Reland "ChannelStatistics used for RTP stats in VoipStatistics."

This is a reland of 444e04be6988fbdcc039d775481ac22481ff9ff4

Reason for reland: resolved the breaks from downstream project

Original change's description:
> ChannelStatistics used for RTP stats in VoipStatistics.
>
> - Added local and remote RTP statistics query API.
> - Change includes simplifying remote SSRC change handling
>   via received RTP and RTCP packets.
>
> Bug: webrtc:11989
> Change-Id: Ia3ee62c1191baaedc67e033ea3c661d8c9301abc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199060
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Tim Na <natim@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32954}

Bug: webrtc:11989
Change-Id: I88620a9f1c037b512821cac9d556905149666870
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201481
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32966}

api/voip/voip_statistics.h

@@ -26,6 +26,51 @@ struct IngressStatistics {
   double total_duration = 0.0;
 };
 
+// Remote statistics obtained via remote RTCP SR/RR report received.
+struct RemoteRtcpStatistics {
+  // Jitter as defined in RFC 3550 [6.4.1] expressed in seconds.
+  double jitter = 0.0;
+
+  // Cumulative packets lost as defined in RFC 3550 [6.4.1]
+  int64_t packets_lost = 0;
+
+  // Fraction lost as defined in RFC 3550 [6.4.1] expressed as a floating
+  // pointer number.
+  double fraction_lost = 0.0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats-roundtriptime
+  absl::optional<double> round_trip_time;
+
+  // Last time (not RTP timestamp) when RTCP report received in milliseconds.
+  int64_t last_report_received_timestamp_ms;
+};
+
+struct ChannelStatistics {
+  // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-packetssent
+  uint64_t packets_sent = 0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcsentrtpstreamstats-bytessent
+  uint64_t bytes_sent = 0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsreceived
+  uint64_t packets_received = 0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-bytesreceived
+  uint64_t bytes_received = 0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-jitter
+  double jitter = 0.0;
+
+  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetslost
+  int64_t packets_lost = 0;
+
+  // SSRC from remote media endpoint as indicated either by RTP header in RFC
+  // 3550 [5.1] or RTCP SSRC of sender in RFC 3550 [6.4.1].
+  absl::optional<uint32_t> remote_ssrc;
+
+  absl::optional<RemoteRtcpStatistics> remote_rtcp;
+};
+
 // VoipStatistics interface provides the interfaces for querying metrics around
 // the jitter buffer (NetEq) performance.
 class VoipStatistics {
@@ -37,6 +82,13 @@ class VoipStatistics {
   virtual VoipResult GetIngressStatistics(ChannelId channel_id,
                                           IngressStatistics& ingress_stats) = 0;
 
+  // Gets the channel statistics by |channel_stats| reference.
+  // Returns following VoipResult;
+  //  kOk - successfully set provided ChannelStatistics reference.
+  //  kInvalidArgument - |channel_id| is invalid.
+  virtual VoipResult GetChannelStatistics(ChannelId channel_id,
+                                          ChannelStatistics& channel_stats) = 0;
+
  protected:
   virtual ~VoipStatistics() = default;
 };

audio/voip/BUILD.gn

@@ -67,6 +67,7 @@ rtc_library("audio_ingress") {
     "../../api:transport_api",
     "../../api/audio:audio_mixer_api",
     "../../api/audio_codecs:audio_codecs_api",
+    "../../api/voip:voip_api",
     "../../modules/audio_coding",
     "../../modules/rtp_rtcp",
     "../../modules/rtp_rtcp:rtp_rtcp_format",

audio/voip/audio_channel.cc

@@ -79,6 +79,12 @@ AudioChannel::~AudioChannel() {
   }
 
   audio_mixer_->RemoveSource(ingress_.get());
+
+  // AudioEgress could hold current global TaskQueueBase that we need to clear
+  // before ProcessThread::DeRegisterModule.
+  egress_.reset();
+  ingress_.reset();
+
   process_thread_->DeRegisterModule(rtp_rtcp_.get());
 }
 
@@ -159,4 +165,17 @@ IngressStatistics AudioChannel::GetIngressStatistics() {
   return ingress_stats;
 }
 
+ChannelStatistics AudioChannel::GetChannelStatistics() {
+  ChannelStatistics channel_stat = ingress_->GetChannelStatistics();
+
+  StreamDataCounters rtp_stats, rtx_stats;
+  rtp_rtcp_->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
+  channel_stat.bytes_sent =
+      rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
+  channel_stat.packets_sent =
+      rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
+
+  return channel_stat;
+}
+
 }  // namespace webrtc

audio/voip/audio_channel.h

@@ -84,6 +84,7 @@ class AudioChannel : public rtc::RefCountInterface {
     ingress_->SetReceiveCodecs(codecs);
   }
   IngressStatistics GetIngressStatistics();
+  ChannelStatistics GetChannelStatistics();
 
   // See comments on the methods used from AudioEgress and AudioIngress.
   // Conversion to double is following what is done in
@@ -106,6 +107,12 @@ class AudioChannel : public rtc::RefCountInterface {
     return ingress_->GetOutputTotalDuration();
   }
 
+  // Internal API for testing purpose.
+  void SendRTCPReportForTesting(RTCPPacketType type) {
+    int32_t result = rtp_rtcp_->SendRTCP(type);
+    RTC_DCHECK(result == 0);
+  }
+
  private:
   // ChannelId that this audio channel belongs for logging purpose.
   ChannelId id_;

audio/voip/audio_ingress.cc

@@ -17,6 +17,10 @@
 #include "api/audio_codecs/audio_format.h"
 #include "audio/utility/audio_frame_operations.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_minmax.h"
 
@@ -153,6 +157,12 @@ void AudioIngress::ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
     rtp_packet_received.set_payload_type_frequency(it->second);
   }
 
+  // Track current remote SSRC.
+  if (rtp_packet_received.Ssrc() != remote_ssrc_) {
+    rtp_rtcp_->SetRemoteSSRC(rtp_packet_received.Ssrc());
+    remote_ssrc_.store(rtp_packet_received.Ssrc());
+  }
+
   rtp_receive_statistics_->OnRtpPacket(rtp_packet_received);
 
   RTPHeader header;
@@ -181,11 +191,28 @@ void AudioIngress::ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
 
 void AudioIngress::ReceivedRTCPPacket(
     rtc::ArrayView<const uint8_t> rtcp_packet) {
-  // Deliver RTCP packet to RTP/RTCP module for parsing.
+  rtcp::CommonHeader rtcp_header;
+  if (rtcp_header.Parse(rtcp_packet.data(), rtcp_packet.size()) &&
+      (rtcp_header.type() == rtcp::SenderReport::kPacketType ||
+       rtcp_header.type() == rtcp::ReceiverReport::kPacketType)) {
+    RTC_DCHECK_GE(rtcp_packet.size(), 8);
+
+    uint32_t sender_ssrc =
+        ByteReader<uint32_t>::ReadBigEndian(rtcp_packet.data() + 4);
+
+    // If we don't have remote ssrc at this point, it's likely that remote
+    // endpoint is receive-only or it could have restarted the media.
+    if (sender_ssrc != remote_ssrc_) {
+      rtp_rtcp_->SetRemoteSSRC(sender_ssrc);
+      remote_ssrc_.store(sender_ssrc);
+    }
+  }
+
+  // Deliver RTCP packet to RTP/RTCP module for parsing and processing.
   rtp_rtcp_->IncomingRtcpPacket(rtcp_packet.data(), rtcp_packet.size());
 
-  absl::optional<int64_t> rtt = GetRoundTripTime();
+  int64_t rtt = 0;
-  if (!rtt.has_value()) {
+  if (rtp_rtcp_->RTT(remote_ssrc_, &rtt, nullptr, nullptr, nullptr) != 0) {
     // Waiting for valid RTT.
     return;
   }
@@ -199,38 +226,69 @@ void AudioIngress::ReceivedRTCPPacket(
 
   {
     MutexLock lock(&lock_);
-    ntp_estimator_.UpdateRtcpTimestamp(*rtt, ntp_secs, ntp_frac, rtp_timestamp);
+    ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
   }
 }
 
