diff --git a/call/BUILD.gn b/call/BUILD.gn
index 04b5e50588..d542f4b5b4 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -86,7 +86,7 @@ rtc_source_set("audio_sender_interface") {
 }
 
 # TODO(nisse): These RTP targets should be moved elsewhere
-# when interfaces have stabilized. See also TODO for |mock_rtp_interfaces|.
+# when interfaces have stabilized. See also TODO for `mock_rtp_interfaces`.
 rtc_library("rtp_interfaces") {
   # Client code SHOULD NOT USE THIS TARGET, but for now it needs to be public
   # because there exists client code that uses it.
@@ -558,7 +558,7 @@ if (rtc_include_tests) {
     }
   }
 
-  # TODO(eladalon): This should be moved, as with the TODO for |rtp_interfaces|.
+  # TODO(eladalon): This should be moved, as with the TODO for `rtp_interfaces`.
   rtc_source_set("mock_rtp_interfaces") {
     testonly = true
 
diff --git a/call/adaptation/resource_adaptation_processor.cc b/call/adaptation/resource_adaptation_processor.cc
index 741575ae38..3c06675a16 100644
--- a/call/adaptation/resource_adaptation_processor.cc
+++ b/call/adaptation/resource_adaptation_processor.cc
@@ -200,7 +200,7 @@ void ResourceAdaptationProcessor::OnResourceUsageStateMeasured(
     ResourceUsageState usage_state) {
   RTC_DCHECK_RUN_ON(task_queue_);
   RTC_DCHECK(resource);
-  // |resource| could have been removed after signalling.
+  // `resource` could have been removed after signalling.
   {
     MutexLock crit(&resources_lock_);
     if (absl::c_find(resources_, resource) == resources_.end()) {
@@ -261,7 +261,7 @@ ResourceAdaptationProcessor::OnResourceUnderuse(
   if (!most_limited_resources.empty() &&
       most_limited_restrictions.counters.Total() >=
           stream_adapter_->adaptation_counters().Total()) {
-    // If |reason_resource| is not one of the most limiting resources then abort
+    // If `reason_resource` is not one of the most limiting resources then abort
     // adaptation.
     if (absl::c_find(most_limited_resources, reason_resource) ==
         most_limited_resources.end()) {
diff --git a/call/adaptation/resource_adaptation_processor.h b/call/adaptation/resource_adaptation_processor.h
index c84d359fec..3e273081f8 100644
--- a/call/adaptation/resource_adaptation_processor.h
+++ b/call/adaptation/resource_adaptation_processor.h
@@ -129,7 +129,7 @@ class ResourceAdaptationProcessor : public ResourceAdaptationProcessorInterface,
                                  const VideoAdaptationCounters& counters)
       RTC_RUN_ON(task_queue_);
 
-  // Searches |adaptation_limits_by_resources_| for each resource with the
+  // Searches `adaptation_limits_by_resources_` for each resource with the
   // highest total adaptation counts. Adaptation up may only occur if the
   // resource performing the adaptation is the only most limited resource. This
   // function returns the list of all most limited resources as well as the
diff --git a/call/adaptation/resource_adaptation_processor_unittest.cc b/call/adaptation/resource_adaptation_processor_unittest.cc
index 5e4f44b221..7020b22daf 100644
--- a/call/adaptation/resource_adaptation_processor_unittest.cc
+++ b/call/adaptation/resource_adaptation_processor_unittest.cc
@@ -290,7 +290,7 @@ TEST_F(ResourceAdaptationProcessorTest, OnlyMostLimitedResourceMayAdaptUp) {
   EXPECT_EQ(2, restrictions_listener_.adaptation_counters().Total());
   RestrictSource(restrictions_listener_.restrictions());
 
-  // |other_resource_| is most limited, resource_ can't adapt up.
+  // `other_resource_` is most limited, resource_ can't adapt up.
   resource_->SetUsageState(ResourceUsageState::kUnderuse);
   EXPECT_EQ(2, restrictions_listener_.adaptation_counters().Total());
   RestrictSource(restrictions_listener_.restrictions());
@@ -298,7 +298,7 @@ TEST_F(ResourceAdaptationProcessorTest, OnlyMostLimitedResourceMayAdaptUp) {
   EXPECT_EQ(1, restrictions_listener_.adaptation_counters().Total());
   RestrictSource(restrictions_listener_.restrictions());
 
-  // |resource_| and |other_resource_| are now most limited, so both must
+  // `resource_` and `other_resource_` are now most limited, so both must
   // signal underuse to adapt up.
   other_resource_->SetUsageState(ResourceUsageState::kUnderuse);
   EXPECT_EQ(1, restrictions_listener_.adaptation_counters().Total());
@@ -440,7 +440,7 @@ TEST_F(ResourceAdaptationProcessorTest,
       DegradationPreference::MAINTAIN_FRAMERATE);
   SetInputStates(true, kDefaultFrameRate, kDefaultFrameSize);
 
-  // Wait for |resource_| to signal oversue first so we know that the delegate
+  // Wait for `resource_` to signal oversue first so we know that the delegate
   // has passed it on to the processor's task queue.
   rtc::Event resource_event;
   TaskQueueForTest resource_task_queue("ResourceTaskQueue");
@@ -466,24 +466,24 @@ TEST_F(ResourceAdaptationProcessorTest,
 
   rtc::Event overuse_event;
   TaskQueueForTest resource_task_queue("ResourceTaskQueue");
-  // Queues task for |resource_| overuse while |processor_| is still listening.
+  // Queues task for `resource_` overuse while `processor_` is still listening.
   resource_task_queue.PostTask(ToQueuedTask([&]() {
     resource_->SetUsageState(ResourceUsageState::kOveruse);
     overuse_event.Set();
   }));
   EXPECT_TRUE(overuse_event.Wait(kDefaultTimeoutMs));
-  // Once we know the overuse task is queued, remove |resource_| so that
-  // |processor_| is not listening to it.
+  // Once we know the overuse task is queued, remove `resource_` so that
+  // `processor_` is not listening to it.
   processor_->RemoveResource(resource_);
 
