diff --git a/webrtc/api/audio_codecs/audio_encoder.cc b/webrtc/api/audio_codecs/audio_encoder.cc
index 3ee371f1ce..d5be26c5a1 100644
--- a/webrtc/api/audio_codecs/audio_encoder.cc
+++ b/webrtc/api/audio_codecs/audio_encoder.cc
@@ -15,6 +15,10 @@
 
 namespace webrtc {
 
+ANAStats::ANAStats() = default;
+ANAStats::~ANAStats() = default;
+ANAStats::ANAStats(const ANAStats&) = default;
+
 AudioEncoder::EncodedInfo::EncodedInfo() = default;
 AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
 AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
@@ -95,4 +99,8 @@ void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
 void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
                                                int max_frame_length_ms) {}
 
+ANAStats AudioEncoder::GetANAStats() const {
+  return ANAStats();
+}
+
 }  // namespace webrtc
diff --git a/webrtc/api/audio_codecs/audio_encoder.h b/webrtc/api/audio_codecs/audio_encoder.h
index 7406d7d653..a77894b7da 100644
--- a/webrtc/api/audio_codecs/audio_encoder.h
+++ b/webrtc/api/audio_codecs/audio_encoder.h
@@ -27,6 +27,33 @@ namespace webrtc {
 class Clock;
 class RtcEventLog;
 
+// Statistics related to Audio Network Adaptation.
+struct ANAStats {
+  ANAStats();
+  ANAStats(const ANAStats&);
+  ~ANAStats();
+  // Number of actions taken by the ANA bitrate controller since the start of
+  // the call. If this value is not set, it indicates that the bitrate
+  // controller is disabled.
+  rtc::Optional<uint32_t> bitrate_action_counter;
+  // Number of actions taken by the ANA channel controller since the start of
+  // the call. If this value is not set, it indicates that the channel
+  // controller is disabled.
+  rtc::Optional<uint32_t> channel_action_counter;
+  // Number of actions taken by the ANA DTX controller since the start of the
+  // call. If this value is not set, it indicates that the DTX controller is
+  // disabled.
+  rtc::Optional<uint32_t> dtx_action_counter;
+  // Number of actions taken by the ANA FEC controller since the start of the
+  // call. If this value is not set, it indicates that the FEC controller is
+  // disabled.
+  rtc::Optional<uint32_t> fec_action_counter;
+  // Number of actions taken by the ANA frame length controller since the start
+  // of the call. If this value is not set, it indicates that the frame length
+  // controller is disabled.
+  rtc::Optional<uint32_t> frame_length_action_counter;
+};
+
 // This is the interface class for encoders in AudioCoding module. Each codec
 // type must have an implementation of this class.
 class AudioEncoder {
@@ -203,6 +230,9 @@ class AudioEncoder {
   virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                            int max_frame_length_ms);
 
+  // Get statistics related to audio network adaptation.
+  virtual ANAStats GetANAStats() const;
+
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
diff --git a/webrtc/api/statstypes.cc b/webrtc/api/statstypes.cc
index 714d36b54b..67bf49e336 100644
--- a/webrtc/api/statstypes.cc
+++ b/webrtc/api/statstypes.cc
@@ -586,6 +586,16 @@ const char* StatsReport::Value::display_name() const {
       return "googResidualEchoLikelihood";
     case kStatsValueNameResidualEchoLikelihoodRecentMax:
       return "googResidualEchoLikelihoodRecentMax";
+    case kStatsValueNameAnaBitrateActionCounter:
+      return "googAnaBitrateActionCounter";
+    case kStatsValueNameAnaChannelActionCounter:
+      return "googAnaChannelActionCounter";
+    case kStatsValueNameAnaDtxActionCounter:
+      return "googAnaDtxActionCounter";
+    case kStatsValueNameAnaFecActionCounter:
+      return "googAnaFecActionCounter";
+    case kStatsValueNameAnaFrameLengthActionCounter:
+      return "googAnaFrameLengthActionCounter";
     case kStatsValueNameRetransmitBitrate:
       return "googRetransmitBitrate";
     case kStatsValueNameRtt:
diff --git a/webrtc/api/statstypes.h b/webrtc/api/statstypes.h
index 42e3601209..219995ed83 100644
--- a/webrtc/api/statstypes.h
+++ b/webrtc/api/statstypes.h
@@ -208,6 +208,11 @@ class StatsReport {
     kStatsValueNameRenderDelayMs,
     kStatsValueNameResidualEchoLikelihood,
     kStatsValueNameResidualEchoLikelihoodRecentMax,
+    kStatsValueNameAnaBitrateActionCounter,
+    kStatsValueNameAnaChannelActionCounter,
+    kStatsValueNameAnaDtxActionCounter,
+    kStatsValueNameAnaFecActionCounter,
+    kStatsValueNameAnaFrameLengthActionCounter,
     kStatsValueNameRetransmitBitrate,
     kStatsValueNameRtt,
     kStatsValueNameSecondaryDecodedRate,
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 5bf6f2b807..cb73e84a23 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -333,6 +333,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   stats.typing_noise_detected = audio_state->typing_noise_detected();
+  stats.ana_statistics = channel_proxy_->GetANAStatistics();
 
