Remove chain of methods in RtpRtcp module to get current payload frequency for RTCP SRs

webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

 /*
  *  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 55 matching lines @@
 std::string NACKStringBuilder::GetResult() {
   if (consecutive_) {
     stream_ << "-" << prevNack_;
     consecutive_ = false;
   }
   return stream_.str();
 }
 
 RTCPSender::FeedbackState::FeedbackState()
     : send_payload_type(0),
-      frequency_hz(0),
       packets_sent(0),
       media_bytes_sent(0),
       send_bitrate(0),
       last_rr_ntp_secs(0),
       last_rr_ntp_frac(0),
       remote_sr(0),
       has_last_xr_rr(false),
       module(nullptr) {}
 
 class PacketContainer : public rtcp::CompoundPacket,
@@ skipping 349 matching lines @@
   return false;
 }
 
 std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) {
   // Timestamp shouldn't be estimated before first media frame.
   RTC_DCHECK_GE(last_frame_capture_time_ms_, 0);
   // The timestamp of this RTCP packet should be estimated as the timestamp of
   // the frame being captured at this moment. We are calculating that
   // timestamp as the last frame's timestamp + the time since the last frame
   // was captured.
+  uint32_t rtp_rate =
+      (audio_ ? kBogusRtpRateForAudioRtcp : kVideoPayloadTypeFrequency) / 1000;
   uint32_t rtp_timestamp =
       timestamp_offset_ + last_rtp_timestamp_ +
-      (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) *
-          (ctx.feedback_state_.frequency_hz / 1000);
+      (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) * rtp_rate;
 
   rtcp::SenderReport* report = new rtcp::SenderReport();
   report->SetSenderSsrc(ssrc_);
   report->SetNtp(NtpTime(ctx.ntp_sec_, ctx.ntp_frac_));
   report->SetRtpTimestamp(rtp_timestamp);
   report->SetPacketCount(ctx.feedback_state_.packets_sent);
   report->SetOctetCount(ctx.feedback_state_.media_bytes_sent);
 
   for (auto it : report_blocks_)
     report->AddReportBlock(it.second);
@@ skipping 582 matching lines @@
     // but we can't because of an incorrect warning (C4822) in MVS 2013.
   } sender(transport_, event_log_);
 
   RTC_DCHECK_LE(max_payload_length_, static_cast<size_t>(IP_PACKET_SIZE));
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, max_payload_length_, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc