Use backticks not vertical bars to denote variables in comments for /sdk

Bug: webrtc:12338
Change-Id: Ifaad29ccb63b0f2f3aeefb77dae061ebc7f87e6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227024
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34561}

sdk/android/src/java/org/webrtc/AndroidVideoDecoder.java

@@ -514,7 +514,7 @@ class AndroidVideoDecoder implements VideoDecoder, VideoSink {
       throw new AssertionError("Stride is not divisible by two: " + stride);
     }
 
-    // Note that the case with odd |sliceHeight| is handled in a special way.
+    // Note that the case with odd `sliceHeight` is handled in a special way.
     // The chroma height contained in the payload is rounded down instead of
     // up, making it one row less than what we expect in WebRTC. Therefore, we
     // have to duplicate the last chroma rows for this case. Also, the offset

sdk/android/src/java/org/webrtc/Camera1Session.java

@@ -133,7 +133,7 @@ class Camera1Session implements CameraSession {
 
   private static CaptureFormat findClosestCaptureFormat(
       android.hardware.Camera.Parameters parameters, int width, int height, int framerate) {
-    // Find closest supported format for |width| x |height| @ |framerate|.
+    // Find closest supported format for `width` x `height` @ `framerate`.
     final List<CaptureFormat.FramerateRange> supportedFramerates =
         Camera1Enumerator.convertFramerates(parameters.getSupportedPreviewFpsRange());
     Logging.d(TAG, "Available fps ranges: " + supportedFramerates);

sdk/android/src/java/org/webrtc/EglBase14Impl.java

@@ -69,7 +69,7 @@ class EglBase14Impl implements EglBase14 {
   }
 
   // Create a new context with the specified config type, sharing data with sharedContext.
-  // |sharedContext| may be null.
+  // `sharedContext` may be null.
   public EglBase14Impl(EGLContext sharedContext, int[] configAttributes) {
     eglDisplay = getEglDisplay();
     eglConfig = getEglConfig(eglDisplay, configAttributes);

sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java

@@ -22,7 +22,7 @@ import org.webrtc.Logging;
 // This class wraps control of three different platform effects. Supported
 // effects are: AcousticEchoCanceler (AEC) and NoiseSuppressor (NS).
 // Calling enable() will active all effects that are
-// supported by the device if the corresponding |shouldEnableXXX| member is set.
+// supported by the device if the corresponding `shouldEnableXXX` member is set.
 class WebRtcAudioEffects {
   private static final boolean DEBUG = false;
 
@@ -71,7 +71,7 @@ class WebRtcAudioEffects {
   }
 
   // Call this method to enable or disable the platform AEC. It modifies
-  // |shouldEnableAec| which is used in enable() where the actual state
+  // `shouldEnableAec` which is used in enable() where the actual state
   // of the AEC effect is modified. Returns true if HW AEC is supported and
   // false otherwise.
   public boolean setAEC(boolean enable) {
@@ -90,7 +90,7 @@ class WebRtcAudioEffects {
   }
 
   // Call this method to enable or disable the platform NS. It modifies
-  // |shouldEnableNs| which is used in enable() where the actual state
+  // `shouldEnableNs` which is used in enable() where the actual state
   // of the NS effect is modified. Returns true if HW NS is supported and
   // false otherwise.
   public boolean setNS(boolean enable) {
@@ -180,7 +180,7 @@ class WebRtcAudioEffects {
     }
   }
 
-  // Returns true for effect types in |type| that are of "VoIP" types:
+  // Returns true for effect types in `type` that are of "VoIP" types:
   // Acoustic Echo Canceler (AEC) or Automatic Gain Control (AGC) or
   // Noise Suppressor (NS). Note that, an extra check for support is needed
   // in each comparison since some devices includes effects in the
@@ -217,7 +217,7 @@ class WebRtcAudioEffects {
   }
 