-absl::optional<int64_t> AudioIngress::GetRoundTripTime() {
+ChannelStatistics AudioIngress::GetChannelStatistics() {
+  ChannelStatistics channel_stats;
+
+  // Get clockrate for current decoder ahead of jitter calculation.
+  uint32_t clockrate_hz = 0;
+  absl::optional<std::pair<int, SdpAudioFormat>> decoder =
+      acm_receiver_.LastDecoder();
+  if (decoder) {
+    clockrate_hz = decoder->second.clockrate_hz;
+  }
+
+  StreamStatistician* statistician =
+      rtp_receive_statistics_->GetStatistician(remote_ssrc_);
+  if (statistician) {
+    RtpReceiveStats stats = statistician->GetStats();
+    channel_stats.packets_lost = stats.packets_lost;
+    channel_stats.packets_received = stats.packet_counter.packets;
+    channel_stats.bytes_received = stats.packet_counter.payload_bytes;
+    channel_stats.remote_ssrc = remote_ssrc_;
+    if (clockrate_hz > 0) {
+      channel_stats.jitter = static_cast<double>(stats.jitter) / clockrate_hz;
+    }
+  }
+
+  // Get RTCP report using remote SSRC.
   const std::vector<ReportBlockData>& report_data =
       rtp_rtcp_->GetLatestReportBlockData();
+  for (const ReportBlockData& block_data : report_data) {
+    const RTCPReportBlock& rtcp_report = block_data.report_block();
+    if (rtp_rtcp_->SSRC() != rtcp_report.source_ssrc ||
+        remote_ssrc_ != rtcp_report.sender_ssrc) {
+      continue;
+    }
+    RemoteRtcpStatistics remote_stat;
+    remote_stat.packets_lost = rtcp_report.packets_lost;
+    remote_stat.fraction_lost =
+        static_cast<double>(rtcp_report.fraction_lost) / (1 << 8);
+    if (clockrate_hz > 0) {
+      remote_stat.jitter =
+          static_cast<double>(rtcp_report.jitter) / clockrate_hz;
+    }
+    if (block_data.has_rtt()) {
+      remote_stat.round_trip_time =
+          static_cast<double>(block_data.last_rtt_ms()) /
+          rtc::kNumMillisecsPerSec;
+    }
+    remote_stat.last_report_received_timestamp_ms =
+        block_data.report_block_timestamp_utc_us() /
+        rtc::kNumMicrosecsPerMillisec;
+    channel_stats.remote_rtcp = remote_stat;
 
-  // If we do not have report block which means remote RTCP hasn't be received
+    // Receive only channel won't send any RTP packets.
-  // yet, return -1 as to indicate uninitialized value.
+    if (!channel_stats.remote_ssrc.has_value()) {
-  if (report_data.empty()) {
+      channel_stats.remote_ssrc = remote_ssrc_;
-    return absl::nullopt;
+    }
+    break;
   }
 
-  // We don't know in advance the remote SSRC used by the other end's receiver
+  return channel_stats;
-  // reports, so use the SSRC of the first report block as remote SSRC for now.
-  // TODO(natim@webrtc.org): handle the case where remote end is changing ssrc
-  // and update accordingly here.
-  const ReportBlockData& block_data = report_data[0];
-
-  const uint32_t sender_ssrc = block_data.report_block().sender_ssrc;
-
-  if (sender_ssrc != remote_ssrc_.load()) {
-    remote_ssrc_.store(sender_ssrc);
-    rtp_rtcp_->SetRemoteSSRC(sender_ssrc);
-  }
-
-  if (!block_data.has_rtt()) {
-    return absl::nullopt;
-  }
-
-  return block_data.last_rtt_ms();
 }
 
 }  // namespace webrtc

audio/voip/audio_ingress.h

@@ -22,6 +22,7 @@
 #include "api/audio/audio_mixer.h"
 #include "api/rtp_headers.h"
 #include "api/scoped_refptr.h"
+#include "api/voip/voip_statistics.h"
 #include "audio/audio_level.h"
 #include "modules/audio_coding/acm2/acm_receiver.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
@@ -86,6 +87,8 @@ class AudioIngress : public AudioMixer::Source {
     return stats;
   }
 
+  ChannelStatistics GetChannelStatistics();
+
   // Implementation of AudioMixer::Source interface.
   AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
       int sampling_rate,
@@ -102,10 +105,6 @@ class AudioIngress : public AudioMixer::Source {
   }
 
  private:
-  // Returns network round trip time (RTT) measued by RTCP exchange with
-  // remote media endpoint. Returns absl::nullopt when it's not initialized.
-  absl::optional<int64_t> GetRoundTripTime();
-
   // Indicates AudioIngress status as caller invokes Start/StopPlaying.
   // If not playing, incoming RTP data processing is skipped, thus
   // producing no data to output device.

