Move all api/ files to use webrtc::RefCountInterface

instead of rtc::RefCountInterface

Bug: webrtc:15622
Change-Id: I085660a097a019c7aa58a7e3f0aceeedd9fcc8c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325460
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41050}

api/BUILD.gn

@@ -141,13 +141,14 @@ rtc_library("media_stream_interface") {
   deps = [
     ":audio_options_api",
     ":make_ref_counted",
+    ":ref_count",
+    ":ref_count",
     ":rtp_parameters",
     ":scoped_refptr",
     ":sequence_checker",
     ":video_track_source_constraints",
     "../modules/audio_processing:audio_processing_statistics",
     "../rtc_base:checks",
-    "../rtc_base:refcount",
     "../rtc_base/system:no_unique_address",
     "../rtc_base/system:rtc_export",
     "video:recordable_encoded_frame",
@@ -195,9 +196,9 @@ rtc_source_set("ice_transport_interface") {
   deps = [
     ":async_dns_resolver",
     ":packet_socket_factory",
+    ":ref_count",
     ":rtc_error",
     ":scoped_refptr",
-    "../rtc_base:refcount",
     "rtc_event_log:rtc_event_log",
   ]
 }
@@ -211,9 +212,9 @@ rtc_library("dtls_transport_interface") {
   ]
   deps = [
     ":ice_transport_interface",
+    ":ref_count",
     ":rtc_error",
     ":scoped_refptr",
-    "../rtc_base:refcount",
     "../rtc_base:ssl",
     "../rtc_base/system:rtc_export",
   ]
@@ -226,7 +227,7 @@ rtc_library("dtmf_sender_interface") {
   sources = [ "dtmf_sender_interface.h" ]
   deps = [
     ":media_stream_interface",
-    "../rtc_base:refcount",
+    ":ref_count",
   ]
 }
 
@@ -242,11 +243,11 @@ rtc_library("rtp_sender_interface") {
     ":dtmf_sender_interface",
     ":frame_transformer_interface",
     ":media_stream_interface",
+    ":ref_count",
     ":rtc_error",
     ":rtp_parameters",
     ":scoped_refptr",
     "../rtc_base:checks",
-    "../rtc_base:refcount",
     "../rtc_base/system:rtc_export",
     "crypto:frame_encryptor_interface",
     "video_codecs:video_codecs_api",
@@ -308,6 +309,7 @@ rtc_library("libjingle_peerconnection_api") {
     ":network_state_predictor_api",
     ":packet_socket_factory",
     ":priority",
+    ":ref_count",
     ":rtc_error",
     ":rtc_stats_api",
     ":rtp_packet_info",
@@ -324,7 +326,6 @@ rtc_library("libjingle_peerconnection_api") {
     "../rtc_base:logging",
     "../rtc_base:network",
     "../rtc_base:network_constants",
-    "../rtc_base:refcount",
     "../rtc_base:rtc_certificate_generator",
     "../rtc_base:ssl",
     "../rtc_base:stringutils",
@@ -378,8 +379,8 @@ rtc_source_set("frame_transformer_interface") {
   sources = [ "frame_transformer_interface.h" ]
   deps = [
     ":make_ref_counted",
+    ":ref_count",
     ":scoped_refptr",
-    "../rtc_base:refcount",
     "video:encoded_frame",
     "video:video_frame_metadata",
   ]
@@ -466,7 +467,10 @@ rtc_source_set("scoped_refptr") {
 rtc_source_set("make_ref_counted") {
   visibility = [ "*" ]
   sources = [ "make_ref_counted.h" ]
-  deps = [ "../rtc_base:refcount" ]
+  deps = [
+    ":ref_count",
+    "../rtc_base:refcount",
+  ]
 }
 
 rtc_source_set("video_quality_analyzer_api") {
@@ -787,6 +791,7 @@ rtc_source_set("rtc_stats_api") {
 
   deps = [
     ":make_ref_counted",
+    ":ref_count",
     ":scoped_refptr",
     "../api:refcountedbase",
     "../rtc_base:checks",
@@ -919,6 +924,7 @@ rtc_source_set("refcountedbase") {
   visibility = [ "*" ]
   sources = [ "ref_counted_base.h" ]
   deps = [
+    ":ref_count",
     "../rtc_base:macromagic",
     "../rtc_base:refcount",
   ]
@@ -1217,6 +1223,7 @@ if (rtc_include_tests) {
       ":array_view",
       ":libjingle_peerconnection_api",
       ":make_ref_counted",
+      ":ref_count",
       ":rtp_parameters",
       "../rtc_base:checks",
       "../rtc_base:refcount",
@@ -1270,6 +1277,7 @@ if (rtc_include_tests) {
 
     deps = [
       ":libjingle_peerconnection_api",
+      ":ref_count",
       "../api:scoped_refptr",
       "../rtc_base:refcount",
       "../test:test_support",
@@ -1410,6 +1418,7 @@ if (rtc_include_tests) {
     sources = [ "test/mock_video_track.h" ]
 
     deps = [
+      ":ref_count",
       "../api:media_stream_interface",
       "../api:scoped_refptr",
       "../rtc_base:refcount",
@@ -1590,10 +1599,10 @@ rtc_library("frame_transformer_factory") {
   ]
   deps = [
     ":frame_transformer_interface",
+    ":ref_count",
     ":scoped_refptr",
     "../audio:audio",
     "../modules/rtp_rtcp",
-    "../rtc_base:refcount",
     "video:encoded_frame",
     "video:video_frame_metadata",
   ]

api/DEPS

@@ -81,7 +81,6 @@ specific_include_rules = {
 
   "data_channel_interface\.h": [
     "+rtc_base/copy_on_write_buffer.h",
-    "+rtc_base/ref_count.h",
   ],
 
   "data_channel_transport_interface\.h": [
@@ -89,33 +88,15 @@ specific_include_rules = {
   ],
 
   "dtls_transport_interface\.h": [
-    "+rtc_base/ref_count.h",
     "+rtc_base/ssl_certificate.h",
   ],
 
-  "dtmf_sender_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
   "fec_controller\.h": [
     "+modules/include/module_fec_types.h",
   ],
 
-  "frame_transformer_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "ice_transport_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "jsep\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
   "media_stream_interface\.h": [
     "+modules/audio_processing/include/audio_processing_statistics.h",
-    "+rtc_base/ref_count.h",
   ],
 
   "packet_socket_factory\.h": [
@@ -132,7 +113,6 @@ specific_include_rules = {
     "+rtc_base/network.h",
     "+rtc_base/network_constants.h",
     "+rtc_base/network_monitor_factory.h",
-    "+rtc_base/ref_count.h",
     "+rtc_base/rtc_certificate.h",
     "+rtc_base/rtc_certificate_generator.h",
     "+rtc_base/socket_address.h",
@@ -148,7 +128,6 @@ specific_include_rules = {
   ],
 
   "ref_counted_base\.h": [
-    "+rtc_base/ref_count.h",
     "+rtc_base/ref_counter.h",
   ],
 
@@ -159,74 +138,19 @@ specific_include_rules = {
     # For private member and constructor.
     "+rtc_base/system/file_wrapper.h",
   ],
-  "rtp_receiver_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "rtp_sender_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "rtp_transceiver_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "sctp_transport_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "set_local_description_observer_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-  "set_remote_description_observer_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
 
   "legacy_stats_types\.h": [
-    "+rtc_base/ref_count.h",
     "+rtc_base/thread_checker.h",
   ],
 
-  "uma_metrics\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "audio_mixer\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
   "audio_decoder\.h": [
     "+rtc_base/buffer.h",
   ],
 
-  "audio_decoder_factory\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
   "audio_encoder\.h": [
     "+rtc_base/buffer.h",
   ],
 
-  "audio_encoder_factory\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "frame_decryptor_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "frame_encryptor_interface\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "rtc_stats_collector_callback\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
-  "rtc_stats_report\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
   "audioproc_float\.h": [
     "+modules/audio_processing/include/audio_processing.h",
   ],
@@ -279,10 +203,6 @@ specific_include_rules = {
     "+modules/video_coding/include/video_codec_interface.h"
   ],
 
