No public description

PiperOrigin-RevId: 753921978

Author: Google Meet Team

cpp/api/BUILD

@@ -29,6 +29,7 @@ cc_library(
         "@com_google_absl//absl/status:statusor",
         "@com_google_absl//absl/strings:string_view",
         "@com_google_absl//absl/types:span",
+        "@media_api_samples//cpp/internal:webrtc_forward_decls",
         "@webrtc",
     ],
 )
@@ -39,6 +40,7 @@ cc_library(
     deps = [
         ":media_api_client_interface",
         "@com_google_absl//absl/status:statusor",
+        "@media_api_samples//cpp/internal:webrtc_forward_decls",
         "@webrtc",
     ],
 )

cpp/api/media_api_client_factory_interface.h

@@ -21,6 +21,8 @@
 
 #include "absl/status/statusor.h"
 #include "cpp/api/media_api_client_interface.h"
+// TODO: Remove once build has updated to a recent WebRTC version.
+#include "cpp/internal/webrtc_forward_decls.h"
 #include "webrtc/api/scoped_refptr.h"
 
 namespace meet {
@@ -33,7 +35,7 @@ class MediaApiClientFactoryInterface {
   /// Creates a `MediaApiClientInterface` instance.
   virtual absl::StatusOr<std::unique_ptr<MediaApiClientInterface>>
   CreateMediaApiClient(const MediaApiClientConfiguration& api_config,
-                       rtc::scoped_refptr<MediaApiClientObserverInterface>
+                       webrtc::scoped_refptr<MediaApiClientObserverInterface>
                            api_session_observer) = 0;
 };
 

cpp/api/media_api_client_interface.h

@@ -72,6 +72,8 @@
 #include "cpp/api/participants_resource.h"
 #include "cpp/api/session_control_resource.h"
 #include "cpp/api/video_assignment_resource.h"
+// TODO: Remove once build has updated to a recent WebRTC version.
+#include "cpp/internal/webrtc_forward_decls.h"
 #include "webrtc/api/ref_count.h"
 #include "webrtc/api/scoped_refptr.h"
 #include "webrtc/api/video/video_frame.h"
@@ -295,7 +297,8 @@ class MediaApiClientInterface {
   /// until the client is destroyed.
   static absl::StatusOr<std::unique_ptr<MediaApiClientInterface>> Create(
       const MediaApiClientConfiguration& api_config,
-      rtc::scoped_refptr<MediaApiClientObserverInterface> api_session_observer);
+      webrtc::scoped_refptr<MediaApiClientObserverInterface>
+          api_session_observer);
 };
 
 }  // namespace meet

cpp/internal/BUILD

@@ -24,6 +24,7 @@ cc_library(
         ":conference_peer_connection_interface",
         ":stats_request_from_report",
         ":variant_utils",
+        ":webrtc_forward_decls",
         "@com_google_absl//absl/base:core_headers",
         "@com_google_absl//absl/container:flat_hash_map",
         "@com_google_absl//absl/container:flat_hash_set",
@@ -58,6 +59,7 @@ cc_library(
         ":participants_resource_handler",
         ":session_control_resource_handler",
         ":video_assignment_resource_handler",
+        ":webrtc_forward_decls",
         "@com_google_absl//absl/functional:any_invocable",
         "@com_google_absl//absl/status",
         "@com_google_absl//absl/status:statusor",
@@ -79,6 +81,7 @@ cc_library(
     srcs = ["media_api_audio_device_module.cc"],
     hdrs = ["media_api_audio_device_module.h"],
     deps = [
+        ":webrtc_forward_decls",
         "@com_google_absl//absl/log:check",
         "@webrtc",
     ],
@@ -88,6 +91,7 @@ cc_library(
     name = "conference_peer_connection_interface",
     hdrs = ["conference_peer_connection_interface.h"],
     deps = [
+        ":webrtc_forward_decls",
         "@com_google_absl//absl/functional:any_invocable",
         "@com_google_absl//absl/status",
         "@com_google_absl//absl/strings:string_view",
@@ -103,6 +107,7 @@ cc_library(
         ":conference_media_tracks",
         ":conference_peer_connection_interface",
         ":http_connector_interface",
+        ":webrtc_forward_decls",
         "@com_google_absl//absl/log",
         "@com_google_absl//absl/status",
         "@com_google_absl//absl/status:statusor",
@@ -204,6 +209,7 @@ cc_library(
     srcs = ["stats_request_from_report.cc"],
     hdrs = ["stats_request_from_report.h"],
     deps = [
+        ":webrtc_forward_decls",
         "@com_google_absl//absl/container:flat_hash_map",
         "@com_google_absl//absl/container:flat_hash_set",
         "@com_google_absl//absl/log",
@@ -229,6 +235,7 @@ cc_library(
     deps = [
         ":conference_data_channel_interface",
         ":resource_handler_interface",
+        ":webrtc_forward_decls",
         "@com_google_absl//absl/log",
         "@com_google_absl//absl/status",
         "@com_google_absl//absl/status:statusor",
@@ -421,6 +428,7 @@ cc_library(
     srcs = ["conference_media_tracks.cc"],
     hdrs = ["conference_media_tracks.h"],
     deps = [
+        ":webrtc_forward_decls",
         "@com_google_absl//absl/functional:any_invocable",
         "@com_google_absl//absl/log",
         "@com_google_absl//absl/types:optional",
@@ -445,3 +453,12 @@ cc_library(
         "@media_api_samples//cpp/api:media_api_client_interface",
     ],
 )
+
+# TODO: Remove once build has updated to a recent WebRTC version.
+cc_library(
+    name = "webrtc_forward_decls",
+    hdrs = ["webrtc_forward_decls.h"],
+    visibility = ["@media_api_samples//cpp:__subpackages__"],
+    deps = [
+    ],
+)

cpp/internal/conference_data_channel.h

@@ -23,9 +23,11 @@
 
 #include "absl/log/log.h"
 #include "absl/status/status.h"
+// TODO: Remove once build has updated to a recent WebRTC version.
 #include "cpp/api/media_api_client_interface.h"
 #include "cpp/internal/conference_data_channel_interface.h"
 #include "cpp/internal/resource_handler_interface.h"
+#include "cpp/internal/webrtc_forward_decls.h"
 #include "webrtc/api/data_channel_interface.h"
 #include "webrtc/api/scoped_refptr.h"
 
@@ -40,7 +42,7 @@ class ConferenceDataChannel : public ConferenceDataChannelInterface,
  public:
   ConferenceDataChannel(
       std::unique_ptr<ResourceHandlerInterface> resource_handler,
-      rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel)
+      webrtc::scoped_refptr<webrtc::DataChannelInterface> data_channel)
       : resource_handler_(std::move(resource_handler)),
         data_channel_(std::move(data_channel)) {
     data_channel_->RegisterObserver(this);
@@ -88,7 +90,7 @@ class ConferenceDataChannel : public ConferenceDataChannelInterface,
 
   ResourceUpdateCallback callback_;
   std::unique_ptr<ResourceHandlerInterface> resource_handler_;
-  rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel_;
+  webrtc::scoped_refptr<webrtc::DataChannelInterface> data_channel_;
 };
 
 }  // namespace meet

cpp/internal/conference_media_tracks.h

@@ -29,6 +29,8 @@
 #include "cpp/api/media_api_client_interface.h"
 #include "webrtc/api/media_stream_interface.h"
 #include "webrtc/api/rtp_receiver_interface.h"
+// TODO: Remove once build has updated to a recent WebRTC version.
+#include "cpp/internal/webrtc_forward_decls.h"
 #include "webrtc/api/scoped_refptr.h"
 #include "webrtc/api/video/video_frame.h"
 #include "webrtc/api/video/video_sink_interface.h"
@@ -45,7 +47,7 @@ class ConferenceAudioTrack : public webrtc::AudioTrackSinkInterface {
 
   ConferenceAudioTrack(
       std::string mid,
-      rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
+      webrtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
       AudioFrameCallback callback)
       : mid_(std::move(mid)),
         receiver_(std::move(receiver)),
@@ -58,14 +60,14 @@ class ConferenceAudioTrack : public webrtc::AudioTrackSinkInterface {
  private:
   // Media line from the SDP offer/answer that identifies this track.
   std::string mid_;
-  rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver_;
+  webrtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver_;
   AudioFrameCallback callback_;
 };
 
-// Adapter class for rtc::VideoSinkInterface that converts
+// Adapter class for webrtc::VideoSinkInterface that converts
 // webrtc::VideoFrames to meet::VideoFrames and calls the callback.
 class ConferenceVideoTrack
-    : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+    : public webrtc::VideoSinkInterface<webrtc::VideoFrame> {
  public:
   using VideoFrameCallback = absl::AnyInvocable<void(VideoFrame frame)>;
 

cpp/internal/conference_media_tracks_test.cc

@@ -29,6 +29,8 @@
 #include "absl/log/globals.h"
 #include "cpp/api/media_api_client_interface.h"
 #include "webrtc/api/rtp_packet_info.h"
+// TODO: Remove once build has updated to a recent WebRTC version.
+#include "cpp/internal/webrtc_forward_decls.h"
 #include "webrtc/api/rtp_packet_infos.h"
 #include "webrtc/api/scoped_refptr.h"
 #include "webrtc/api/test/mock_rtpreceiver.h"
@@ -51,7 +53,7 @@ using ::testing::SizeIs;
 using ::testing::UnorderedElementsAre;
 
