Bug 1676855: Support RTCRtpEncodingParameters.active r=jib,mjf

Differential Revision: https://phabricator.services.mozilla.com/D156837

dom/media/webrtc/jsapi/RTCRtpSender.cpp

@@ -859,6 +859,7 @@ bool operator==(const RTCRtpEncodingParameters& a1,
 void RTCRtpSender::ApplyJsEncodingToConduitEncoding(
     const RTCRtpEncodingParameters& aJsEncoding,
     VideoCodecConfig::Encoding* aConduitEncoding) {
+  aConduitEncoding->active = aJsEncoding.mActive;
   if (aJsEncoding.mMaxBitrate.WasPassed()) {
     aConduitEncoding->constraints.maxBr = aJsEncoding.mMaxBitrate.Value();
   }

dom/media/webrtc/libwebrtcglue/CodecConfig.h

@@ -121,8 +121,11 @@ class VideoCodecConfig {
   struct Encoding {
     std::string rid;
     EncodingConstraints constraints;
+    bool active = true;
+    // TODO(bug 1744116): Use = default here
     bool operator==(const Encoding& aOther) const {
-      return rid == aOther.rid && constraints == aOther.constraints;
+      return rid == aOther.rid && constraints == aOther.constraints &&
+             active == aOther.active;
     }
   };
   std::vector<Encoding> mEncodings;
@@ -133,6 +136,7 @@ class VideoCodecConfig {
   uint8_t mPacketizationMode;
   // TODO: add external negotiated SPS/PPS
 
+  // TODO(bug 1744116): Use = default here
   bool operator==(const VideoCodecConfig& aRhs) const {
     return mType == aRhs.mType && mName == aRhs.mName &&
            mAckFbTypes == aRhs.mAckFbTypes &&

dom/media/webrtc/libwebrtcglue/VideoConduit.cpp

@@ -1309,6 +1309,22 @@ MediaConduitErrorCode WebrtcVideoConduit::SendVideoFrame(
                     __FUNCTION__);
       return kMediaConduitNoError;
     }
+
+    // Workaround for bug in libwebrtc where all encodings are transmitted
+    // if they are all inactive.
+    bool anyActive = false;
+    for (const auto& encoding : mCurSendCodecConfig->mEncodings) {
+      if (encoding.active) {
+        anyActive = true;
+        break;
+      }
+    }
+    if (!anyActive) {
+      CSFLogVerbose(LOGTAG, "WebrtcVideoConduit %p %s No active encodings",
+                    this, __FUNCTION__);
+      return kMediaConduitNoError;
+    }
+
     CSFLogVerbose(LOGTAG, "WebrtcVideoConduit %p %s (send SSRC %u (0x%x))",
                   this, __FUNCTION__, mSendStreamConfig.rtp.ssrcs.front(),
                   mSendStreamConfig.rtp.ssrcs.front());

dom/media/webrtc/libwebrtcglue/VideoStreamFactory.cpp

@@ -150,6 +150,7 @@ std::vector<webrtc::VideoStream> VideoStreamFactory::CreateEncoderStreams(
   for (int idx = streamCount - 1; idx >= 0; --idx) {
     webrtc::VideoStream video_stream;
     auto& encoding = mCodecConfig.mEncodings[idx];
+    video_stream.active = encoding.active;
     MOZ_ASSERT(encoding.constraints.scaleDownBy >= 1.0);
 
     gfx::IntSize newSize(0, 0);

testing/web-platform/meta/webrtc/simulcast/setParameters-active.https.html.ini

 [setParameters-active.https.html]
-  expected:
-    if (os == "android") and fission: [TIMEOUT, OK]
+  expected: [OK, TIMEOUT]
+  bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1787474
+  [Simulcast setParameters active=false on first encoding stops sending frames for that encoding]
+    expected: [PASS, TIMEOUT]
+  [Simulcast setParameters active=false on second encoding stops sending frames for that encoding]
+    expected: [PASS, TIMEOUT, NOTRUN]
   [Simulcast setParameters active=false stops sending frames]
-    expected: FAIL
+    expected: [PASS, TIMEOUT, NOTRUN]

testing/web-platform/tests/webrtc/simulcast/setParameters-active.https.html

@@ -24,6 +24,56 @@ async function queryReceiverStats(pc) {
   return inboundStats.map(s => s.framesDecoded);
 }
 
+promise_test(async t => {
+  const rids = [0, 1];
+  const pc1 = new RTCPeerConnection();
+  t.add_cleanup(() => pc1.close());
+  const pc2 = new RTCPeerConnection();
+  t.add_cleanup(() => pc2.close());
+
+  await negotiateSimulcastAndWaitForVideo(t, rids, pc1, pc2);
+
+  // Deactivate first sender.
+  const parameters = pc1.getSenders()[0].getParameters();
+  parameters.encodings[0].active = false;
+  await pc1.getSenders()[0].setParameters(parameters);
+
+  // Assert (almost) no new frames are received on the first encoding.
+  // Without any action we would expect to have received around 30fps.
+  await new Promise(resolve => t.step_timeout(resolve, 100)); // Wait a bit.
+  const initialStats = await queryReceiverStats(pc2);
+  await new Promise(resolve => t.step_timeout(resolve, 1000)); // Wait more.
+  const subsequentStats = await queryReceiverStats(pc2);
+
+  assert_equals(subsequentStats[0], initialStats[0]);
+  assert_greater_than(subsequentStats[1], initialStats[1]);
+}, 'Simulcast setParameters active=false on first encoding stops sending frames for that encoding');
+
+promise_test(async t => {
+  const rids = [0, 1];
+  const pc1 = new RTCPeerConnection();
+  t.add_cleanup(() => pc1.close());
+  const pc2 = new RTCPeerConnection();
+  t.add_cleanup(() => pc2.close());
+
+  await negotiateSimulcastAndWaitForVideo(t, rids, pc1, pc2);
+
+  // Deactivate second sender.
+  const parameters = pc1.getSenders()[0].getParameters();
+  parameters.encodings[1].active = false;
+  await pc1.getSenders()[0].setParameters(parameters);
+
+  // Assert (almost) no new frames are received on the second encoding.
+  // Without any action we would expect to have received around 30fps.
+  await new Promise(resolve => t.step_timeout(resolve, 100)); // Wait a bit.
+  const initialStats = await queryReceiverStats(pc2);
+  await new Promise(resolve => t.step_timeout(resolve, 1000)); // Wait more.
+  const subsequentStats = await queryReceiverStats(pc2);
+
+  assert_equals(subsequentStats[1], initialStats[1]);
+  assert_greater_than(subsequentStats[0], initialStats[0]);
+}, 'Simulcast setParameters active=false on second encoding stops sending frames for that encoding');
+
 promise_test(async t => {
   const rids = [0, 1];
   const pc1 = new RTCPeerConnection();