   // Returns true if an effect of the specified type is available. Functionally
-  // equivalent to (NoiseSuppressor|AutomaticGainControl|...).isAvailable(), but
+  // equivalent to (NoiseSuppressor`AutomaticGainControl`...).isAvailable(), but
   // faster as it avoids the expensive OS call to enumerate effects.
   private static boolean isEffectTypeAvailable(UUID effectType, UUID blockListedUuid) {
     Descriptor[] effects = getAvailableEffects();

sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java

@@ -237,7 +237,7 @@ class WebRtcAudioRecord {
 
   // Returns true if verifyAudioConfig() succeeds. This value is set after a specific delay when
   // startRecording() has been called. Hence, should preferably be called in combination with
-  // stopRecording() to ensure that it has been set properly. |isAudioConfigVerified| is
+  // stopRecording() to ensure that it has been set properly. `isAudioConfigVerified` is
   // enabled in WebRtcAudioRecord to ensure that the returned value is valid.
   @CalledByNative
   boolean isAudioSourceMatchingRecordingSession() {
@@ -491,7 +491,7 @@ class WebRtcAudioRecord {
       long nativeAudioRecordJni, ByteBuffer byteBuffer);
   private native void nativeDataIsRecorded(long nativeAudioRecordJni, int bytes);
 
-  // Sets all recorded samples to zero if |mute| is true, i.e., ensures that
+  // Sets all recorded samples to zero if `mute` is true, i.e., ensures that
   // the microphone is muted.
   public void setMicrophoneMute(boolean mute) {
     Logging.w(TAG, "setMicrophoneMute(" + mute + ")");

sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java

@@ -76,7 +76,7 @@ class WebRtcAudioTrack {
   private @Nullable AudioTrackThread audioThread;
   private final VolumeLogger volumeLogger;
 
-  // Samples to be played are replaced by zeros if |speakerMute| is set to true.
+  // Samples to be played are replaced by zeros if `speakerMute` is set to true.
   // Can be used to ensure that the speaker is fully muted.
   private volatile boolean speakerMute;
   private byte[] emptyBytes;
@@ -218,9 +218,9 @@ class WebRtcAudioTrack {
     Logging.d(TAG, "minBufferSizeInBytes: " + minBufferSizeInBytes);
     // For the streaming mode, data must be written to the audio sink in
     // chunks of size (given by byteBuffer.capacity()) less than or equal
-    // to the total buffer size |minBufferSizeInBytes|. But, we have seen
+    // to the total buffer size `minBufferSizeInBytes`. But, we have seen
     // reports of "getMinBufferSize(): error querying hardware". Hence, it
-    // can happen that |minBufferSizeInBytes| contains an invalid value.
+    // can happen that `minBufferSizeInBytes` contains an invalid value.
     if (minBufferSizeInBytes < byteBuffer.capacity()) {
       reportWebRtcAudioTrackInitError("AudioTrack.getMinBufferSize returns an invalid value.");
       return -1;
@@ -559,7 +559,7 @@ class WebRtcAudioTrack {
       long nativeAudioTrackJni, ByteBuffer byteBuffer);
   private static native void nativeGetPlayoutData(long nativeAudioTrackJni, int bytes);
 
-  // Sets all samples to be played out to zero if |mute| is true, i.e.,
+  // Sets all samples to be played out to zero if `mute` is true, i.e.,
   // ensures that the speaker is muted.
   public void setSpeakerMute(boolean mute) {
     Logging.w(TAG, "setSpeakerMute(" + mute + ")");

sdk/android/src/jni/android_metrics.cc

@@ -31,14 +31,14 @@ static ScopedJavaLocalRef<jobject> JNI_Metrics_GetAndReset(JNIEnv* jni) {
   std::map<std::string, std::unique_ptr<metrics::SampleInfo>> histograms;
   metrics::GetAndReset(&histograms);
   for (const auto& kv : histograms) {
-    // Create and add samples to |HistogramInfo|.
+    // Create and add samples to `HistogramInfo`.
     ScopedJavaLocalRef<jobject> j_info = Java_HistogramInfo_Constructor(
         jni, kv.second->min, kv.second->max,
         static_cast<int>(kv.second->bucket_count));
     for (const auto& sample : kv.second->samples) {
       Java_HistogramInfo_addSample(jni, j_info, sample.first, sample.second);
     }
-    // Add |HistogramInfo| to |Metrics|.
+    // Add `HistogramInfo` to `Metrics`.
     ScopedJavaLocalRef<jstring> j_name = NativeToJavaString(jni, kv.first);
     Java_Metrics_add(jni, j_metrics, j_name, j_info);
   }

sdk/android/src/jni/android_network_monitor.cc

@@ -376,7 +376,7 @@ rtc::NetworkBindingResult AndroidNetworkMonitor::BindSocketToNetwork(
     rv = lollipopSetNetworkForSocket(*network_handle, socket_fd);
   }
 