audio/voip/test/BUILD.gn

@@ -9,6 +9,16 @@
 import("../../../webrtc.gni")
 
 if (rtc_include_tests) {
+  rtc_source_set("mock_task_queue") {
+    testonly = true
+    visibility = [ "*" ]
+    sources = [ "mock_task_queue.h" ]
+    deps = [
+      "../../../api/task_queue:task_queue",
+      "../../../test:test_support",
+    ]
+  }
+
   rtc_library("voip_core_unittests") {
     testonly = true
     sources = [ "voip_core_unittest.cc" ]
@@ -30,18 +40,18 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "audio_channel_unittest.cc" ]
     deps = [
+      ":mock_task_queue",
       "..:audio_channel",
       "../../../api:transport_api",
       "../../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../../api/audio_codecs:builtin_audio_encoder_factory",
-      "../../../api/task_queue:default_task_queue_factory",
+      "../../../api/task_queue:task_queue",
       "../../../modules/audio_mixer:audio_mixer_impl",
       "../../../modules/audio_mixer:audio_mixer_test_utils",
       "../../../modules/rtp_rtcp:rtp_rtcp",
       "../../../modules/rtp_rtcp:rtp_rtcp_format",
       "../../../modules/utility",
       "../../../rtc_base:logging",
-      "../../../rtc_base:rtc_event",
       "../../../test:mock_transport",
       "../../../test:test_support",
     ]

audio/voip/test/audio_channel_unittest.cc

@@ -12,12 +12,12 @@
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/call/transport.h"
-#include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "audio/voip/test/mock_task_queue.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "modules/audio_mixer/sine_wave_generator.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/utility/include/process_thread.h"
-#include "rtc_base/event.h"
 #include "rtc_base/logging.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -41,11 +41,16 @@ class AudioChannelTest : public ::testing::Test {
 
   AudioChannelTest()
       : fake_clock_(kStartTime), wave_generator_(1000.0, kAudioLevel) {
+    task_queue_factory_ = std::make_unique<MockTaskQueueFactory>(&task_queue_);
     process_thread_ = ProcessThread::Create("ModuleProcessThread");
     audio_mixer_ = AudioMixerImpl::Create();
-    task_queue_factory_ = CreateDefaultTaskQueueFactory();
     encoder_factory_ = CreateBuiltinAudioEncoderFactory();
     decoder_factory_ = CreateBuiltinAudioDecoderFactory();
+
+    // By default, run the queued task immediately.
+    ON_CALL(task_queue_, PostTask)
+        .WillByDefault(
+            Invoke([&](std::unique_ptr<QueuedTask> task) { task->Run(); }));
   }
 
   void SetUp() override {
@@ -80,6 +85,7 @@ class AudioChannelTest : public ::testing::Test {
   SimulatedClock fake_clock_;
   SineWaveGenerator wave_generator_;
   NiceMock<MockTransport> transport_;
+  NiceMock<MockTaskQueue> task_queue_;
   std::unique_ptr<TaskQueueFactory> task_queue_factory_;
   rtc::scoped_refptr<AudioMixer> audio_mixer_;
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
@@ -92,11 +98,9 @@ class AudioChannelTest : public ::testing::Test {
 // Resulted RTP packet is looped back into AudioChannel and gets decoded into
 // audio frame to see if it has some signal to indicate its validity.
 TEST_F(AudioChannelTest, PlayRtpByLocalLoop) {
-  rtc::Event event;
   auto loop_rtp = [&](const uint8_t* packet, size_t length, Unused) {
     audio_channel_->ReceivedRTPPacket(
         rtc::ArrayView<const uint8_t>(packet, length));
-    event.Set();
     return true;
   };
   EXPECT_CALL(transport_, SendRtp).WillOnce(Invoke(loop_rtp));
@@ -105,8 +109,6 @@ TEST_F(AudioChannelTest, PlayRtpByLocalLoop) {
   audio_sender->SendAudioData(GetAudioFrame(0));
   audio_sender->SendAudioData(GetAudioFrame(1));
 
