Merge branch 'upstream-master'

Bug: 153469641
Test: none
Change-Id: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802

modules/audio_coding/DEPS

@@ -0,0 +1,7 @@
+include_rules = [
+  "+call",
+  "+common_audio",
+  "+logging/rtc_event_log",
+  "+audio_coding/neteq/neteq_unittest.pb.h",  # Different path.
+  "+system_wrappers",
+]

modules/audio_coding/OWNERS.webrtc

@@ -0,0 +1,4 @@
+henrik.lundin@webrtc.org
+kwiberg@webrtc.org
+minyue@webrtc.org
+ivoc@webrtc.org

modules/audio_coding/acm2/acm_receive_test.cc

@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#include "modules/audio_coding/acm2/acm_receive_test.h"
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+AudioCodingModule::Config MakeAcmConfig(
+    Clock* clock,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
+  AudioCodingModule::Config config;
+  config.clock = clock;
+  config.decoder_factory = std::move(decoder_factory);
+  return config;
+}
+}  // namespace
+
+AcmReceiveTestOldApi::AcmReceiveTestOldApi(
+    PacketSource* packet_source,
+    AudioSink* audio_sink,
+    int output_freq_hz,
+    NumOutputChannels exptected_output_channels,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
+    : clock_(0),
+      acm_(webrtc::AudioCodingModule::Create(
+          MakeAcmConfig(&clock_, std::move(decoder_factory)))),
+      packet_source_(packet_source),
+      audio_sink_(audio_sink),
+      output_freq_hz_(output_freq_hz),
+      exptected_output_channels_(exptected_output_channels) {}
+
+AcmReceiveTestOldApi::~AcmReceiveTestOldApi() = default;
+
+void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
+  acm_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
+                          {104, {"ISAC", 32000, 1}},
+                          {107, {"L16", 8000, 1}},
+                          {108, {"L16", 16000, 1}},
+                          {109, {"L16", 32000, 1}},
+                          {111, {"L16", 8000, 2}},
+                          {112, {"L16", 16000, 2}},
+                          {113, {"L16", 32000, 2}},
+                          {0, {"PCMU", 8000, 1}},
+                          {110, {"PCMU", 8000, 2}},
+                          {8, {"PCMA", 8000, 1}},
+                          {118, {"PCMA", 8000, 2}},
+                          {102, {"ILBC", 8000, 1}},
+                          {9, {"G722", 8000, 1}},
+                          {119, {"G722", 8000, 2}},
+                          {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+                          {13, {"CN", 8000, 1}},
+                          {98, {"CN", 16000, 1}},
+                          {99, {"CN", 32000, 1}}});
+}
+
+// Remaps payload types from ACM's default to those used in the resource file
+// neteq_universal_new.rtp.
+void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
+  acm_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
+                          {104, {"ISAC", 32000, 1}},
+                          {93, {"L16", 8000, 1}},
+                          {94, {"L16", 16000, 1}},
+                          {95, {"L16", 32000, 1}},
+                          {0, {"PCMU", 8000, 1}},
+                          {8, {"PCMA", 8000, 1}},
+                          {102, {"ILBC", 8000, 1}},
+                          {9, {"G722", 8000, 1}},
+                          {120, {"OPUS", 48000, 2}},
+                          {13, {"CN", 8000, 1}},
+                          {98, {"CN", 16000, 1}},
+                          {99, {"CN", 32000, 1}}});
+}
+
+void AcmReceiveTestOldApi::Run() {
+  for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
+       packet = packet_source_->NextPacket()) {
+    // Pull audio until time to insert packet.
+    while (clock_.TimeInMilliseconds() < packet->time_ms()) {
+      AudioFrame output_frame;
+      bool muted;
+      EXPECT_EQ(0,
+                acm_->PlayoutData10Ms(output_freq_hz_, &output_frame, &muted));
+      ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
+      ASSERT_FALSE(muted);
+      const size_t samples_per_block =
+          static_cast<size_t>(output_freq_hz_ * 10 / 1000);
+      EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
+      if (exptected_output_channels_ != kArbitraryChannels) {
+        if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
+          // Don't check number of channels for PLC output, since each test run
+          // usually starts with a short period of mono PLC before decoding the
+          // first packet.
+        } else {
+          EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
+        }
+      }
+      ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
+      clock_.AdvanceTimeMilliseconds(10);
+      AfterGetAudio();
+    }
+
+    EXPECT_EQ(0, acm_->IncomingPacket(
+                     packet->payload(),
+                     static_cast<int32_t>(packet->payload_length_bytes()),
+                     packet->header()))
+        << "Failure when inserting packet:" << std::endl
+        << "  PT = " << static_cast<int>(packet->header().payloadType)
+        << std::endl
+        << "  TS = " << packet->header().timestamp << std::endl
+        << "  SN = " << packet->header().sequenceNumber;
+  }
+}
+
+AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
+    PacketSource* packet_source,
+    AudioSink* audio_sink,
+    int output_freq_hz_1,
+    int output_freq_hz_2,
+    int toggle_period_ms,
+    NumOutputChannels exptected_output_channels)
+    : AcmReceiveTestOldApi(packet_source,
+                           audio_sink,
+                           output_freq_hz_1,
+                           exptected_output_channels,
+                           CreateBuiltinAudioDecoderFactory()),
+      output_freq_hz_1_(output_freq_hz_1),
+      output_freq_hz_2_(output_freq_hz_2),
+      toggle_period_ms_(toggle_period_ms),
+      last_toggle_time_ms_(clock_.TimeInMilliseconds()) {}
+
+void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
+  if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
+    output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
+                          ? output_freq_hz_2_
+                          : output_freq_hz_1_;
+    last_toggle_time_ms_ = clock_.TimeInMilliseconds();
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc

modules/audio_coding/acm2/acm_receive_test.h

@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2014 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 MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
+
+#include <stddef.h>  // for size_t
+
+#include <memory>
+#include <string>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/scoped_refptr.h"
+#include "rtc_base/constructor_magic.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+class AudioCodingModule;
+class AudioDecoder;
+
+namespace test {
+class AudioSink;
+class PacketSource;
+
+class AcmReceiveTestOldApi {
+ public:
+  enum NumOutputChannels : size_t {
+    kArbitraryChannels = 0,
+    kMonoOutput = 1,
+    kStereoOutput = 2,
+    kQuadOutput = 4
+  };
+
+  AcmReceiveTestOldApi(PacketSource* packet_source,
+                       AudioSink* audio_sink,
+                       int output_freq_hz,
+                       NumOutputChannels exptected_output_channels,
+                       rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
+  virtual ~AcmReceiveTestOldApi();
+
+  // Registers the codecs with default parameters from ACM.
+  void RegisterDefaultCodecs();
+
+  // Registers codecs with payload types matching the pre-encoded NetEq test
+  // files.
+  void RegisterNetEqTestCodecs();
+
+  // Runs the test and returns true if successful.
+  void Run();
+
+  AudioCodingModule* get_acm() { return acm_.get(); }
+
+ protected:
+  // Method is called after each block of output audio is received from ACM.
+  virtual void AfterGetAudio() {}
+
+  SimulatedClock clock_;
+  std::unique_ptr<AudioCodingModule> acm_;
+  PacketSource* packet_source_;
+  AudioSink* audio_sink_;
+  int output_freq_hz_;
+  NumOutputChannels exptected_output_channels_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
+};
+
+// This test toggles the output frequency every |toggle_period_ms|. The test
+// starts with |output_freq_hz_1|. Except for the toggling, it does the same
+// thing as AcmReceiveTestOldApi.
+class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
+ public:
+  AcmReceiveTestToggleOutputFreqOldApi(
+      PacketSource* packet_source,
+      AudioSink* audio_sink,
+      int output_freq_hz_1,
+      int output_freq_hz_2,
+      int toggle_period_ms,
+      NumOutputChannels exptected_output_channels);
+
+ protected:
+  void AfterGetAudio() override;
+
+  const int output_freq_hz_1_;
+  const int output_freq_hz_2_;
+  const int toggle_period_ms_;
+  int64_t last_toggle_time_ms_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_