-  // Runs the queued task so |processor_| gets signalled kOveruse from
-  // |resource_| even though |processor_| was not listening.
+  // Runs the queued task so `processor_` gets signalled kOveruse from
+  // `resource_` even though `processor_` was not listening.
   WaitUntilTaskQueueIdle();
 
-  // No restrictions should change even though |resource_| signaled |kOveruse|.
+  // No restrictions should change even though `resource_` signaled `kOveruse`.
   EXPECT_EQ(0u, restrictions_listener_.restrictions_updated_count());
 
-  // Delete |resource_| for cleanup.
+  // Delete `resource_` for cleanup.
   resource_ = nullptr;
 }
 
@@ -500,7 +500,7 @@ TEST_F(ResourceAdaptationProcessorTest,
   processor_->RemoveResource(resource_);
   EXPECT_EQ(0, restrictions_listener_.adaptation_counters().Total());
 
-  // Delete |resource_| for cleanup.
+  // Delete `resource_` for cleanup.
   resource_ = nullptr;
 }
 
@@ -522,14 +522,14 @@ TEST_F(ResourceAdaptationProcessorTest,
   RestrictSource(restrictions_listener_.restrictions());
   EXPECT_EQ(2, restrictions_listener_.adaptation_counters().Total());
 
-  // Removing most limited |resource_| should revert us back to
+  // Removing most limited `resource_` should revert us back to
   processor_->RemoveResource(resource_);
   EXPECT_EQ(1, restrictions_listener_.adaptation_counters().Total());
   EXPECT_EQ(next_limited_restrictions, restrictions_listener_.restrictions());
   EXPECT_EQ(next_limited_counters,
             restrictions_listener_.adaptation_counters());
 
-  // Delete |resource_| for cleanup.
+  // Delete `resource_` for cleanup.
   resource_ = nullptr;
 }
 
@@ -556,8 +556,8 @@ TEST_F(ResourceAdaptationProcessorTest,
   resource_->SetUsageState(ResourceUsageState::kUnderuse);
   EXPECT_EQ(2, restrictions_listener_.adaptation_counters().Total());
 
-  // Removing most limited |resource_| should revert us back to, even though we
-  // did not call RestrictSource() after |resource_| was overused. Normally
+  // Removing most limited `resource_` should revert us back to, even though we
+  // did not call RestrictSource() after `resource_` was overused. Normally
   // adaptation for MAINTAIN_FRAMERATE would be blocked here but for removal we
   // allow this anyways.
   processor_->RemoveResource(resource_);
@@ -566,7 +566,7 @@ TEST_F(ResourceAdaptationProcessorTest,
   EXPECT_EQ(next_limited_counters,
             restrictions_listener_.adaptation_counters());
 
-  // Delete |resource_| for cleanup.
+  // Delete `resource_` for cleanup.
   resource_ = nullptr;
 }
 
@@ -588,12 +588,12 @@ TEST_F(ResourceAdaptationProcessorTest,
       restrictions_listener_.adaptation_counters();
   EXPECT_EQ(2, restrictions_listener_.adaptation_counters().Total());
 
-  // Removing most limited |resource_| should revert us back to
+  // Removing most limited `resource_` should revert us back to
   processor_->RemoveResource(other_resource_);
   EXPECT_EQ(current_restrictions, restrictions_listener_.restrictions());
   EXPECT_EQ(current_counters, restrictions_listener_.adaptation_counters());
 
-  // Delete |other_resource_| for cleanup.
+  // Delete `other_resource_` for cleanup.
   other_resource_ = nullptr;
 }
 
@@ -617,7 +617,7 @@ TEST_F(ResourceAdaptationProcessorTest,
   RestrictSource(restrictions_listener_.restrictions());
   EXPECT_EQ(2, restrictions_listener_.adaptation_counters().Total());
 
-  // Revert to |other_resource_| when removing |resource_| even though the
+  // Revert to `other_resource_` when removing `resource_` even though the
   // degradation preference was different when it was overused.
   processor_->RemoveResource(resource_);
   EXPECT_EQ(next_limited_counters,
@@ -629,7 +629,7 @@ TEST_F(ResourceAdaptationProcessorTest,
       DegradationPreference::MAINTAIN_FRAMERATE);
   EXPECT_EQ(next_limited_restrictions, restrictions_listener_.restrictions());
 
-  // Delete |resource_| for cleanup.
+  // Delete `resource_` for cleanup.
   resource_ = nullptr;
 }
 
@@ -653,7 +653,7 @@ TEST_F(ResourceAdaptationProcessorTest,
   video_stream_adapter_->SetDegradationPreference(
       DegradationPreference::DISABLED);
 
-  // Revert to |other_resource_| when removing |resource_| even though the
+  // Revert to `other_resource_` when removing `resource_` even though the
   // current degradataion preference is disabled.
   processor_->RemoveResource(resource_);
 
@@ -665,7 +665,7 @@ TEST_F(ResourceAdaptationProcessorTest,
   EXPECT_EQ(next_limited_counters,
             restrictions_listener_.adaptation_counters());
 
-  // Delete |resource_| for cleanup.
+  // Delete `resource_` for cleanup.
   resource_ = nullptr;
 }
 
@@ -679,7 +679,7 @@ TEST_F(ResourceAdaptationProcessorTest,
   resource_->SetUsageState(ResourceUsageState::kOveruse);
   EXPECT_EQ(0u, restrictions_listener_.restrictions_updated_count());
 
-  // Delete |resource_| for cleanup.
+  // Delete `resource_` for cleanup.
   resource_ = nullptr;
 }
 