-  event.Wait(/*ms=*/1000);
-
   AudioFrame empty_frame, audio_frame;
   empty_frame.Mute();
   empty_frame.mutable_data();  // This will zero out the data.
@@ -122,10 +124,8 @@ TEST_F(AudioChannelTest, PlayRtpByLocalLoop) {
 // Validate assigned local SSRC is resulted in RTP packet.
 TEST_F(AudioChannelTest, VerifyLocalSsrcAsAssigned) {
   RtpPacketReceived rtp;
-  rtc::Event event;
   auto loop_rtp = [&](const uint8_t* packet, size_t length, Unused) {
     rtp.Parse(packet, length);
-    event.Set();
     return true;
   };
   EXPECT_CALL(transport_, SendRtp).WillOnce(Invoke(loop_rtp));
@@ -134,18 +134,14 @@ TEST_F(AudioChannelTest, VerifyLocalSsrcAsAssigned) {
   audio_sender->SendAudioData(GetAudioFrame(0));
   audio_sender->SendAudioData(GetAudioFrame(1));
 
-  event.Wait(/*ms=*/1000);
-
   EXPECT_EQ(rtp.Ssrc(), kLocalSsrc);
 }
 
 // Check metrics after processing an RTP packet.
 TEST_F(AudioChannelTest, TestIngressStatistics) {
-  auto event = std::make_unique<rtc::Event>();
   auto loop_rtp = [&](const uint8_t* packet, size_t length, Unused) {
     audio_channel_->ReceivedRTPPacket(
         rtc::ArrayView<const uint8_t>(packet, length));
-    event->Set();
     return true;
   };
   EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(loop_rtp));
@@ -153,7 +149,6 @@ TEST_F(AudioChannelTest, TestIngressStatistics) {
   auto audio_sender = audio_channel_->GetAudioSender();
   audio_sender->SendAudioData(GetAudioFrame(0));
   audio_sender->SendAudioData(GetAudioFrame(1));
-  event->Wait(/*give_up_after_ms=*/1000);
 
   AudioFrame audio_frame;
   audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
@@ -182,10 +177,8 @@ TEST_F(AudioChannelTest, TestIngressStatistics) {
   audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
 
   // Send another RTP packet to intentionally break PLC.
-  event = std::make_unique<rtc::Event>();
   audio_sender->SendAudioData(GetAudioFrame(2));
   audio_sender->SendAudioData(GetAudioFrame(3));
-  event->Wait(/*give_up_after_ms=*/1000);
 
   ingress_stats = audio_channel_->GetIngressStatistics();
   EXPECT_TRUE(ingress_stats);
@@ -222,5 +215,59 @@ TEST_F(AudioChannelTest, TestIngressStatistics) {
   EXPECT_DOUBLE_EQ(ingress_stats->total_duration, 0.06);
 }
 