-  "video_encoder_config\.h": [
-    "+rtc_base/ref_count.h",
-  ],
-
   "sequence_checker\.h": [
     "+rtc_base/synchronization/sequence_checker_internal.h",
     "+rtc_base/thread_annotations.h",

api/data_channel_interface.h

@@ -22,10 +22,10 @@
 #include "absl/functional/any_invocable.h"
 #include "absl/types/optional.h"
 #include "api/priority.h"
+#include "api/ref_count.h"
 #include "api/rtc_error.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {

api/dtls_transport_interface.h

@@ -16,9 +16,9 @@
 
 #include "absl/types/optional.h"
 #include "api/ice_transport_interface.h"
+#include "api/ref_count.h"
 #include "api/rtc_error.h"
 #include "api/scoped_refptr.h"
-#include "rtc_base/ref_count.h"
 #include "rtc_base/ssl_certificate.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -107,7 +107,7 @@ class DtlsTransportObserverInterface {
 // accessed on that thread, except for functions explicitly marked otherwise.
 // References can be held by other threads, and destruction can therefore
 // be initiated by other threads.
-class DtlsTransportInterface : public rtc::RefCountInterface {
+class DtlsTransportInterface : public webrtc::RefCountInterface {
  public:
   // Returns a pointer to the ICE transport that is owned by the DTLS transport.
   virtual rtc::scoped_refptr<IceTransportInterface> ice_transport() = 0;

api/dtmf_sender_interface.h

@@ -14,7 +14,7 @@
 #include <string>
 
 #include "api/media_stream_interface.h"
-#include "rtc_base/ref_count.h"
+#include "api/ref_count.h"
 
 namespace webrtc {
 
@@ -42,7 +42,7 @@ class DtmfSenderObserverInterface {
 // The interface of native implementation of the RTCDTMFSender defined by the
 // WebRTC W3C Editor's Draft.
 // See: https://www.w3.org/TR/webrtc/#peer-to-peer-dtmf
-class DtmfSenderInterface : public rtc::RefCountInterface {
+class DtmfSenderInterface : public webrtc::RefCountInterface {
  public:
   // Provides the spec compliant default 2 second delay for the ',' character.
   static const int kDtmfDefaultCommaDelayMs = 2000;

api/frame_transformer_interface.h

@@ -15,10 +15,10 @@
 #include <string>
 #include <vector>
 
+#include "api/ref_count.h"
 #include "api/scoped_refptr.h"
 #include "api/video/encoded_frame.h"
 #include "api/video/video_frame_metadata.h"
-#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 

api/ice_transport_interface.h

@@ -15,10 +15,10 @@
 
 #include "api/async_dns_resolver.h"
 #include "api/async_resolver_factory.h"
+#include "api/ref_count.h"
 #include "api/rtc_error.h"
 #include "api/rtc_event_log/rtc_event_log.h"
 #include "api/scoped_refptr.h"
-#include "rtc_base/ref_count.h"
 
 namespace cricket {
 class IceTransportInternal;
@@ -33,7 +33,7 @@ class FieldTrialsView;
 // An ICE transport, as represented to the outside world.
 // This object is refcounted, and is therefore alive until the
 // last holder has released it.
-class IceTransportInterface : public rtc::RefCountInterface {
+class IceTransportInterface : public webrtc::RefCountInterface {
  public:
   // Accessor for the internal representation of an ICE transport.
   // The returned object can only be safely used on the signalling thread.

api/jsep.h

@@ -27,8 +27,8 @@
 #include <vector>
 
 #include "absl/types/optional.h"
+#include "api/ref_count.h"
 #include "api/rtc_error.h"
-#include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace cricket {
@@ -219,7 +219,7 @@ std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
 
 // CreateOffer and CreateAnswer callback interface.
 class RTC_EXPORT CreateSessionDescriptionObserver
-    : public rtc::RefCountInterface {
+    : public webrtc::RefCountInterface {
  public:
   // This callback transfers the ownership of the `desc`.
   // TODO(deadbeef): Make this take an std::unique_ptr<> to avoid confusion
@@ -238,7 +238,8 @@ class RTC_EXPORT CreateSessionDescriptionObserver
 };
 
 // SetLocalDescription and SetRemoteDescription callback interface.
-class RTC_EXPORT SetSessionDescriptionObserver : public rtc::RefCountInterface {
+class RTC_EXPORT SetSessionDescriptionObserver
+    : public webrtc::RefCountInterface {
  public:
   virtual void OnSuccess() = 0;
   // See description in CreateSessionDescriptionObserver for OnFailure.

