Revert "ChannelStatistics used for RTP stats in VoipStatistics."
This reverts commit 444e04be6988fbdcc039d775481ac22481ff9ff4.
Reason for revert: breaks 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}
TBR=mbonadei@webrtc.org,saza@webrtc.org,hta@webrtc.org,natim@webrtc.org
Change-Id: I5ce6a698c1216c7d56e32fce3308c16daac852f4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11989
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201460
Reviewed-by: Alex Loiko <aleloi@google.com>
Commit-Queue: Alex Loiko <aleloi@google.com>
Cr-Commit-Position: refs/heads/master@{#32956}
diff --git a/api/voip/voip_statistics.h b/api/voip/voip_statistics.h
index 1b9b164..08f4cb7 100644
--- a/api/voip/voip_statistics.h
+++ b/api/voip/voip_statistics.h
@@ -26,51 +26,6 @@
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 {
@@ -82,13 +37,6 @@
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 ed0508f..dd5267f 100644
--- a/audio/voip/BUILD.gn
+++ b/audio/voip/BUILD.gn
@@ -67,7 +67,6 @@
"../../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 d11e6d7..dc53acf 100644
--- a/audio/voip/audio_channel.cc
+++ b/audio/voip/audio_channel.cc
@@ -79,12 +79,6 @@
}
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());
}
@@ -165,17 +159,4 @@
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 7b9fa6f..5bc7483 100644
--- a/audio/voip/audio_channel.h
+++ b/audio/voip/audio_channel.h
@@ -84,7 +84,6 @@
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
@@ -107,12 +106,6 @@
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 8aa552b..3be4718 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -17,10 +17,6 @@
#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"
@@ -157,12 +153,6 @@
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;
@@ -191,28 +181,11 @@
void AudioIngress::ReceivedRTCPPacket(
rtc::ArrayView<const uint8_t> rtcp_packet) {
- 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.
+ // Deliver RTCP packet to RTP/RTCP module for parsing.
rtp_rtcp_->IncomingRtcpPacket(rtcp_packet.data(), rtcp_packet.size());
- int64_t rtt = 0;
- if (rtp_rtcp_->RTT(remote_ssrc_, &rtt, nullptr, nullptr, nullptr) != 0) {
+ absl::optional<int64_t> rtt = GetRoundTripTime();
+ if (!rtt.has_value()) {
// Waiting for valid RTT.
return;
}
@@ -226,69 +199,38 @@
{
MutexLock lock(&lock_);
- ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
+ ntp_estimator_.UpdateRtcpTimestamp(*rtt, ntp_secs, ntp_frac, rtp_timestamp);
}
}
-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.
+absl::optional<int64_t> AudioIngress::GetRoundTripTime() {
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;
- // Receive only channel won't send any RTP packets.
- if (!channel_stats.remote_ssrc.has_value()) {
- channel_stats.remote_ssrc = remote_ssrc_;
- }
- break;
+ // 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;
}
- return channel_stats;
+ // 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();
}
} // namespace webrtc
diff --git a/audio/voip/audio_ingress.h b/audio/voip/audio_ingress.h
index 9a36a46..663b59b 100644
--- a/audio/voip/audio_ingress.h
+++ b/audio/voip/audio_ingress.h
@@ -22,7 +22,6 @@
#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"
@@ -87,8 +86,6 @@
return stats;
}
- ChannelStatistics GetChannelStatistics();
-
// Implementation of AudioMixer::Source interface.
AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
int sampling_rate,
@@ -105,6 +102,10 @@
}
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 ab074f7..ade1076 100644
--- a/audio/voip/test/BUILD.gn
+++ b/audio/voip/test/BUILD.gn
@@ -9,16 +9,6 @@
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" ]
@@ -40,18 +30,18 @@
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:task_queue",
+ "../../../api/task_queue:default_task_queue_factory",
"../../../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 1a79d84..34b595c 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/task_queue_factory.h"
-#include "audio/voip/test/mock_task_queue.h"
+#include "api/task_queue/default_task_queue_factory.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"
@@ -41,16 +41,11 @@
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 {
@@ -85,7 +80,6 @@
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_;
@@ -98,9 +92,11 @@
// 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));
@@ -109,6 +105,8 @@
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.
@@ -124,8 +122,10 @@
// 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));
@@ -134,14 +134,18 @@
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));
@@ -149,6 +153,7 @@
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);
@@ -177,8 +182,10 @@
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);
@@ -215,59 +222,5 @@
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
deleted file mode 100644
index c3553a2..0000000
--- a/audio/voip/test/mock_task_queue.h
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * 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 33dadbc..f65352c 100644
--- a/audio/voip/voip_core.cc
+++ b/audio/voip/voip_core.cc
@@ -458,19 +458,6 @@
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 b7c1f29..194f8fb 100644
--- a/audio/voip/voip_core.h
+++ b/audio/voip/voip_core.h
@@ -109,8 +109,6 @@
// 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;
