Add jitterBufferTarget attribute to RTCRtpReceiver interface accordin…

…g to the spec. Audio test case in WPT RTCRtpReceiver-jitterBufferTarget-stats.html remains failing due to a problem that without audio tag webrtc receive streams can not be played back and because of that jitter buffer stats are empty. Otherwise it was manually verified that when adding audio tag the test case would start passing. Bug: 324276557 Change-Id: I06cbba95a7f78c20cc7bf803819fcfb5ef2f6be5 Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/5278176 Commit-Queue: Philipp Hancke <[email protected]> Reviewed-by: Henrik Boström <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Reviewed-by: Rick Byers <[email protected]> Cr-Commit-Position: refs/heads/main@{#1262622}
anhtai9710 · Feb 20, 2024 · 8734f19 · 8734f19
1 parent 88b0eca
commit 8734f19
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Showing 11 changed files with 46 additions and 30 deletions.
diff --git a/AUTHORS b/AUTHORS
@@ -389,6 +389,7 @@ Egor Starkov <[email protected]>
 Ehsan Akhgari <[email protected]>
 Ehsan Akhgari <[email protected]>
 Elan Ruusamäe <[email protected]>
+Eldar Rello <[email protected]>
 Ely Ronnen <[email protected]>
 Emil Suleymanov <[email protected]>
 Ergun Erdogmus <[email protected]>

diff --git a/third_party/blink/public/mojom/use_counter/metrics/web_feature.mojom b/third_party/blink/public/mojom/use_counter/metrics/web_feature.mojom
@@ -4223,6 +4223,8 @@ enum WebFeature {
   kV8PointerEvent_GetCoalescedEvents_Method = 4859,
   kCSSFunctions = 4861,
   kCSSPageRule = 4862,
+  kV8RTCRtpReceiver_JitterBufferTarget_AttributeGetter = 4863,
+  kV8RTCRtpReceiver_JitterBufferTarget_AttributeSetter = 4864,
 
   // Add new features immediately above this line. Don't change assigned
   // numbers of any item, and don't reuse removed slots.

diff --git a/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc b/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.cc
@@ -124,6 +124,29 @@ void RTCRtpReceiver::setPlayoutDelayHint(std::optional<double> hint,
   receiver_->SetJitterBufferMinimumDelay(playout_delay_hint_);
 }
 
+std::optional<double> RTCRtpReceiver::jitterBufferTarget() const {
+  DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
+  return jitter_buffer_target_;
+}
+
+void RTCRtpReceiver::setJitterBufferTarget(std::optional<double> target,
+                                           ExceptionState& exception_state) {
+  DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
+  if (target.has_value() && (target.value() < 0.0 || target.value() > 4000.0)) {
+    exception_state.ThrowRangeError(
+        "jitterBufferTarget is out of expected range 0 to 4000 ms");
+    return;
+  }
+
+  jitter_buffer_target_ = target;
+  if (jitter_buffer_target_.has_value()) {
+    receiver_->SetJitterBufferMinimumDelay(jitter_buffer_target_.value() /
+                                           1000.0);
+  } else {
+    receiver_->SetJitterBufferMinimumDelay(std::nullopt);
+  }
+}
+
 HeapVector<Member<RTCRtpSynchronizationSource>>
 RTCRtpReceiver::getSynchronizationSources(ScriptState* script_state,
                                           ExceptionState& exception_state) {

diff --git a/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.h b/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.h
@@ -70,6 +70,8 @@ class RTCRtpReceiver final : public ScriptWrappable,
   RTCDtlsTransport* rtcpTransport();
   std::optional<double> playoutDelayHint() const;
   void setPlayoutDelayHint(std::optional<double>, ExceptionState&);
+  std::optional<double> jitterBufferTarget() const;
+  void setJitterBufferTarget(std::optional<double>, ExceptionState&);
   RTCRtpReceiveParameters* getParameters();
   HeapVector<Member<RTCRtpSynchronizationSource>> getSynchronizationSources(
       ScriptState*,
@@ -141,6 +143,7 @@ class RTCRtpReceiver final : public ScriptWrappable,
   // observed delay may differ depending on the congestion control. |nullopt|
   // means default value must be used.
   std::optional<double> playout_delay_hint_;
+  std::optional<double> jitter_buffer_target_;
 
   THREAD_CHECKER(thread_checker_);
 

diff --git a/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.idl b/third_party/blink/renderer/modules/peerconnection/rtc_rtp_receiver.idl
@@ -10,6 +10,8 @@ interface RTCRtpReceiver {
     readonly attribute RTCDtlsTransport? rtcpTransport;
     // https://henbos.github.io/webrtc-extensions/#dom-rtcrtpreceiver-playoutdelayhint
     [RaisesException=Setter, Measure] attribute double? playoutDelayHint;
+    // https://w3c.github.io/webrtc-extensions/#dom-rtcrtpreceiver-jitterbuffertarget
+    [RaisesException=Setter, Measure, RuntimeEnabled=RTCJitterBufferTarget] attribute double? jitterBufferTarget;
     [CallWith=ScriptState, HighEntropy, Measure] static RTCRtpCapabilities? getCapabilities(DOMString kind);
     RTCRtpReceiveParameters               getParameters();
     [CallWith=ScriptState, RaisesException] sequence<RTCRtpSynchronizationSource> getSynchronizationSources();

diff --git a/third_party/blink/renderer/platform/runtime_enabled_features.json5 b/third_party/blink/renderer/platform/runtime_enabled_features.json5
@@ -3152,6 +3152,12 @@
       name: "RTCEncodedVideoFrameAdditionalMetadata",
       status: "experimental",
     },
+    // Enables the use of jitterBufferTarget attribute in WebRTC.
+    // Spec: https://w3c.github.io/webrtc-extensions/#dom-rtcrtpreceiver-jitterbuffertarget
+    {
+      name: "RTCJitterBufferTarget",
+      status: "stable",
+    },
     // Legacy callback-based getStats() has limited availability unless this
     // Deprecation Trial is enabled.
     // TODO(https://crbug.com/822696): Remove when origin trial ends.

diff --git a/...k/web_tests/external/wpt/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-expected.txt b/...k/web_tests/external/wpt/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-expected.txt
diff --git a/...tests/external/wpt/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-expected.txt b/...tests/external/wpt/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-expected.txt
@@ -1,7 +1,5 @@
 This is a testharness.js-based test.
-[FAIL] measure raising and lowering video jitterBufferTarget
-  assert_equals: jitterBufferTarget supported for video expected (object) null but got (undefined) undefined
 [FAIL] measure raising and lowering audio jitterBufferTarget
-  assert_equals: jitterBufferTarget supported for audio expected (object) null but got (undefined) undefined
+  assert_greater_than: audio increased delay NaN  greater than base delay NaN expected a number greater than NaN but got NaN
 Harness: the test ran to completion.
 
diff --git a/third_party/blink/web_tests/virtual/stable/webexposed/global-interface-listing-expected.txt b/third_party/blink/web_tests/virtual/stable/webexposed/global-interface-listing-expected.txt
@@ -6739,6 +6739,7 @@ interface RTCPeerConnectionIceEvent : Event
 interface RTCRtpReceiver
     static method getCapabilities
     attribute @@toStringTag
+    getter jitterBufferTarget
     getter playoutDelayHint
     getter rtcpTransport
     getter track
@@ -6749,6 +6750,7 @@ interface RTCRtpReceiver
     method getParameters
     method getStats
     method getSynchronizationSources
+    setter jitterBufferTarget
     setter playoutDelayHint
 interface RTCRtpSender
     static method getCapabilities

diff --git a/third_party/blink/web_tests/webexposed/global-interface-listing-expected.txt b/third_party/blink/web_tests/webexposed/global-interface-listing-expected.txt
@@ -7746,6 +7746,7 @@ interface RTCPeerConnectionIceEvent : Event
 interface RTCRtpReceiver
     static method getCapabilities
     attribute @@toStringTag
+    getter jitterBufferTarget
     getter playoutDelayHint
     getter rtcpTransport
     getter track
@@ -7756,6 +7757,7 @@ interface RTCRtpReceiver
     method getParameters
     method getStats
     method getSynchronizationSources
+    setter jitterBufferTarget
     setter playoutDelayHint
 interface RTCRtpSender
     static method getCapabilities

diff --git a/tools/metrics/histograms/enums.xml b/tools/metrics/histograms/enums.xml
@@ -11240,6 +11240,10 @@ Called by update_use_counter_feature_enum.py.-->
   <int value="4859" label="V8PointerEvent_GetCoalescedEvents_Method"/>
   <int value="4861" label="CSSFunctions"/>
   <int value="4862" label="CSSPageRule"/>
+  <int value="4863"
+      label="V8RTCRtpReceiver_JitterBufferTarget_AttributeGetter"/>
+  <int value="4864"
+      label="V8RTCRtpReceiver_JitterBufferTarget_AttributeSetter"/>
 </enum>
 
 <enum name="FeaturePolicyFeature">