Clean up unused members in WebRtcVideoEngine classes

Following the split of the two-way video channel into separate send and
receive channels, several members and methods became unused or redundant.

This CL:
- Removes unused private methods Construct and
  FillBandwidthEstimationStats from WebRtcVideoSendChannel.
- Removes unused member variables last_unsignalled_ssrc_creation_time_ms_
  and rtp_config_ from WebRtcVideoSendChannel.
- Removes unused member variable rtp_config_ from
  WebRtcVideoReceiveChannel.
- Removes redundant initialization of rtcp_mode in
  WebRtcVideoReceiveChannel::ConfigureReceiverRtp.

Bug: None
Change-Id: Ice3aa1788f60a2c192c271e42d503e17fc4c649c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454702
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47098}

media/engine/webrtc_video_engine.cc

diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 8bfffdc..fc04a6c 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -3076,10 +3076,6 @@
 
   config->rtp.remote_ssrc = ssrc;
 
-  // The mode and rtx time is determined by a call to the configuration
-  // function.
-  config->rtp.rtcp_mode = rtp_config_.rtcp_mode;
-
   sp.GetFidSsrc(ssrc, &config->rtp.rtx_ssrc);
 
   // TODO(brandtr): Generalize when we add support for multistream protection.

media/engine/webrtc_video_engine.h

diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 307768f..d79174f 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -432,8 +432,6 @@
     const bool disable_automatic_resize_;
   };
 
-  void Construct(Call* call, WebRtcVideoEngine* engine);
-
   // Get all codecs that are compatible with the receiver.
   std::vector<VideoCodecSettings> SelectSendVideoCodecs(
       const std::vector<VideoCodecSettings>& remote_mapped_codecs) const
@@ -441,9 +439,6 @@
 
   void FillSenderStats(VideoMediaSendInfo* info, bool log_stats)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
-  void FillBandwidthEstimationStats(const Call::Stats& stats,
-                                    VideoMediaInfo* info)
-      RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
   void FillSendCodecStats(VideoMediaSendInfo* video_media_info)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
 
@@ -471,8 +466,6 @@
   // Using primary-ssrc (first ssrc) as key.
   std::map<uint32_t, WebRtcVideoSendStream*> send_streams_
       RTC_GUARDED_BY(thread_checker_);
-  std::optional<int64_t> last_unsignalled_ssrc_creation_time_ms_
-      RTC_GUARDED_BY(thread_checker_);
   std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(thread_checker_);
 
   std::optional<VideoCodecSettings> send_codec_ RTC_GUARDED_BY(thread_checker_);
@@ -489,8 +482,6 @@
   // See reason for keeping track of the FlexFEC payload type separately in
   // comment in WebRtcVideoChannel::ChangedReceiverParameters.
   BitrateConstraints bitrate_config_ RTC_GUARDED_BY(thread_checker_);
-  // TODO(deadbeef): Don't duplicate information between
-  // send_params/recv_params, rtp_extensions, options, etc.
   VideoSenderParameters send_params_ RTC_GUARDED_BY(thread_checker_);
   VideoOptions default_send_options_ RTC_GUARDED_BY(thread_checker_);
   int64_t last_send_stats_log_ms_ RTC_GUARDED_BY(thread_checker_);
@@ -498,11 +489,6 @@
   // connection.
   const CryptoOptions crypto_options_ RTC_GUARDED_BY(thread_checker_);
 
-  // RTP parameters that need to be set when creating a video receive stream.
-  // Only used in Receiver mode - in Both mode, it reads those things from the
-  // codec.
-  VideoReceiveStreamInterface::Config::Rtp rtp_config_;
-
   // Callback invoked whenever the list of SSRCs changes.
   absl::AnyInvocable<void(const std::set<uint32_t>&)>
       ssrc_list_changed_callback_;
@@ -785,11 +771,6 @@
   scoped_refptr<FrameTransformerInterface> unsignaled_frame_transformer_
       RTC_GUARDED_BY(thread_checker_);
 
-  // RTP parameters that need to be set when creating a video receive stream.
-  // Only used in Receiver mode - in Both mode, it reads those things from the
-  // codec.
-  VideoReceiveStreamInterface::Config::Rtp rtp_config_;
-
   const int receive_buffer_size_;
 };