Remove RTPSenderInterface

webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 13 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class RTPSenderVideo {
  public:
-  RTPSenderVideo(Clock* clock, RTPSenderInterface* rtp_sender);
+  RTPSenderVideo(Clock* clock, RTPSender* rtpSender);
   virtual ~RTPSenderVideo();
 
   virtual RtpVideoCodecTypes VideoCodecType() const;
 
   size_t FECPacketOverhead() const;
 
   static RtpUtility::Payload* CreateVideoPayload(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       int8_t payload_type);
 
@@ skipping 40 matching lines @@
 
   void SendVideoPacketAsRed(uint8_t* data_buffer,
                             size_t payload_length,
                             size_t rtp_header_length,
                             uint16_t video_seq_num,
                             uint32_t capture_timestamp,
                             int64_t capture_time_ms,
                             StorageType media_packet_storage,
                             bool protect);
 
-  RTPSenderInterface* const rtp_sender_;
+  RTPSender* const rtp_sender_;
   Clock* const clock_;
 
   // Should never be held when calling out of this class.
   rtc::CriticalSection crit_;
 
   RtpVideoCodecTypes video_type_ = kRtpVideoGeneric;
   int32_t retransmission_settings_ GUARDED_BY(crit_) = kRetransmitBaseLayer;
 
   // FEC
   ForwardErrorCorrection fec_;
@@ skipping 11 matching lines @@
   // and any padding overhead.
   RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_);
   // Bitrate used for video payload and RTP headers.
   RateStatistics video_bitrate_ GUARDED_BY(stats_crit_);
   OneTimeEvent first_frame_sent_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_