-  // If |network| has since disconnected, |rv| will be ENONET. Surface this as
+  // If `network` has since disconnected, `rv` will be ENONET. Surface this as
   // ERR_NETWORK_CHANGED, rather than MapSystemError(ENONET) which gives back
   // the less descriptive ERR_FAILED.
   if (rv == 0) {

sdk/android/src/jni/android_network_monitor.h

@@ -76,7 +76,7 @@ class AndroidNetworkMonitor : public rtc::NetworkMonitorInterface {
   void Start() override;
   void Stop() override;
 
-  // Does |this| NetworkMonitorInterface implement BindSocketToNetwork?
+  // Does `this` NetworkMonitorInterface implement BindSocketToNetwork?
   // Only Android returns true.
   virtual bool SupportsBindSocketToNetwork() const override { return true; }
 

sdk/android/src/jni/audio_device/aaudio_player.cc

@@ -200,7 +200,7 @@ aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
   }
 
   // Read audio data from the WebRTC source using the FineAudioBuffer object
-  // and write that data into |audio_data| to be played out by AAudio.
+  // and write that data into `audio_data` to be played out by AAudio.
   // Prime output with zeros during a short initial phase to avoid distortion.
   // TODO(henrika): do more work to figure out of if the initial forced silence
   // period is really needed.

sdk/android/src/jni/audio_device/aaudio_player.h

@@ -81,8 +81,8 @@ class AAudioPlayer final : public AudioOutput,
  protected:
   // AAudioObserverInterface implementation.
 
-  // For an output stream, this function should render and write |num_frames|
-  // of data in the streams current data format to the |audio_data| buffer.
+  // For an output stream, this function should render and write `num_frames`
+  // of data in the streams current data format to the `audio_data` buffer.
   // Called on a real-time thread owned by AAudio.
   aaudio_data_callback_result_t OnDataCallback(void* audio_data,
                                                int32_t num_frames) override;

sdk/android/src/jni/audio_device/aaudio_recorder.cc

@@ -157,7 +157,7 @@ void AAudioRecorder::OnErrorCallback(aaudio_result_t error) {
   }
 }
 
-// Read and process |num_frames| of data from the |audio_data| buffer.
+// Read and process `num_frames` of data from the `audio_data` buffer.
 // TODO(henrika): possibly add trace here to be included in systrace.
 // See https://developer.android.com/studio/profile/systrace-commandline.html.
 aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
@@ -191,7 +191,7 @@ aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
     RTC_DLOG(INFO) << "input latency: " << latency_millis_
                    << ", num_frames: " << num_frames;
   }
-  // Copy recorded audio in |audio_data| to the WebRTC sink using the
+  // Copy recorded audio in `audio_data` to the WebRTC sink using the
   // FineAudioBuffer object.
   fine_audio_buffer_->DeliverRecordedData(
       rtc::MakeArrayView(static_cast<const int16_t*>(audio_data),

sdk/android/src/jni/audio_device/aaudio_recorder.h

@@ -72,8 +72,8 @@ class AAudioRecorder : public AudioInput,
  protected:
   // AAudioObserverInterface implementation.
 
-  // For an input stream, this function should read |num_frames| of recorded
-  // data, in the stream's current data format, from the |audio_data| buffer.
+  // For an input stream, this function should read `num_frames` of recorded
+  // data, in the stream's current data format, from the `audio_data` buffer.
   // Called on a real-time thread owned by AAudio.
   aaudio_data_callback_result_t OnDataCallback(void* audio_data,
                                                int32_t num_frames) override;

sdk/android/src/jni/audio_device/audio_record_jni.cc

@@ -253,8 +253,8 @@ void AudioRecordJni::DataIsRecorded(JNIEnv* env,
   audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
                                           frames_per_buffer_);
   // We provide one (combined) fixed delay estimate for the APM and use the
-  // |playDelayMs| parameter only. Components like the AEC only sees the sum
-  // of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
+  // `playDelayMs` parameter only. Components like the AEC only sees the sum
+  // of `playDelayMs` and `recDelayMs`, hence the distributions does not matter.
   audio_device_buffer_->SetVQEData(total_delay_ms_, 0);
   if (audio_device_buffer_->DeliverRecordedData() == -1) {
     RTC_LOG(INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";

sdk/android/src/jni/audio_device/audio_record_jni.h

@@ -74,8 +74,8 @@ class AudioRecordJni : public AudioInput {
   int32_t EnableBuiltInNS(bool enable) override;
 