+// Check ChannelStatistics metric after processing RTP and RTCP packets.
+TEST_F(AudioChannelTest, TestChannelStatistics) {
+  auto loop_rtp = [&](const uint8_t* packet, size_t length, Unused) {
+    audio_channel_->ReceivedRTPPacket(
+        rtc::ArrayView<const uint8_t>(packet, length));
+    return true;
+  };
+  auto loop_rtcp = [&](const uint8_t* packet, size_t length) {
+    audio_channel_->ReceivedRTCPPacket(
+        rtc::ArrayView<const uint8_t>(packet, length));
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(loop_rtp));
+  EXPECT_CALL(transport_, SendRtcp).WillRepeatedly(Invoke(loop_rtcp));
+
+  // Simulate microphone giving audio frame (10 ms). This will trigger tranport
+  // to send RTP as handled in loop_rtp above.
+  auto audio_sender = audio_channel_->GetAudioSender();
+  audio_sender->SendAudioData(GetAudioFrame(0));
+  audio_sender->SendAudioData(GetAudioFrame(1));
+
+  // Simulate speaker requesting audio frame (10 ms). This will trigger VoIP
+  // engine to fetch audio samples from RTP packets stored in jitter buffer.
+  AudioFrame audio_frame;
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+
+  // Force sending RTCP SR report in order to have remote_rtcp field available
+  // in channel statistics. This will trigger tranport to send RTCP as handled
+  // in loop_rtcp above.
+  audio_channel_->SendRTCPReportForTesting(kRtcpSr);
+
+  absl::optional<ChannelStatistics> channel_stats =
+      audio_channel_->GetChannelStatistics();
+  EXPECT_TRUE(channel_stats);
+
+  EXPECT_EQ(channel_stats->packets_sent, 1ULL);
+  EXPECT_EQ(channel_stats->bytes_sent, 160ULL);
+
+  EXPECT_EQ(channel_stats->packets_received, 1ULL);
+  EXPECT_EQ(channel_stats->bytes_received, 160ULL);
+  EXPECT_EQ(channel_stats->jitter, 0);
+  EXPECT_EQ(channel_stats->packets_lost, 0);
+  EXPECT_EQ(channel_stats->remote_ssrc.value(), kLocalSsrc);
+
+  EXPECT_TRUE(channel_stats->remote_rtcp.has_value());
+
+  EXPECT_EQ(channel_stats->remote_rtcp->jitter, 0);
+  EXPECT_EQ(channel_stats->remote_rtcp->packets_lost, 0);
+  EXPECT_EQ(channel_stats->remote_rtcp->fraction_lost, 0);
+  EXPECT_GT(channel_stats->remote_rtcp->last_report_received_timestamp_ms, 0);
+  EXPECT_FALSE(channel_stats->remote_rtcp->round_trip_time.has_value());
+}
+
 }  // namespace
 }  // namespace webrtc

audio/voip/test/mock_task_queue.h

@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2020 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.
+ */
+
+#ifndef AUDIO_VOIP_TEST_MOCK_TASK_QUEUE_H_
+#define AUDIO_VOIP_TEST_MOCK_TASK_QUEUE_H_
+
+#include <memory>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+// MockTaskQueue enables immediate task run from global TaskQueueBase.
+// It's necessary for some tests depending on TaskQueueBase internally.
+class MockTaskQueue : public TaskQueueBase {
+ public:
+  MockTaskQueue() : current_(this) {}
+
+  // Delete is deliberately defined as no-op as MockTaskQueue is expected to
+  // hold onto current global TaskQueueBase throughout the testing.
+  void Delete() override {}
+
+  MOCK_METHOD(void, PostTask, (std::unique_ptr<QueuedTask>), (override));
+  MOCK_METHOD(void,
+              PostDelayedTask,
+              (std::unique_ptr<QueuedTask>, uint32_t),
+              (override));
+
+ private:
+  CurrentTaskQueueSetter current_;
+};
+
+class MockTaskQueueFactory : public TaskQueueFactory {
+ public:
+  explicit MockTaskQueueFactory(MockTaskQueue* task_queue)
+      : task_queue_(task_queue) {}
+
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
+      absl::string_view name,
+      Priority priority) const override {
+    // Default MockTaskQueue::Delete is no-op, therefore it's safe to pass the
+    // raw pointer.
+    return std::unique_ptr<TaskQueueBase, TaskQueueDeleter>(task_queue_);
+  }
+
+ private:
+  MockTaskQueue* task_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_TEST_MOCK_TASK_QUEUE_H_

audio/voip/voip_core.cc

@@ -458,6 +458,19 @@ VoipResult VoipCore::GetIngressStatistics(ChannelId channel_id,
   return VoipResult::kOk;
 }
 
+VoipResult VoipCore::GetChannelStatistics(ChannelId channel_id,
+                                          ChannelStatistics& channel_stats) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel_stats = channel->GetChannelStatistics();
+
+  return VoipResult::kOk;
+}
+
 VoipResult VoipCore::SetInputMuted(ChannelId channel_id, bool enable) {
   rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
 

audio/voip/voip_core.h

@@ -109,6 +109,8 @@ class VoipCore : public VoipEngine,
   // Implements VoipStatistics interfaces.
   VoipResult GetIngressStatistics(ChannelId channel_id,
                                   IngressStatistics& ingress_stats) override;
+  VoipResult GetChannelStatistics(ChannelId channe_id,
+                                  ChannelStatistics& channel_stats) override;
 
   // Implements VoipVolumeControl interfaces.
   VoipResult SetInputMuted(ChannelId channel_id, bool enable) override;