modules/audio_coding/acm2/acm_receiver.cc

@@ -0,0 +1,336 @@
+/*
+ *  Copyright (c) 2013 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.
+ */
+
+#include "modules/audio_coding/acm2/acm_receiver.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/acm2/call_statistics.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/audio_format_to_string.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+namespace {
+
+std::unique_ptr<NetEq> CreateNetEq(
+    NetEqFactory* neteq_factory,
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+  if (neteq_factory) {
+    return neteq_factory->CreateNetEq(config, decoder_factory, clock);
+  }
+  return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
+}
+
+}  // namespace
+
+AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
+    : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
+      neteq_(CreateNetEq(config.neteq_factory,
+                         config.neteq_config,
+                         config.clock,
+                         config.decoder_factory)),
+      clock_(config.clock),
+      resampled_last_output_frame_(true) {
+  RTC_DCHECK(clock_);
+  memset(last_audio_buffer_.get(), 0,
+         sizeof(int16_t) * AudioFrame::kMaxDataSizeSamples);
+}
+
+AcmReceiver::~AcmReceiver() = default;
+
+int AcmReceiver::SetMinimumDelay(int delay_ms) {
+  if (neteq_->SetMinimumDelay(delay_ms))
+    return 0;
+  RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
+  return -1;
+}
+
+int AcmReceiver::SetMaximumDelay(int delay_ms) {
+  if (neteq_->SetMaximumDelay(delay_ms))
+    return 0;
+  RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
+  return -1;
+}
+
+bool AcmReceiver::SetBaseMinimumDelayMs(int delay_ms) {
+  return neteq_->SetBaseMinimumDelayMs(delay_ms);
+}
+
+int AcmReceiver::GetBaseMinimumDelayMs() const {
+  return neteq_->GetBaseMinimumDelayMs();
+}
+
+absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
+  rtc::CritScope lock(&crit_sect_);
+  if (!last_decoder_) {
+    return absl::nullopt;
+  }
+  return last_decoder_->sample_rate_hz;
+}
+
+int AcmReceiver::last_output_sample_rate_hz() const {
+  return neteq_->last_output_sample_rate_hz();
+}
+
+int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
+                              rtc::ArrayView<const uint8_t> incoming_payload) {
+  if (incoming_payload.empty()) {
+    neteq_->InsertEmptyPacket(rtp_header);
+    return 0;
+  }
+
+  int payload_type = rtp_header.payloadType;
+  auto format = neteq_->GetDecoderFormat(payload_type);
+  if (format && absl::EqualsIgnoreCase(format->sdp_format.name, "red")) {
+    // This is a RED packet. Get the format of the audio codec.
+    payload_type = incoming_payload[0] & 0x7f;
+    format = neteq_->GetDecoderFormat(payload_type);
+  }
+  if (!format) {
+    RTC_LOG_F(LS_ERROR) << "Payload-type " << payload_type
+                        << " is not registered.";
+    return -1;
+  }
+
+  {
+    rtc::CritScope lock(&crit_sect_);
+    if (absl::EqualsIgnoreCase(format->sdp_format.name, "cn")) {
+      if (last_decoder_ && last_decoder_->num_channels > 1) {
+        // This is a CNG and the audio codec is not mono, so skip pushing in
+        // packets into NetEq.
+        return 0;
+      }
+    } else {
+      last_decoder_ = DecoderInfo{/*payload_type=*/payload_type,
+                                  /*sample_rate_hz=*/format->sample_rate_hz,
+                                  /*num_channels=*/format->num_channels,
+                                  /*sdp_format=*/std::move(format->sdp_format)};
+    }
+  }  // |crit_sect_| is released.
+
+  if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) {
+    RTC_LOG(LERROR) << "AcmReceiver::InsertPacket "
+                    << static_cast<int>(rtp_header.payloadType)
+                    << " Failed to insert packet";
+    return -1;
+  }
+  return 0;
+}
+
+int AcmReceiver::GetAudio(int desired_freq_hz,
+                          AudioFrame* audio_frame,
+                          bool* muted) {
+  RTC_DCHECK(muted);
+  // Accessing members, take the lock.
+  rtc::CritScope lock(&crit_sect_);
+
+  if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
+    RTC_LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
+    return -1;
+  }
+
+  const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
+
+  // Update if resampling is required.
+  const bool need_resampling =
+      (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
+
+  if (need_resampling && !resampled_last_output_frame_) {
+    // Prime the resampler with the last frame.
+    int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+        audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
+        temp_output);
+    if (samples_per_channel_int < 0) {
+      RTC_LOG(LERROR) << "AcmReceiver::GetAudio - "
+                         "Resampling last_audio_buffer_ failed.";
+      return -1;
+    }
+  }
+
+  // TODO(henrik.lundin) Glitches in the output may appear if the output rate
+  // from NetEq changes. See WebRTC issue 3923.
+  if (need_resampling) {
+    // TODO(yujo): handle this more efficiently for muted frames.
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        audio_frame->data(), current_sample_rate_hz, desired_freq_hz,
+        audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
+        audio_frame->mutable_data());
+    if (samples_per_channel_int < 0) {
+      RTC_LOG(LERROR)
+          << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
+      return -1;
+    }
+    audio_frame->samples_per_channel_ =
+        static_cast<size_t>(samples_per_channel_int);
+    audio_frame->sample_rate_hz_ = desired_freq_hz;
+    RTC_DCHECK_EQ(
+        audio_frame->sample_rate_hz_,
+        rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
+    resampled_last_output_frame_ = true;
+  } else {
+    resampled_last_output_frame_ = false;
+    // We might end up here ONLY if codec is changed.
+  }
+
+  // Store current audio in |last_audio_buffer_| for next time.
+  memcpy(last_audio_buffer_.get(), audio_frame->data(),
+         sizeof(int16_t) * audio_frame->samples_per_channel_ *
+             audio_frame->num_channels_);
+
+  call_stats_.DecodedByNetEq(audio_frame->speech_type_, *muted);
+  return 0;
+}
+
+void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+  neteq_->SetCodecs(codecs);
+}
+
+void AcmReceiver::FlushBuffers() {
+  neteq_->FlushBuffers();
+}
+
+void AcmReceiver::RemoveAllCodecs() {
+  rtc::CritScope lock(&crit_sect_);
+  neteq_->RemoveAllPayloadTypes();
+  last_decoder_ = absl::nullopt;
+}
+
+absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
+  return neteq_->GetPlayoutTimestamp();
+}
+
+int AcmReceiver::FilteredCurrentDelayMs() const {
+  return neteq_->FilteredCurrentDelayMs();
+}
+
+int AcmReceiver::TargetDelayMs() const {
+  return neteq_->TargetDelayMs();
+}
+
+absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
+    const {
+  rtc::CritScope lock(&crit_sect_);
+  if (!last_decoder_) {
+    return absl::nullopt;
+  }
+  RTC_DCHECK_NE(-1, last_decoder_->payload_type);
+  return std::make_pair(last_decoder_->payload_type, last_decoder_->sdp_format);
+}
+
+void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) const {
+  NetEqNetworkStatistics neteq_stat;
+  // NetEq function always returns zero, so we don't check the return value.
+  neteq_->NetworkStatistics(&neteq_stat);
+
+  acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
+  acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
+  acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
+  acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
+  acm_stat->currentExpandRate = neteq_stat.expand_rate;
+  acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
+  acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
+  acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
+  acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
+  acm_stat->currentSecondaryDiscardedRate = neteq_stat.secondary_discarded_rate;
+  acm_stat->addedSamples = neteq_stat.added_zero_samples;
+  acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
+  acm_stat->medianWaitingTimeMs = neteq_stat.median_waiting_time_ms;
+  acm_stat->minWaitingTimeMs = neteq_stat.min_waiting_time_ms;
+  acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
+
+  NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
+  acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
+  acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
+  acm_stat->silentConcealedSamples =
+      neteq_lifetime_stat.silent_concealed_samples;
+  acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
+  acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
+  acm_stat->jitterBufferTargetDelayMs =
+      neteq_lifetime_stat.jitter_buffer_target_delay_ms;
+  acm_stat->jitterBufferEmittedCount =
+      neteq_lifetime_stat.jitter_buffer_emitted_count;
+  acm_stat->delayedPacketOutageSamples =
+      neteq_lifetime_stat.delayed_packet_outage_samples;
+  acm_stat->relativePacketArrivalDelayMs =
+      neteq_lifetime_stat.relative_packet_arrival_delay_ms;
+  acm_stat->interruptionCount = neteq_lifetime_stat.interruption_count;
+  acm_stat->totalInterruptionDurationMs =
+      neteq_lifetime_stat.total_interruption_duration_ms;
+  acm_stat->insertedSamplesForDeceleration =
+      neteq_lifetime_stat.inserted_samples_for_deceleration;
+  acm_stat->removedSamplesForAcceleration =
+      neteq_lifetime_stat.removed_samples_for_acceleration;
+  acm_stat->fecPacketsReceived = neteq_lifetime_stat.fec_packets_received;
+  acm_stat->fecPacketsDiscarded = neteq_lifetime_stat.fec_packets_discarded;
+
+  NetEqOperationsAndState neteq_operations_and_state =
+      neteq_->GetOperationsAndState();
+  acm_stat->packetBufferFlushes =
+      neteq_operations_and_state.packet_buffer_flushes;
+}
+
+int AcmReceiver::EnableNack(size_t max_nack_list_size) {
+  neteq_->EnableNack(max_nack_list_size);
+  return 0;
+}
+
+void AcmReceiver::DisableNack() {
+  neteq_->DisableNack();
+}
+
+std::vector<uint16_t> AcmReceiver::GetNackList(
+    int64_t round_trip_time_ms) const {
+  return neteq_->GetNackList(round_trip_time_ms);
+}
+
+void AcmReceiver::ResetInitialDelay() {
+  neteq_->SetMinimumDelay(0);
+  // TODO(turajs): Should NetEq Buffer be flushed?
+}
+
+uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
+  // Down-cast the time to (32-6)-bit since we only care about
+  // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
+  // We masked 6 most significant bits of 32-bit so there is no overflow in
+  // the conversion from milliseconds to timestamp.
+  const uint32_t now_in_ms =
+      static_cast<uint32_t>(clock_->TimeInMilliseconds() & 0x03ffffff);
+  return static_cast<uint32_t>((decoder_sampling_rate / 1000) * now_in_ms);
+}
+
+void AcmReceiver::GetDecodingCallStatistics(
+    AudioDecodingCallStats* stats) const {
+  rtc::CritScope lock(&crit_sect_);
+  *stats = call_stats_.GetDecodingStatistics();
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc

modules/audio_coding/acm2/acm_receiver.h

@@ -0,0 +1,229 @@
+/*
+ *  Copyright (c) 2013 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 MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+
+#include <stdint.h>
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/acm2/call_statistics.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "rtc_base/critical_section.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class Clock;
+class NetEq;
+struct RTPHeader;
+
+namespace acm2 {
+
+class AcmReceiver {
+ public:
+  // Constructor of the class
+  explicit AcmReceiver(const AudioCodingModule::Config& config);
+
+  // Destructor of the class.
+  ~AcmReceiver();
+
+  //
+  // Inserts a payload with its associated RTP-header into NetEq.
+  //
+  // Input:
+  //   - rtp_header           : RTP header for the incoming payload containing
+  //                            information about payload type, sequence number,
+  //                            timestamp, SSRC and marker bit.
+  //   - incoming_payload     : Incoming audio payload.
+  //   - length_payload       : Length of incoming audio payload in bytes.
+  //
+  // Return value             : 0 if OK.
+  //                           <0 if NetEq returned an error.
+  //
+  int InsertPacket(const RTPHeader& rtp_header,
+                   rtc::ArrayView<const uint8_t> incoming_payload);
+
+  //
+  // Asks NetEq for 10 milliseconds of decoded audio.
+  //
+  // Input:
+  //   -desired_freq_hz       : specifies the sampling rate [Hz] of the output
+  //                            audio. If set -1 indicates to resampling is
+  //                            is required and the audio returned at the
+  //                            sampling rate of the decoder.
+  //
+  // Output:
+  //   -audio_frame           : an audio frame were output data and
+  //                            associated parameters are written to.
+  //   -muted                 : if true, the sample data in audio_frame is not
+  //                            populated, and must be interpreted as all zero.
+  //
+  // Return value             : 0 if OK.
+  //                           -1 if NetEq returned an error.
+  //
+  int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
+
+  // Replace the current set of decoders with the specified set.
+  void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
+
+  //
+  // Sets a minimum delay for packet buffer. The given delay is maintained,
+  // unless channel condition dictates a higher delay.
+  //
+  // Input:
+  //   - delay_ms             : minimum delay in milliseconds.
+  //
+  // Return value             : 0 if OK.
+  //                           <0 if NetEq returned an error.
+  //
+  int SetMinimumDelay(int delay_ms);
+
+  //
+  // Sets a maximum delay [ms] for the packet buffer. The target delay does not
+  // exceed the given value, even if channel condition requires so.
+  //
+  // Input:
+  //   - delay_ms             : maximum delay in milliseconds.
+  //
+  // Return value             : 0 if OK.
+  //                           <0 if NetEq returned an error.
+  //
+  int SetMaximumDelay(int delay_ms);
+
+  // Sets a base minimum delay in milliseconds for the packet buffer.
+  // Base minimum delay sets lower bound minimum delay value which
+  // is set via SetMinimumDelay.
+  //
+  // Returns true if value was successfully set, false overwise.
+  bool SetBaseMinimumDelayMs(int delay_ms);
+
+  // Returns current value of base minimum delay in milliseconds.
+  int GetBaseMinimumDelayMs() const;
+
+  //
+  // Resets the initial delay to zero.
+  //
+  void ResetInitialDelay();
+
+  // Returns the sample rate of the decoder associated with the last incoming
+  // packet. If no packet of a registered non-CNG codec has been received, the
+  // return value is empty. Also, if the decoder was unregistered since the last
+  // packet was inserted, the return value is empty.
+  absl::optional<int> last_packet_sample_rate_hz() const;
+
+  // Returns last_output_sample_rate_hz from the NetEq instance.
+  int last_output_sample_rate_hz() const;
+
+  //
+  // Get the current network statistics from NetEq.
+  //
+  // Output:
+  //   - statistics           : The current network statistics.
+  //
+  void GetNetworkStatistics(NetworkStatistics* statistics) const;
+
+  //
+  // Flushes the NetEq packet and speech buffers.
+  //
+  void FlushBuffers();
+
+  //
+  // Remove all registered codecs.
+  //
+  void RemoveAllCodecs();
+
+  // Returns the RTP timestamp for the last sample delivered by GetAudio().
+  // The return value will be empty if no valid timestamp is available.
+  absl::optional<uint32_t> GetPlayoutTimestamp();
+
+  // Returns the current total delay from NetEq (packet buffer and sync buffer)
+  // in ms, with smoothing applied to even out short-time fluctuations due to
+  // jitter. The packet buffer part of the delay is not updated during DTX/CNG
+  // periods.
+  //
+  int FilteredCurrentDelayMs() const;
+
+  // Returns the current target delay for NetEq in ms.
+  //
+  int TargetDelayMs() const;
+
+  //
+  // Get payload type and format of the last non-CNG/non-DTMF received payload.
+  // If no non-CNG/non-DTMF packet is received absl::nullopt is returned.
+  //
+  absl::optional<std::pair<int, SdpAudioFormat>> LastDecoder() const;
+
+  //
+  // Enable NACK and set the maximum size of the NACK list. If NACK is already
+  // enabled then the maximum NACK list size is modified accordingly.
+  //
+  // If the sequence number of last received packet is N, the sequence numbers
+  // of NACK list are in the range of [N - |max_nack_list_size|, N).
+  //
+  // |max_nack_list_size| should be positive (none zero) and less than or
+  // equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
+  // is returned. 0 is returned at success.
+  //
+  int EnableNack(size_t max_nack_list_size);
+
+  // Disable NACK.
+  void DisableNack();
+
+  //
+  // Get a list of packets to be retransmitted. |round_trip_time_ms| is an
+  // estimate of the round-trip-time (in milliseconds). Missing packets which
+  // will be playout in a shorter time than the round-trip-time (with respect
+  // to the time this API is called) will not be included in the list.
+  //
+  // Negative |round_trip_time_ms| results is an error message and empty list
+  // is returned.
+  //
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
+
+  //
+  // Get statistics of calls to GetAudio().
+  void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
+
+ private:
+  struct DecoderInfo {
+    int payload_type;
+    int sample_rate_hz;
+    int num_channels;
+    SdpAudioFormat sdp_format;
+  };
+
+  uint32_t NowInTimestamp(int decoder_sampling_rate) const;
+
+  rtc::CriticalSection crit_sect_;
+  absl::optional<DecoderInfo> last_decoder_ RTC_GUARDED_BY(crit_sect_);
+  ACMResampler resampler_ RTC_GUARDED_BY(crit_sect_);
+  std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(crit_sect_);
+  CallStatistics call_stats_ RTC_GUARDED_BY(crit_sect_);
+  const std::unique_ptr<NetEq> neteq_;  // NetEq is thread-safe; no lock needed.
+  Clock* const clock_;
+  bool resampled_last_output_frame_ RTC_GUARDED_BY(crit_sect_);
+};
+
+}  // namespace acm2
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_