   return stats;
 }
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 1308657d0e..4c29f6ac63 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -301,6 +301,8 @@ struct ConfigHelper {
         .WillRepeatedly(Return(kCallStats));
     EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
         .WillRepeatedly(Return(report_blocks));
+    EXPECT_CALL(*channel_proxy_, GetANAStatistics())
+        .WillRepeatedly(Return(ANAStats()));
     EXPECT_CALL(voice_engine_, transmit_mixer())
         .WillRepeatedly(Return(&transmit_mixer_));
 
diff --git a/webrtc/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h
index 4f7f8c0f84..ac962c5ada 100644
--- a/webrtc/call/audio_send_stream.h
+++ b/webrtc/call/audio_send_stream.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_encoder_factory.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/call/transport.h"
@@ -61,6 +62,7 @@ class AudioSendStream {
     float residual_echo_likelihood = -1.0f;
     float residual_echo_likelihood_recent_max = -1.0f;
     bool typing_noise_detected = false;
+    ANAStats ana_statistics;
   };
 
   struct Config {
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index cea85e6c3d..7e7b815821 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/optional.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/api/rtpreceiverinterface.h"
@@ -641,6 +642,7 @@ struct VoiceSenderInfo : public MediaSenderInfo {
   float residual_echo_likelihood;
   float residual_echo_likelihood_recent_max;
   bool typing_noise_detected;
+  webrtc::ANAStats ana_statistics;
 };
 
 struct VoiceReceiverInfo : public MediaReceiverInfo {
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 8c45e953e4..d202106be0 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -2260,6 +2260,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     sinfo.residual_echo_likelihood_recent_max =
         stats.residual_echo_likelihood_recent_max;
     sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false);
+    sinfo.ana_statistics = stats.ana_statistics;
     info->senders.push_back(sinfo);
   }
 
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 5ccd7d6bb6..67fde12194 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -546,6 +546,12 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     stats.echo_return_loss_enhancement = 1234;
     stats.residual_echo_likelihood = 0.432f;
     stats.residual_echo_likelihood_recent_max = 0.6f;
+    stats.ana_statistics.bitrate_action_counter = rtc::Optional<uint32_t>(321);
+    stats.ana_statistics.channel_action_counter = rtc::Optional<uint32_t>(432);
+    stats.ana_statistics.dtx_action_counter = rtc::Optional<uint32_t>(543);
+    stats.ana_statistics.fec_action_counter = rtc::Optional<uint32_t>(654);
+    stats.ana_statistics.frame_length_action_counter =
+        rtc::Optional<uint32_t>(765);
     stats.typing_noise_detected = true;
     return stats;
   }
@@ -577,6 +583,16 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     EXPECT_EQ(info.residual_echo_likelihood, stats.residual_echo_likelihood);
     EXPECT_EQ(info.residual_echo_likelihood_recent_max,
               stats.residual_echo_likelihood_recent_max);
+    EXPECT_EQ(info.ana_statistics.bitrate_action_counter,
+              stats.ana_statistics.bitrate_action_counter);
+    EXPECT_EQ(info.ana_statistics.channel_action_counter,
+              stats.ana_statistics.channel_action_counter);
+    EXPECT_EQ(info.ana_statistics.dtx_action_counter,
+              stats.ana_statistics.dtx_action_counter);
+    EXPECT_EQ(info.ana_statistics.fec_action_counter,
+              stats.ana_statistics.fec_action_counter);
+    EXPECT_EQ(info.ana_statistics.frame_length_action_counter,
+              stats.ana_statistics.frame_length_action_counter);
     EXPECT_EQ(info.typing_noise_detected,
               stats.typing_noise_detected && is_sending);
   }
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 32d5336ce9..898723b8ca 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -913,6 +913,7 @@ rtc_static_library("audio_network_adaptor") {
   deps = [
     "../..:webrtc_common",
     "../../api:optional",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../common_audio",
     "../../logging:rtc_event_log_api",
     "../../rtc_base:protobuf_utils",
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index f7607c6521..0244882de6 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -204,6 +204,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
 