@@ -692,7 +692,7 @@ TEST_F(ResourceAdaptationProcessorTest,
   other_resource_->SetUsageState(ResourceUsageState::kOveruse);
   RestrictSource(restrictions_listener_.restrictions());
   EXPECT_EQ(1, restrictions_listener_.adaptation_counters().Total());
-  // Adapt |resource_| up and then down so that both resource's are most
+  // Adapt `resource_` up and then down so that both resource's are most
   // limited at 1 adaptation.
   resource_->SetUsageState(ResourceUsageState::kOveruse);
   RestrictSource(restrictions_listener_.restrictions());
@@ -700,12 +700,12 @@ TEST_F(ResourceAdaptationProcessorTest,
   RestrictSource(restrictions_listener_.restrictions());
   EXPECT_EQ(1, restrictions_listener_.adaptation_counters().Total());
 
-  // Removing |resource_| has no effect since both |resource_| and
-  // |other_resource_| are most limited.
+  // Removing `resource_` has no effect since both `resource_` and
+  // `other_resource_` are most limited.
   processor_->RemoveResource(resource_);
   EXPECT_EQ(1, restrictions_listener_.adaptation_counters().Total());
 
-  // Delete |resource_| for cleanup.
+  // Delete `resource_` for cleanup.
   resource_ = nullptr;
 }
 
@@ -727,8 +727,8 @@ TEST_F(ResourceAdaptationProcessorTest,
   EXPECT_TRUE(has_reached_min_pixels);
   auto last_update_count = restrictions_listener_.restrictions_updated_count();
   other_resource_->SetUsageState(ResourceUsageState::kOveruse);
-  // Now both |resource_| and |other_resource_| are most limited. Underuse of
-  // |resource_| will not adapt up.
+  // Now both `resource_` and `other_resource_` are most limited. Underuse of
+  // `resource_` will not adapt up.
   resource_->SetUsageState(ResourceUsageState::kUnderuse);
   EXPECT_EQ(last_update_count,
             restrictions_listener_.restrictions_updated_count());
diff --git a/call/adaptation/video_source_restrictions.h b/call/adaptation/video_source_restrictions.h
index 7f79a48e5d..004cc09055 100644
--- a/call/adaptation/video_source_restrictions.h
+++ b/call/adaptation/video_source_restrictions.h
@@ -61,8 +61,8 @@ class VideoSourceRestrictions {
   void set_max_frame_rate(absl::optional<double> max_frame_rate);
 
  private:
-  // These map to rtc::VideoSinkWants's |max_pixel_count| and
-  // |target_pixel_count|.
+  // These map to rtc::VideoSinkWants's `max_pixel_count` and
+  // `target_pixel_count`.
   absl::optional<size_t> max_pixels_per_frame_;
   absl::optional<size_t> target_pixels_per_frame_;
   absl::optional<double> max_frame_rate_;
diff --git a/call/adaptation/video_stream_adapter.cc b/call/adaptation/video_stream_adapter.cc
index 64e1a77786..49a4d6a1ce 100644
--- a/call/adaptation/video_stream_adapter.cc
+++ b/call/adaptation/video_stream_adapter.cc
@@ -640,7 +640,7 @@ VideoStreamAdapter::GetAdaptDownResolutionStepForBalanced(
     return first_step;
   }
   // We didn't decrease resolution so force it; amend a resolution resuction
-  // to the existing framerate reduction in |first_restrictions|.
+  // to the existing framerate reduction in `first_restrictions`.
   auto second_step = DecreaseResolution(input_state, first_restrictions);
   if (absl::holds_alternative<RestrictionsWithCounters>(second_step)) {
     return second_step;
diff --git a/call/adaptation/video_stream_adapter.h b/call/adaptation/video_stream_adapter.h
index 3c876b8970..7bf424a17e 100644
--- a/call/adaptation/video_stream_adapter.h
+++ b/call/adaptation/video_stream_adapter.h
@@ -39,8 +39,8 @@ class VideoSourceRestrictionsListener {
  public:
   virtual ~VideoSourceRestrictionsListener();
 
-  // The |restrictions| are filtered by degradation preference but not the
-  // |adaptation_counters|, which are currently only reported for legacy stats
+  // The `restrictions` are filtered by degradation preference but not the
+  // `adaptation_counters`, which are currently only reported for legacy stats
   // calculation purposes.
   virtual void OnVideoSourceRestrictionsUpdated(
       VideoSourceRestrictions restrictions,
diff --git a/call/bitrate_allocator.cc b/call/bitrate_allocator.cc
index 8e2006defa..1693661ef5 100644
--- a/call/bitrate_allocator.cc
+++ b/call/bitrate_allocator.cc
@@ -70,9 +70,9 @@ bool EnoughBitrateForAllObservers(
   return true;
 }
 
-// Splits |bitrate| evenly to observers already in |allocation|.
-// |include_zero_allocations| decides if zero allocations should be part of
-// the distribution or not. The allowed max bitrate is |max_multiplier| x
+// Splits `bitrate` evenly to observers already in `allocation`.
+// `include_zero_allocations` decides if zero allocations should be part of
+// the distribution or not. The allowed max bitrate is `max_multiplier` x
 // observer max bitrate.
 void DistributeBitrateEvenly(
     const std::vector<AllocatableTrack>& allocatable_tracks,
@@ -110,7 +110,7 @@ void DistributeBitrateEvenly(
   }
 }
 
-// From the available |bitrate|, each observer will be allocated a
+// From the available `bitrate`, each observer will be allocated a
 // proportional amount based upon its bitrate priority. If that amount is
 // more than the observer's capacity, it will be allocated its capacity, and
 // the excess bitrate is still allocated proportionally to other observers.
diff --git a/call/bitrate_allocator.h b/call/bitrate_allocator.h
index c0d664b6f0..204fc6f94d 100644
--- a/call/bitrate_allocator.h
+++ b/call/bitrate_allocator.h
@@ -53,9 +53,9 @@ struct MediaStreamAllocationConfig {
   uint32_t pad_up_bitrate_bps;
   int64_t priority_bitrate_bps;
   // True means track may not be paused by allocating 0 bitrate will allocate at
-  // least |min_bitrate_bps| for this observer, even if the BWE is too low,
+  // least `min_bitrate_bps` for this observer, even if the BWE is too low,
   // false will allocate 0 to the observer if BWE doesn't allow
-  // |min_bitrate_bps|.
+  // `min_bitrate_bps`.
   bool enforce_min_bitrate;
   // The amount of bitrate allocated to this observer relative to all other
   // observers. If an observer has twice the bitrate_priority of other
@@ -119,12 +119,12 @@ class BitrateAllocator : public BitrateAllocatorInterface {
   void OnNetworkEstimateChanged(TargetTransferRate msg);
 
   // Set the configuration used by the bandwidth management.
-  // |observer| updates bitrates if already in use.
-  // |config| is the configuration to use for allocation.
-  // Note that |observer|->OnBitrateUpdated() will be called
+  // `observer` updates bitrates if already in use.
+  // `config` is the configuration to use for allocation.
+  // Note that `observer`->OnBitrateUpdated() will be called
   // within the scope of this method with the current rtt, fraction_loss and
   // available bitrate and that the bitrate in OnBitrateUpdated will be zero if
-  // the |observer| is currently not allowed to send data.
+  // the `observer` is currently not allowed to send data.
   void AddObserver(BitrateAllocatorObserver* observer,
                    MediaStreamAllocationConfig config) override;
 
@@ -132,7 +132,7 @@ class BitrateAllocator : public BitrateAllocatorInterface {
   // allocation.
   void RemoveObserver(BitrateAllocatorObserver* observer) override;
 
-  // Returns initial bitrate allocated for |observer|. If |observer| is not in
+  // Returns initial bitrate allocated for `observer`. If `observer` is not in
   // the list of added observers, a best guess is returned.
   int GetStartBitrate(BitrateAllocatorObserver* observer) const override;
 
diff --git a/call/bitrate_allocator_unittest.cc b/call/bitrate_allocator_unittest.cc
index 00fb236948..e4f0566827 100644
--- a/call/bitrate_allocator_unittest.cc
+++ b/call/bitrate_allocator_unittest.cc
@@ -197,7 +197,7 @@ TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
       CreateTargetRateMessage(4000000, 0, 0, kDefaultProbingIntervalMs));
   EXPECT_EQ(3000000, allocator_->GetStartBitrate(&bitrate_observer));
 
-  // Expect |max_padding_bitrate_bps| to change to 0 if the observer is updated.
+  // Expect `max_padding_bitrate_bps` to change to 0 if the observer is updated.
   EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(
                                    AllocationLimitsEq(kMinSendBitrateBps, 0)));
   AddObserver(&bitrate_observer, kMinSendBitrateBps, 4000000, 0, true,
@@ -320,8 +320,8 @@ class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
 // intended.
 TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserver) {
   TestBitrateObserver bitrate_observer_1;
-  // Expect OnAllocationLimitsChanged with |min_send_bitrate_bps| = 0 since
-  // AddObserver is called with |enforce_min_bitrate| = false.
+  // Expect OnAllocationLimitsChanged with `min_send_bitrate_bps` = 0 since
+  // AddObserver is called with `enforce_min_bitrate` = false.
   EXPECT_CALL(limit_observer_,
               OnAllocationLimitsChanged(AllocationLimitsEq(0, 0)));
   EXPECT_CALL(limit_observer_,
@@ -421,8 +421,8 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserverWithPacketLoss) {
   const uint32_t kMinStartBitrateBps =
       kMinBitrateBps + std::max(20000u, kMinBitrateBps / 10);
 
-  // Expect OnAllocationLimitsChanged with |min_send_bitrate_bps| = 0 since
-  // AddObserver is called with |enforce_min_bitrate| = false.
+  // Expect OnAllocationLimitsChanged with `min_send_bitrate_bps` = 0 since
+  // AddObserver is called with `enforce_min_bitrate` = false.
   TestBitrateObserver bitrate_observer;
   EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(
                                    AllocationLimitsEq(0, 0, kMaxBitrateBps)));
@@ -494,7 +494,7 @@ TEST_F(BitrateAllocatorTest,
   const uint32_t kMinBitrateBps = 100000;
   const uint32_t kMaxBitrateBps = 400000;
 
-  // Register |bitrate_observer| and expect total allocation limits to change.
+  // Register `bitrate_observer` and expect total allocation limits to change.
   EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(AllocationLimitsEq(
                                    kMinBitrateBps, 0, kMaxBitrateBps)))
       .Times(1);
diff --git a/call/call.cc b/call/call.cc
index a0c33ff756..00f58d30a0 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -411,7 +411,7 @@ class Call final : public webrtc::Call,
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_
       RTC_GUARDED_BY(worker_thread_);
   std::set<VideoSendStream*> video_send_streams_ RTC_GUARDED_BY(worker_thread_);
-  // True if |video_send_streams_| is empty, false if not. The atomic variable
+  // True if `video_send_streams_` is empty, false if not. The atomic variable
   // is used to decide UMA send statistics behavior and enables avoiding a
   // PostTask().
   std::atomic<bool> video_send_streams_empty_{true};
@@ -434,7 +434,7 @@ class Call final : public webrtc::Call,
   // thread.
   ReceiveStats receive_stats_ RTC_GUARDED_BY(worker_thread_);
   SendStats send_stats_ RTC_GUARDED_BY(send_transport_sequence_checker_);
-  // |last_bandwidth_bps_| and |configured_max_padding_bitrate_bps_| being
+  // `last_bandwidth_bps_` and `configured_max_padding_bitrate_bps_` being
   // atomic avoids a PostTask. The variables are used for stats gathering.
   std::atomic<uint32_t> last_bandwidth_bps_{0};
   std::atomic<uint32_t> configured_max_padding_bitrate_bps_{0};
@@ -446,8 +446,8 @@ class Call final : public webrtc::Call,
   const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
   const Timestamp start_of_call_;
 
