar, only alloc bits to one (Bug 1249859)
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if (new_width != width || new_height != height) {
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if (vie_codec.numberOfSimulcastStreams == 1) {
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// Use less strict scaling in unicast. That way 320x240 / 3 = 106x79.
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ConstrainPreservingAspectRatio(new_width, new_height,
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&stream.width, &stream.height);
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} else {
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// webrtc.org supposedly won't tolerate simulcast unless every stream
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// is exactly the same aspect ratio. 320x240 / 3 = 80x60.
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ConstrainPreservingAspectRatioExact(new_width*new_height,
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&stream.width, &stream.height);
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}
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}
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// Give each layer default appropriate bandwidth limits based on the
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// resolution/framerate of that layer
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SelectBitrates(stream.width, stream.height, stream.jsMaxBitrate,
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mLastFramerateTenths,
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stream.minBitrate,
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stream.targetBitrate,
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stream.maxBitrate);
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vie_codec.minBitrate = std::min(stream.minBitrate, vie_codec.minBitrate);
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vie_codec.startBitrate += stream.targetBitrate;
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vie_codec.maxBitrate = std::max(stream.maxBitrate, vie_codec.maxBitrate);
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vie_codec.minBitrate = std::min(stream.minBitrate, vie_codec.minBitrate);
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vie_codec.startBitrate += stream.targetBitrate;
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vie_codec.maxBitrate = std::max(stream.maxBitrate, vie_codec.maxBitrate);
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// webrtc.org expects the last, highest fidelity, simulcast stream to
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// always have the same resolution as vie_codec
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if (i == vie_codec.numberOfSimulcastStreams) {
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vie_codec.width = stream.width;
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vie_codec.height = stream.height;
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}
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}
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if (vie_codec.numberOfSimulcastStreams != 0) {
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vie_codec.startBitrate /= vie_codec.numberOfSimulcastStreams;
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}
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if ((err = mPtrViECodec->SetSendCodec(mChannel, vie_codec)) != 0)
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{
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CSFLogError(logTag, "%s: SetSendCodec(%ux%u) failed, err %d",
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__FUNCTION__, width, height, err);
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return NS_ERROR_FAILURE;
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}
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if (mMinBitrateEstimate != 0) {
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mPtrViENetwork->SetBitrateConfig(mChannel,
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mMinBitrateEstimate,
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std::max(vie_codec.startBitrate,
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mMinBitrateEstimate),
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std::max(vie_codec.maxBitrate,
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mMinBitrateEstimate));
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}
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// webrtc.org expects the last, highest fidelity, simulcast stream to
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// always have the same resolution as vie_codec
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if (i == vie_codec.numberOfSimulcastStreams) {
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vie_codec.width = stream.width;
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vie_codec.height = stream.height;
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}
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}
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if (vie_codec.numberOfSimulcastStreams != 0) {
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vie_codec.startBitrate /= vie_codec.numberOfSimulcastStreams;
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}
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if ((err = mPtrViECodec->SetSendCodec(mChannel, vie_codec)) != 0)
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{
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CSFLogError(logTag, "%s: SetSendCodec(%ux%u) failed, err %d",
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__FUNCTION__, width, height, err);
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return NS_ERROR_FAILURE;
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}
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if (mMinBitrateEstimate != 0) {
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mPtrViENetwork->SetBitrateConfig(mChannel,
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mMinBitrateEstimate,
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std::max(vie_codec.startBitrate,
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mMinBitrateEstimate),
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std::max(vie_codec.maxBitrate,
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mMinBitrateEstimate));
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}
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CSFLogDebug(logTag, "%s: Encoder resolution changed to %ux%u @ %ufps, bitrate %u:%u",
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__FUNCTION__, width, height, mSendingFramerate,
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vie_codec.minBitrate, vie_codec.maxBitrate);
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} // else no change; mSendingWidth likely was 0
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CSFLogDebug(logTag, "%s: Encoder resolution changed to %ux%u @ %ufps, bitrate %u:%u",
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__FUNCTION__, width, height, mSendingFramerate,
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vie_codec.minBitrate, vie_codec.maxBitrate);
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if (frame) {
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// XXX I really don't like doing this from MainThread...
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mPtrExtCapture->IncomingFrame(*frame);
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@ -1864,57 +1860,57 @@ WebrtcVideoConduit::CodecConfigToWebRTCCodec(const VideoCodecConfig* codecInfo,
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cinst.codecSpecific.H264.spsLen = 0;
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cinst.codecSpecific.H264.ppsData = nullptr;
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cinst.codecSpecific.H264.ppsLen = 0;
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} else {
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// TODO(bug 1210175): H264 doesn't support simulcast yet.
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for (size_t i = 0; i < codecInfo->mSimulcastEncodings.size(); ++i) {
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const VideoCodecConfig::SimulcastEncoding& encoding =
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codecInfo->mSimulcastEncodings[i];
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// Make sure the constraints on the whole stream are reflected.
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webrtc::SimulcastStream stream;
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memset(&stream, 0, sizeof(stream));
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stream.width = cinst.width;
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stream.height = cinst.height;
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stream.numberOfTemporalLayers = 1;
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stream.maxBitrate = cinst.maxBitrate;
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stream.targetBitrate = cinst.targetBitrate;
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stream.minBitrate = cinst.minBitrate;
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stream.qpMax = cinst.qpMax;
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strncpy(stream.rid, encoding.rid.c_str(), sizeof(stream.rid)-1);
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stream.rid[sizeof(stream.rid) - 1] = 0;
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// Apply encoding-specific constraints.
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stream.width = MinIgnoreZero(
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stream.width,
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(unsigned short)encoding.constraints.maxWidth);
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stream.height = MinIgnoreZero(
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stream.height,
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(unsigned short)encoding.constraints.maxHeight);
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// webrtc.org uses kbps, we use bps
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stream.jsMaxBitrate = encoding.constraints.maxBr/1000;
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stream.jsScaleDownBy = encoding.constraints.scaleDownBy;
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MOZ_ASSERT(stream.jsScaleDownBy >= 1.0);
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uint32_t width = stream.width? stream.width : 640;
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uint32_t height = stream.height? stream.height : 480;
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uint32_t new_width = uint32_t(width / stream.jsScaleDownBy);
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uint32_t new_height = uint32_t(height / stream.jsScaleDownBy);
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if (new_width != width || new_height != height) {
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// Estimate. Overridden on first frame.
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SelectBitrates(new_width, new_height, stream.jsMaxBitrate,
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mLastFramerateTenths,
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stream.minBitrate,
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stream.targetBitrate,
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stream.maxBitrate);
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}
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// webrtc.org expects simulcast streams to be ordered by increasing
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// fidelity, our jsep code does the opposite.
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cinst.simulcastStream[codecInfo->mSimulcastEncodings.size()-i-1] = stream;
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}
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cinst.numberOfSimulcastStreams = codecInfo->mSimulcastEncodings.size();
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}
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// Init mSimulcastEncodings always since they hold info from setParameters.
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// TODO(bug 1210175): H264 doesn't support simulcast yet.
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for (size_t i = 0; i < codecInfo->mSimulcastEncodings.size(); ++i) {
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const VideoCodecConfig::SimulcastEncoding& encoding =
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codecInfo->mSimulcastEncodings[i];
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// Make sure the constraints on the whole stream are reflected.
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webrtc::SimulcastStream stream;
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memset(&stream, 0, sizeof(stream));
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stream.width = cinst.width;
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stream.height = cinst.height;
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stream.numberOfTemporalLayers = 1;
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stream.maxBitrate = cinst.maxBitrate;
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stream.targetBitrate = cinst.targetBitrate;
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stream.minBitrate = cinst.minBitrate;
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stream.qpMax = cinst.qpMax;
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strncpy(stream.rid, encoding.rid.c_str(), sizeof(stream.rid)-1);
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stream.rid[sizeof(stream.rid) - 1] = 0;
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// Apply encoding-specific constraints.
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stream.width = MinIgnoreZero(
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stream.width,
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(unsigned short)encoding.constraints.maxWidth);
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stream.height = MinIgnoreZero(
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stream.height,
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(unsigned short)encoding.constraints.maxHeight);
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// webrtc.org uses kbps, we use bps
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stream.jsMaxBitrate = encoding.constraints.maxBr/1000;
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stream.jsScaleDownBy = encoding.constraints.scaleDownBy;
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MOZ_ASSERT(stream.jsScaleDownBy >= 1.0);
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uint32_t width = stream.width? stream.width : 640;
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uint32_t height = stream.height? stream.height : 480;
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uint32_t new_width = uint32_t(width / stream.jsScaleDownBy);
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uint32_t new_height = uint32_t(height / stream.jsScaleDownBy);
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if (new_width != width || new_height != height) {
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// Estimate. Overridden on first frame.
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SelectBitrates(new_width, new_height, stream.jsMaxBitrate,
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mLastFramerateTenths,
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stream.minBitrate,
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stream.targetBitrate,
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stream.maxBitrate);
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}
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// webrtc.org expects simulcast streams to be ordered by increasing
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// fidelity, our jsep code does the opposite.
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cinst.simulcastStream[codecInfo->mSimulcastEncodings.size()-i-1] = stream;
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}
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cinst.numberOfSimulcastStreams = codecInfo->mSimulcastEncodings.size();
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}
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//Copy the codec passed into Conduit's database
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