Move RtpTransportControllerSend to a new file.

Also move RtpTransportControllerSendInterface to its own header file.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2808043002
Cr-Commit-Position: refs/heads/master@{#17840}

webrtc/audio/audio_send_stream.cc

@@ -20,7 +20,7 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/call/rtp_transport_controller_send.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"

webrtc/audio/audio_send_stream_unittest.cc

@@ -15,7 +15,7 @@
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/task_queue.h"
-#include "webrtc/call/rtp_transport_controller_send.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h"

webrtc/call/BUILD.gn

@@ -16,7 +16,7 @@ rtc_source_set("call_interfaces") {
     "audio_state.h",
     "call.h",
     "flexfec_receive_stream.h",
-    "rtp_transport_controller_send.h",
+    "rtp_transport_controller_send_interface.h",
     "syncable.cc",
     "syncable.h",
   ]
@@ -38,6 +38,8 @@ rtc_static_library("call") {
     "call.cc",
     "flexfec_receive_stream_impl.cc",
     "flexfec_receive_stream_impl.h",
+    "rtp_transport_controller_send.cc",
+    "rtp_transport_controller_send.h",
   ]
 
   if (!build_with_chromium && is_clang) {

webrtc/call/call.cc

@@ -38,7 +38,6 @@
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h"
-#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
@@ -87,40 +86,6 @@ bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {
   return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);
 }
 
-class RtpTransportControllerSend : public RtpTransportControllerSendInterface {
- public:
-  RtpTransportControllerSend(Clock* clock, webrtc::RtcEventLog* event_log);
-
-  void RegisterNetworkObserver(
-      SendSideCongestionController::Observer* observer);
-
-  // Implements RtpTransportControllerSendInterface
-  PacketRouter* packet_router() override { return &packet_router_; }
-  SendSideCongestionController* send_side_cc() override {
-    return &send_side_cc_;
-  }
-  TransportFeedbackObserver* transport_feedback_observer() override {
-    return &send_side_cc_;
-  }
-  RtpPacketSender* packet_sender() override { return send_side_cc_.pacer(); }
-
- private:
-  PacketRouter packet_router_;
-  SendSideCongestionController send_side_cc_;
-};
-
-RtpTransportControllerSend::RtpTransportControllerSend(
-    Clock* clock,
-    webrtc::RtcEventLog* event_log)
-    : send_side_cc_(clock, nullptr /* observer */, event_log, &packet_router_) {
-}
-
-void RtpTransportControllerSend::RegisterNetworkObserver(
-    SendSideCongestionController::Observer* observer) {
-  // Must be called only once.
-  send_side_cc_.RegisterNetworkObserver(observer);
-}
-
 }  // namespace
 
 namespace internal {

webrtc/call/rtp_transport_controller_send.cc

@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/call/rtp_transport_controller_send.h"
+
+namespace webrtc {
+
+RtpTransportControllerSend::RtpTransportControllerSend(
+    Clock* clock,
+    webrtc::RtcEventLog* event_log)
+    : send_side_cc_(clock, nullptr /* observer */, event_log, &packet_router_) {
+}
+
+void RtpTransportControllerSend::RegisterNetworkObserver(
+    SendSideCongestionController::Observer* observer) {
+  // Must be called only once.
+  send_side_cc_.RegisterNetworkObserver(observer);
+}
+
+}  // namespace webrtc

webrtc/call/rtp_transport_controller_send.h

@@ -11,46 +11,39 @@
 #ifndef WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
 #define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
 
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
+#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
+
 namespace webrtc {
+class Clock;
+class RtcEventLog;
 