 TEST(ConferenceAudioTrackTest, CallsObserverWithAudioFrameFromLoudestSpeaker) {
-  rtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
+  webrtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
       new webrtc::MockRtpReceiver());
   webrtc::RtpSource csrc_rtp_source(
       webrtc::Timestamp::Micros(1234567890),
@@ -103,7 +105,7 @@ TEST(ConferenceAudioTrackTest, CallsObserverWithAudioFrameFromLoudestSpeaker) {
 
 TEST(ConferenceAudioTrackTest,
      CallsObserverWithAudioFrameFromNonLoudestSpeaker) {
-  rtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
+  webrtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
       new webrtc::MockRtpReceiver());
   webrtc::RtpSource csrc_rtp_source(
       webrtc::Timestamp::Micros(1234567890),
@@ -168,7 +170,7 @@ TEST(ConferenceAudioTrackTest, LogsErrorWithUnsupportedBitsPerSample) {
 }
 
 TEST(ConferenceAudioTrackTest, LogsErrorWithMissingCsrc) {
-  rtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
+  webrtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
       new webrtc::MockRtpReceiver());
   webrtc::RtpSource ssrc_rtp_source(
       webrtc::Timestamp::Micros(1234567890),
@@ -201,7 +203,7 @@ TEST(ConferenceAudioTrackTest, LogsErrorWithMissingCsrc) {
 }
 
 TEST(ConferenceAudioTrackTest, LogsErrorWithMissingSsrc) {
-  rtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
+  webrtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
       new webrtc::MockRtpReceiver());
   webrtc::RtpSource csrc_rtp_source(
       webrtc::Timestamp::Micros(1234567890),
@@ -233,7 +235,7 @@ TEST(ConferenceAudioTrackTest, LogsErrorWithMissingSsrc) {
   EXPECT_EQ(message, "AudioFrame is missing SSRC for mid: mid");
 }
 TEST(ConferenceAudioTrackTest, LogsErrorWithMissingCsrcAndSsrc) {
-  rtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
+  webrtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
       new webrtc::MockRtpReceiver());
   EXPECT_CALL(*mock_receiver, GetSources)
       .WillOnce(Return(std::vector<webrtc::RtpSource>()));
@@ -264,7 +266,7 @@ TEST(ConferenceAudioTrackTest, LogsErrorWithMissingCsrcAndSsrc) {
 }
 
 TEST(ConferenceAudioTrackTest, LogsErrorWithOnlyLoudestSpeakerCsrc) {
-  rtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
+  webrtc::scoped_refptr<webrtc::MockRtpReceiver> mock_receiver(
       new webrtc::MockRtpReceiver());
   webrtc::RtpSource csrc_rtp_source(
       webrtc::Timestamp::Micros(1234567890),

cpp/internal/conference_peer_connection.cc

@@ -29,6 +29,8 @@
 #include "nlohmann/json.hpp"
 #include "webrtc/api/jsep.h"
 #include "webrtc/api/make_ref_counted.h"
+// TODO: Remove once build has updated to a recent WebRTC version.
+#include "cpp/internal/webrtc_forward_decls.h"
 #include "webrtc/api/peer_connection_interface.h"
 #include "webrtc/api/rtc_error.h"
 #include "webrtc/api/rtp_transceiver_interface.h"
@@ -121,7 +123,7 @@ void ConferencePeerConnection::OnConnectionChange(
 }
 
 void ConferencePeerConnection::OnTrack(
-    rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
+    webrtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
   if (track_signaled_callback_ == nullptr) {
     LOG(WARNING)
         << "ConferencePeerConnection::OnTrack called without callback.";

