[audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo

This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was
split up into smaller pieces for reviewability. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats

In [1], ReportBlockData was implemented and tested.
This CL adds the plumbing to make it available in MediaSenderInfo for
audio streams, but the code is not wired up to make use of these stats.

In follow-up CL [2], ReportBlockData will be used to implement
RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration
tests exercising the full code path.

[1] https://webrtc-review.googlesource.com/c/src/+/136584
[2] https://webrtc-review.googlesource.com/c/src/+/138067

Bug: webrtc:10455
Change-Id: Id8940090cc9c121e8f06ccdb068a22ce24c07092
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138066
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28072}

audio/audio_send_stream.cc

@@ -434,6 +434,8 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
   stats.apm_statistics =
       audio_state_->audio_processing()->GetStatistics(has_remote_tracks);
 
+  stats.report_block_datas = std::move(call_stats.report_block_datas);
+
   return stats;
 }
 

audio/channel_send.cc

@@ -1079,6 +1079,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
   stats.packetsSent =
       rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
   stats.retransmitted_packets_sent = rtp_stats.retransmitted.packets;
+  stats.report_block_datas = _rtpRtcpModule->GetLatestReportBlockData();
 
   return stats;
 }

audio/channel_send.h

@@ -22,6 +22,7 @@
 #include "api/media_transport_config.h"
 #include "api/media_transport_interface.h"
 #include "api/task_queue/task_queue_factory.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/source/rtp_sender_audio.h"
 
@@ -41,6 +42,11 @@ struct CallSendStatistics {
   int packetsSent;
   // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
   uint64_t retransmitted_packets_sent;
+  // A snapshot of Report Blocks with additional data of interest to statistics.
+  // Within this list, the sender-source SSRC pair is unique and per-pair the
+  // ReportBlockData represents the latest Report Block that was received for
+  // that pair.
+  std::vector<ReportBlockData> report_block_datas;
 };
 
 // See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.

call/BUILD.gn

@@ -44,6 +44,7 @@ rtc_source_set("call_interfaces") {
     "../modules/audio_processing",
     "../modules/audio_processing:api",
     "../modules/audio_processing:audio_processing_statistics",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
     "../modules/utility",
     "../rtc_base",
     "../rtc_base:audio_format_to_string",

call/audio_send_stream.h

@@ -29,6 +29,7 @@
 #include "api/scoped_refptr.h"
 #include "call/rtp_config.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
 
 namespace webrtc {
 
@@ -66,6 +67,11 @@ class AudioSendStream {
     AudioProcessingStats apm_statistics;
 
     int64_t target_bitrate_bps = 0;
+    // A snapshot of Report Blocks with additional data of interest to
+    // statistics. Within this list, the sender-source SSRC pair is unique and
+    // per-pair the ReportBlockData represents the latest Report Block that was
+    // received for that pair.
+    std::vector<ReportBlockData> report_block_datas;
   };
 
   struct Config {

media/engine/webrtc_voice_engine.cc

@@ -2215,6 +2215,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false);
     sinfo.ana_statistics = stats.ana_statistics;
     sinfo.apm_statistics = stats.apm_statistics;
+    sinfo.report_block_datas = std::move(stats.report_block_datas);
     info->senders.push_back(sinfo);
   }
 

modules/rtp_rtcp/include/rtp_rtcp.h

@@ -350,8 +350,15 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
 
   // Returns received RTCP report block.
   // Returns -1 on failure else 0.
+  // TODO(https://crbug.com/webrtc/10678): Remove this in favor of
+  // GetLatestReportBlockData().
   virtual int32_t RemoteRTCPStat(
       std::vector<RTCPReportBlock>* receive_blocks) const = 0;
+  // A snapshot of Report Blocks with additional data of interest to statistics.
+  // Within this list, the sender-source SSRC pair is unique and per-pair the
+  // ReportBlockData represents the latest Report Block that was received for
+  // that pair.
+  virtual std::vector<ReportBlockData> GetLatestReportBlockData() const = 0;
 
   // (APP) Sets application specific data.
   // Returns -1 on failure else 0.

modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

@@ -126,6 +126,7 @@ class MockRtpRtcp : public RtpRtcp {
                      void(bool, uint32_t, struct RtpPacketLossStats*));
   MOCK_CONST_METHOD1(RemoteRTCPStat,
                      int32_t(std::vector<RTCPReportBlock>* receive_blocks));
+  MOCK_CONST_METHOD0(GetLatestReportBlockData, std::vector<ReportBlockData>());
   MOCK_METHOD4(SetRTCPApplicationSpecificData,
                int32_t(uint8_t sub_type,
                        uint32_t name,

modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -589,6 +589,11 @@ int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(
   return rtcp_receiver_.StatisticsReceived(receive_blocks);
 }
 
+std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
+    const {
+  return rtcp_receiver_.GetLatestReportBlockData();
+}
+
 // (REMB) Receiver Estimated Max Bitrate.
 void ModuleRtpRtcpImpl::SetRemb(int64_t bitrate_bps,
                                 std::vector<uint32_t> ssrcs) {

modules/rtp_rtcp/source/rtp_rtcp_impl.h

@@ -200,6 +200,11 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
   // Get received RTCP report, report block.
   int32_t RemoteRTCPStat(
       std::vector<RTCPReportBlock>* receive_blocks) const override;
+  // A snapshot of the most recent Report Block with additional data of
+  // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
+  // Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
+  // which is the SSRC of the corresponding outbound RTP stream, is unique.
+  std::vector<ReportBlockData> GetLatestReportBlockData() const override;
 
   // (REMB) Receiver Estimated Max Bitrate.
   void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;