modules/audio_coding/acm2/acm_receiver_unittest.cc

@@ -0,0 +1,464 @@
+/*
+ *  Copyright (c) 2013 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.
+ */
+
+#include "modules/audio_coding/acm2/acm_receiver.h"
+
+#include <algorithm>  // std::min
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+class AcmReceiverTestOldApi : public AudioPacketizationCallback,
+                              public ::testing::Test {
+ protected:
+  AcmReceiverTestOldApi()
+      : timestamp_(0),
+        packet_sent_(false),
+        last_packet_send_timestamp_(timestamp_),
+        last_frame_type_(AudioFrameType::kEmptyFrame) {
+    config_.decoder_factory = decoder_factory_;
+  }
+
+  ~AcmReceiverTestOldApi() {}
+
+  void SetUp() override {
+    acm_.reset(AudioCodingModule::Create(config_));
+    receiver_.reset(new AcmReceiver(config_));
+    ASSERT_TRUE(receiver_.get() != NULL);
+    ASSERT_TRUE(acm_.get() != NULL);
+    acm_->InitializeReceiver();
+    acm_->RegisterTransportCallback(this);
+
+    rtp_header_.sequenceNumber = 0;
+    rtp_header_.timestamp = 0;
+    rtp_header_.markerBit = false;
+    rtp_header_.ssrc = 0x12345678;  // Arbitrary.
+    rtp_header_.numCSRCs = 0;
+    rtp_header_.payloadType = 0;
+  }
+
+  void TearDown() override {}
+
+  AudioCodecInfo SetEncoder(int payload_type,
+                            const SdpAudioFormat& format,
+                            const std::map<int, int> cng_payload_types = {}) {
+    // Create the speech encoder.
+    AudioCodecInfo info = encoder_factory_->QueryAudioEncoder(format).value();
+    std::unique_ptr<AudioEncoder> enc =
+        encoder_factory_->MakeAudioEncoder(payload_type, format, absl::nullopt);
+
+    // If we have a compatible CN specification, stack a CNG on top.
+    auto it = cng_payload_types.find(info.sample_rate_hz);
+    if (it != cng_payload_types.end()) {
+      AudioEncoderCngConfig config;
+      config.speech_encoder = std::move(enc);
+      config.num_channels = 1;
+      config.payload_type = it->second;
+      config.vad_mode = Vad::kVadNormal;
+      enc = CreateComfortNoiseEncoder(std::move(config));
+    }
+
+    // Actually start using the new encoder.
+    acm_->SetEncoder(std::move(enc));
+    return info;
+  }
+
+  int InsertOnePacketOfSilence(const AudioCodecInfo& info) {
+    // Frame setup according to the codec.
+    AudioFrame frame;
+    frame.sample_rate_hz_ = info.sample_rate_hz;
+    frame.samples_per_channel_ = info.sample_rate_hz / 100;  // 10 ms.
+    frame.num_channels_ = info.num_channels;
+    frame.Mute();
+    packet_sent_ = false;
+    last_packet_send_timestamp_ = timestamp_;
+    int num_10ms_frames = 0;
+    while (!packet_sent_) {
+      frame.timestamp_ = timestamp_;
+      timestamp_ += rtc::checked_cast<uint32_t>(frame.samples_per_channel_);
+      EXPECT_GE(acm_->Add10MsData(frame), 0);
+      ++num_10ms_frames;
+    }
+    return num_10ms_frames;
+  }
+
+  int SendData(AudioFrameType frame_type,
+               uint8_t payload_type,
+               uint32_t timestamp,
+               const uint8_t* payload_data,
+               size_t payload_len_bytes,
+               int64_t absolute_capture_timestamp_ms) override {
+    if (frame_type == AudioFrameType::kEmptyFrame)
+      return 0;
+
+    rtp_header_.payloadType = payload_type;
+    rtp_header_.timestamp = timestamp;
+
+    int ret_val = receiver_->InsertPacket(
+        rtp_header_,
+        rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
+    if (ret_val < 0) {
+      assert(false);
+      return -1;
+    }
+    rtp_header_.sequenceNumber++;
+    packet_sent_ = true;
+    last_frame_type_ = frame_type;
+    return 0;
+  }
+
+  const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_ =
+      CreateBuiltinAudioEncoderFactory();
+  const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_ =
+      CreateBuiltinAudioDecoderFactory();
+  AudioCodingModule::Config config_;
+  std::unique_ptr<AcmReceiver> receiver_;
+  std::unique_ptr<AudioCodingModule> acm_;
+  RTPHeader rtp_header_;
+  uint32_t timestamp_;
+  bool packet_sent_;  // Set when SendData is called reset when inserting audio.
+  uint32_t last_packet_send_timestamp_;
+  AudioFrameType last_frame_type_;
+};
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_SampleRate DISABLED_SampleRate
+#else
+#define MAYBE_SampleRate SampleRate
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
+  const std::map<int, SdpAudioFormat> codecs = {{0, {"ISAC", 16000, 1}},
+                                                {1, {"ISAC", 32000, 1}}};
+  receiver_->SetCodecs(codecs);
+
+  constexpr int kOutSampleRateHz = 8000;  // Different than codec sample rate.
+  for (size_t i = 0; i < codecs.size(); ++i) {
+    const int payload_type = rtc::checked_cast<int>(i);
+    const int num_10ms_frames =
+        InsertOnePacketOfSilence(SetEncoder(payload_type, codecs.at(i)));
+    for (int k = 0; k < num_10ms_frames; ++k) {
+      AudioFrame frame;
+      bool muted;
+      EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame, &muted));
+    }
+    EXPECT_EQ(encoder_factory_->QueryAudioEncoder(codecs.at(i))->sample_rate_hz,
+              receiver_->last_output_sample_rate_hz());
+  }
+}
+
+class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
+ protected:
+  AcmReceiverTestFaxModeOldApi() {
+    config_.neteq_config.for_test_no_time_stretching = true;
+  }
+
+  void RunVerifyAudioFrame(const SdpAudioFormat& codec) {
+    // Make sure "fax mode" is enabled. This will avoid delay changes unless the
+    // packet-loss concealment is made. We do this in order to make the
+    // timestamp increments predictable; in normal mode, NetEq may decide to do
+    // accelerate or pre-emptive expand operations after some time, offsetting
+    // the timestamp.
+    EXPECT_TRUE(config_.neteq_config.for_test_no_time_stretching);
+
+    constexpr int payload_type = 17;
+    receiver_->SetCodecs({{payload_type, codec}});
+
+    const AudioCodecInfo info = SetEncoder(payload_type, codec);
+    const int output_sample_rate_hz = info.sample_rate_hz;
+    const size_t output_channels = info.num_channels;
+    const size_t samples_per_ms = rtc::checked_cast<size_t>(
+        rtc::CheckedDivExact(output_sample_rate_hz, 1000));
+    const AudioFrame::VADActivity expected_vad_activity =
+        output_sample_rate_hz > 16000 ? AudioFrame::kVadActive
+                                      : AudioFrame::kVadPassive;
+
+    // Expect the first output timestamp to be 5*fs/8000 samples before the
+    // first inserted timestamp (because of NetEq's look-ahead). (This value is
+    // defined in Expand::overlap_length_.)
+    uint32_t expected_output_ts =
+        last_packet_send_timestamp_ -
+        rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);
+
+    AudioFrame frame;
+    bool muted;
+    EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
+    // Expect timestamp = 0 before first packet is inserted.
+    EXPECT_EQ(0u, frame.timestamp_);
+    for (int i = 0; i < 5; ++i) {
+      const int num_10ms_frames = InsertOnePacketOfSilence(info);
+      for (int k = 0; k < num_10ms_frames; ++k) {
+        EXPECT_EQ(0,
+                  receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
+        EXPECT_EQ(expected_output_ts, frame.timestamp_);
+        expected_output_ts += rtc::checked_cast<uint32_t>(10 * samples_per_ms);
+        EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
+        EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
+        EXPECT_EQ(output_channels, frame.num_channels_);
+        EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
+        EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
+        EXPECT_FALSE(muted);
+      }
+    }
+  }
+};
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyAudioFramePCMU DISABLED_VerifyAudioFramePCMU
+#else
+#define MAYBE_VerifyAudioFramePCMU VerifyAudioFramePCMU
+#endif
+TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFramePCMU) {
+  RunVerifyAudioFrame({"PCMU", 8000, 1});
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyAudioFrameISAC DISABLED_VerifyAudioFrameISAC
+#else
+#define MAYBE_VerifyAudioFrameISAC VerifyAudioFrameISAC
+#endif
+TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameISAC) {
+  RunVerifyAudioFrame({"ISAC", 16000, 1});
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyAudioFrameOpus DISABLED_VerifyAudioFrameOpus
+#else
+#define MAYBE_VerifyAudioFrameOpus VerifyAudioFrameOpus
+#endif
+TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameOpus) {
+  RunVerifyAudioFrame({"opus", 48000, 2});
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
+#else
+#define MAYBE_PostdecodingVad PostdecodingVad
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
+  EXPECT_TRUE(config_.neteq_config.enable_post_decode_vad);
+  constexpr int payload_type = 34;
+  const SdpAudioFormat codec = {"L16", 16000, 1};
+  const AudioCodecInfo info = SetEncoder(payload_type, codec);
+  receiver_->SetCodecs({{payload_type, codec}});
+  constexpr int kNumPackets = 5;
+  AudioFrame frame;
+  for (int n = 0; n < kNumPackets; ++n) {
+    const int num_10ms_frames = InsertOnePacketOfSilence(info);
+    for (int k = 0; k < num_10ms_frames; ++k) {
+      bool muted;
+      ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
+    }
+  }
+  EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
+}
+
+class AcmReceiverTestPostDecodeVadPassiveOldApi : public AcmReceiverTestOldApi {
+ protected:
+  AcmReceiverTestPostDecodeVadPassiveOldApi() {
+    config_.neteq_config.enable_post_decode_vad = false;
+  }
+};
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
+#else
+#define MAYBE_PostdecodingVad PostdecodingVad
+#endif
+TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
+  EXPECT_FALSE(config_.neteq_config.enable_post_decode_vad);
+  constexpr int payload_type = 34;
+  const SdpAudioFormat codec = {"L16", 16000, 1};
+  const AudioCodecInfo info = SetEncoder(payload_type, codec);
+  auto const value = encoder_factory_->QueryAudioEncoder(codec);
+  ASSERT_TRUE(value.has_value());
+  receiver_->SetCodecs({{payload_type, codec}});
+  const int kNumPackets = 5;
+  AudioFrame frame;
+  for (int n = 0; n < kNumPackets; ++n) {
+    const int num_10ms_frames = InsertOnePacketOfSilence(info);
+    for (int k = 0; k < num_10ms_frames; ++k) {
+      bool muted;
+      ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
+    }
+  }
+  EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
+#else
+#define MAYBE_LastAudioCodec LastAudioCodec
+#endif
+#if defined(WEBRTC_CODEC_ISAC)
+TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
+  const std::map<int, SdpAudioFormat> codecs = {{0, {"ISAC", 16000, 1}},
+                                                {1, {"PCMA", 8000, 1}},
+                                                {2, {"ISAC", 32000, 1}},
+                                                {3, {"L16", 32000, 1}}};
+  const std::map<int, int> cng_payload_types = {
+      {8000, 100}, {16000, 101}, {32000, 102}};
+  {
+    std::map<int, SdpAudioFormat> receive_codecs = codecs;
+    for (const auto& cng_type : cng_payload_types) {
+      receive_codecs.emplace(std::make_pair(
+          cng_type.second, SdpAudioFormat("CN", cng_type.first, 1)));
+    }
+    receiver_->SetCodecs(receive_codecs);
+  }
+
+  // No audio payload is received.
+  EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
+
+  // Start with sending DTX.
+  packet_sent_ = false;
+  InsertOnePacketOfSilence(
+      SetEncoder(0, codecs.at(0), cng_payload_types));  // Enough to test
+                                                        // with one codec.
+  ASSERT_TRUE(packet_sent_);
+  EXPECT_EQ(AudioFrameType::kAudioFrameCN, last_frame_type_);
+
+  // Has received, only, DTX. Last Audio codec is undefined.
+  EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
+  EXPECT_EQ(absl::nullopt, receiver_->last_packet_sample_rate_hz());
+
+  for (size_t i = 0; i < codecs.size(); ++i) {
+    // Set DTX off to send audio payload.
+    packet_sent_ = false;
+    const int payload_type = rtc::checked_cast<int>(i);
+    const AudioCodecInfo info_without_cng =
+        SetEncoder(payload_type, codecs.at(i));
+    InsertOnePacketOfSilence(info_without_cng);
+
+    // Sanity check if Actually an audio payload received, and it should be
+    // of type "speech."
+    ASSERT_TRUE(packet_sent_);
+    ASSERT_EQ(AudioFrameType::kAudioFrameSpeech, last_frame_type_);
+    EXPECT_EQ(info_without_cng.sample_rate_hz,
+              receiver_->last_packet_sample_rate_hz());
+
+    // Set VAD on to send DTX. Then check if the "Last Audio codec" returns
+    // the expected codec. Encode repeatedly until a DTX is sent.
+    const AudioCodecInfo info_with_cng =
+        SetEncoder(payload_type, codecs.at(i), cng_payload_types);
+    while (last_frame_type_ != AudioFrameType::kAudioFrameCN) {
+      packet_sent_ = false;
+      InsertOnePacketOfSilence(info_with_cng);
+      ASSERT_TRUE(packet_sent_);
+    }
+    EXPECT_EQ(info_with_cng.sample_rate_hz,
+              receiver_->last_packet_sample_rate_hz());
+    EXPECT_EQ(codecs.at(i), receiver_->LastDecoder()->second);
+  }
+}
+#endif
+
+// Check if the statistics are initialized correctly. Before any call to ACM
+// all fields have to be zero.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_InitializedToZero DISABLED_InitializedToZero
+#else
+#define MAYBE_InitializedToZero InitializedToZero
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_InitializedToZero) {
+  AudioDecodingCallStats stats;
+  receiver_->GetDecodingCallStatistics(&stats);
+  EXPECT_EQ(0, stats.calls_to_neteq);
+  EXPECT_EQ(0, stats.calls_to_silence_generator);
+  EXPECT_EQ(0, stats.decoded_normal);
+  EXPECT_EQ(0, stats.decoded_cng);
+  EXPECT_EQ(0, stats.decoded_neteq_plc);
+  EXPECT_EQ(0, stats.decoded_plc_cng);
+  EXPECT_EQ(0, stats.decoded_muted_output);
+}
+
+// Insert some packets and pull audio. Check statistics are valid. Then,
+// simulate packet loss and check if PLC and PLC-to-CNG statistics are
+// correctly updated.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_NetEqCalls DISABLED_NetEqCalls
+#else
+#define MAYBE_NetEqCalls NetEqCalls
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_NetEqCalls) {
+  AudioDecodingCallStats stats;
+  const int kNumNormalCalls = 10;
+  const int kSampleRateHz = 16000;
+  const int kNumSamples10ms = kSampleRateHz / 100;
+  const int kFrameSizeMs = 10;  // Multiple of 10.
+  const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
+  const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
+  const uint8_t kPayloadType = 111;
+  RTPHeader rtp_header;
+  AudioFrame audio_frame;
+  bool muted;
+
+  receiver_->SetCodecs(
+      {{kPayloadType, SdpAudioFormat("L16", kSampleRateHz, 1)}});
+  rtp_header.sequenceNumber = 0xABCD;
+  rtp_header.timestamp = 0xABCDEF01;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.markerBit = false;
+  rtp_header.ssrc = 0x1234;
+  rtp_header.numCSRCs = 0;
+  rtp_header.payload_type_frequency = kSampleRateHz;
+
+  for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) {
+    const uint8_t kPayload[kPayloadSizeBytes] = {0};
+    ASSERT_EQ(0, receiver_->InsertPacket(rtp_header, kPayload));
+    ++rtp_header.sequenceNumber;
+    rtp_header.timestamp += kFrameSizeSamples;
+    ASSERT_EQ(0, receiver_->GetAudio(-1, &audio_frame, &muted));
+    EXPECT_FALSE(muted);
+  }
+  receiver_->GetDecodingCallStatistics(&stats);
+  EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq);
+  EXPECT_EQ(0, stats.calls_to_silence_generator);
+  EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
+  EXPECT_EQ(0, stats.decoded_cng);
+  EXPECT_EQ(0, stats.decoded_neteq_plc);
+  EXPECT_EQ(0, stats.decoded_plc_cng);
+  EXPECT_EQ(0, stats.decoded_muted_output);
+
+  const int kNumPlc = 3;
+  const int kNumPlcCng = 5;
+
+  // Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG.
+  for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) {
+    ASSERT_EQ(0, receiver_->GetAudio(-1, &audio_frame, &muted));
+    EXPECT_FALSE(muted);
+  }
+  receiver_->GetDecodingCallStatistics(&stats);
+  EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq);
+  EXPECT_EQ(0, stats.calls_to_silence_generator);
+  EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
+  EXPECT_EQ(0, stats.decoded_cng);
+  EXPECT_EQ(kNumPlc, stats.decoded_neteq_plc);
+  EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
+  EXPECT_EQ(0, stats.decoded_muted_output);
+  // TODO(henrik.lundin) Add a test with muted state enabled.
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc

modules/audio_coding/acm2/acm_remixing.cc

@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2019 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.
+ */
+
+#include "modules/audio_coding/acm2/acm_remixing.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output) {
+  RTC_DCHECK_EQ(input.num_channels_, 2);
+  RTC_DCHECK_EQ(output.size(), input.samples_per_channel_);
+
+  if (input.muted()) {
+    std::fill(output.begin(), output.begin() + input.samples_per_channel_, 0);
+  } else {
+    const int16_t* const input_data = input.data();
+    for (size_t n = 0; n < input.samples_per_channel_; ++n) {
+      output[n] = rtc::dchecked_cast<int16_t>(
+          (int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
+    }
+  }
+}
+
+void ReMixFrame(const AudioFrame& input,
+                size_t num_output_channels,
+                std::vector<int16_t>* output) {
+  const size_t output_size = num_output_channels * input.samples_per_channel_;
+  RTC_DCHECK(!(input.num_channels_ == 0 && num_output_channels > 0 &&
+               input.samples_per_channel_ > 0));
+
+  if (output->size() != output_size) {
+    output->resize(output_size);
+  }
+
+  // For muted frames, fill the frame with zeros.
+  if (input.muted()) {
+    std::fill(output->begin(), output->end(), 0);
+    return;
+  }
+
+  // Ensure that the special case of zero input channels is handled correctly
+  // (zero samples per channel is already handled correctly in the code below).
+  if (input.num_channels_ == 0) {
+    return;
+  }
+
+  const int16_t* const input_data = input.data();
+  size_t out_index = 0;
+
+  // When upmixing is needed and the input is mono copy the left channel
+  // into the left and right channels, and set any remaining channels to zero.
+  if (input.num_channels_ == 1 && input.num_channels_ < num_output_channels) {
+    for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+      (*output)[out_index++] = input_data[k];
+      (*output)[out_index++] = input_data[k];
+      for (size_t j = 2; j < num_output_channels; ++j) {
+        (*output)[out_index++] = 0;
+      }
+      RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
+    }
+    RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
+    return;
+  }
+
+  size_t in_index = 0;
+
+  // When upmixing is needed and the output is surround, copy the available
+  // channels directly, and set the remaining channels to zero.
+  if (input.num_channels_ < num_output_channels) {
+    for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+      for (size_t j = 0; j < input.num_channels_; ++j) {
+        (*output)[out_index++] = input_data[in_index++];
+      }
+      for (size_t j = input.num_channels_; j < num_output_channels; ++j) {
+        (*output)[out_index++] = 0;
+      }
+      RTC_DCHECK_EQ(in_index, (k + 1) * input.num_channels_);
+      RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
+    }
+    RTC_DCHECK_EQ(in_index, input.samples_per_channel_ * input.num_channels_);
+    RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
+
+    return;
+  }
+
+  // When downmixing is needed, and the input is stereo, average the channels.
+  if (input.num_channels_ == 2) {
+    for (size_t n = 0; n < input.samples_per_channel_; ++n) {
+      (*output)[n] = rtc::dchecked_cast<int16_t>(
+          (int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
+    }
+    return;
+  }
+
+  // When downmixing is needed, and the input is multichannel, drop the surplus
+  // channels.
+  const size_t num_channels_to_drop = input.num_channels_ - num_output_channels;
+  for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+    for (size_t j = 0; j < num_output_channels; ++j) {
+      (*output)[out_index++] = input_data[in_index++];
+    }
+    in_index += num_channels_to_drop;
+  }
+}
+
+}  // namespace webrtc

modules/audio_coding/acm2/acm_remixing.h

@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2019 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 MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
+
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+
+// Stereo-to-mono downmixing. The length of the output must equal to the number
+// of samples per channel in the input.
+void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output);
+
+// Remixes the interleaved input frame to an interleaved output data vector. The
+// remixed data replaces the data in the output vector which is resized if
+// needed. The remixing supports any combination of input and output channels,
+// as well as any number of samples per channel.
+void ReMixFrame(const AudioFrame& input,
+                size_t num_output_channels,
+                std::vector<int16_t>* output);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_

modules/audio_coding/acm2/acm_remixing_unittest.cc

@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2013 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.
+ */
+
+#include "modules/audio_coding/acm2/acm_remixing.h"
+
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using ::testing::AllOf;
+using ::testing::Each;
+using ::testing::ElementsAreArray;
+using ::testing::SizeIs;
+
+namespace webrtc {
+
+TEST(AcmRemixing, DownMixFrame) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 2;
+    in_data[2 * k + 1] = 0;
+  }
+
+  DownMixFrame(in, out);
+
+  EXPECT_THAT(out, AllOf(SizeIs(480), Each(1)));
+}
+
+TEST(AcmRemixing, DownMixMutedFrame) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 2;
+    in_data[2 * k + 1] = 0;
+  }
+
+  in.Mute();
+
+  DownMixFrame(in, out);
+
+  EXPECT_THAT(out, AllOf(SizeIs(480), Each(0)));
+}
+
+TEST(AcmRemixing, RemixMutedStereoFrameTo6Channels) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 1;
+    in_data[2 * k + 1] = 2;
+  }
+  in.Mute();
+
+  ReMixFrame(in, 6, &out);
+  EXPECT_EQ(6 * 480u, out.size());
+
+  EXPECT_THAT(out, AllOf(SizeIs(in.samples_per_channel_ * 6), Each(0)));
+}
+
+TEST(AcmRemixing, RemixStereoFrameTo6Channels) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 1;
+    in_data[2 * k + 1] = 2;
+  }
+
+  ReMixFrame(in, 6, &out);
+  EXPECT_EQ(6 * 480u, out.size());
+
+  std::vector<int16_t> expected_output(in.samples_per_channel_ * 6);
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    expected_output[6 * k] = 1;
+    expected_output[6 * k + 1] = 2;
+  }
+
+  EXPECT_THAT(out, ElementsAreArray(expected_output));
+}
+
+TEST(AcmRemixing, RemixMonoFrameTo6Channels) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 1;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[k] = 1;
+  }
+
+  ReMixFrame(in, 6, &out);
+  EXPECT_EQ(6 * 480u, out.size());
+
+  std::vector<int16_t> expected_output(in.samples_per_channel_ * 6, 0);
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    expected_output[6 * k] = 1;
+    expected_output[6 * k + 1] = 1;
+  }
+
+  EXPECT_THAT(out, ElementsAreArray(expected_output));
+}
+
+TEST(AcmRemixing, RemixStereoFrameToMono) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 2;
+    in_data[2 * k + 1] = 0;
+  }
+
+  ReMixFrame(in, 1, &out);
+  EXPECT_EQ(480u, out.size());
+
+  EXPECT_THAT(out, AllOf(SizeIs(in.samples_per_channel_), Each(1)));
+}
+
+TEST(AcmRemixing, RemixMonoFrameToStereo) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 1;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[k] = 1;
+  }
+
+  ReMixFrame(in, 2, &out);
+  EXPECT_EQ(960u, out.size());
+
+  EXPECT_THAT(out, AllOf(SizeIs(2 * in.samples_per_channel_), Each(1)));
+}
+
+TEST(AcmRemixing, Remix3ChannelFrameToStereo) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 3;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    for (size_t j = 0; j < 3; ++j) {
+      in_data[3 * k + j] = j;
+    }
+  }
+
+  ReMixFrame(in, 2, &out);
+  EXPECT_EQ(2 * 480u, out.size());
+
+  std::vector<int16_t> expected_output(in.samples_per_channel_ * 2);
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    for (size_t j = 0; j < 2; ++j) {
+      expected_output[2 * k + j] = static_cast<int>(j);
+    }
+  }
+
+  EXPECT_THAT(out, ElementsAreArray(expected_output));
+}
+
+}  // namespace webrtc

modules/audio_coding/acm2/acm_resampler.cc

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#include "modules/audio_coding/acm2/acm_resampler.h"
+
+#include <assert.h>
+#include <string.h>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace acm2 {
+
+ACMResampler::ACMResampler() {}
+
+ACMResampler::~ACMResampler() {}
+
+int ACMResampler::Resample10Msec(const int16_t* in_audio,
+                                 int in_freq_hz,
+                                 int out_freq_hz,
+                                 size_t num_audio_channels,
+                                 size_t out_capacity_samples,
+                                 int16_t* out_audio) {
+  size_t in_length = in_freq_hz * num_audio_channels / 100;
+  if (in_freq_hz == out_freq_hz) {
+    if (out_capacity_samples < in_length) {
+      assert(false);
+      return -1;
+    }
+    memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
+    return static_cast<int>(in_length / num_audio_channels);
+  }
+
+  if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
+                                    num_audio_channels) != 0) {
+    RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
+                      << out_freq_hz << ", " << num_audio_channels
+                      << ") failed.";
+    return -1;
+  }
+
+  int out_length =
+      resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
+  if (out_length == -1) {
+    RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
+                      << out_audio << ", " << out_capacity_samples
+                      << ") failed.";
+    return -1;
+  }
+
+  return static_cast<int>(out_length / num_audio_channels);
+}
+
+}  // namespace acm2
+}  // namespace webrtc

modules/audio_coding/acm2/acm_resampler.h

@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2012 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 MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "common_audio/resampler/include/push_resampler.h"
+
+namespace webrtc {
+namespace acm2 {
+
+class ACMResampler {
+ public:
+  ACMResampler();
+  ~ACMResampler();
+
+  int Resample10Msec(const int16_t* in_audio,
+                     int in_freq_hz,
+                     int out_freq_hz,
+                     size_t num_audio_channels,
+                     size_t out_capacity_samples,
+                     int16_t* out_audio);
+
+ private:
+  PushResampler<int16_t> resampler_;
+};
+
+}  // namespace acm2
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_