   // Called from Java side so we can cache the address of the Java-manged
-  // |byte_buffer| in |direct_buffer_address_|. The size of the buffer
-  // is also stored in |direct_buffer_capacity_in_bytes_|.
+  // `byte_buffer` in `direct_buffer_address_`. The size of the buffer
+  // is also stored in `direct_buffer_capacity_in_bytes_`.
   // This method will be called by the WebRtcAudioRecord constructor, i.e.,
   // on the same thread that this object is created on.
   void CacheDirectBufferAddress(JNIEnv* env,
@@ -83,8 +83,8 @@ class AudioRecordJni : public AudioInput {
                                 const JavaParamRef<jobject>& byte_buffer);
 
   // Called periodically by the Java based WebRtcAudioRecord object when
-  // recording has started. Each call indicates that there are |length| new
-  // bytes recorded in the memory area |direct_buffer_address_| and it is
+  // recording has started. Each call indicates that there are `length` new
+  // bytes recorded in the memory area `direct_buffer_address_` and it is
   // now time to send these to the consumer.
   // This method is called on a high-priority thread from Java. The name of
   // the thread is 'AudioRecordThread'.
@@ -111,10 +111,10 @@ class AudioRecordJni : public AudioInput {
   // possible values. See audio_common.h for details.
   const int total_delay_ms_;
 
-  // Cached copy of address to direct audio buffer owned by |j_audio_record_|.
+  // Cached copy of address to direct audio buffer owned by `j_audio_record_`.
   void* direct_buffer_address_;
 
-  // Number of bytes in the direct audio buffer owned by |j_audio_record_|.
+  // Number of bytes in the direct audio buffer owned by `j_audio_record_`.
   size_t direct_buffer_capacity_in_bytes_;
 
   // Number audio frames per audio buffer. Each audio frame corresponds to

sdk/android/src/jni/audio_device/audio_track_jni.h

@@ -71,14 +71,14 @@ class AudioTrackJni : public AudioOutput {
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 
   // Called from Java side so we can cache the address of the Java-manged
-  // |byte_buffer| in |direct_buffer_address_|. The size of the buffer
-  // is also stored in |direct_buffer_capacity_in_bytes_|.
+  // `byte_buffer` in `direct_buffer_address_`. The size of the buffer
+  // is also stored in `direct_buffer_capacity_in_bytes_`.
   // Called on the same thread as the creating thread.
   void CacheDirectBufferAddress(JNIEnv* env,
                                 const JavaParamRef<jobject>& byte_buffer);
   // Called periodically by the Java based WebRtcAudioTrack object when
-  // playout has started. Each call indicates that |length| new bytes should
-  // be written to the memory area |direct_buffer_address_| for playout.
+  // playout has started. Each call indicates that `length` new bytes should
+  // be written to the memory area `direct_buffer_address_` for playout.
   // This method is called on a high-priority thread from Java. The name of
   // the thread is 'AudioTrackThread'.
   void GetPlayoutData(JNIEnv* env, size_t length);
@@ -99,10 +99,10 @@ class AudioTrackJni : public AudioOutput {
   // AudioManager.
   const AudioParameters audio_parameters_;
 
-  // Cached copy of address to direct audio buffer owned by |j_audio_track_|.
+  // Cached copy of address to direct audio buffer owned by `j_audio_track_`.
   void* direct_buffer_address_;
 
-  // Number of bytes in the direct audio buffer owned by |j_audio_track_|.
+  // Number of bytes in the direct audio buffer owned by `j_audio_track_`.
   size_t direct_buffer_capacity_in_bytes_;
 