api/legacy_stats_types.h

@@ -20,9 +20,9 @@
 #include <string>
 #include <vector>
 
+#include "api/ref_count.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
-#include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -240,7 +240,7 @@ class RTC_EXPORT StatsReport {
     kStatsValueNameLocalCandidateRelayProtocol,
   };
 
-  class RTC_EXPORT IdBase : public rtc::RefCountInterface {
+  class RTC_EXPORT IdBase : public webrtc::RefCountInterface {
    public:
     ~IdBase() override;
     StatsType type() const;

api/media_stream_interface.h

@@ -23,6 +23,7 @@
 
 #include "absl/types/optional.h"
 #include "api/audio_options.h"
+#include "api/ref_count.h"
 #include "api/scoped_refptr.h"
 #include "api/video/recordable_encoded_frame.h"
 #include "api/video/video_frame.h"
@@ -30,7 +31,6 @@
 #include "api/video/video_source_interface.h"
 #include "api/video_track_source_constraints.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
-#include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -54,7 +54,7 @@ class NotifierInterface {
 
 // Base class for sources. A MediaStreamTrack has an underlying source that
 // provides media. A source can be shared by multiple tracks.
-class RTC_EXPORT MediaSourceInterface : public rtc::RefCountInterface,
+class RTC_EXPORT MediaSourceInterface : public webrtc::RefCountInterface,
                                         public NotifierInterface {
  public:
   enum SourceState { kInitializing, kLive, kEnded, kMuted };
@@ -69,7 +69,7 @@ class RTC_EXPORT MediaSourceInterface : public rtc::RefCountInterface,
 
 // C++ version of MediaStreamTrack.
 // See: https://www.w3.org/TR/mediacapture-streams/#mediastreamtrack
-class RTC_EXPORT MediaStreamTrackInterface : public rtc::RefCountInterface,
+class RTC_EXPORT MediaStreamTrackInterface : public webrtc::RefCountInterface,
                                              public NotifierInterface {
  public:
   enum TrackState {
@@ -267,7 +267,7 @@ class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {
 
 // Interface of the audio processor used by the audio track to collect
 // statistics.
-class AudioProcessorInterface : public rtc::RefCountInterface {
+class AudioProcessorInterface : public webrtc::RefCountInterface {
  public:
   struct AudioProcessorStatistics {
     bool typing_noise_detected = false;
@@ -321,7 +321,7 @@ typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> > VideoTrackVector;
 // must be pushed down.
 //
 // Thus, this interface acts as simply a container for tracks.
-class MediaStreamInterface : public rtc::RefCountInterface,
+class MediaStreamInterface : public webrtc::RefCountInterface,
                              public NotifierInterface {
  public:
   virtual std::string id() const = 0;

api/peer_connection_interface.h

@@ -128,11 +128,11 @@
 // TODO(bugs.webrtc.org/7447): We plan to provide a way to let applications
 // inject a PacketSocketFactory and/or NetworkManager, and not expose
 // PortAllocator in the PeerConnection api.
+#include "api/ref_count.h"
 #include "p2p/base/port_allocator.h"
 #include "rtc_base/network.h"
 #include "rtc_base/network_constants.h"
 #include "rtc_base/network_monitor_factory.h"
-#include "rtc_base/ref_count.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/rtc_certificate_generator.h"
 #include "rtc_base/socket_address.h"
@@ -151,7 +151,7 @@ namespace webrtc {
 class MediaFactory;
 
 // MediaStream container interface.
-class StreamCollectionInterface : public rtc::RefCountInterface {
+class StreamCollectionInterface : public webrtc::RefCountInterface {
  public:
   // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
   virtual size_t count() = 0;
@@ -165,7 +165,7 @@ class StreamCollectionInterface : public rtc::RefCountInterface {
   ~StreamCollectionInterface() override = default;
 };
 
-class StatsObserver : public rtc::RefCountInterface {
+class StatsObserver : public webrtc::RefCountInterface {
  public:
   virtual void OnComplete(const StatsReports& reports) = 0;
 
@@ -180,7 +180,7 @@ enum class SdpSemantics {
   kUnifiedPlan,
 };
 