-  // Note that |task_safety_| needs to be at a greater scope than the task queue
-  // owned by |transport_send_| since calls might arrive on the network thread
+  // Note that `task_safety_` needs to be at a greater scope than the task queue
+  // owned by `transport_send_` since calls might arrive on the network thread
   // while Call is being deleted and the task queue is being torn down.
   const ScopedTaskSafety task_safety_;
 
@@ -554,7 +554,7 @@ class SharedModuleThread::Impl {
 
     if (ref_count_ == 1 && on_one_ref_remaining_) {
       auto moved_fn = std::move(on_one_ref_remaining_);
-      // NOTE: after this function returns, chances are that |this| has been
+      // NOTE: after this function returns, chances are that `this` has been
       // deleted - do not touch any member variables.
       // If the owner of the last reference implements a lambda that releases
       // that last reference inside of the callback (which is legal according
@@ -781,8 +781,8 @@ Call::Call(Clock* clock,
     : clock_(clock),
       task_queue_factory_(task_queue_factory),
       worker_thread_(GetCurrentTaskQueueOrThread()),
-      // If |network_task_queue_| was set to nullptr, network related calls
-      // must be made on |worker_thread_| (i.e. they're one and the same).
+      // If `network_task_queue_` was set to nullptr, network related calls
+      // must be made on `worker_thread_` (i.e. they're one and the same).
       network_thread_(config.network_task_queue_ ? config.network_task_queue_
                                                  : worker_thread_),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
@@ -1000,7 +1000,7 @@ void Call::DestroyAudioReceiveStream(
   receive_rtp_config_.erase(ssrc);
 
   UpdateAggregateNetworkState();
-  // TODO(bugs.webrtc.org/11993): Consider if deleting |audio_receive_stream|
+  // TODO(bugs.webrtc.org/11993): Consider if deleting `audio_receive_stream`
   // on the network thread would be better or if we'd need to tear down the
   // state in two phases.
   delete audio_receive_stream;
@@ -1025,7 +1025,7 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
-  // Copy ssrcs from |config| since |config| is moved.
+  // Copy ssrcs from `config` since `config` is moved.
   std::vector<uint32_t> ssrcs = config.rtp.ssrcs;
 
   VideoSendStream* send_stream = new VideoSendStream(
@@ -1120,10 +1120,10 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
 
   EnsureStarted();
 
-  // TODO(bugs.webrtc.org/11993): Move the registration between |receive_stream|
-  // and |video_receiver_controller_| out of VideoReceiveStream2 construction
+  // TODO(bugs.webrtc.org/11993): Move the registration between `receive_stream`
+  // and `video_receiver_controller_` out of VideoReceiveStream2 construction
   // and set it up asynchronously on the network thread (the registration and
-  // |video_receiver_controller_| need to live on the network thread).
+  // `video_receiver_controller_` need to live on the network thread).
   VideoReceiveStream2* receive_stream = new VideoReceiveStream2(
       task_queue_factory_, this, num_cpu_cores_,
       transport_send_->packet_router(), std::move(configuration),
@@ -1190,7 +1190,7 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
   FlexfecReceiveStreamImpl* receive_stream;
 
   // Unlike the video and audio receive streams, FlexfecReceiveStream implements
-  // RtpPacketSinkInterface itself, and hence its constructor passes its |this|
+  // RtpPacketSinkInterface itself, and hence its constructor passes its `this`
   // pointer to video_receiver_controller_->CreateStream(). Calling the
   // constructor while on the worker thread ensures that we don't call
   // OnRtpPacket until the constructor is finished and the object is
@@ -1407,9 +1407,9 @@ void Call::OnTargetTransferRate(TargetTransferRate msg) {
 
   // Ignore updates if bitrate is zero (the aggregate network state is
   // down) or if we're not sending video.
-  // Using |video_send_streams_empty_| is racy but as the caller can't
-  // reasonably expect synchronize with changes in |video_send_streams_| (being
-  // on |send_transport_sequence_checker|), we can avoid a PostTask this way.
+  // Using `video_send_streams_empty_` is racy but as the caller can't
+  // reasonably expect synchronize with changes in `video_send_streams_` (being
+  // on `send_transport_sequence_checker`), we can avoid a PostTask this way.
   if (target_bitrate_bps == 0 ||
       video_send_streams_empty_.load(std::memory_order_relaxed)) {
     send_stats_.PauseSendAndPacerBitrateCounters();
@@ -1565,8 +1565,8 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
     RTC_LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc "
                       << parsed_packet.Ssrc();
     // Destruction of the receive stream, including deregistering from the
-    // RtpDemuxer, is not protected by the |worker_thread_|.
-    // But deregistering in the |receive_rtp_config_| map is. So by not passing
+    // RtpDemuxer, is not protected by the `worker_thread_`.
+    // But deregistering in the `receive_rtp_config_` map is. So by not passing
     // the packet on to demuxing in this case, we prevent incoming packets to be
     // passed on via the demuxer to a receive stream which is being torned down.
     return DELIVERY_UNKNOWN_SSRC;
@@ -1617,7 +1617,7 @@ PacketReceiver::DeliveryStatus Call::DeliverPacket(
 
 void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
   // TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
-  // This method is called synchronously via |OnRtpPacket()| (see DeliverRtp)
+  // This method is called synchronously via `OnRtpPacket()` (see DeliverRtp)
   // on the same thread.
   RTC_DCHECK_RUN_ON(worker_thread_);
   RtpPacketReceived parsed_packet;
@@ -1631,8 +1631,8 @@ void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
     RTC_LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc "
                       << parsed_packet.Ssrc();
     // Destruction of the receive stream, including deregistering from the
-    // RtpDemuxer, is not protected by the |worker_thread_|.
-    // But deregistering in the |receive_rtp_config_| map is.
+    // RtpDemuxer, is not protected by the `worker_thread_`.
+    // But deregistering in the `receive_rtp_config_` map is.
     // So by not passing the packet on to demuxing in this case, we prevent
     // incoming packets to be passed on via the demuxer to a receive stream
     // which is being torn down.
diff --git a/call/call_config.h b/call/call_config.h
index ba6dec3ad6..f149790150 100644
--- a/call/call_config.h
+++ b/call/call_config.h
@@ -28,7 +28,7 @@ class AudioProcessing;
 class RtcEventLog;
 
