Add remote-outbound stats for audio streams
Add missing members needed to surface `RTCRemoteOutboundRtpStreamStats` via `ChannelReceive::GetRTCPStatistics()` - i.e., audio streams. `GetSenderReportStats()` is added to both `ModuleRtpRtcpImpl` and `ModuleRtpRtcpImpl2` and used by `ChannelReceive::GetRTCPStatistics()`. Bug: webrtc:12529 Change-Id: Ia8f5dfe2e4cfc43e3ddd28f2f1149f5c00f9269d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211041 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33452}
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@ -196,6 +196,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
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// Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
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// which is the SSRC of the corresponding outbound RTP stream, is unique.
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std::vector<ReportBlockData> GetLatestReportBlockData() const override;
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absl::optional<SenderReportStats> GetSenderReportStats() const override;
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// (REMB) Receiver Estimated Max Bitrate.
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void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
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