diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index e0b15dc..b407508 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -189,7 +189,7 @@
 
   BitrateAllocatorInterface* const bitrate_allocator_
       RTC_GUARDED_BY(rtp_transport_queue_);
-  // Constrains cached to be accessed from |rtp_transport_queue_|.
+  // Constrains cached to be accessed from `rtp_transport_queue_`.
   absl::optional<AudioSendStream::TargetAudioBitrateConstraints>
       cached_constraints_ RTC_GUARDED_BY(rtp_transport_queue_) = absl::nullopt;
   RtpTransportControllerSendInterface* const rtp_transport_;
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 357e080..db42efc 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -172,7 +172,7 @@
     SetupMockForSetupSendCodec(expect_set_encoder_call);
     SetupMockForCallEncoder();
 
-    // Use ISAC as default codec so as to prevent unnecessary |channel_proxy_|
+    // Use ISAC as default codec so as to prevent unnecessary `channel_proxy_`
     // calls from the default ctor behavior.
     stream_config_.send_codec_spec =
         AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
@@ -336,7 +336,7 @@
   ::testing::NiceMock<MockRtpRtcpInterface> rtp_rtcp_;
   ::testing::NiceMock<MockLimitObserver> limit_observer_;
   BitrateAllocator bitrate_allocator_;
-  // |worker_queue| is defined last to ensure all pending tasks are cancelled
+  // `worker_queue` is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   TaskQueueForTest worker_queue_;
   std::unique_ptr<AudioEncoder> audio_encoder_;
diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc
index 8710ced..2a80ea8 100644
--- a/audio/audio_transport_impl.cc
+++ b/audio/audio_transport_impl.cc
@@ -64,8 +64,8 @@
   }
 }
 
-// Resample audio in |frame| to given sample rate preserving the
-// channel count and place the result in |destination|.
+// Resample audio in `frame` to given sample rate preserving the
+// channel count and place the result in `destination`.
 int Resample(const AudioFrame& frame,
              const int destination_sample_rate,
              PushResampler<int16_t>* resampler,
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 57269cd..3ca3b51b 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -429,8 +429,8 @@
   }
 
   // Measure audio level (0-9)
-  // TODO(henrik.lundin) Use the |muted| information here too.
-  // TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| (see
+  // TODO(henrik.lundin) Use the `muted` information here too.
+  // TODO(deadbeef): Use RmsLevel for `_outputAudioLevel` (see
   // https://crbug.com/webrtc/7517).
   _outputAudioLevel.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds);
 
@@ -454,10 +454,10 @@
       // Compute ntp time.
       audio_frame->ntp_time_ms_ =
           ntp_estimator_.Estimate(audio_frame->timestamp_);
-      // |ntp_time_ms_| won't be valid until at least 2 RTCP SRs are received.
+      // `ntp_time_ms_` won't be valid until at least 2 RTCP SRs are received.
       if (audio_frame->ntp_time_ms_ > 0) {
-        // Compute |capture_start_ntp_time_ms_| so that
-        // |capture_start_ntp_time_ms_| + |elapsed_time_ms_| == |ntp_time_ms_|
+        // Compute `capture_start_ntp_time_ms_` so that
+        // `capture_start_ntp_time_ms_` + `elapsed_time_ms_` == `ntp_time_ms_`
         capture_start_ntp_time_ms_ =
             audio_frame->ntp_time_ms_ - audio_frame->elapsed_time_ms_;
       }
diff --git a/audio/channel_receive_frame_transformer_delegate.h b/audio/channel_receive_frame_transformer_delegate.h
index f59834d..04ad7c4 100644
--- a/audio/channel_receive_frame_transformer_delegate.h
+++ b/audio/channel_receive_frame_transformer_delegate.h
@@ -23,7 +23,7 @@
 
