[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}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 5c88e01..24f6fe1 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -434,6 +434,8 @@
stats.apm_statistics =
audio_state_->audio_processing()->GetStatistics(has_remote_tracks);
+ stats.report_block_datas = std::move(call_stats.report_block_datas);
+
return stats;
}
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 38e89d8..793a7dd 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -1079,6 +1079,7 @@
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;
}
diff --git a/audio/channel_send.h b/audio/channel_send.h
index fb98be3..2762f53 100644
--- a/audio/channel_send.h
+++ b/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 @@
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.
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 64cba50..1696478 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -44,6 +44,7 @@
"../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",
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index b21b2ef..d8fdddb 100644
--- a/call/audio_send_stream.h
+++ b/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 @@
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 {
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 435b3ab..55a2826 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2215,6 +2215,7 @@
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);
}
diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h
index 216f758..22de74e 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -350,8 +350,15 @@
// 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.
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index eaad03d..b1ec2ca 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -126,6 +126,7 @@
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,
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 1d7f048..4855775 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -589,6 +589,11 @@
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) {
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 71a856c..286a8af 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -200,6 +200,11 @@
// 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;