   // Number of audio frames per audio buffer. Each audio frame corresponds to

sdk/android/src/jni/audio_device/opensles_player.h

@@ -95,7 +95,7 @@ class OpenSLESPlayer : public AudioOutput {
   // Reads audio data in PCM format using the AudioDeviceBuffer.
   // Can be called both on the main thread (during Start()) and from the
   // internal audio thread while output streaming is active.
-  // If the |silence| flag is set, the audio is filled with zeros instead of
+  // If the `silence` flag is set, the audio is filled with zeros instead of
   // asking the WebRTC layer for real audio data. This procedure is also known
   // as audio priming.
   void EnqueuePlayoutData(bool silence);
@@ -106,7 +106,7 @@ class OpenSLESPlayer : public AudioOutput {
 
   // Obtaines the SL Engine Interface from the existing global Engine object.
   // The interface exposes creation methods of all the OpenSL ES object types.
-  // This method defines the |engine_| member variable.
+  // This method defines the `engine_` member variable.
   bool ObtainEngineInterface();
 
   // Creates/destroys the output mix object.

sdk/android/src/jni/audio_device/opensles_recorder.h

@@ -88,7 +88,7 @@ class OpenSLESRecorder : public AudioInput {
  private:
   // Obtaines the SL Engine Interface from the existing global Engine object.
   // The interface exposes creation methods of all the OpenSL ES object types.
-  // This method defines the |engine_| member variable.
+  // This method defines the `engine_` member variable.
   bool ObtainEngineInterface();
 
   // Creates/destroys the audio recorder and the simple-buffer queue object.
@@ -109,7 +109,7 @@ class OpenSLESRecorder : public AudioInput {
   // Wraps calls to SLAndroidSimpleBufferQueueState::Enqueue() and it can be
   // called both on the main thread (but before recording has started) and from
   // the internal audio thread while input streaming is active. It uses
-  // |simple_buffer_queue_| but no lock is needed since the initial calls from
+  // `simple_buffer_queue_` but no lock is needed since the initial calls from
   // the main thread and the native callback thread are mutually exclusive.
   bool EnqueueAudioBuffer();
 

sdk/android/src/jni/jni_generator_helper.cc

@@ -15,7 +15,7 @@
 
 namespace webrtc {
 
-// If |atomic_class_id| set, it'll return immediately. Otherwise, it will look
+// If `atomic_class_id` set, it'll return immediately. Otherwise, it will look
 // up the class and store it. If there's a race, we take care to only store one
 // global reference (and the duplicated effort will happen only once).
 jclass LazyGetClass(JNIEnv* env,
@@ -29,18 +29,18 @@ jclass LazyGetClass(JNIEnv* env,
   jclass cas_result = nullptr;
   if (std::atomic_compare_exchange_strong(atomic_class_id, &cas_result,
                                           clazz.obj())) {
-    // We sucessfully stored |clazz| in |atomic_class_id|, so we are
+    // We sucessfully stored `clazz` in `atomic_class_id`, so we are
     // intentionally leaking the global ref since it's now stored there.
     return clazz.Release();
   } else {
     // Some other thread came before us and stored a global pointer in
-    // |atomic_class_id|. Relase our global ref and return the ref from the
+    // `atomic_class_id`. Relase our global ref and return the ref from the
     // other thread.
     return cas_result;
   }
 }
 
-// If |atomic_method_id| set, it'll return immediately. Otherwise, it will look
+// If `atomic_method_id` set, it'll return immediately. Otherwise, it will look
 // up the method id and store it. If there's a race, it's ok since the values
 // are the same (and the duplicated effort will happen only once).
 template <MethodID::Type type>

sdk/android/src/jni/jni_generator_helper.h

@@ -44,11 +44,11 @@
 
 namespace webrtc {
 
-// This function will initialize |atomic_class_id| to contain a global ref to
+// This function will initialize `atomic_class_id` to contain a global ref to
 // the given class, and will return that ref on subsequent calls. The caller is
-// responsible to zero-initialize |atomic_class_id|. It's fine to
+// responsible to zero-initialize `atomic_class_id`. It's fine to
 // simultaneously call this on multiple threads referencing the same
-// |atomic_method_id|.
+// `atomic_method_id`.
 jclass LazyGetClass(JNIEnv* env,
                     const char* class_name,
                     std::atomic<jclass>* atomic_class_id);
@@ -61,11 +61,11 @@ class MethodID {
     TYPE_INSTANCE,
   };
 
-  // This function will initialize |atomic_method_id| to contain a ref to
+  // This function will initialize `atomic_method_id` to contain a ref to
   // the given method, and will return that ref on subsequent calls. The caller
-  // is responsible to zero-initialize |atomic_method_id|. It's fine to
+  // is responsible to zero-initialize `atomic_method_id`. It's fine to
   // simultaneously call this on multiple threads referencing the same
-  // |atomic_method_id|.
+  // `atomic_method_id`.
   template <Type type>
   static jmethodID LazyGet(JNIEnv* env,
                            jclass clazz,
@@ -151,7 +151,7 @@ struct BASE_EXPORT JniJavaCallContextChecked {
             const char* jni_signature,
             std::atomic<jmethodID>* atomic_method_id) {
     base.Init<type>(env, clazz, method_name, jni_signature, atomic_method_id);
-    // Reset |pc| to correct caller.
+    // Reset `pc` to correct caller.
     base.pc = reinterpret_cast<uintptr_t>(__builtin_return_address(0));
   }
 

sdk/android/src/jni/jvm.cc

@@ -27,7 +27,7 @@ static JavaVM* g_jvm = nullptr;
 
 static pthread_once_t g_jni_ptr_once = PTHREAD_ONCE_INIT;
 
-// Key for per-thread JNIEnv* data.  Non-NULL in threads attached to |g_jvm| by
+// Key for per-thread JNIEnv* data.  Non-NULL in threads attached to `g_jvm` by
 // AttachCurrentThreadIfNeeded(), NULL in unattached threads and threads that
 // were attached by the JVM because of a Java->native call.
 static pthread_key_t g_jni_ptr;
@@ -48,7 +48,7 @@ JNIEnv* GetEnv() {
 }
 
 static void ThreadDestructor(void* prev_jni_ptr) {
-  // This function only runs on threads where |g_jni_ptr| is non-NULL, meaning
+  // This function only runs on threads where `g_jni_ptr` is non-NULL, meaning
   // we were responsible for originally attaching the thread, so are responsible
   // for detaching it now.  However, because some JVM implementations (notably
   // Oracle's http://goo.gl/eHApYT) also use the pthread_key_create mechanism,
@@ -102,7 +102,7 @@ static std::string GetThreadName() {
   return std::string(name);
 }
 
-// Return a |JNIEnv*| usable on this thread.  Attaches to |g_jvm| if necessary.
+// Return a |JNIEnv*| usable on this thread.  Attaches to `g_jvm` if necessary.
 JNIEnv* AttachCurrentThreadIfNeeded() {
   JNIEnv* jni = GetEnv();
   if (jni)

sdk/android/src/jni/jvm.h

@@ -23,7 +23,7 @@ JNIEnv* GetEnv();
 
 JavaVM* GetJVM();
 
-// Return a |JNIEnv*| usable on this thread.  Attaches to |g_jvm| if necessary.
+// Return a |JNIEnv*| usable on this thread.  Attaches to `g_jvm` if necessary.
 JNIEnv* AttachCurrentThreadIfNeeded();
 
 }  // namespace jni

sdk/android/src/jni/pc/media_stream.cc

@@ -49,7 +49,7 @@ JavaMediaStream::JavaMediaStream(
   observer_->SignalVideoTrackAdded.connect(
       this, &JavaMediaStream::OnVideoTrackAddedToStream);
 