 struct CallConfig {
-  // If |network_task_queue| is set to nullptr, Call will assume that network
+  // If `network_task_queue` is set to nullptr, Call will assume that network
   // related callbacks will be made on the same TQ as the Call instance was
   // constructed on.
   explicit CallConfig(RtcEventLog* event_log,
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 701dda7e62..71bd490851 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -424,7 +424,7 @@ void CallPerfTest::TestCaptureNtpTime(
       int64_t now_ms = clock_->TimeInMilliseconds();
       int64_t time_since_creation = now_ms - creation_time_ms_;
       if (time_since_creation < start_time_ms_) {
-        // Wait for |start_time_ms_| before start measuring.
+        // Wait for `start_time_ms_` before start measuring.
         return;
       }
 
@@ -797,7 +797,7 @@ TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
       ++encoder_inits_;
       if (encoder_inits_ == 1) {
         // First time initialization. Frame size is known.
-        // |expected_bitrate| is affected by bandwidth estimation before the
+        // `expected_bitrate` is affected by bandwidth estimation before the
         // first frame arrives to the encoder.
         uint32_t expected_bitrate = last_set_bitrate_kbps_ > 0
                                         ? last_set_bitrate_kbps_
@@ -888,9 +888,9 @@ TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
 // contrained to the test bitrate.
 //
 // |test_bitrate_from test_bitrate_to| bitrate constraint range
-// |test_bitrate_step| bitrate constraint update step during the test
+// `test_bitrate_step` bitrate constraint update step during the test
 // |min_bwe max_bwe| BWE range
-// |start_bwe| initial BWE
+// `start_bwe` initial BWE
 void CallPerfTest::TestMinAudioVideoBitrate(int test_bitrate_from,
                                             int test_bitrate_to,
                                             int test_bitrate_step,
diff --git a/call/call_unittest.cc b/call/call_unittest.cc
index 92a037f157..c84cc51951 100644
--- a/call/call_unittest.cc
+++ b/call/call_unittest.cc
@@ -375,7 +375,7 @@ TEST(CallTest, AddAdaptationResourceAfterCreatingVideoSendStream) {
   // Add a fake resource.
   auto fake_resource = FakeResource::Create("FakeResource");
   call->AddAdaptationResource(fake_resource);
-  // An adapter resource mirroring the |fake_resource| should now be present on
+  // An adapter resource mirroring the `fake_resource` should now be present on
   // both streams.
   auto injected_resource1 = FindResourceWhoseNameContains(
       stream1->GetAdaptationResources(), fake_resource->Name());
@@ -437,7 +437,7 @@ TEST(CallTest, AddAdaptationResourceBeforeCreatingVideoSendStream) {
   VideoSendStream* stream2 =
       call->CreateVideoSendStream(config.Copy(), encoder_config.Copy());
   EXPECT_NE(stream2, nullptr);
-  // An adapter resource mirroring the |fake_resource| should be present on both
+  // An adapter resource mirroring the `fake_resource` should be present on both
   // streams.
   auto injected_resource1 = FindResourceWhoseNameContains(
       stream1->GetAdaptationResources(), fake_resource->Name());
@@ -506,7 +506,7 @@ TEST(CallTest, SharedModuleThread) {
   };
 
   // Create our test instance and pass a lambda to it that gets executed when
-  // the reference count goes back to 1 - meaning |shared| again is the only
+  // the reference count goes back to 1 - meaning `shared` again is the only
   // reference, which means we can free the variable and deallocate the thread.
   rtc::scoped_refptr<SharedModuleThread> shared;
   shared =
diff --git a/call/fake_network_pipe.cc b/call/fake_network_pipe.cc
index 324a7bd793..4b5579dfc8 100644
--- a/call/fake_network_pipe.cc
+++ b/call/fake_network_pipe.cc
@@ -307,7 +307,7 @@ void FakeNetworkPipe::Process() {
             delivery_info.receive_time_us - packet.send_time();
         packet.IncrementArrivalTime(added_delay_us);
         packets_to_deliver.emplace(std::move(packet));
-        // |time_now_us| might be later than when the packet should have
+        // `time_now_us` might be later than when the packet should have
         // arrived, due to NetworkProcess being called too late. For stats, use
         // the time it should have been on the link.
         total_packet_delay_us_ += added_delay_us;
diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h
index 1e5bb513bf..fadae337f5 100644
--- a/call/fake_network_pipe.h
+++ b/call/fake_network_pipe.h
@@ -44,7 +44,7 @@ class NetworkPacket {
                 absl::optional<int64_t> packet_time_us,
                 Transport* transport);
 
-  // Disallow copy constructor and copy assignment (no deep copies of |data_|).
+  // Disallow copy constructor and copy assignment (no deep copies of `data_`).
   NetworkPacket(const NetworkPacket&) = delete;
   ~NetworkPacket();
   NetworkPacket& operator=(const NetworkPacket&) = delete;
@@ -91,7 +91,7 @@ class NetworkPacket {
 // SimulatedNetworkInterface to simulate network behavior.
 class FakeNetworkPipe : public SimulatedPacketReceiverInterface {
  public:
-  // Will keep |network_behavior| alive while pipe is alive itself.
+  // Will keep `network_behavior` alive while pipe is alive itself.
   FakeNetworkPipe(Clock* clock,
                   std::unique_ptr<NetworkBehaviorInterface> network_behavior);
   FakeNetworkPipe(Clock* clock,
@@ -130,7 +130,7 @@ class FakeNetworkPipe : public SimulatedPacketReceiverInterface {
   bool SendRtcp(const uint8_t* packet, size_t length);
 