cpp/internal/conference_peer_connection.h

@@ -30,6 +30,8 @@
 #include "cpp/internal/conference_peer_connection_interface.h"
 #include "cpp/internal/http_connector_interface.h"
 #include "webrtc/api/data_channel_interface.h"
+// TODO: Remove once build has updated to a recent WebRTC version.
+#include "cpp/internal/webrtc_forward_decls.h"
 #include "webrtc/api/jsep.h"
 #include "webrtc/api/media_stream_interface.h"
 #include "webrtc/api/peer_connection_interface.h"
@@ -50,7 +52,7 @@ class ConferencePeerConnection : public ConferencePeerConnectionInterface,
                                  public webrtc::PeerConnectionObserver {
  public:
   ConferencePeerConnection(
-      std::unique_ptr<rtc::Thread> signaling_thread,
+      std::unique_ptr<webrtc::Thread> signaling_thread,
       std::unique_ptr<HttpConnectorInterface> http_connector)
       : signaling_thread_(std::move(signaling_thread)),
         http_connector_(std::move(http_connector)) {};
@@ -67,17 +69,19 @@ class ConferencePeerConnection : public ConferencePeerConnectionInterface,
   };
 
   void OnAddStream(
-      rtc::scoped_refptr<webrtc::MediaStreamInterface> /* stream */) override {
+      webrtc::scoped_refptr<webrtc::MediaStreamInterface> /* stream */)
+      override {
     VLOG(1) << "OnAddStream called.";
   }
 
   void OnRemoveStream(
-      rtc::scoped_refptr<webrtc::MediaStreamInterface> /* stream */) override {
+      webrtc::scoped_refptr<webrtc::MediaStreamInterface> /* stream */)
+      override {
     VLOG(1) << "OnRemoveStream called.";
   }
 
-  void OnDataChannel(
-      rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override {
+  void OnDataChannel(webrtc::scoped_refptr<webrtc::DataChannelInterface>
+                         data_channel) override {
     LOG(ERROR) << "OnDataChannel opened from server: " << data_channel->label();
     // The Meet servers should never open a data channel; all data channels are
     // opened by the client.
@@ -120,7 +124,7 @@ class ConferencePeerConnection : public ConferencePeerConnectionInterface,
   }
 
   void OnIceCandidatesRemoved(
-      const std::vector<cricket::Candidate>& candidates) override {
+      const std::vector<webrtc::Candidate>& candidates) override {
     VLOG(1) << "OnIceCandidatesRemoved: " << candidates.size();
   }
 
@@ -129,20 +133,20 @@ class ConferencePeerConnection : public ConferencePeerConnectionInterface,
   }
 
   void OnIceSelectedCandidatePairChanged(
-      const cricket::CandidatePairChangeEvent& /* event */) override {
+      const webrtc::CandidatePairChangeEvent& /* event */) override {
     VLOG(1) << "OnIceSelectedCandidatePairChanged called.";
   }
 
   void OnAddTrack(
-      rtc::scoped_refptr<webrtc::RtpReceiverInterface> /* receiver */,
+      webrtc::scoped_refptr<webrtc::RtpReceiverInterface> /* receiver */,
       const std::vector<
-          rtc::scoped_refptr<webrtc::MediaStreamInterface>>& /* streams */)
+          webrtc::scoped_refptr<webrtc::MediaStreamInterface>>& /* streams */)
       override {
     VLOG(1) << "OnAddTrack called.";
   }
 
   void OnRemoveTrack(
-      rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) override {
+      webrtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) override {
     VLOG(1) << "OnRemoveTrack called.";
   };
 
@@ -153,8 +157,8 @@ class ConferencePeerConnection : public ConferencePeerConnectionInterface,
   void OnConnectionChange(
       webrtc::PeerConnectionInterface::PeerConnectionState new_state) override;
 
-  void OnTrack(
-      rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) override;
+  void OnTrack(webrtc::scoped_refptr<webrtc::RtpTransceiverInterface>
+                   transceiver) override;
 
   // Sets the disconnect callback for the conference peer connection. Conference
   // peer connections can only have one disconnect callback at a time, and the
@@ -218,7 +222,7 @@ class ConferencePeerConnection : public ConferencePeerConnectionInterface,
   // Calling this is not thread-safe, so it should only be called before the
   // conference peer connection is used.
   void SetPeerConnection(
-      rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection) {
+      webrtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection) {
     peer_connection_ = std::move(peer_connection);
   }
 
@@ -229,9 +233,9 @@ class ConferencePeerConnection : public ConferencePeerConnectionInterface,
 
   DisconnectCallback disconnect_callback_;
   TrackSignaledCallback track_signaled_callback_;
-  std::unique_ptr<rtc::Thread> signaling_thread_;
+  std::unique_ptr<webrtc::Thread> signaling_thread_;
   std::unique_ptr<HttpConnectorInterface> http_connector_;
-  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
+  webrtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
 };
 
 }  // namespace meet