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override;
 
+  ANAStats GetANAStats() const override;
+
  private:
   struct InputData {
     uint32_t input_timestamp;
@@ -1273,6 +1275,14 @@ void AudioCodingModuleImpl::GetDecodingCallStatistics(
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
+ANAStats AudioCodingModuleImpl::GetANAStats() const {
+  rtc::CritScope lock(&acm_crit_sect_);
+  if (encoder_stack_)
+    return encoder_stack_->GetANAStats();
+  // If no encoder is set, return default stats.
+  return ANAStats();
+}
+
 }  // namespace
 
 AudioCodingModule::Config::Config()
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index 7645c4a1f2..5eff4b3571 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -135,6 +135,12 @@ void AudioNetworkAdaptorImpl::StopDebugDump() {
   debug_dump_writer_.reset(nullptr);
 }
 
+ANAStats AudioNetworkAdaptorImpl::GetStats() const {
+  // TODO(ivoc): Actually implement the stat.
+  // Tracking bug: https://crbug.com/webrtc/8127
+  return ANAStats();
+}
+
 void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {
   if (debug_dump_writer_)
     debug_dump_writer_->DumpNetworkMetrics(last_metrics_, rtc::TimeMillis());
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index 7d7a2cb086..bd2a250f66 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -58,6 +58,8 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
 
   void StopDebugDump() override;
 
+  ANAStats GetStats() const override;
+
  private:
   void DumpNetworkMetrics();
 
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index 1914939566..c29b1e8c17 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -270,4 +270,36 @@ TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) {
   states.audio_network_adaptor->GetEncoderRuntimeConfig();
 }
 
+TEST(AudioNetworkAdaptorImplTest, TestANAStats) {
+  auto states = CreateAudioNetworkAdaptor();
+
+  // Simulate some adaptation, otherwise the stats will not show anything.
+  AudioEncoderRuntimeConfig config1, config2;
+  config1.bitrate_bps = rtc::Optional<int>(32000);
+  config1.enable_fec = rtc::Optional<bool>(true);
+  config2.bitrate_bps = rtc::Optional<int>(16000);
+  config2.enable_fec = rtc::Optional<bool>(false);
+
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config1));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config2));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+
+  auto ana_stats = states.audio_network_adaptor->GetStats();
+
+  // Check that the default stats are returned, as these have not been
+  // implemented yet). Tracking bug: https://crbug.com/8127
+  auto default_stats = ANAStats();
+  EXPECT_EQ(ana_stats.bitrate_action_counter,
+            default_stats.bitrate_action_counter);
+  EXPECT_EQ(ana_stats.channel_action_counter,
+            default_stats.channel_action_counter);
+  EXPECT_EQ(ana_stats.dtx_action_counter, default_stats.dtx_action_counter);
+  EXPECT_EQ(ana_stats.fec_action_counter, default_stats.fec_action_counter);
+  EXPECT_EQ(ana_stats.frame_length_action_counter,
+            default_stats.frame_length_action_counter);
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
index fa742e69f6..0097d70bd8 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
 
+#include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/optional.h"
 
 namespace webrtc {
@@ -38,7 +39,6 @@ struct AudioEncoderRuntimeConfig {
 // encoder based on network metrics.
 class AudioNetworkAdaptor {
  public:
-
   virtual ~AudioNetworkAdaptor() = default;
 
   virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
@@ -60,6 +60,8 @@ class AudioNetworkAdaptor {
   virtual void StartDebugDump(FILE* file_handle) = 0;
 
   virtual void StopDebugDump() = 0;
+
+  virtual ANAStats GetStats() const = 0;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
index 4b9a4772a1..f58a48285b 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
@@ -40,6 +40,8 @@ class MockAudioNetworkAdaptor : public AudioNetworkAdaptor {
   MOCK_METHOD1(StartDebugDump, void(FILE* file_handle));
 
