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}
diff --git a/api/voip/voip_statistics.h b/api/voip/voip_statistics.h
index 08f4cb7..1b9b164 100644
--- a/api/voip/voip_statistics.h
+++ b/api/voip/voip_statistics.h
@@ -26,6 +26,51 @@
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 @@
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;
};
diff --git a/audio/voip/BUILD.gn b/audio/voip/BUILD.gn
index dd5267f..ed0508f 100644
--- a/audio/voip/BUILD.gn
+++ b/audio/voip/BUILD.gn
@@ -67,6 +67,7 @@
"../../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",
diff --git a/audio/voip/audio_channel.cc b/audio/voip/audio_channel.cc
index dc53acf..d11e6d7 100644
--- a/audio/voip/audio_channel.cc
+++ b/audio/voip/audio_channel.cc
@@ -79,6 +79,12 @@
}
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 @@
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
diff --git a/audio/voip/audio_channel.h b/audio/voip/audio_channel.h
index 5bc7483..7b9fa6f 100644
--- a/audio/voip/audio_channel.h
+++ b/audio/voip/audio_channel.h
@@ -84,6 +84,7 @@
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 @@
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_;
diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index 3be4718..8aa552b 100644
--- a/audio/voip/audio_ingress.cc
+++ b/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 @@
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::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();
- if (!rtt.has_value()) {
+ int64_t rtt = 0;
+ if (rtp_rtcp_->RTT(remote_ssrc_, &rtt, nullptr, nullptr, nullptr) != 0) {
// Waiting for valid RTT.
return;
}
@@ -199,38 +226,69 @@
{
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
- // yet, return -1 as to indicate uninitialized value.
- if (report_data.empty()) {
- return absl::nullopt;
+ // Receive only channel won't send any RTP packets.
+ if (!channel_stats.remote_ssrc.has_value()) {
+ channel_stats.remote_ssrc = remote_ssrc_;
+ }
+ break;
}
- // We don't know in advance the remote SSRC used by the other end's receiver
- // 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();
+ return channel_stats;
}
} // namespace webrtc
diff --git a/audio/voip/audio_ingress.h b/audio/voip/audio_ingress.h
index 663b59b..9a36a46 100644
--- a/audio/voip/audio_ingress.h
+++ b/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 @@
return stats;
}
+ ChannelStatistics GetChannelStatistics();
+
// Implementation of AudioMixer::Source interface.
AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sampling_rate,
@@ -102,10 +105,6 @@
}
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.
diff --git a/audio/voip/test/BUILD.gn b/audio/voip/test/BUILD.gn
index ade1076..ab074f7 100644
--- a/audio/voip/test/BUILD.gn
+++ b/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 @@
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",
]
diff --git a/audio/voip/test/audio_channel_unittest.cc b/audio/voip/test/audio_channel_unittest.cc
index 34b595c..1a79d84 100644
--- a/audio/voip/test/audio_channel_unittest.cc
+++ b/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 @@
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 @@
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 @@
// 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 @@
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 @@
// 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 @@
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 @@
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 @@
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 @@
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
diff --git a/audio/voip/test/mock_task_queue.h b/audio/voip/test/mock_task_queue.h
new file mode 100644
index 0000000..c3553a2
--- /dev/null
+++ b/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_
diff --git a/audio/voip/voip_core.cc b/audio/voip/voip_core.cc
index f65352c..33dadbc 100644
--- a/audio/voip/voip_core.cc
+++ b/audio/voip/voip_core.cc
@@ -458,6 +458,19 @@
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);
diff --git a/audio/voip/voip_core.h b/audio/voip/voip_core.h
index 194f8fb..b7c1f29 100644
--- a/audio/voip/voip_core.h
+++ b/audio/voip/voip_core.h
@@ -109,6 +109,8 @@
// 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;
