Bug 1570212 - Convert media.webrtc.net.force_disable_rtcp_reception to a static pref. r=jya

Differential Revision: https://phabricator.services.mozilla.com/D40154

--HG--
extra : moz-landing-system : lando

media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp

@@ -34,6 +34,7 @@
 #include "mozilla/Preferences.h"
 #include "mozilla/SharedThreadPool.h"
 #include "mozilla/Sprintf.h"
+#include "mozilla/StaticPrefs_media.h"
 #include "mozilla/TaskQueue.h"
 #include "mozilla/UniquePtr.h"
 #include "mozilla/UniquePtrExtensions.h"
@@ -64,14 +65,6 @@ mozilla::LazyLogModule gMediaPipelineLog("MediaPipeline");
 
 namespace mozilla {
 
-// When enabled, this pref disables the reception of RTCP. This is used
-// for testing.
-static const auto kQuashRtcpRxPref =
-    NS_LITERAL_CSTRING("media.webrtc.net.force_disable_rtcp_reception");
-Atomic<bool, ReleaseAcquire> MediaPipeline::sPrefsRegistered(false);
-Atomic<bool, ReleaseAcquire> MediaPipeline::sForceDisableRtcpReceptionPref(
-    false);
-
 // An async inserter for audio data, to avoid running audio codec encoders
 // on the MSG/input audio thread.  Basically just bounces all the audio
 // data to a single audio processing/input queue.  We could if we wanted to
@@ -271,13 +264,6 @@ MediaPipeline::MediaPipeline(const std::string& aPc,
   } else {
     mConduit->SetTransmitterTransport(mTransport);
   }
-  if (!sPrefsRegistered.exchange(true)) {
-    MOZ_ASSERT(Preferences::IsServiceAvailable());
-    Preferences::AddAtomicBoolVarCache(&sForceDisableRtcpReceptionPref,
-                                       kQuashRtcpRxPref, false);
-    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
-            ("Creating pref cache: %s succeeded", kQuashRtcpRxPref.get()));
-  }
 }
 
 MediaPipeline::~MediaPipeline() {
@@ -619,7 +605,7 @@ void MediaPipeline::RtcpPacketReceived(MediaPacket& packet) {
   mPacketDumper->Dump(mLevel, dom::mozPacketDumpType::Rtcp, false,
                       packet.data(), packet.len());
 
-  if (sForceDisableRtcpReceptionPref) {
+  if (StaticPrefs::media_webrtc_net_force_disable_rtcp_reception()) {
     MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
             ("%s RTCP packet forced to be dropped", mDescription.c_str()));
     return;

media/webrtc/signaling/src/mediapipeline/MediaPipeline.h

@@ -260,11 +260,6 @@ class MediaPipeline : public sigslot::has_slots<> {
   bool IsRtp(const unsigned char* aData, size_t aLen) const;
   // Must be called on the STS thread.  Must be called after DetachMedia().
   void DetachTransport_s();
-
-  // Cached preferences are not tolerant of being registered more than once
-  static Atomic<bool, ReleaseAcquire> sPrefsRegistered;
-  // Cached pref media.webrtc.net.force_disable_rtcp_reception
-  static Atomic<bool, ReleaseAcquire> sForceDisableRtcpReceptionPref;
 };
 
 // A specialization of pipeline for reading from an input device

modules/libpref/init/StaticPrefList.yaml

@@ -4918,6 +4918,12 @@
   value: false
   mirror: always
 
+# This pref disables the reception of RTCP. It is used for testing.
+- name: media.webrtc.net.force_disable_rtcp_reception
+  type: ReleaseAcquireAtomicBool
+  value: false
+  mirror: always
+
 # TextTrack WebVTT Region extension support.
 - name: media.webvtt.regions.enabled
   type: bool