-  // |j_media_stream| holds one reference. Corresponding Release() is in
+  // `j_media_stream` holds one reference. Corresponding Release() is in
   // MediaStream_free, triggered by MediaStream.dispose().
   media_stream.release();
 }

sdk/android/src/jni/pc/peer_connection.cc

@@ -499,7 +499,7 @@ static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetLocalDescription(
     const JavaParamRef<jobject>& j_pc) {
   PeerConnectionInterface* pc = ExtractNativePC(jni, j_pc);
   // It's only safe to operate on SessionDescriptionInterface on the
-  // signaling thread, but |jni| may only be used on the current thread, so we
+  // signaling thread, but `jni` may only be used on the current thread, so we
   // must do this odd dance.
   std::string sdp;
   std::string type;
@@ -518,7 +518,7 @@ static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetRemoteDescription(
     const JavaParamRef<jobject>& j_pc) {
   PeerConnectionInterface* pc = ExtractNativePC(jni, j_pc);
   // It's only safe to operate on SessionDescriptionInterface on the
-  // signaling thread, but |jni| may only be used on the current thread, so we
+  // signaling thread, but `jni` may only be used on the current thread, so we
   // must do this odd dance.
   std::string sdp;
   std::string type;

sdk/android/src/jni/pc/peer_connection_factory.cc

@@ -242,9 +242,9 @@ static void JNI_PeerConnectionFactory_ShutdownInternalTracer(JNIEnv* jni) {
 }
 
 // Following parameters are optional:
-// |audio_device_module|, |jencoder_factory|, |jdecoder_factory|,
-// |audio_processor|, |fec_controller_factory|,
-// |network_state_predictor_factory|, |neteq_factory|.
+// `audio_device_module`, `jencoder_factory`, `jdecoder_factory`,
+// `audio_processor`, `fec_controller_factory`,
+// `network_state_predictor_factory`, `neteq_factory`.
 ScopedJavaLocalRef<jobject> CreatePeerConnectionFactoryForJava(
     JNIEnv* jni,
     const JavaParamRef<jobject>& jcontext,

sdk/android/src/jni/pc/peer_connection_factory.h

@@ -18,7 +18,7 @@
 namespace webrtc {
 namespace jni {
 
-// Creates java PeerConnectionFactory with specified |pcf|.
+// Creates java PeerConnectionFactory with specified `pcf`.
 jobject NativeToJavaPeerConnectionFactory(
     JNIEnv* jni,
     rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf,

sdk/android/src/jni/pc/rtp_receiver.h

@@ -23,7 +23,7 @@ ScopedJavaLocalRef<jobject> NativeToJavaRtpReceiver(
     JNIEnv* env,
     rtc::scoped_refptr<RtpReceiverInterface> receiver);
 
-// Takes ownership of the passed |j_receiver| and stores it as a global
+// Takes ownership of the passed `j_receiver` and stores it as a global
 // reference. Will call dispose() in the dtor.
 class JavaRtpReceiverGlobalOwner {
  public:

sdk/android/src/jni/pc/rtp_transceiver.h

@@ -27,7 +27,7 @@ ScopedJavaLocalRef<jobject> NativeToJavaRtpTransceiver(
     JNIEnv* env,
     rtc::scoped_refptr<RtpTransceiverInterface> transceiver);
 
-// This takes ownership of the of the |j_transceiver| and stores it as a global
+// This takes ownership of the of the `j_transceiver` and stores it as a global
 // reference. This calls the Java Transceiver's dispose() method with the dtor.
 class JavaRtpTransceiverGlobalOwner {
  public:

sdk/android/src/jni/video_encoder_wrapper.cc

@@ -267,7 +267,7 @@ void VideoEncoderWrapper::OnEncodedFrame(
     frame_extra_infos_.pop_front();
   }
 
-  // This is a bit subtle. The |frame| variable from the lambda capture is
+  // This is a bit subtle. The `frame` variable from the lambda capture is
   // const. Which implies that (i) we need to make a copy to be able to
   // write to the metadata, and (ii) we should avoid using the .data()
   // method (including implicit conversion to ArrayView) on the non-const

sdk/android/src/jni/video_frame.cc

@@ -41,8 +41,8 @@ class AndroidVideoBuffer : public VideoFrameBuffer {
 
   const ScopedJavaGlobalRef<jobject>& video_frame_buffer() const;
 
-  // Crops a region defined by |crop_x|, |crop_y|, |crop_width| and
-  // |crop_height|. Scales it to size |scale_width| x |scale_height|.
+  // Crops a region defined by `crop_x`, `crop_y`, `crop_width` and
+  // `crop_height`. Scales it to size `scale_width` x `scale_height`.
   rtc::scoped_refptr<VideoFrameBuffer> CropAndScale(int crop_x,
                                                     int crop_y,
                                                     int crop_width,