   // Methods for use with Transport interface. When/if packets are delivered,
-  // they will be passed to the instance specified by the |transport| parameter.
+  // they will be passed to the instance specified by the `transport` parameter.
   // Note that that instance must be in the map of active transports.
   bool SendRtp(const uint8_t* packet,
                size_t length,
@@ -203,13 +203,13 @@ class FakeNetworkPipe : public SimulatedPacketReceiverInterface {
   bool HasReceiver() const;
 
   Clock* const clock_;
-  // |config_lock| guards the mostly constant things like the callbacks.
+  // `config_lock` guards the mostly constant things like the callbacks.
   mutable Mutex config_lock_;
   const std::unique_ptr<NetworkBehaviorInterface> network_behavior_;
   PacketReceiver* receiver_ RTC_GUARDED_BY(config_lock_);
   Transport* const global_transport_;
 
-  // |process_lock| guards the data structures involved in delay and loss
+  // `process_lock` guards the data structures involved in delay and loss
   // processes, such as the packet queues.
   Mutex process_lock_;
   // Packets  are added at the back of the deque, this makes the deque ordered
diff --git a/call/fake_network_pipe_unittest.cc b/call/fake_network_pipe_unittest.cc
index 852a427222..b9c69c9b74 100644
--- a/call/fake_network_pipe_unittest.cc
+++ b/call/fake_network_pipe_unittest.cc
@@ -379,7 +379,7 @@ TEST_F(FakeNetworkPipeTest, BurstLoss) {
   fake_clock_.AdvanceTimeMilliseconds(1000);
   pipe->Process();
 
-  // Check that the average loss is |kLossPercent| percent.
+  // Check that the average loss is `kLossPercent` percent.
   int lost_packets = kNumPackets - receiver.delivered_sequence_numbers_.size();
   double loss_fraction = lost_packets / static_cast<double>(kNumPackets);
 
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index 688efb7b5e..7230a4a56d 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -189,7 +189,7 @@ void FlexfecReceiveStreamImpl::OnRtpPacket(const RtpPacketReceived& packet) {
 
   receiver_->OnRtpPacket(packet);
 
-  // Do not report media packets in the RTCP RRs generated by |rtp_rtcp_|.
+  // Do not report media packets in the RTCP RRs generated by `rtp_rtcp_`.
   if (packet.Ssrc() == config_.rtp.remote_ssrc) {
     rtp_receive_statistics_->OnRtpPacket(packet);
   }
diff --git a/call/rtp_bitrate_configurator.h b/call/rtp_bitrate_configurator.h
index 73c4e41f63..7ad83f8b0b 100644
--- a/call/rtp_bitrate_configurator.h
+++ b/call/rtp_bitrate_configurator.h
@@ -49,8 +49,8 @@ class RtpBitrateConfigurator {
   absl::optional<BitrateConstraints> UpdateWithRelayCap(DataRate cap);
 
  private:
-  // Applies update to the BitrateConstraints cached in |config_|, resetting
-  // with |new_start| if set.
+  // Applies update to the BitrateConstraints cached in `config_`, resetting
+  // with `new_start` if set.
   absl::optional<BitrateConstraints> UpdateConstraints(
       const absl::optional<int>& new_start);
 
diff --git a/call/rtp_config.h b/call/rtp_config.h
index 298644930c..ae5ae3b85f 100644
--- a/call/rtp_config.h
+++ b/call/rtp_config.h
@@ -78,7 +78,7 @@ struct RtpConfig {
 
   // The Rtp Stream Ids (aka RIDs) to send in the RID RTP header extension
   // if the extension is included in the list of extensions.
-  // If rids are specified, they should correspond to the |ssrcs| vector.
+  // If rids are specified, they should correspond to the `ssrcs` vector.
   // This means that:
   // 1. rids.size() == 0 || rids.size() == ssrcs.size().
   // 2. If rids is not empty, then |rids[i]| should use |ssrcs[i]|.
diff --git a/call/rtp_demuxer_unittest.cc b/call/rtp_demuxer_unittest.cc
index a4abab73ed..0c50b81035 100644
--- a/call/rtp_demuxer_unittest.cc
+++ b/call/rtp_demuxer_unittest.cc
@@ -631,13 +631,13 @@ TEST_F(RtpDemuxerTest, FirstSsrcAssociatedWithAnRsidIsNotForgotten) {
 
   constexpr uint32_t shared_ssrc = 100;
 
-  // First a packet with |rsid_a| is received, and |sink_a| is associated with
+  // First a packet with `rsid_a` is received, and `sink_a` is associated with
   // its SSRC.
   auto packet_a = CreatePacketWithSsrcRsid(shared_ssrc, rsid_a);
   EXPECT_CALL(sink_a, OnRtpPacket(SamePacketAs(*packet_a))).Times(1);
   EXPECT_TRUE(demuxer_.OnRtpPacket(*packet_a));
 
-  // Second, a packet with |rsid_b| is received. We guarantee that |sink_b|
+  // Second, a packet with `rsid_b` is received. We guarantee that `sink_b`
   // receives it.
   auto packet_b = CreatePacketWithSsrcRsid(shared_ssrc, rsid_b);
   EXPECT_CALL(sink_a, OnRtpPacket(_)).Times(0);
@@ -645,7 +645,7 @@ TEST_F(RtpDemuxerTest, FirstSsrcAssociatedWithAnRsidIsNotForgotten) {
   EXPECT_TRUE(demuxer_.OnRtpPacket(*packet_b));
 
   // Known edge-case; adding a new RSID association makes us re-examine all
-  // SSRCs. |sink_b| may or may not be associated with the SSRC now; we make
+  // SSRCs. `sink_b` may or may not be associated with the SSRC now; we make
   // no promises on that. However, since the RSID is specified and it cannot be
   // found the packet should be dropped.
   MockRtpPacketSink sink_c;
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index 7455060945..f1b90c7f52 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -201,8 +201,8 @@ class RtpTransportControllerSend final
   // Protected by internal locks.
   RateLimiter retransmission_rate_limiter_;
 
-  // TODO(perkj): |task_queue_| is supposed to replace |process_thread_|.
-  // |task_queue_| is defined last to ensure all pending tasks are cancelled
+  // TODO(perkj): `task_queue_` is supposed to replace `process_thread_`.
+  // `task_queue_` is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   rtc::TaskQueue task_queue_;
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend);
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 7fad89b20b..5b6712d87e 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -398,7 +398,7 @@ RtpVideoSender::RtpVideoSender(
   RTC_DCHECK_EQ(rtp_config_.ssrcs.size(), rtp_streams_.size());
   if (send_side_bwe_with_overhead_ && has_packet_feedback_)
     transport_->IncludeOverheadInPacedSender();
-  // SSRCs are assumed to be sorted in the same order as |rtp_modules|.
+  // SSRCs are assumed to be sorted in the same order as `rtp_modules`.
   for (uint32_t ssrc : rtp_config_.ssrcs) {
     // Restore state if it previously existed.
     const RtpPayloadState* state = nullptr;
@@ -817,14 +817,14 @@ void RtpVideoSender::OnBitrateUpdated(BitrateAllocationUpdate update,
   if (!fec_allowed_) {
     encoder_target_rate_bps_ = payload_bitrate_bps;
     // fec_controller_->UpdateFecRates() was still called so as to allow
-    // |fec_controller_| to update whatever internal state it might have,
-    // since |fec_allowed_| may be toggled back on at any moment.
+    // `fec_controller_` to update whatever internal state it might have,
+    // since `fec_allowed_` may be toggled back on at any moment.
   }
 
-    // Subtract packetization overhead from the encoder target. If target rate
-    // is really low, cap the overhead at 50%. This also avoids the case where
-    // |encoder_target_rate_bps_| is 0 due to encoder pause event while the
-    // packetization rate is positive since packets are still flowing.
+  // Subtract packetization overhead from the encoder target. If target rate
+  // is really low, cap the overhead at 50%. This also avoids the case where
+  // `encoder_target_rate_bps_` is 0 due to encoder pause event while the
+  // packetization rate is positive since packets are still flowing.
   uint32_t packetization_rate_bps =
       std::min(GetPacketizationOverheadRate(), encoder_target_rate_bps_ / 2);
   encoder_target_rate_bps_ -= packetization_rate_bps;
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index 991276fe79..c725214d8e 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -189,7 +189,7 @@ class RtpVideoSender : public RtpVideoSenderInterface,
 
   // When using the generic descriptor we want all simulcast streams to share
   // one frame id space (so that the SFU can switch stream without having to
-  // rewrite the frame id), therefore |shared_frame_id| has to live in a place
+  // rewrite the frame id), therefore `shared_frame_id` has to live in a place
   // where we are aware of all the different streams.
   int64_t shared_frame_id_ = 0;
   std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(mutex_);
diff --git a/call/simulated_network.h b/call/simulated_network.h
index 68d066cb82..b781b4658f 100644
--- a/call/simulated_network.h
+++ b/call/simulated_network.h
@@ -98,7 +98,7 @@ class SimulatedNetwork : public SimulatedNetworkInterface {
 
   mutable Mutex config_lock_;
 
-  // |process_checker_| guards the data structures involved in delay and loss
+  // `process_checker_` guards the data structures involved in delay and loss
   // processes, such as the packet queues.
   rtc::RaceChecker process_checker_;
   CoDelSimulation codel_controller_ RTC_GUARDED_BY(process_checker_);
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 86e5052151..7c70acddd3 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -266,10 +266,10 @@ class VideoReceiveStream : public MediaReceiveStream {
   virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
 
   // Sets and returns recording state. The old state is moved out
-  // of the video receive stream and returned to the caller, and |state|
+  // of the video receive stream and returned to the caller, and `state`
   // is moved in. If the state's callback is set, it will be called with
   // recordable encoded frames as they arrive.
-  // If |generate_key_frame| is true, the method will generate a key frame.
+  // If `generate_key_frame` is true, the method will generate a key frame.
   // When the function returns, it's guaranteed that all old callouts
   // to the returned callback has ceased.
   // Note: the client should not interpret the returned state's attributes, but
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index fd7a101b0a..65c2aba328 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -54,10 +54,10 @@ class VideoSendStream {
       // references to this media stream's SSRC.
       kMedia,
       // RTX streams are streams dedicated to retransmissions. They have a
-      // dependency on a single kMedia stream: |referenced_media_ssrc|.
+      // dependency on a single kMedia stream: `referenced_media_ssrc`.
       kRtx,
       // FlexFEC streams are streams dedicated to FlexFEC. They have a
-      // dependency on a single kMedia stream: |referenced_media_ssrc|.
+      // dependency on a single kMedia stream: `referenced_media_ssrc`.
       kFlexfec,
     };
 
@@ -67,9 +67,9 @@ class VideoSendStream {
     std::string ToString() const;
 
     StreamType type = StreamType::kMedia;
-    // If |type| is kRtx or kFlexfec this value is present. The referenced SSRC
+    // If `type` is kRtx or kFlexfec this value is present. The referenced SSRC
     // is the kMedia stream that this stream is performing retransmissions or
-    // FEC for. If |type| is kMedia, this value is null.
+    // FEC for. If `type` is kMedia, this value is null.
     absl::optional<uint32_t> referenced_media_ssrc;
     FrameCounts frame_counts;
     int width = 0;
@@ -170,7 +170,7 @@ class VideoSendStream {
 
     // Expected delay needed by the renderer, i.e. the frame will be delivered
     // this many milliseconds, if possible, earlier than expected render time.
-    // Only valid if |local_renderer| is set.
+    // Only valid if `local_renderer` is set.
     int render_delay_ms = 0;
 
     // Target delay in milliseconds. A positive value indicates this stream is