   MOCK_METHOD0(StopDebugDump, void());
+
+  MOCK_CONST_METHOD0(GetStats, ANAStats());
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index cd47bf4265..3b4473a921 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -773,4 +773,11 @@ void AudioEncoderOpus::MaybeUpdateUplinkBandwidth() {
   }
 }
 
+ANAStats AudioEncoderOpus::GetANAStats() const {
+  if (audio_network_adaptor_) {
+    return audio_network_adaptor_->GetStats();
+  }
+  return ANAStats();
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 588f268e7f..f966dbe74d 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -116,6 +116,7 @@ class AudioEncoderOpus final : public AudioEncoder {
   void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
   void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                    int max_frame_length_ms) override;
+  ANAStats GetANAStats() const override;
   rtc::ArrayView<const int> supported_frame_lengths_ms() const {
     return config_.supported_frame_lengths_ms;
   }
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index 4ceade25d0..eb59404dd1 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -823,6 +823,8 @@ class AudioCodingModule {
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
+
+  virtual ANAStats GetANAStats() const = 0;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/pc/statscollector.cc b/webrtc/pc/statscollector.cc
index 08cf464568..b96428353a 100644
--- a/webrtc/pc/statscollector.cc
+++ b/webrtc/pc/statscollector.cc
@@ -224,6 +224,26 @@ void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
     }
   }
   report->AddString(StatsReport::kStatsValueNameMediaType, "audio");
+  if (info.ana_statistics.bitrate_action_counter) {
+    report->AddInt(StatsReport::kStatsValueNameAnaBitrateActionCounter,
+                   *info.ana_statistics.bitrate_action_counter);
+  }
+  if (info.ana_statistics.channel_action_counter) {
+    report->AddInt(StatsReport::kStatsValueNameAnaChannelActionCounter,
+                   *info.ana_statistics.channel_action_counter);
+  }
+  if (info.ana_statistics.dtx_action_counter) {
+    report->AddInt(StatsReport::kStatsValueNameAnaDtxActionCounter,
+                   *info.ana_statistics.dtx_action_counter);
+  }
+  if (info.ana_statistics.fec_action_counter) {
+    report->AddInt(StatsReport::kStatsValueNameAnaFecActionCounter,
+                   *info.ana_statistics.fec_action_counter);
+  }
+  if (info.ana_statistics.frame_length_action_counter) {
+    report->AddInt(StatsReport::kStatsValueNameAnaFrameLengthActionCounter,
+                   *info.ana_statistics.frame_length_action_counter);
+  }
 }
 
 void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {
diff --git a/webrtc/pc/statscollector_unittest.cc b/webrtc/pc/statscollector_unittest.cc
index 56d4fca14d..57346760e7 100644
--- a/webrtc/pc/statscollector_unittest.cc
+++ b/webrtc/pc/statscollector_unittest.cc
@@ -430,6 +430,37 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report,
       report, StatsReport::kStatsValueNameTypingNoiseState, &value_in_report));
   std::string typing_detected = sinfo.typing_noise_detected ? "true" : "false";
   EXPECT_EQ(typing_detected, value_in_report);
+  EXPECT_TRUE(GetValue(report,
+                       StatsReport::kStatsValueNameAnaBitrateActionCounter,
+                       &value_in_report));
+  ASSERT_TRUE(sinfo.ana_statistics.bitrate_action_counter);
+  EXPECT_EQ(
+      rtc::ToString<uint32_t>(*sinfo.ana_statistics.bitrate_action_counter),
+      value_in_report);
+  EXPECT_TRUE(GetValue(report,
+                       StatsReport::kStatsValueNameAnaChannelActionCounter,
+                       &value_in_report));
+  ASSERT_TRUE(sinfo.ana_statistics.channel_action_counter);
+  EXPECT_EQ(
+      rtc::ToString<uint32_t>(*sinfo.ana_statistics.channel_action_counter),
+      value_in_report);
+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAnaDtxActionCounter,
+                       &value_in_report));
+  ASSERT_TRUE(sinfo.ana_statistics.dtx_action_counter);
+  EXPECT_EQ(rtc::ToString<uint32_t>(*sinfo.ana_statistics.dtx_action_counter),
+            value_in_report);
+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAnaFecActionCounter,
+                       &value_in_report));
+  ASSERT_TRUE(sinfo.ana_statistics.fec_action_counter);
+  EXPECT_EQ(rtc::ToString<uint32_t>(*sinfo.ana_statistics.fec_action_counter),
+            value_in_report);
+  EXPECT_TRUE(GetValue(report,
+                       StatsReport::kStatsValueNameAnaFrameLengthActionCounter,
+                       &value_in_report));
+  ASSERT_TRUE(sinfo.ana_statistics.frame_length_action_counter);
+  EXPECT_EQ(rtc::ToString<uint32_t>(
+                *sinfo.ana_statistics.frame_length_action_counter),
+            value_in_report);
 }
 