modules/audio_coding/acm2/acm_send_test.cc

@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#include "modules/audio_coding/acm2/acm_send_test.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "absl/strings/match.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_encode.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
+                                     int source_rate_hz,
+                                     int test_duration_ms)
+    : clock_(0),
+      acm_(webrtc::AudioCodingModule::Create([this] {
+        AudioCodingModule::Config config;
+        config.clock = &clock_;
+        config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+        return config;
+      }())),
+      audio_source_(audio_source),
+      source_rate_hz_(source_rate_hz),
+      input_block_size_samples_(
+          static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),
+      codec_registered_(false),
+      test_duration_ms_(test_duration_ms),
+      frame_type_(AudioFrameType::kAudioFrameSpeech),
+      payload_type_(0),
+      timestamp_(0),
+      sequence_number_(0) {
+  input_frame_.sample_rate_hz_ = source_rate_hz_;
+  input_frame_.num_channels_ = 1;
+  input_frame_.samples_per_channel_ = input_block_size_samples_;
+  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
+         AudioFrame::kMaxDataSizeSamples);
+  acm_->RegisterTransportCallback(this);
+}
+
+AcmSendTestOldApi::~AcmSendTestOldApi() = default;
+
+bool AcmSendTestOldApi::RegisterCodec(const char* payload_name,
+                                      int clockrate_hz,
+                                      int num_channels,
+                                      int payload_type,
+                                      int frame_size_samples) {
+  SdpAudioFormat format(payload_name, clockrate_hz, num_channels);
+  if (absl::EqualsIgnoreCase(payload_name, "g722")) {
+    RTC_CHECK_EQ(16000, clockrate_hz);
+    format.clockrate_hz = 8000;
+  } else if (absl::EqualsIgnoreCase(payload_name, "opus")) {
+    RTC_CHECK(num_channels == 1 || num_channels == 2);
+    if (num_channels == 2) {
+      format.parameters["stereo"] = "1";
+    }
+    format.num_channels = 2;
+  }
+  format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
+      frame_size_samples, rtc::CheckedDivExact(clockrate_hz, 1000)));
+  auto factory = CreateBuiltinAudioEncoderFactory();
+  acm_->SetEncoder(
+      factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
+  codec_registered_ = true;
+  input_frame_.num_channels_ = num_channels;
+  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
+         AudioFrame::kMaxDataSizeSamples);
+  return codec_registered_;
+}
+
+void AcmSendTestOldApi::RegisterExternalCodec(
+    std::unique_ptr<AudioEncoder> external_speech_encoder) {
+  input_frame_.num_channels_ = external_speech_encoder->NumChannels();
+  acm_->SetEncoder(std::move(external_speech_encoder));
+  assert(input_block_size_samples_ * input_frame_.num_channels_ <=
+         AudioFrame::kMaxDataSizeSamples);
+  codec_registered_ = true;
+}
+
+std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
+  assert(codec_registered_);
+  if (filter_.test(static_cast<size_t>(payload_type_))) {
+    // This payload type should be filtered out. Since the payload type is the
+    // same throughout the whole test run, no packet at all will be delivered.
+    // We can just as well signal that the test is over by returning NULL.
+    return nullptr;
+  }
+  // Insert audio and process until one packet is produced.
+  while (clock_.TimeInMilliseconds() < test_duration_ms_) {
+    clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
+    RTC_CHECK(audio_source_->Read(
+        input_block_size_samples_ * input_frame_.num_channels_,
+        input_frame_.mutable_data()));
+    data_to_send_ = false;
+    RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
+    input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
+    if (data_to_send_) {
+      // Encoded packet received.
+      return CreatePacket();
+    }
+  }
+  // Test ended.
+  return nullptr;
+}
+
+// This method receives the callback from ACM when a new packet is produced.
+int32_t AcmSendTestOldApi::SendData(AudioFrameType frame_type,
+                                    uint8_t payload_type,
+                                    uint32_t timestamp,
+                                    const uint8_t* payload_data,
+                                    size_t payload_len_bytes,
+                                    int64_t absolute_capture_timestamp_ms) {
+  // Store the packet locally.
+  frame_type_ = frame_type;
+  payload_type_ = payload_type;
+  timestamp_ = timestamp;
+  last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
+  assert(last_payload_vec_.size() == payload_len_bytes);
+  data_to_send_ = true;
+  return 0;
+}
+
+std::unique_ptr<Packet> AcmSendTestOldApi::CreatePacket() {
+  const size_t kRtpHeaderSize = 12;
+  size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize;
+  uint8_t* packet_memory = new uint8_t[allocated_bytes];
+  // Populate the header bytes.
+  packet_memory[0] = 0x80;
+  packet_memory[1] = static_cast<uint8_t>(payload_type_);
+  packet_memory[2] = (sequence_number_ >> 8) & 0xFF;
+  packet_memory[3] = (sequence_number_)&0xFF;
+  packet_memory[4] = (timestamp_ >> 24) & 0xFF;
+  packet_memory[5] = (timestamp_ >> 16) & 0xFF;
+  packet_memory[6] = (timestamp_ >> 8) & 0xFF;
+  packet_memory[7] = timestamp_ & 0xFF;
+  // Set SSRC to 0x12345678.
+  packet_memory[8] = 0x12;
+  packet_memory[9] = 0x34;
+  packet_memory[10] = 0x56;
+  packet_memory[11] = 0x78;
+
+  ++sequence_number_;
+
+  // Copy the payload data.
+  memcpy(packet_memory + kRtpHeaderSize, &last_payload_vec_[0],
+         last_payload_vec_.size());
+  std::unique_ptr<Packet> packet(
+      new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()));
+  RTC_DCHECK(packet);
+  RTC_DCHECK(packet->valid_header());
+  return packet;
+}
+
+}  // namespace test
+}  // namespace webrtc

modules/audio_coding/acm2/acm_send_test.h

@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2014 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 MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "rtc_base/constructor_magic.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+class AudioEncoder;
+
+namespace test {
+class InputAudioFile;
+class Packet;
+
+class AcmSendTestOldApi : public AudioPacketizationCallback,
+                          public PacketSource {
+ public:
+  AcmSendTestOldApi(InputAudioFile* audio_source,
+                    int source_rate_hz,
+                    int test_duration_ms);
+  ~AcmSendTestOldApi() override;
+
+  // Registers the send codec. Returns true on success, false otherwise.
+  bool RegisterCodec(const char* payload_name,
+                     int sampling_freq_hz,
+                     int channels,
+                     int payload_type,
+                     int frame_size_samples);
+
+  // Registers an external send codec.
+  void RegisterExternalCodec(
+      std::unique_ptr<AudioEncoder> external_speech_encoder);
+
+  // Inherited from PacketSource.
+  std::unique_ptr<Packet> NextPacket() override;
+
+  // Inherited from AudioPacketizationCallback.
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_len_bytes,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  AudioCodingModule* acm() { return acm_.get(); }
+
+ private:
+  static const int kBlockSizeMs = 10;
+
+  // Creates a Packet object from the last packet produced by ACM (and received
+  // through the SendData method as a callback).
+  std::unique_ptr<Packet> CreatePacket();
+
+  SimulatedClock clock_;
+  std::unique_ptr<AudioCodingModule> acm_;
+  InputAudioFile* audio_source_;
+  int source_rate_hz_;
+  const size_t input_block_size_samples_;
+  AudioFrame input_frame_;
+  bool codec_registered_;
+  int test_duration_ms_;
+  // The following member variables are set whenever SendData() is called.
+  AudioFrameType frame_type_;
+  int payload_type_;
+  uint32_t timestamp_;
+  uint16_t sequence_number_;
+  std::vector<uint8_t> last_payload_vec_;
+  bool data_to_send_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_