-class Module;
-class PacketRouter;
-class RtpPacketSender;
-class SendSideCongestionController;
-class TransportFeedbackObserver;
-
-// An RtpTransportController should own everything related to the RTP
-// transport to/from a remote endpoint. We should have separate
-// interfaces for send and receive side, even if they are implemented
-// by the same class. This is an ongoing refactoring project. At some
-// point, this class should be promoted to a public api under
-// webrtc/api/rtp/.
-//
-// For a start, this object is just a collection of the objects needed
-// by the VideoSendStream constructor. The plan is to move ownership
-// of all RTP-related objects here, and add methods to create per-ssrc
-// objects which would then be passed to VideoSendStream. Eventually,
-// direct accessors like packet_router() should be removed.
-//
-// This should also have a reference to the underlying
-// webrtc::Transport(s). Currently, webrtc::Transport is implemented by
-// WebRtcVideoChannel2 and WebRtcVoiceMediaChannel, and owned by
-// WebrtcSession. Video and audio always uses different transport
-// objects, even in the common case where they are bundled over the
-// same underlying transport.
-//
-// Extracting the logic of the webrtc::Transport from BaseChannel and
-// subclasses into a separate class seems to be a prerequesite for
-// moving the transport here.
-class RtpTransportControllerSendInterface {
+// TODO(nisse): When we get the underlying transports here, we should
+// have one object implementing RtpTransportControllerSendInterface
+// per transport, sharing the same congestion controller.
+class RtpTransportControllerSend : public RtpTransportControllerSendInterface {
  public:
-  virtual ~RtpTransportControllerSendInterface() {}
-  virtual PacketRouter* packet_router() = 0;
-  // Currently returning the same pointer, but with different types.
-  virtual SendSideCongestionController* send_side_cc() = 0;
-  virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
+  RtpTransportControllerSend(Clock* clock, webrtc::RtcEventLog* event_log);
 
-  virtual RtpPacketSender* packet_sender() = 0;
+  void RegisterNetworkObserver(
+      SendSideCongestionController::Observer* observer);
+
+  // Implements RtpTransportControllerSendInterface
+  PacketRouter* packet_router() override { return &packet_router_; }
+  SendSideCongestionController* send_side_cc() override {
+    return &send_side_cc_;
+  }
+  TransportFeedbackObserver* transport_feedback_observer() override {
+    return &send_side_cc_;
+  }
+  RtpPacketSender* packet_sender() override { return send_side_cc_.pacer(); }
+
+ private:
+  PacketRouter packet_router_;
+  SendSideCongestionController send_side_cc_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend);
 };
 
 }  // namespace webrtc

webrtc/call/rtp_transport_controller_send_interface.h

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
+#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
+
+namespace webrtc {
+
+class PacketRouter;
+class RtpPacketSender;
+class SendSideCongestionController;
+class TransportFeedbackObserver;
+
+// An RtpTransportController should own everything related to the RTP
+// transport to/from a remote endpoint. We should have separate
+// interfaces for send and receive side, even if they are implemented
+// by the same class. This is an ongoing refactoring project. At some
+// point, this class should be promoted to a public api under
+// webrtc/api/rtp/.
+//
+// For a start, this object is just a collection of the objects needed
+// by the VideoSendStream constructor. The plan is to move ownership
+// of all RTP-related objects here, and add methods to create per-ssrc
+// objects which would then be passed to VideoSendStream. Eventually,
+// direct accessors like packet_router() should be removed.
+//
+// This should also have a reference to the underlying
+// webrtc::Transport(s). Currently, webrtc::Transport is implemented by
+// WebRtcVideoChannel2 and WebRtcVoiceMediaChannel, and owned by
+// WebrtcSession. Video and audio always uses different transport
+// objects, even in the common case where they are bundled over the
+// same underlying transport.
+//
+// Extracting the logic of the webrtc::Transport from BaseChannel and
+// subclasses into a separate class seems to be a prerequesite for
+// moving the transport here.
+class RtpTransportControllerSendInterface {
+ public:
+  virtual ~RtpTransportControllerSendInterface() {}
+  virtual PacketRouter* packet_router() = 0;
+  // Currently returning the same pointer, but with different types.
+  virtual SendSideCongestionController* send_side_cc() = 0;
+  virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
+
+  virtual RtpPacketSender* packet_sender() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_

webrtc/video/video_send_stream.cc

@@ -22,7 +22,7 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/base/weak_ptr.h"
-#include "webrtc/call/rtp_transport_controller_send.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/video_bitrate_allocator.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"

webrtc/voice_engine/channel.cc

@@ -24,7 +24,7 @@
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/call/rtp_transport_controller_send.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"

webrtc/voice_engine/channel_proxy.cc

@@ -15,7 +15,7 @@
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/call/rtp_transport_controller_send.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/voice_engine/channel.h"
 
 namespace webrtc {