 // Helper methods to avoid duplication of code.
@@ -450,6 +481,16 @@ void InitVoiceSenderInfo(cricket::VoiceSenderInfo* voice_sender_info) {
   voice_sender_info->echo_delay_std_ms = 111;
   voice_sender_info->aec_quality_min = 112.0f;
   voice_sender_info->typing_noise_detected = false;
+  voice_sender_info->ana_statistics.bitrate_action_counter =
+      rtc::Optional<uint32_t>(113);
+  voice_sender_info->ana_statistics.channel_action_counter =
+      rtc::Optional<uint32_t>(114);
+  voice_sender_info->ana_statistics.dtx_action_counter =
+      rtc::Optional<uint32_t>(115);
+  voice_sender_info->ana_statistics.fec_action_counter =
+      rtc::Optional<uint32_t>(116);
+  voice_sender_info->ana_statistics.frame_length_action_counter =
+      rtc::Optional<uint32_t>(117);
 }
 
 void UpdateVoiceSenderInfoFromAudioTrack(
@@ -1988,6 +2029,7 @@ TEST_F(StatsCollectorTest, TwoLocalTracksWithSameSsrc) {
   stats.AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
 
   cricket::VoiceSenderInfo voice_sender_info;
+  InitVoiceSenderInfo(&voice_sender_info);
   voice_sender_info.add_ssrc(kSsrcOfTrack);
 
   cricket::VoiceMediaInfo stats_read;
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index 6aa60857d1..b151aa8e0e 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -52,6 +52,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
   MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
   MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
+  MOCK_CONST_METHOD0(GetANAStatistics, ANAStats());
   MOCK_CONST_METHOD0(GetSpeechOutputLevel, int());
   MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int());
   MOCK_CONST_METHOD0(GetTotalOutputEnergy, double());
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 1a56e81181..0e83989ccf 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -2856,6 +2856,10 @@ void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
   audio_coding_->GetDecodingCallStatistics(stats);
 }
 
+ANAStats Channel::GetANAStatistics() const {
+  return audio_coding_->GetANAStats();
+}
+
 uint32_t Channel::GetDelayEstimate() const {
   rtc::CritScope lock(&video_sync_lock_);
   return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_;
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 0d71a41d82..89c2b430ec 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -268,6 +268,7 @@ class Channel
   // Stats.
   int GetNetworkStatistics(NetworkStatistics& stats);
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
+  ANAStats GetANAStatistics() const;
 
   // Audio+Video Sync.
   uint32_t GetDelayEstimate() const;
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index cdd4b6624f..5f4f98a78f 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -144,6 +144,11 @@ AudioDecodingCallStats ChannelProxy::GetDecodingCallStatistics() const {
   return stats;
 }
 
+ANAStats ChannelProxy::GetANAStatistics() const {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  return channel()->GetANAStatistics();
+}
+
 int ChannelProxy::GetSpeechOutputLevel() const {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return channel()->GetSpeechOutputLevel();
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 5aa2839feb..826117b9aa 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -81,6 +81,7 @@ class ChannelProxy : public RtpPacketSinkInterface {
   virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
   virtual NetworkStatistics GetNetworkStatistics() const;
   virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
+  virtual ANAStats GetANAStatistics() const;
   virtual int GetSpeechOutputLevel() const;
   virtual int GetSpeechOutputLevelFullRange() const;
   // See description of "totalAudioEnergy" in the WebRTC stats spec: