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;