 // Delegates calls to FrameTransformerInterface to transform frames, and to
 // ChannelReceive to receive the transformed frames using the
-// |receive_frame_callback_| on the |channel_receive_thread_|.
+// `receive_frame_callback_` on the `channel_receive_thread_`.
 class ChannelReceiveFrameTransformerDelegate : public TransformedFrameCallback {
  public:
   using ReceiveFrameCallback =
@@ -34,12 +34,12 @@
       rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
       TaskQueueBase* channel_receive_thread);
 
-  // Registers |this| as callback for |frame_transformer_|, to get the
+  // Registers `this` as callback for `frame_transformer_`, to get the
   // transformed frames.
   void Init();
 
-  // Unregisters and releases the |frame_transformer_| reference, and resets
-  // |receive_frame_callback_| on |channel_receive_thread_|. Called from
+  // Unregisters and releases the `frame_transformer_` reference, and resets
+  // `receive_frame_callback_` on `channel_receive_thread_`. Called from
   // ChannelReceive destructor to prevent running the callback on a dangling
   // channel.
   void Reset();
@@ -55,7 +55,7 @@
       std::unique_ptr<TransformableFrameInterface> frame) override;
 
   // Delegates the call to ChannelReceive::OnReceivedPayloadData on the
-  // |channel_receive_thread_|, by calling |receive_frame_callback_|.
+  // `channel_receive_thread_`, by calling `receive_frame_callback_`.
   void ReceiveFrame(std::unique_ptr<TransformableFrameInterface> frame) const;
 
  protected:
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 67391af..663b947 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -98,12 +98,12 @@
       std::unique_ptr<AudioFrame> audio_frame) = 0;
   virtual RtpRtcpInterface* GetRtpRtcp() const = 0;
 
-  // In RTP we currently rely on RTCP packets (|ReceivedRTCPPacket|) to inform
+  // In RTP we currently rely on RTCP packets (`ReceivedRTCPPacket`) to inform
   // about RTT.
   // In media transport we rely on the TargetTransferRateObserver instead.
   // In other words, if you are using RTP, you should expect
-  // |ReceivedRTCPPacket| to be called, if you are using media transport,
-  // |OnTargetTransferRate| will be called.
+  // `ReceivedRTCPPacket` to be called, if you are using media transport,
+  // `OnTargetTransferRate` will be called.
   //
   // In future, RTP media will move to the media transport implementation and
   // these conditions will be removed.
diff --git a/audio/channel_send_frame_transformer_delegate.h b/audio/channel_send_frame_transformer_delegate.h
index 9b7eb33..6d9f0a8 100644
--- a/audio/channel_send_frame_transformer_delegate.h
+++ b/audio/channel_send_frame_transformer_delegate.h
@@ -23,8 +23,8 @@
 namespace webrtc {
 
 // Delegates calls to FrameTransformerInterface to transform frames, and to
-// ChannelSend to send the transformed frames using |send_frame_callback_| on
-// the |encoder_queue_|.
+// ChannelSend to send the transformed frames using `send_frame_callback_` on
+// the `encoder_queue_`.
 // OnTransformedFrame() can be called from any thread, the delegate ensures
 // thread-safe access to the ChannelSend callback.
 class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback {
@@ -40,12 +40,12 @@
       rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
       rtc::TaskQueue* encoder_queue);
 
-  // Registers |this| as callback for |frame_transformer_|, to get the
+  // Registers `this` as callback for `frame_transformer_`, to get the
   // transformed frames.
   void Init();
 
-  // Unregisters and releases the |frame_transformer_| reference, and resets
-  // |send_frame_callback_| under lock. Called from ChannelSend destructor to
+  // Unregisters and releases the `frame_transformer_` reference, and resets
+  // `send_frame_callback_` under lock. Called from ChannelSend destructor to
   // prevent running the callback on a dangling channel.
   void Reset();
 
@@ -64,8 +64,8 @@
   void OnTransformedFrame(
       std::unique_ptr<TransformableFrameInterface> frame) override;
 
-  // Delegates the call to ChannelSend::SendRtpAudio on the |encoder_queue_|,
-  // by calling |send_audio_callback_|.
+  // Delegates the call to ChannelSend::SendRtpAudio on the `encoder_queue_`,
+  // by calling `send_audio_callback_`.
   void SendFrame(std::unique_ptr<TransformableFrameInterface> frame) const;
 
  protected:
diff --git a/audio/null_audio_poller.cc b/audio/null_audio_poller.cc
index 22f575d..de2c5ca 100644
--- a/audio/null_audio_poller.cc
+++ b/audio/null_audio_poller.cc
@@ -47,7 +47,7 @@
 
   // Buffer to hold the audio samples.
   int16_t buffer[kNumSamples * kNumChannels];
-  // Output variables from |NeedMorePlayData|.
+  // Output variables from `NeedMorePlayData`.
   size_t n_samples;
   int64_t elapsed_time_ms;
   int64_t ntp_time_ms;
diff --git a/audio/remix_resample.h b/audio/remix_resample.h
index a45270b..bd8da76 100644
--- a/audio/remix_resample.h
+++ b/audio/remix_resample.h
@@ -17,19 +17,19 @@
 namespace webrtc {
 namespace voe {
 
-// Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|
+// Upmix or downmix and resample the audio to `dst_frame`. Expects `dst_frame`
 // to have its sample rate and channels members set to the desired values.
-// Updates the |samples_per_channel_| member accordingly.
+// Updates the `samples_per_channel_` member accordingly.
 //
-// This version has an AudioFrame |src_frame| as input and sets the output
-// |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the
+// This version has an AudioFrame `src_frame` as input and sets the output
+// `timestamp_`, `elapsed_time_ms_` and `ntp_time_ms_` members equals to the
 // input ones.
 void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame);
 
-// This version has a pointer to the samples |src_data| as input and receives
-// |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
+// This version has a pointer to the samples `src_data` as input and receives
+// `samples_per_channel`, `num_channels` and `sample_rate_hz` of the data as
 // parameters.
 void RemixAndResample(const int16_t* src_data,
                       size_t samples_per_channel,
diff --git a/audio/remix_resample_unittest.cc b/audio/remix_resample_unittest.cc
index d2155a6..a80476e 100644
--- a/audio/remix_resample_unittest.cc
+++ b/audio/remix_resample_unittest.cc
@@ -43,7 +43,7 @@
   AudioFrame golden_frame_;
 };
 
-// Sets the signal value to increase by |data| with every sample. Floats are
+// Sets the signal value to increase by `data` with every sample. Floats are
 // used so non-integer values result in rounding error, but not an accumulating
 // error.
 void SetMonoFrame(float data, int sample_rate_hz, AudioFrame* frame) {
@@ -62,7 +62,7 @@
   SetMonoFrame(data, frame->sample_rate_hz_, frame);
 }
 
-// Sets the signal value to increase by |left| and |right| with every sample in
+// Sets the signal value to increase by `left` and `right` with every sample in
 // each channel respectively.
 void SetStereoFrame(float left,
                     float right,
@@ -84,7 +84,7 @@
   SetStereoFrame(left, right, frame->sample_rate_hz_, frame);
 }
 
-// Sets the signal value to increase by |ch1|, |ch2|, |ch3|, |ch4| with every
+// Sets the signal value to increase by `ch1`, `ch2`, `ch3`, `ch4` with every
 // sample in each channel respectively.
 void SetQuadFrame(float ch1,
                   float ch2,
@@ -111,8 +111,8 @@
   EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_);
 }
 
-// Computes the best SNR based on the error between |ref_frame| and
-// |test_frame|. It allows for up to a |max_delay| in samples between the
+// Computes the best SNR based on the error between `ref_frame` and
+// `test_frame`. It allows for up to a `max_delay` in samples between the
 // signals to compensate for the resampling delay.
 float ComputeSNR(const AudioFrame& ref_frame,
                  const AudioFrame& test_frame,
diff --git a/audio/utility/audio_frame_operations.cc b/audio/utility/audio_frame_operations.cc
index e13a09b..8f3f37a 100644
--- a/audio/utility/audio_frame_operations.cc
+++ b/audio/utility/audio_frame_operations.cc
@@ -222,14 +222,14 @@
     size_t end = count;
     float start_g = 0.0f;
     if (current_frame_muted) {
-      // Fade out the last |count| samples of frame.
+      // Fade out the last `count` samples of frame.
       RTC_DCHECK(!previous_frame_muted);
       start = frame->samples_per_channel_ - count;
       end = frame->samples_per_channel_;
       start_g = 1.0f;
       inc = -inc;
     } else {
-      // Fade in the first |count| samples of frame.
+      // Fade in the first `count` samples of frame.
       RTC_DCHECK(previous_frame_muted);
     }
 
diff --git a/audio/utility/audio_frame_operations.h b/audio/utility/audio_frame_operations.h
index 2f1540b..7e954df 100644
--- a/audio/utility/audio_frame_operations.h
+++ b/audio/utility/audio_frame_operations.h
@@ -24,40 +24,40 @@
 // than a class.
 class AudioFrameOperations {
  public:
-  // Add samples in |frame_to_add| with samples in |result_frame|
-  // putting the results in |results_frame|.  The fields
-  // |vad_activity_| and |speech_type_| of the result frame are
-  // updated. If |result_frame| is empty (|samples_per_channel_|==0),
-  // the samples in |frame_to_add| are added to it.  The number of
+  // Add samples in `frame_to_add` with samples in `result_frame`
+  // putting the results in `results_frame`.  The fields
+  // `vad_activity_` and `speech_type_` of the result frame are
+  // updated. If `result_frame` is empty (`samples_per_channel_`==0),
+  // the samples in `frame_to_add` are added to it.  The number of
   // channels and number of samples per channel must match except when
-  // |result_frame| is empty.
+  // `result_frame` is empty.
   static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
 
   // |frame.num_channels_| will be updated. This version checks for sufficient
-  // buffer size and that |num_channels_| is mono. Use UpmixChannels
+  // buffer size and that `num_channels_` is mono. Use UpmixChannels
   // instead. TODO(bugs.webrtc.org/8649): remove.
   ABSL_DEPRECATED("bugs.webrtc.org/8649")
   static int MonoToStereo(AudioFrame* frame);
 
   // |frame.num_channels_| will be updated. This version checks that
-  // |num_channels_| is stereo. Use DownmixChannels
+  // `num_channels_` is stereo. Use DownmixChannels
   // instead. TODO(bugs.webrtc.org/8649): remove.
   ABSL_DEPRECATED("bugs.webrtc.org/8649")
   static int StereoToMono(AudioFrame* frame);
 
-  // Downmixes 4 channels |src_audio| to stereo |dst_audio|. This is an in-place
-  // operation, meaning |src_audio| and |dst_audio| may point to the same
+  // Downmixes 4 channels `src_audio` to stereo `dst_audio`. This is an in-place
+  // operation, meaning `src_audio` and `dst_audio` may point to the same
   // buffer.
   static void QuadToStereo(const int16_t* src_audio,
                            size_t samples_per_channel,
                            int16_t* dst_audio);
 
   // |frame.num_channels_| will be updated. This version checks that
-  // |num_channels_| is 4 channels.
+  // `num_channels_` is 4 channels.
   static int QuadToStereo(AudioFrame* frame);
 
-  // Downmixes |src_channels| |src_audio| to |dst_channels| |dst_audio|.
-  // This is an in-place operation, meaning |src_audio| and |dst_audio|
+  // Downmixes `src_channels` `src_audio` to `dst_channels` `dst_audio`.
+  // This is an in-place operation, meaning `src_audio` and `dst_audio`
   // may point to the same buffer. Supported channel combinations are
   // Stereo to Mono, Quad to Mono, and Quad to Stereo.
   static void DownmixChannels(const int16_t* src_audio,
@@ -67,26 +67,26 @@
                               int16_t* dst_audio);
 