-class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
+class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
  public:
   // See https://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
   enum SignalingState {
@@ -1487,7 +1487,7 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
 // CreatePeerConnectionFactory method which accepts threads as input, and use
 // the CreatePeerConnection version that takes a PortAllocator as an argument.
 class RTC_EXPORT PeerConnectionFactoryInterface
-    : public rtc::RefCountInterface {
+    : public webrtc::RefCountInterface {
  public:
   class Options {
    public:

api/rtp_receiver_interface.h

@@ -22,10 +22,10 @@
 #include "api/frame_transformer_interface.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
+#include "api/ref_count.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
 #include "api/transport/rtp/rtp_source.h"
-#include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -44,7 +44,7 @@ class RtpReceiverObserverInterface {
   virtual ~RtpReceiverObserverInterface() {}
 };
 
-class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
+class RTC_EXPORT RtpReceiverInterface : public webrtc::RefCountInterface {
  public:
   virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
 

api/rtp_sender_interface.h

@@ -25,18 +25,18 @@
 #include "api/frame_transformer_interface.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
+#include "api/ref_count.h"
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
 #include "api/video_codecs/video_encoder_factory.h"
-#include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
 
 using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
 
-class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
+class RTC_EXPORT RtpSenderInterface : public webrtc::RefCountInterface {
  public:
   // Returns true if successful in setting the track.
   // Fails if an audio track is set on a video RtpSender, or vice-versa.

api/rtp_transceiver_interface.h

@@ -18,12 +18,12 @@
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/media_types.h"
+#include "api/ref_count.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_receiver_interface.h"
 #include "api/rtp_sender_interface.h"
 #include "api/rtp_transceiver_direction.h"
 #include "api/scoped_refptr.h"
-#include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -58,7 +58,7 @@ struct RTC_EXPORT RtpTransceiverInit final {
 //
 // WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
-class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
+class RTC_EXPORT RtpTransceiverInterface : public webrtc::RefCountInterface {
  public:
   // Media type of the transceiver. Any sender(s)/receiver(s) will have this
   // type as well.

api/sctp_transport_interface.h

@@ -13,9 +13,9 @@
 
 #include "absl/types/optional.h"
 #include "api/dtls_transport_interface.h"
+#include "api/ref_count.h"
 #include "api/rtc_error.h"
 #include "api/scoped_refptr.h"
-#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -75,7 +75,7 @@ class SctpTransportObserverInterface {
 // accessed on that thread, except for functions explicitly marked otherwise.
 // References can be held by other threads, and destruction can therefore
 // be initiated by other threads.
-class SctpTransportInterface : public rtc::RefCountInterface {
+class SctpTransportInterface : public webrtc::RefCountInterface {
  public:
   // This function can be called from other threads.
   virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0;

api/set_local_description_observer_interface.h

@@ -11,15 +11,15 @@
 #ifndef API_SET_LOCAL_DESCRIPTION_OBSERVER_INTERFACE_H_
 #define API_SET_LOCAL_DESCRIPTION_OBSERVER_INTERFACE_H_
 
+#include "api/ref_count.h"
 #include "api/rtc_error.h"
-#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
 // OnSetLocalDescriptionComplete() invokes as soon as
 // PeerConnectionInterface::SetLocalDescription() operation completes, allowing
 // the observer to examine the effects of the operation without delay.
-class SetLocalDescriptionObserverInterface : public rtc::RefCountInterface {
+class SetLocalDescriptionObserverInterface : public webrtc::RefCountInterface {
  public:
   // On success, `error.ok()` is true.
   virtual void OnSetLocalDescriptionComplete(RTCError error) = 0;

api/set_remote_description_observer_interface.h

@@ -11,8 +11,8 @@
 #ifndef API_SET_REMOTE_DESCRIPTION_OBSERVER_INTERFACE_H_
 #define API_SET_REMOTE_DESCRIPTION_OBSERVER_INTERFACE_H_
 
+#include "api/ref_count.h"
 #include "api/rtc_error.h"
-#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -20,7 +20,7 @@ namespace webrtc {
 // callback is invoked such that the state of the peer connection can be
 // examined to accurately reflect the effects of the SetRemoteDescription
 // operation.
-class SetRemoteDescriptionObserverInterface : public rtc::RefCountInterface {
+class SetRemoteDescriptionObserverInterface : public webrtc::RefCountInterface {
  public:
   // On success, `error.ok()` is true.
   virtual void OnSetRemoteDescriptionComplete(RTCError error) = 0;