   // |frame.num_channels_| will be updated. This version checks that
-  // |num_channels_| and |dst_channels| are valid and performs relevant downmix.
+  // `num_channels_` and `dst_channels` are valid and performs relevant downmix.
   // Supported channel combinations are N channels to Mono, and Quad to Stereo.
   static void DownmixChannels(size_t dst_channels, AudioFrame* frame);
 
   // |frame.num_channels_| will be updated. This version checks that
-  // |num_channels_| and |dst_channels| are valid and performs relevant
+  // `num_channels_` and `dst_channels` are valid and performs relevant
   // downmix. Supported channel combinations are Mono to N
   // channels. The single channel is replicated.
   static void UpmixChannels(size_t target_number_of_channels,
                             AudioFrame* frame);
 
-  // Swap the left and right channels of |frame|. Fails silently if |frame| is
+  // Swap the left and right channels of `frame`. Fails silently if `frame` is
   // not stereo.
   static void SwapStereoChannels(AudioFrame* frame);
 
-  // Conditionally zero out contents of |frame| for implementing audio mute:
-  //  |previous_frame_muted| &&  |current_frame_muted| - Zero out whole frame.
-  //  |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
-  // !|previous_frame_muted| &&  |current_frame_muted| - Fade-out at frame end.
-  // !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
+  // Conditionally zero out contents of `frame` for implementing audio mute:
+  //  `previous_frame_muted` &&  `current_frame_muted` - Zero out whole frame.
+  //  `previous_frame_muted` && !`current_frame_muted` - Fade-in at frame start.
+  // !`previous_frame_muted` &&  `current_frame_muted` - Fade-out at frame end.
+  // !`previous_frame_muted` && !`current_frame_muted` - Leave frame untouched.
   static void Mute(AudioFrame* frame,
                    bool previous_frame_muted,
                    bool current_frame_muted);
@@ -94,7 +94,7 @@
   // Zero out contents of frame.
   static void Mute(AudioFrame* frame);
 
-  // Halve samples in |frame|.
+  // Halve samples in `frame`.
   static void ApplyHalfGain(AudioFrame* frame);
 
   static int Scale(float left, float right, AudioFrame* frame);
diff --git a/audio/utility/channel_mixer.cc b/audio/utility/channel_mixer.cc
index 8867a3e..0f1e663 100644
--- a/audio/utility/channel_mixer.cc
+++ b/audio/utility/channel_mixer.cc
@@ -90,7 +90,7 @@
   frame->num_channels_ = output_channels_;
   frame->channel_layout_ = output_layout_;
 
-  // Copy the output result to the audio frame in |frame|.
+  // Copy the output result to the audio frame in `frame`.
   memcpy(
       frame->mutable_data(), out_audio,
       sizeof(int16_t) * frame->samples_per_channel() * frame->num_channels());
diff --git a/audio/utility/channel_mixer.h b/audio/utility/channel_mixer.h
index 8b6b7f5..2dea8eb 100644
--- a/audio/utility/channel_mixer.h
+++ b/audio/utility/channel_mixer.h
@@ -38,8 +38,8 @@
   ChannelMixer(ChannelLayout input_layout, ChannelLayout output_layout);
   ~ChannelMixer();
 
-  // Transforms all input channels corresponding to the selected |input_layout|
-  // to the number of channels in the selected |output_layout|.
+  // Transforms all input channels corresponding to the selected `input_layout`
+  // to the number of channels in the selected `output_layout`.
   // Example usage (downmix from stereo to mono):
   //
   //   ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO);
@@ -69,11 +69,11 @@
   // 1D array used as temporary storage during the transformation.
   std::unique_ptr<int16_t[]> audio_vector_;
 
-  // Number of elements allocated for |audio_vector_|.
+  // Number of elements allocated for `audio_vector_`.
   size_t audio_vector_size_ = 0;
 
   // Optimization case for when we can simply remap the input channels to output
-  // channels, i.e., when all scaling factors in |matrix_| equals 1.0.
+  // channels, i.e., when all scaling factors in `matrix_` equals 1.0.
   bool remapping_;
 
   // Delete the copy constructor and assignment operator.
diff --git a/audio/utility/channel_mixing_matrix.cc b/audio/utility/channel_mixing_matrix.cc
index 4baff8b..1244653 100644
--- a/audio/utility/channel_mixing_matrix.cc
+++ b/audio/utility/channel_mixing_matrix.cc
@@ -274,7 +274,7 @@
   // All channels should now be accounted for.
   RTC_DCHECK(unaccounted_inputs_.empty());
 
-  // See if the output |matrix_| is simply a remapping matrix.  If each input
+  // See if the output `matrix_` is simply a remapping matrix.  If each input
   // channel maps to a single output channel we can simply remap.  Doing this
   // programmatically is less fragile than logic checks on channel mappings.
   for (int output_ch = 0; output_ch < output_channels_; ++output_ch) {
@@ -287,7 +287,7 @@
     }
   }
 
-  // If we've gotten here, |matrix_| is simply a remapping.
+  // If we've gotten here, `matrix_` is simply a remapping.
   return true;
 }
 
diff --git a/audio/utility/channel_mixing_matrix.h b/audio/utility/channel_mixing_matrix.h
index 7aef47b..ee00860 100644
--- a/audio/utility/channel_mixing_matrix.h
+++ b/audio/utility/channel_mixing_matrix.h
@@ -29,7 +29,7 @@
   // Create the transformation matrix of input channels to output channels.
   // Updates the empty matrix with the transformation, and returns true
   // if the transformation is just a remapping of channels (no mixing).
-  // The size of |matrix| is |output_channels| x |input_channels|, i.e., the
+  // The size of `matrix` is `output_channels` x `input_channels`, i.e., the
   // number of rows equals the number of output channels and the number of
   // columns corresponds to the number of input channels.
   // This file is derived from Chromium's media/base/channel_mixing_matrix.h.
@@ -55,14 +55,14 @@
   void AccountFor(Channels ch);
   bool IsUnaccounted(Channels ch) const;
 
-  // Helper methods for checking if |ch| exists in either |input_layout_| or
-  // |output_layout_| respectively.
+  // Helper methods for checking if `ch` exists in either `input_layout_` or
+  // `output_layout_` respectively.
   bool HasInputChannel(Channels ch) const;
   bool HasOutputChannel(Channels ch) const;
 
-  // Helper methods for updating |matrix_| with the proper value for
-  // mixing |input_ch| into |output_ch|.  MixWithoutAccounting() does not
-  // remove the channel from |unaccounted_inputs_|.
+  // Helper methods for updating `matrix_` with the proper value for
+  // mixing `input_ch` into `output_ch`.  MixWithoutAccounting() does not
+  // remove the channel from `unaccounted_inputs_`.
   void Mix(Channels input_ch, Channels output_ch, float scale);
   void MixWithoutAccounting(Channels input_ch, Channels output_ch, float scale);
 
diff --git a/audio/voip/audio_channel.cc b/audio/voip/audio_channel.cc
index b4a50ee..4650d19 100644
--- a/audio/voip/audio_channel.cc
+++ b/audio/voip/audio_channel.cc
@@ -75,7 +75,7 @@
 
   audio_mixer_->RemoveSource(ingress_.get());
 
-  // TODO(bugs.webrtc.org/11581): unclear if we still need to clear |egress_|
+  // TODO(bugs.webrtc.org/11581): unclear if we still need to clear `egress_`
   // here.
   egress_.reset();
   ingress_.reset();
diff --git a/audio/voip/audio_egress.h b/audio/voip/audio_egress.h
index a39c7e2..989e5bd 100644
--- a/audio/voip/audio_egress.h
+++ b/audio/voip/audio_egress.h
@@ -52,7 +52,7 @@
 
   // Set the encoder format and payload type for AudioCodingModule.
   // It's possible to change the encoder type during its active usage.
-  // |payload_type| must be the type that is negotiated with peer through
+  // `payload_type` must be the type that is negotiated with peer through
   // offer/answer.
   void SetEncoder(int payload_type,
                   const SdpAudioFormat& encoder_format,
@@ -84,7 +84,7 @@
 
   // Send DTMF named event as specified by
   // https://tools.ietf.org/html/rfc4733#section-3.2
-  // |duration_ms| specifies the duration of DTMF packets that will be emitted
+  // `duration_ms` specifies the duration of DTMF packets that will be emitted
   // in place of real RTP packets instead.
   // This will return true when requested dtmf event is successfully scheduled
   // otherwise false when the dtmf queue reached maximum of 20 events.
@@ -139,7 +139,7 @@
     // newly received audio frame from AudioTransport.
     uint32_t frame_rtp_timestamp_ = 0;
 
-    // Flag to track mute state from caller. |previously_muted_| is used to
+    // Flag to track mute state from caller. `previously_muted_` is used to
     // track previous state as part of input to AudioFrameOperations::Mute
     // to implement fading effect when (un)mute is invoked.
     bool mute_ = false;
diff --git a/audio/voip/voip_core.cc b/audio/voip/voip_core.cc
index fd66379..8df1c59 100644
--- a/audio/voip/voip_core.cc
+++ b/audio/voip/voip_core.cc
@@ -55,7 +55,7 @@
 }
 
 bool VoipCore::InitializeIfNeeded() {
-  // |audio_device_module_| internally owns a lock and the whole logic here
+  // `audio_device_module_` internally owns a lock and the whole logic here
   // needs to be executed atomically once using another lock in VoipCore.
   // Further changes in this method will need to make sure that no deadlock is
   // introduced in the future.
@@ -178,7 +178,7 @@
   }
 
   if (no_channels_after_release) {
-    // TODO(bugs.webrtc.org/11581): unclear if we still need to clear |channel|
+    // TODO(bugs.webrtc.org/11581): unclear if we still need to clear `channel`
     // here.
     channel = nullptr;
 
diff --git a/audio/voip/voip_core.h b/audio/voip/voip_core.h
index 359e072..4393935 100644
--- a/audio/voip/voip_core.h
+++ b/audio/voip/voip_core.h
@@ -53,7 +53,7 @@
                  public VoipVolumeControl {
  public:
   // Construct VoipCore with provided arguments.
-  // ProcessThread implementation can be injected by |process_thread|
+  // ProcessThread implementation can be injected by `process_thread`
   // (mainly for testing purpose) and when set to nullptr, default
   // implementation will be used.
   VoipCore(rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
@@ -128,7 +128,7 @@
   // mode. Therefore it would be better to delay the logic as late as possible.
   bool InitializeIfNeeded();
 
-  // Fetches the corresponding AudioChannel assigned with given |channel|.
+  // Fetches the corresponding AudioChannel assigned with given `channel`.
   // Returns nullptr if not found.
   rtc::scoped_refptr<AudioChannel> GetChannel(ChannelId channel_id);
 
@@ -144,15 +144,15 @@
   std::unique_ptr<TaskQueueFactory> task_queue_factory_;
 
   // Synchronization is handled internally by AudioProcessing.
-  // Must be placed before |audio_device_module_| for proper destruction.
+  // Must be placed before `audio_device_module_` for proper destruction.
   rtc::scoped_refptr<AudioProcessing> audio_processing_;
 
   // Synchronization is handled internally by AudioMixer.
-  // Must be placed before |audio_device_module_| for proper destruction.
+  // Must be placed before `audio_device_module_` for proper destruction.
   rtc::scoped_refptr<AudioMixer> audio_mixer_;
 
   // Synchronization is handled internally by AudioTransportImpl.
-  // Must be placed before |audio_device_module_| for proper destruction.
+  // Must be placed before `audio_device_module_` for proper destruction.
   std::unique_ptr<AudioTransportImpl> audio_transport_;
 
   // Synchronization is handled internally by AudioDeviceModule.