modules/audio_coding/acm2/audio_coding_module.cc

@@ -0,0 +1,626 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+
+#include <assert.h>
+#include <algorithm>
+#include <cstdint>
+
+#include "absl/strings/match.h"
+#include "api/array_view.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_remixing.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/include/module_common_types.h"
+#include "modules/include/module_common_types_public.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/critical_section.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+// Initial size for the buffer in InputBuffer. This matches 6 channels of 10 ms
+// 48 kHz data.
+constexpr size_t kInitialInputDataBufferSize = 6 * 480;
+
+constexpr int32_t kMaxInputSampleRateHz = 192000;
+
+class AudioCodingModuleImpl final : public AudioCodingModule {
+ public:
+  explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
+  ~AudioCodingModuleImpl() override;
+
+  /////////////////////////////////////////
+  //   Sender
+  //
+
+  void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
+                         modifier) override;
+
+  // Register a transport callback which will be
+  // called to deliver the encoded buffers.
+  int RegisterTransportCallback(AudioPacketizationCallback* transport) override;
+
+  // Add 10 ms of raw (PCM) audio data to the encoder.
+  int Add10MsData(const AudioFrame& audio_frame) override;
+
+  /////////////////////////////////////////
+  // (FEC) Forward Error Correction (codec internal)
+  //
+
+  // Set target packet loss rate
+  int SetPacketLossRate(int loss_rate) override;
+
+  /////////////////////////////////////////
+  //   Receiver
+  //
+
+  // Initialize receiver, resets codec database etc.
+  int InitializeReceiver() override;
+
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
+
+  // Incoming packet from network parsed and ready for decode.
+  int IncomingPacket(const uint8_t* incoming_payload,
+                     const size_t payload_length,
+                     const RTPHeader& rtp_info) override;
+
+  // Get 10 milliseconds of raw audio data to play out, and
+  // automatic resample to the requested frequency if > 0.
+  int PlayoutData10Ms(int desired_freq_hz,
+                      AudioFrame* audio_frame,
+                      bool* muted) override;
+
+  /////////////////////////////////////////
+  //   Statistics
+  //
+
+  int GetNetworkStatistics(NetworkStatistics* statistics) override;
+
+  ANAStats GetANAStats() const override;
+
+ private:
+  struct InputData {
+    InputData() : buffer(kInitialInputDataBufferSize) {}
+    uint32_t input_timestamp;
+    const int16_t* audio;
+    size_t length_per_channel;
+    size_t audio_channel;
+    // If a re-mix is required (up or down), this buffer will store a re-mixed
+    // version of the input.
+    std::vector<int16_t> buffer;
+  };
+
+  InputData input_data_ RTC_GUARDED_BY(acm_crit_sect_);
+
+  // This member class writes values to the named UMA histogram, but only if
+  // the value has changed since the last time (and always for the first call).
+  class ChangeLogger {
+   public:
+    explicit ChangeLogger(const std::string& histogram_name)
+        : histogram_name_(histogram_name) {}
+    // Logs the new value if it is different from the last logged value, or if
+    // this is the first call.
+    void MaybeLog(int value);
+
+   private:
+    int last_value_ = 0;
+    int first_time_ = true;
+    const std::string histogram_name_;
+  };
+
+  int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+
+  // TODO(bugs.webrtc.org/10739): change |absolute_capture_timestamp_ms| to
+  // int64_t when it always receives a valid value.
+  int Encode(const InputData& input_data,
+             absl::optional<int64_t> absolute_capture_timestamp_ms)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+
+  int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+
+  bool HaveValidEncoder(const char* caller_name) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+
+  // Preprocessing of input audio, including resampling and down-mixing if
+  // required, before pushing audio into encoder's buffer.
+  //
+  // in_frame: input audio-frame
+  // ptr_out: pointer to output audio_frame. If no preprocessing is required
+  //          |ptr_out| will be pointing to |in_frame|, otherwise pointing to
+  //          |preprocess_frame_|.
+  //
+  // Return value:
+  //   -1: if encountering an error.
+  //    0: otherwise.
+  int PreprocessToAddData(const AudioFrame& in_frame,
+                          const AudioFrame** ptr_out)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+
+  // Change required states after starting to receive the codec corresponding
+  // to |index|.
+  int UpdateUponReceivingCodec(int index);
+
+  rtc::CriticalSection acm_crit_sect_;
+  rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_crit_sect_);
+  uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_crit_sect_);
+  uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_crit_sect_);
+  acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_crit_sect_);
+  acm2::AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
+  ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_crit_sect_);
+
+  // Current encoder stack, provided by a call to RegisterEncoder.
+  std::unique_ptr<AudioEncoder> encoder_stack_ RTC_GUARDED_BY(acm_crit_sect_);
+
+  // This is to keep track of CN instances where we can send DTMFs.
+  uint8_t previous_pltype_ RTC_GUARDED_BY(acm_crit_sect_);
+
+  bool receiver_initialized_ RTC_GUARDED_BY(acm_crit_sect_);
+
+  AudioFrame preprocess_frame_ RTC_GUARDED_BY(acm_crit_sect_);
+  bool first_10ms_data_ RTC_GUARDED_BY(acm_crit_sect_);
+
+  bool first_frame_ RTC_GUARDED_BY(acm_crit_sect_);
+  uint32_t last_timestamp_ RTC_GUARDED_BY(acm_crit_sect_);
+  uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(acm_crit_sect_);
+
+  rtc::CriticalSection callback_crit_sect_;
+  AudioPacketizationCallback* packetization_callback_
+      RTC_GUARDED_BY(callback_crit_sect_);
+
+  int codec_histogram_bins_log_[static_cast<size_t>(
+      AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)];
+  int number_of_consecutive_empty_packets_;
+};
+
+// Adds a codec usage sample to the histogram.
+void UpdateCodecTypeHistogram(size_t codec_type) {
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.Audio.Encoder.CodecType", static_cast<int>(codec_type),
+      static_cast<int>(
+          webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
+}
+
+void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
+  if (value != last_value_ || first_time_) {
+    first_time_ = false;
+    last_value_ = value;
+    RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, value);
+  }
+}
+
+AudioCodingModuleImpl::AudioCodingModuleImpl(
+    const AudioCodingModule::Config& config)
+    : expected_codec_ts_(0xD87F3F9F),
+      expected_in_ts_(0xD87F3F9F),
+      receiver_(config),
+      bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
+      encoder_stack_(nullptr),
+      previous_pltype_(255),
+      receiver_initialized_(false),
+      first_10ms_data_(false),
+      first_frame_(true),
+      packetization_callback_(NULL),
+      codec_histogram_bins_log_(),
+      number_of_consecutive_empty_packets_(0) {
+  if (InitializeReceiverSafe() < 0) {
+    RTC_LOG(LS_ERROR) << "Cannot initialize receiver";
+  }
+  RTC_LOG(LS_INFO) << "Created";
+}
+
+AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
+
+int32_t AudioCodingModuleImpl::Encode(
+    const InputData& input_data,
+    absl::optional<int64_t> absolute_capture_timestamp_ms) {
+  // TODO(bugs.webrtc.org/10739): add dcheck that
+  // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
+  AudioEncoder::EncodedInfo encoded_info;
+  uint8_t previous_pltype;
+
+  // Check if there is an encoder before.
+  if (!HaveValidEncoder("Process"))
+    return -1;
+
+  if (!first_frame_) {
+    RTC_DCHECK(IsNewerTimestamp(input_data.input_timestamp, last_timestamp_))
+        << "Time should not move backwards";
+  }
+
+  // Scale the timestamp to the codec's RTP timestamp rate.
+  uint32_t rtp_timestamp =
+      first_frame_
+          ? input_data.input_timestamp
+          : last_rtp_timestamp_ +
+                rtc::dchecked_cast<uint32_t>(rtc::CheckedDivExact(
+                    int64_t{input_data.input_timestamp - last_timestamp_} *
+                        encoder_stack_->RtpTimestampRateHz(),
+                    int64_t{encoder_stack_->SampleRateHz()}));
+
+  last_timestamp_ = input_data.input_timestamp;
+  last_rtp_timestamp_ = rtp_timestamp;
+  first_frame_ = false;
+
+  // Clear the buffer before reuse - encoded data will get appended.
+  encode_buffer_.Clear();
+  encoded_info = encoder_stack_->Encode(
+      rtp_timestamp,
+      rtc::ArrayView<const int16_t>(
+          input_data.audio,
+          input_data.audio_channel * input_data.length_per_channel),
+      &encode_buffer_);
+
+  bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000);
+  if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {
+    // Not enough data.
+    return 0;
+  }
+  previous_pltype = previous_pltype_;  // Read it while we have the critsect.
+
+  // Log codec type to histogram once every 500 packets.
+  if (encoded_info.encoded_bytes == 0) {
+    ++number_of_consecutive_empty_packets_;
+  } else {
+    size_t codec_type = static_cast<size_t>(encoded_info.encoder_type);
+    codec_histogram_bins_log_[codec_type] +=
+        number_of_consecutive_empty_packets_ + 1;
+    number_of_consecutive_empty_packets_ = 0;
+    if (codec_histogram_bins_log_[codec_type] >= 500) {
+      codec_histogram_bins_log_[codec_type] -= 500;
+      UpdateCodecTypeHistogram(codec_type);
+    }
+  }
+
+  AudioFrameType frame_type;
+  if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
+    frame_type = AudioFrameType::kEmptyFrame;
+    encoded_info.payload_type = previous_pltype;
+  } else {
+    RTC_DCHECK_GT(encode_buffer_.size(), 0);
+    frame_type = encoded_info.speech ? AudioFrameType::kAudioFrameSpeech
+                                     : AudioFrameType::kAudioFrameCN;
+  }
+
+  {
+    rtc::CritScope lock(&callback_crit_sect_);
+    if (packetization_callback_) {
+      packetization_callback_->SendData(
+          frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
+          encode_buffer_.data(), encode_buffer_.size(),
+          absolute_capture_timestamp_ms.value_or(-1));
+    }
+  }
+  previous_pltype_ = encoded_info.payload_type;
+  return static_cast<int32_t>(encode_buffer_.size());
+}
+
+/////////////////////////////////////////
+//   Sender
+//
+
+void AudioCodingModuleImpl::ModifyEncoder(
+    rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
+  rtc::CritScope lock(&acm_crit_sect_);
+  modifier(&encoder_stack_);
+}
+
+// Register a transport callback which will be called to deliver
+// the encoded buffers.
+int AudioCodingModuleImpl::RegisterTransportCallback(
+    AudioPacketizationCallback* transport) {
+  rtc::CritScope lock(&callback_crit_sect_);
+  packetization_callback_ = transport;
+  return 0;
+}
+
+// Add 10MS of raw (PCM) audio data to the encoder.
+int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
+  rtc::CritScope lock(&acm_crit_sect_);
+  int r = Add10MsDataInternal(audio_frame, &input_data_);
+  // TODO(bugs.webrtc.org/10739): add dcheck that
+  // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
+  return r < 0
+             ? r
+             : Encode(input_data_, audio_frame.absolute_capture_timestamp_ms());
+}
+
+int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
+                                               InputData* input_data) {
+  if (audio_frame.samples_per_channel_ == 0) {
+    assert(false);
+    RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, payload length is zero";
+    return -1;
+  }
+
+  if (audio_frame.sample_rate_hz_ > kMaxInputSampleRateHz) {
+    assert(false);
+    RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, input frequency not valid";
+    return -1;
+  }
+
+  // If the length and frequency matches. We currently just support raw PCM.
+  if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) !=
+      audio_frame.samples_per_channel_) {
+    RTC_LOG(LS_ERROR)
+        << "Cannot Add 10 ms audio, input frequency and length doesn't match";
+    return -1;
+  }
+
+  if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2 &&
+      audio_frame.num_channels_ != 4 && audio_frame.num_channels_ != 6 &&
+      audio_frame.num_channels_ != 8) {
+    RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, invalid number of channels.";
+    return -1;
+  }
+
+  // Do we have a codec registered?
+  if (!HaveValidEncoder("Add10MsData")) {
+    return -1;
+  }
+
+  const AudioFrame* ptr_frame;
+  // Perform a resampling, also down-mix if it is required and can be
+  // performed before resampling (a down mix prior to resampling will take
+  // place if both primary and secondary encoders are mono and input is in
+  // stereo).
+  if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) {
+    return -1;
+  }
+
+  // Check whether we need an up-mix or down-mix?
+  const size_t current_num_channels = encoder_stack_->NumChannels();
+  const bool same_num_channels =
+      ptr_frame->num_channels_ == current_num_channels;
+
+  // TODO(yujo): Skip encode of muted frames.
+  input_data->input_timestamp = ptr_frame->timestamp_;
+  input_data->length_per_channel = ptr_frame->samples_per_channel_;
+  input_data->audio_channel = current_num_channels;
+
+  if (!same_num_channels) {
+    // Remixes the input frame to the output data and in the process resize the
+    // output data if needed.
+    ReMixFrame(*ptr_frame, current_num_channels, &input_data->buffer);
+
+    // For pushing data to primary, point the |ptr_audio| to correct buffer.
+    input_data->audio = input_data->buffer.data();
+    RTC_DCHECK_GE(input_data->buffer.size(),
+                  input_data->length_per_channel * input_data->audio_channel);
+  } else {
+    // When adding data to encoders this pointer is pointing to an audio buffer
+    // with correct number of channels.
+    input_data->audio = ptr_frame->data();
+  }
+
+  return 0;
+}
+
+// Perform a resampling and down-mix if required. We down-mix only if
+// encoder is mono and input is stereo. In case of dual-streaming, both
+// encoders has to be mono for down-mix to take place.
+// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing
+// is required, |*ptr_out| points to |in_frame|.
+// TODO(yujo): Make this more efficient for muted frames.
+int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
+                                               const AudioFrame** ptr_out) {
+  const bool resample =
+      in_frame.sample_rate_hz_ != encoder_stack_->SampleRateHz();
+
+  // This variable is true if primary codec and secondary codec (if exists)
+  // are both mono and input is stereo.
+  // TODO(henrik.lundin): This condition should probably be
+  //   in_frame.num_channels_ > encoder_stack_->NumChannels()
+  const bool down_mix =
+      in_frame.num_channels_ == 2 && encoder_stack_->NumChannels() == 1;
+
+  if (!first_10ms_data_) {
+    expected_in_ts_ = in_frame.timestamp_;
+    expected_codec_ts_ = in_frame.timestamp_;
+    first_10ms_data_ = true;
+  } else if (in_frame.timestamp_ != expected_in_ts_) {
+    RTC_LOG(LS_WARNING) << "Unexpected input timestamp: " << in_frame.timestamp_
+                        << ", expected: " << expected_in_ts_;
+    expected_codec_ts_ +=
+        (in_frame.timestamp_ - expected_in_ts_) *
+        static_cast<uint32_t>(
+            static_cast<double>(encoder_stack_->SampleRateHz()) /
+            static_cast<double>(in_frame.sample_rate_hz_));
+    expected_in_ts_ = in_frame.timestamp_;
+  }
+
+  if (!down_mix && !resample) {
+    // No pre-processing is required.
+    if (expected_in_ts_ == expected_codec_ts_) {
+      // If we've never resampled, we can use the input frame as-is
+      *ptr_out = &in_frame;
+    } else {
+      // Otherwise we'll need to alter the timestamp. Since in_frame is const,
+      // we'll have to make a copy of it.
+      preprocess_frame_.CopyFrom(in_frame);
+      preprocess_frame_.timestamp_ = expected_codec_ts_;
+      *ptr_out = &preprocess_frame_;
+    }
+
+    expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+    expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+    return 0;
+  }
+
+  *ptr_out = &preprocess_frame_;
+  preprocess_frame_.num_channels_ = in_frame.num_channels_;
+  preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
+  std::array<int16_t, AudioFrame::kMaxDataSizeSamples> audio;
+  const int16_t* src_ptr_audio;
+  if (down_mix) {
+    // If a resampling is required, the output of a down-mix is written into a
+    // local buffer, otherwise, it will be written to the output frame.
+    int16_t* dest_ptr_audio =
+        resample ? audio.data() : preprocess_frame_.mutable_data();
+    RTC_DCHECK_GE(audio.size(), preprocess_frame_.samples_per_channel_);
+    RTC_DCHECK_GE(audio.size(), in_frame.samples_per_channel_);
+    DownMixFrame(in_frame,
+                 rtc::ArrayView<int16_t>(
+                     dest_ptr_audio, preprocess_frame_.samples_per_channel_));
+    preprocess_frame_.num_channels_ = 1;
+
+    // Set the input of the resampler to the down-mixed signal.
+    src_ptr_audio = audio.data();
+  } else {
+    // Set the input of the resampler to the original data.
+    src_ptr_audio = in_frame.data();
+  }
+
+  preprocess_frame_.timestamp_ = expected_codec_ts_;
+  preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_;
+  // If it is required, we have to do a resampling.
+  if (resample) {
+    // The result of the resampler is written to output frame.
+    int16_t* dest_ptr_audio = preprocess_frame_.mutable_data();
+
+    int samples_per_channel = resampler_.Resample10Msec(
+        src_ptr_audio, in_frame.sample_rate_hz_, encoder_stack_->SampleRateHz(),
+        preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
+        dest_ptr_audio);
+
+    if (samples_per_channel < 0) {
+      RTC_LOG(LS_ERROR) << "Cannot add 10 ms audio, resampling failed";
+      return -1;
+    }
+    preprocess_frame_.samples_per_channel_ =
+        static_cast<size_t>(samples_per_channel);
+    preprocess_frame_.sample_rate_hz_ = encoder_stack_->SampleRateHz();
+  }
+
+  expected_codec_ts_ +=
+      static_cast<uint32_t>(preprocess_frame_.samples_per_channel_);
+  expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+
+  return 0;
+}
+
+/////////////////////////////////////////
+//   (FEC) Forward Error Correction (codec internal)
+//
+
+int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
+  rtc::CritScope lock(&acm_crit_sect_);
+  if (HaveValidEncoder("SetPacketLossRate")) {
+    encoder_stack_->OnReceivedUplinkPacketLossFraction(loss_rate / 100.0);
+  }
+  return 0;
+}
+
+/////////////////////////////////////////
+//   Receiver
+//
+
+int AudioCodingModuleImpl::InitializeReceiver() {
+  rtc::CritScope lock(&acm_crit_sect_);
+  return InitializeReceiverSafe();
+}
+
+// Initialize receiver, resets codec database etc.
+int AudioCodingModuleImpl::InitializeReceiverSafe() {
+  // If the receiver is already initialized then we want to destroy any
+  // existing decoders. After a call to this function, we should have a clean
+  // start-up.
+  if (receiver_initialized_)
+    receiver_.RemoveAllCodecs();
+  receiver_.FlushBuffers();
+
+  receiver_initialized_ = true;
+  return 0;
+}
+
+void AudioCodingModuleImpl::SetReceiveCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  rtc::CritScope lock(&acm_crit_sect_);
+  receiver_.SetCodecs(codecs);
+}
+
+// Incoming packet from network parsed and ready for decode.
+int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
+                                          const size_t payload_length,
+                                          const RTPHeader& rtp_header) {
+  RTC_DCHECK_EQ(payload_length == 0, incoming_payload == nullptr);
+  return receiver_.InsertPacket(
+      rtp_header,
+      rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
+}
+
+// Get 10 milliseconds of raw audio data to play out.
+// Automatic resample to the requested frequency.
+int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
+                                           AudioFrame* audio_frame,
+                                           bool* muted) {
+  // GetAudio always returns 10 ms, at the requested sample rate.
+  if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
+    RTC_LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
+    return -1;
+  }
+  return 0;
+}
+
+/////////////////////////////////////////
+//   Statistics
+//
+
+// TODO(turajs) change the return value to void. Also change the corresponding
+// NetEq function.
+int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
+  receiver_.GetNetworkStatistics(statistics);
+  return 0;
+}
+
+bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
+  if (!encoder_stack_) {
+    RTC_LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";
+    return false;
+  }
+  return true;
+}
+
+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(
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
+    : neteq_config(),
+      clock(Clock::GetRealTimeClock()),
+      decoder_factory(decoder_factory) {
+  // Post-decode VAD is disabled by default in NetEq, however, Audio
+  // Conference Mixer relies on VAD decisions and fails without them.
+  neteq_config.enable_post_decode_vad = true;
+}
+
+AudioCodingModule::Config::Config(const Config&) = default;
+AudioCodingModule::Config::~Config() = default;
+
+AudioCodingModule* AudioCodingModule::Create(const Config& config) {
+  return new AudioCodingModuleImpl(config);
+}
+
+}  // namespace webrtc

modules/audio_coding/acm2/call_statistics.cc

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2013 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.
+ */
+
+#include "modules/audio_coding/acm2/call_statistics.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type,
+                                    bool muted) {
+  ++decoding_stat_.calls_to_neteq;
+  if (muted) {
+    ++decoding_stat_.decoded_muted_output;
+  }
+  switch (speech_type) {
+    case AudioFrame::kNormalSpeech: {
+      ++decoding_stat_.decoded_normal;
+      break;
+    }
+    case AudioFrame::kPLC: {
+      ++decoding_stat_.decoded_neteq_plc;
+      break;
+    }
+    case AudioFrame::kCodecPLC: {
+      ++decoding_stat_.decoded_codec_plc;
+      break;
+    }
+    case AudioFrame::kCNG: {
+      ++decoding_stat_.decoded_cng;
+      break;
+    }
+    case AudioFrame::kPLCCNG: {
+      ++decoding_stat_.decoded_plc_cng;
+      break;
+    }
+    case AudioFrame::kUndefined: {
+      // If the audio is decoded by NetEq, |kUndefined| is not an option.
+      RTC_NOTREACHED();
+    }
+  }
+}
+
+void CallStatistics::DecodedBySilenceGenerator() {
+  ++decoding_stat_.calls_to_silence_generator;
+}
+
+const AudioDecodingCallStats& CallStatistics::GetDecodingStatistics() const {
+  return decoding_stat_;
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc

modules/audio_coding/acm2/call_statistics.h

@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2013 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 MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+#define MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+
+#include "api/audio/audio_frame.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+
+//
+// This class is for book keeping of calls to ACM. It is not useful to log API
+// calls which are supposed to be called every 10ms, e.g. PlayoutData10Ms(),
+// however, it is useful to know the number of such calls in a given time
+// interval. The current implementation covers calls to PlayoutData10Ms() with
+// detailed accounting of the decoded speech type.
+//
+// Thread Safety
+// =============
+// Please note that this class in not thread safe. The class must be protected
+// if different APIs are called from different threads.
+//
+
+namespace webrtc {
+
+namespace acm2 {
+
+class CallStatistics {
+ public:
+  CallStatistics() {}
+  ~CallStatistics() {}
+
+  // Call this method to indicate that NetEq engaged in decoding. |speech_type|
+  // is the audio-type according to NetEq, and |muted| indicates if the decoded
+  // frame was produced in muted state.
+  void DecodedByNetEq(AudioFrame::SpeechType speech_type, bool muted);
+
+  // Call this method to indicate that a decoding call resulted in generating
+  // silence, i.e. call to NetEq is bypassed and the output audio is zero.
+  void DecodedBySilenceGenerator();
+
+  // Get statistics for decoding. The statistics include the number of calls to
+  // NetEq and silence generator, as well as the type of speech pulled of off
+  // NetEq, c.f. declaration of AudioDecodingCallStats for detailed description.
+  const AudioDecodingCallStats& GetDecodingStatistics() const;
+
+ private:
+  // Reset the decoding statistics.
+  void ResetDecodingStatistics();
+
+  AudioDecodingCallStats decoding_stat_;
+};
+
+}  // namespace acm2
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_

modules/audio_coding/acm2/call_statistics_unittest.cc

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2013 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.
+ */
+
+#include "modules/audio_coding/acm2/call_statistics.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+TEST(CallStatisticsTest, InitializedZero) {
+  CallStatistics call_stats;
+  AudioDecodingCallStats stats;
+
+  stats = call_stats.GetDecodingStatistics();
+  EXPECT_EQ(0, stats.calls_to_neteq);
+  EXPECT_EQ(0, stats.calls_to_silence_generator);
+  EXPECT_EQ(0, stats.decoded_normal);
+  EXPECT_EQ(0, stats.decoded_cng);
+  EXPECT_EQ(0, stats.decoded_neteq_plc);
+  EXPECT_EQ(0, stats.decoded_plc_cng);
+  EXPECT_EQ(0, stats.decoded_muted_output);
+}
+
+TEST(CallStatisticsTest, AllCalls) {
+  CallStatistics call_stats;
+  AudioDecodingCallStats stats;
+
+  call_stats.DecodedBySilenceGenerator();
+  call_stats.DecodedByNetEq(AudioFrame::kNormalSpeech, false);
+  call_stats.DecodedByNetEq(AudioFrame::kPLC, false);
+  call_stats.DecodedByNetEq(AudioFrame::kCodecPLC, false);
+  call_stats.DecodedByNetEq(AudioFrame::kPLCCNG, true);  // Let this be muted.
+  call_stats.DecodedByNetEq(AudioFrame::kCNG, false);
+
+  stats = call_stats.GetDecodingStatistics();
+  EXPECT_EQ(5, stats.calls_to_neteq);
+  EXPECT_EQ(1, stats.calls_to_silence_generator);
+  EXPECT_EQ(1, stats.decoded_normal);
+  EXPECT_EQ(1, stats.decoded_cng);
+  EXPECT_EQ(1, stats.decoded_neteq_plc);
+  EXPECT_EQ(1, stats.decoded_codec_plc);
+  EXPECT_EQ(1, stats.decoded_plc_cng);
+  EXPECT_EQ(1, stats.decoded_muted_output);
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc

modules/audio_coding/audio_coding.gni

@@ -0,0 +1,33 @@
+# Copyright (c) 2016 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.
+
+import("../../webrtc.gni")
+
+audio_codec_defines = []
+if (rtc_include_ilbc) {
+  audio_codec_defines += [ "WEBRTC_CODEC_ILBC" ]
+}
+if (rtc_include_opus) {
+  audio_codec_defines += [ "WEBRTC_CODEC_OPUS" ]
+}
+if (rtc_opus_support_120ms_ptime) {
+  audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
+} else {
+  audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
+}
+if (current_cpu == "arm") {
+  audio_codec_defines += [ "WEBRTC_CODEC_ISACFX" ]
+} else {
+  audio_codec_defines += [ "WEBRTC_CODEC_ISAC" ]
+}
+if (!build_with_mozilla && !build_with_chromium) {
+  audio_codec_defines += [ "WEBRTC_CODEC_RED" ]
+}
+
+audio_coding_defines = audio_codec_defines
+neteq_defines = audio_codec_defines

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc

@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+
+AudioEncoderRuntimeConfig::AudioEncoderRuntimeConfig() = default;
+
+AudioEncoderRuntimeConfig::AudioEncoderRuntimeConfig(
+    const AudioEncoderRuntimeConfig& other) = default;
+
+AudioEncoderRuntimeConfig::~AudioEncoderRuntimeConfig() = default;
+
+AudioEncoderRuntimeConfig& AudioEncoderRuntimeConfig::operator=(
+    const AudioEncoderRuntimeConfig& other) = default;
+
+bool AudioEncoderRuntimeConfig::operator==(
+    const AudioEncoderRuntimeConfig& other) const {
+  return bitrate_bps == other.bitrate_bps &&
+         frame_length_ms == other.frame_length_ms &&
+         uplink_packet_loss_fraction == other.uplink_packet_loss_fraction &&
+         enable_fec == other.enable_fec && enable_dtx == other.enable_dtx &&
+         num_channels == other.num_channels;
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc

@@ -0,0 +1,171 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kEventLogMinBitrateChangeBps = 5000;
+constexpr float kEventLogMinBitrateChangeFraction = 0.25;
+constexpr float kEventLogMinPacketLossChangeFraction = 0.5;
+}  // namespace
+
+AudioNetworkAdaptorImpl::Config::Config() : event_log(nullptr) {}
+
+AudioNetworkAdaptorImpl::Config::~Config() = default;
+
+AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
+    const Config& config,
+    std::unique_ptr<ControllerManager> controller_manager,
+    std::unique_ptr<DebugDumpWriter> debug_dump_writer)
+    : config_(config),
+      controller_manager_(std::move(controller_manager)),
+      debug_dump_writer_(std::move(debug_dump_writer)),
+      event_log_writer_(
+          config.event_log
+              ? new EventLogWriter(config.event_log,
+                                   kEventLogMinBitrateChangeBps,
+                                   kEventLogMinBitrateChangeFraction,
+                                   kEventLogMinPacketLossChangeFraction)
+              : nullptr) {
+  RTC_DCHECK(controller_manager_);
+}
+
+AudioNetworkAdaptorImpl::~AudioNetworkAdaptorImpl() = default;
+
+void AudioNetworkAdaptorImpl::SetUplinkBandwidth(int uplink_bandwidth_bps) {
+  last_metrics_.uplink_bandwidth_bps = uplink_bandwidth_bps;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+void AudioNetworkAdaptorImpl::SetUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  last_metrics_.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) {
+  last_metrics_.rtt_ms = rtt_ms;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.rtt_ms = rtt_ms;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+void AudioNetworkAdaptorImpl::SetTargetAudioBitrate(
+    int target_audio_bitrate_bps) {
+  last_metrics_.target_audio_bitrate_bps = target_audio_bitrate_bps;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+void AudioNetworkAdaptorImpl::SetOverhead(size_t overhead_bytes_per_packet) {
+  last_metrics_.overhead_bytes_per_packet = overhead_bytes_per_packet;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+AudioEncoderRuntimeConfig AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
+  AudioEncoderRuntimeConfig config;
+  for (auto& controller :
+       controller_manager_->GetSortedControllers(last_metrics_))
+    controller->MakeDecision(&config);
+
+  // Update ANA stats.
+  auto increment_opt = [](absl::optional<uint32_t>& a) {
+    a = a.value_or(0) + 1;
+  };
+  if (prev_config_) {
+    if (config.bitrate_bps != prev_config_->bitrate_bps) {
+      increment_opt(stats_.bitrate_action_counter);
+    }
+    if (config.enable_dtx != prev_config_->enable_dtx) {
+      increment_opt(stats_.dtx_action_counter);
+    }
+    if (config.enable_fec != prev_config_->enable_fec) {
+      increment_opt(stats_.fec_action_counter);
+    }
+    if (config.frame_length_ms && prev_config_->frame_length_ms) {
+      if (*config.frame_length_ms > *prev_config_->frame_length_ms) {
+        increment_opt(stats_.frame_length_increase_counter);
+      } else if (*config.frame_length_ms < *prev_config_->frame_length_ms) {
+        increment_opt(stats_.frame_length_decrease_counter);
+      }
+    }
+    if (config.num_channels != prev_config_->num_channels) {
+      increment_opt(stats_.channel_action_counter);
+    }
+    if (config.uplink_packet_loss_fraction) {
+      stats_.uplink_packet_loss_fraction = *config.uplink_packet_loss_fraction;
+    }
+  }
+  prev_config_ = config;
+
+  if (debug_dump_writer_)
+    debug_dump_writer_->DumpEncoderRuntimeConfig(config, rtc::TimeMillis());
+
+  if (event_log_writer_)
+    event_log_writer_->MaybeLogEncoderConfig(config);
+
+  return config;
+}
+
+void AudioNetworkAdaptorImpl::StartDebugDump(FILE* file_handle) {
+  debug_dump_writer_ = DebugDumpWriter::Create(file_handle);
+}
+
+void AudioNetworkAdaptorImpl::StopDebugDump() {
+  debug_dump_writer_.reset(nullptr);
+}
+
+ANAStats AudioNetworkAdaptorImpl::GetStats() const {
+  return stats_;
+}
+
+void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {
+  if (debug_dump_writer_)
+    debug_dump_writer_->DumpNetworkMetrics(last_metrics_, rtc::TimeMillis());
+}
+
+void AudioNetworkAdaptorImpl::UpdateNetworkMetrics(
+    const Controller::NetworkMetrics& network_metrics) {
+  for (auto& controller : controller_manager_->GetControllers())
+    controller->UpdateNetworkMetrics(network_metrics);
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h

@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class ControllerManager;
+class EventLogWriter;
+class RtcEventLog;
+
+class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
+ public:
+  struct Config {
+    Config();
+    ~Config();
+    RtcEventLog* event_log;
+  };
+
+  AudioNetworkAdaptorImpl(
+      const Config& config,
+      std::unique_ptr<ControllerManager> controller_manager,
+      std::unique_ptr<DebugDumpWriter> debug_dump_writer = nullptr);
+
+  ~AudioNetworkAdaptorImpl() override;
+
+  void SetUplinkBandwidth(int uplink_bandwidth_bps) override;
+
+  void SetUplinkPacketLossFraction(float uplink_packet_loss_fraction) override;
+
+  void SetRtt(int rtt_ms) override;
+
+  void SetTargetAudioBitrate(int target_audio_bitrate_bps) override;
+
+  void SetOverhead(size_t overhead_bytes_per_packet) override;
+
+  AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() override;
+
+  void StartDebugDump(FILE* file_handle) override;
+
+  void StopDebugDump() override;
+
+  ANAStats GetStats() const override;
+
+ private:
+  void DumpNetworkMetrics();
+
+  void UpdateNetworkMetrics(const Controller::NetworkMetrics& network_metrics);
+
+  const Config config_;
+
+  std::unique_ptr<ControllerManager> controller_manager_;
+
+  std::unique_ptr<DebugDumpWriter> debug_dump_writer_;
+
+  const std::unique_ptr<EventLogWriter> event_log_writer_;
+
+  Controller::NetworkMetrics last_metrics_;
+
+  absl::optional<AudioEncoderRuntimeConfig> prev_config_;
+
+  ANAStats stats_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioNetworkAdaptorImpl);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc

@@ -0,0 +1,306 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+
+#include <utility>
+#include <vector>
+
+#include "api/rtc_event_log/rtc_event.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
+#include "rtc_base/fake_clock.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+namespace {
+
+constexpr size_t kNumControllers = 2;
+
+constexpr int64_t kClockInitialTimeMs = 12345678;
+
+MATCHER_P(NetworkMetricsIs, metric, "") {
+  return arg.uplink_bandwidth_bps == metric.uplink_bandwidth_bps &&
+         arg.target_audio_bitrate_bps == metric.target_audio_bitrate_bps &&
+         arg.rtt_ms == metric.rtt_ms &&
+         arg.overhead_bytes_per_packet == metric.overhead_bytes_per_packet &&
+         arg.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction;
+}
+
+MATCHER_P(IsRtcEventAnaConfigEqualTo, config, "") {
+  if (arg->GetType() != RtcEvent::Type::AudioNetworkAdaptation) {
+    return false;
+  }
+  auto ana_event = static_cast<RtcEventAudioNetworkAdaptation*>(arg);
+  return ana_event->config() == config;
+}
+
+MATCHER_P(EncoderRuntimeConfigIs, config, "") {
+  return arg.bitrate_bps == config.bitrate_bps &&
+         arg.frame_length_ms == config.frame_length_ms &&
+         arg.uplink_packet_loss_fraction ==
+             config.uplink_packet_loss_fraction &&
+         arg.enable_fec == config.enable_fec &&
+         arg.enable_dtx == config.enable_dtx &&
+         arg.num_channels == config.num_channels;
+}
+
+struct AudioNetworkAdaptorStates {
+  std::unique_ptr<AudioNetworkAdaptorImpl> audio_network_adaptor;
+  std::vector<std::unique_ptr<MockController>> mock_controllers;
+  std::unique_ptr<MockRtcEventLog> event_log;
+  MockDebugDumpWriter* mock_debug_dump_writer;
+};
+
+AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() {
+  AudioNetworkAdaptorStates states;
+  std::vector<Controller*> controllers;
+  for (size_t i = 0; i < kNumControllers; ++i) {
+    auto controller =
+        std::unique_ptr<MockController>(new NiceMock<MockController>());
+    EXPECT_CALL(*controller, Die());
+    controllers.push_back(controller.get());
+    states.mock_controllers.push_back(std::move(controller));
+  }
+
+  auto controller_manager = std::unique_ptr<MockControllerManager>(
+      new NiceMock<MockControllerManager>());
+
+  EXPECT_CALL(*controller_manager, Die());
+  EXPECT_CALL(*controller_manager, GetControllers())
+      .WillRepeatedly(Return(controllers));
+  EXPECT_CALL(*controller_manager, GetSortedControllers(_))
+      .WillRepeatedly(Return(controllers));
+
+  states.event_log.reset(new NiceMock<MockRtcEventLog>());
+
+  auto debug_dump_writer =
+      std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
+  EXPECT_CALL(*debug_dump_writer, Die());
+  states.mock_debug_dump_writer = debug_dump_writer.get();
+
+  AudioNetworkAdaptorImpl::Config config;
+  config.event_log = states.event_log.get();
+  // AudioNetworkAdaptorImpl governs the lifetime of controller manager.
+  states.audio_network_adaptor.reset(new AudioNetworkAdaptorImpl(
+      config, std::move(controller_manager), std::move(debug_dump_writer)));
+
+  return states;
+}
+
+void SetExpectCallToUpdateNetworkMetrics(
+    const std::vector<std::unique_ptr<MockController>>& controllers,
+    const Controller::NetworkMetrics& check) {
+  for (auto& mock_controller : controllers) {
+    EXPECT_CALL(*mock_controller,
+                UpdateNetworkMetrics(NetworkMetricsIs(check)));
+  }
+}
+
+}  // namespace
+
+TEST(AudioNetworkAdaptorImplTest,
+     UpdateNetworkMetricsIsCalledOnSetUplinkBandwidth) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr int kBandwidth = 16000;
+  Controller::NetworkMetrics check;
+  check.uplink_bandwidth_bps = kBandwidth;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     UpdateNetworkMetricsIsCalledOnSetUplinkPacketLossFraction) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr float kPacketLoss = 0.7f;
+  Controller::NetworkMetrics check;
+  check.uplink_packet_loss_fraction = kPacketLoss;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
+}
+
+TEST(AudioNetworkAdaptorImplTest, UpdateNetworkMetricsIsCalledOnSetRtt) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr int kRtt = 100;
+  Controller::NetworkMetrics check;
+  check.rtt_ms = kRtt;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetRtt(kRtt);
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     UpdateNetworkMetricsIsCalledOnSetTargetAudioBitrate) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr int kTargetAudioBitrate = 15000;
+  Controller::NetworkMetrics check;
+  check.target_audio_bitrate_bps = kTargetAudioBitrate;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
+}
+
+TEST(AudioNetworkAdaptorImplTest, UpdateNetworkMetricsIsCalledOnSetOverhead) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr size_t kOverhead = 64;
+  Controller::NetworkMetrics check;
+  check.overhead_bytes_per_packet = kOverhead;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetOverhead(kOverhead);
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     MakeDecisionIsCalledOnGetEncoderRuntimeConfig) {
+  auto states = CreateAudioNetworkAdaptor();
+  for (auto& mock_controller : states.mock_controllers)
+    EXPECT_CALL(*mock_controller, MakeDecision(_));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
+      "Enabled/");
+  rtc::ScopedFakeClock fake_clock;
+  fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
+  auto states = CreateAudioNetworkAdaptor();
+  AudioEncoderRuntimeConfig config;
+  config.bitrate_bps = 32000;
+  config.enable_fec = true;
+
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config));
+
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpEncoderRuntimeConfig(EncoderRuntimeConfigIs(config),
+                                       kClockInitialTimeMs));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
+  rtc::ScopedFakeClock fake_clock;
+  fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
+
+  auto states = CreateAudioNetworkAdaptor();
+
+  constexpr int kBandwidth = 16000;
+  constexpr float kPacketLoss = 0.7f;
+  constexpr int kRtt = 100;
+  constexpr int kTargetAudioBitrate = 15000;
+  constexpr size_t kOverhead = 64;
+
+  Controller::NetworkMetrics check;
+  check.uplink_bandwidth_bps = kBandwidth;
+  int64_t timestamp_check = kClockInitialTimeMs;
+
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(100));
+  timestamp_check += 100;
+  check.uplink_packet_loss_fraction = kPacketLoss;
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(50));
+  timestamp_check += 50;
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(200));
+  timestamp_check += 200;
+  check.rtt_ms = kRtt;
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetRtt(kRtt);
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(150));
+  timestamp_check += 150;
+  check.target_audio_bitrate_bps = kTargetAudioBitrate;
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(50));
+  timestamp_check += 50;
+  check.overhead_bytes_per_packet = kOverhead;
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetOverhead(kOverhead);
+}
+
+TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
+      "Enabled/");
+  auto states = CreateAudioNetworkAdaptor();
+
+  AudioEncoderRuntimeConfig config;
+  config.bitrate_bps = 32000;
+  config.enable_fec = true;
+
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config));
+
+  EXPECT_CALL(*states.event_log, LogProxy(IsRtcEventAnaConfigEqualTo(config)))
+      .Times(1);
+  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 = 32000;
+  config1.num_channels = 2;
+  config1.enable_fec = true;
+  config1.enable_dtx = true;
+  config1.frame_length_ms = 120;
+  config1.uplink_packet_loss_fraction = 0.1f;
+  config2.bitrate_bps = 16000;
+  config2.num_channels = 1;
+  config2.enable_fec = false;
+  config2.enable_dtx = false;
+  config2.frame_length_ms = 60;
+  config1.uplink_packet_loss_fraction = 0.1f;
+
+  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();
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config1));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+
+  auto ana_stats = states.audio_network_adaptor->GetStats();
+
+  EXPECT_EQ(ana_stats.bitrate_action_counter, 2u);
+  EXPECT_EQ(ana_stats.channel_action_counter, 2u);
+  EXPECT_EQ(ana_stats.dtx_action_counter, 2u);
+  EXPECT_EQ(ana_stats.fec_action_counter, 2u);
+  EXPECT_EQ(ana_stats.frame_length_increase_counter, 1u);
+  EXPECT_EQ(ana_stats.frame_length_decrease_counter, 1u);
+  EXPECT_EQ(ana_stats.uplink_packet_loss_fraction, 0.1f);
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/bitrate_controller.cc

@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace audio_network_adaptor {
+
+BitrateController::Config::Config(int initial_bitrate_bps,
+                                  int initial_frame_length_ms,
+                                  int fl_increase_overhead_offset,
+                                  int fl_decrease_overhead_offset)
+    : initial_bitrate_bps(initial_bitrate_bps),
+      initial_frame_length_ms(initial_frame_length_ms),
+      fl_increase_overhead_offset(fl_increase_overhead_offset),
+      fl_decrease_overhead_offset(fl_decrease_overhead_offset) {}
+
+BitrateController::Config::~Config() = default;
+
+BitrateController::BitrateController(const Config& config)
+    : config_(config),
+      bitrate_bps_(config_.initial_bitrate_bps),
+      frame_length_ms_(config_.initial_frame_length_ms) {
+  RTC_DCHECK_GT(bitrate_bps_, 0);
+  RTC_DCHECK_GT(frame_length_ms_, 0);
+}
+
+BitrateController::~BitrateController() = default;
+
+void BitrateController::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.target_audio_bitrate_bps)
+    target_audio_bitrate_bps_ = network_metrics.target_audio_bitrate_bps;
+  if (network_metrics.overhead_bytes_per_packet) {
+    RTC_DCHECK_GT(*network_metrics.overhead_bytes_per_packet, 0);
+    overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet;
+  }
+}
+
+void BitrateController::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  // Decision on |bitrate_bps| should not have been made.
+  RTC_DCHECK(!config->bitrate_bps);
+  if (target_audio_bitrate_bps_ && overhead_bytes_per_packet_) {
+    // Current implementation of BitrateController can only work when
+    // |metrics.target_audio_bitrate_bps| includes overhead is enabled. This is
+    // currently governed by the following field trial.
+    RTC_DCHECK(
+        webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead"));
+    if (config->frame_length_ms)
+      frame_length_ms_ = *config->frame_length_ms;
+    int offset = config->last_fl_change_increase
+                     ? config_.fl_increase_overhead_offset
+                     : config_.fl_decrease_overhead_offset;
+    // Check that
+    // -(*overhead_bytes_per_packet_) <= offset <= (*overhead_bytes_per_packet_)
+    RTC_DCHECK_GE(*overhead_bytes_per_packet_, -offset);
+    RTC_DCHECK_LE(offset, *overhead_bytes_per_packet_);
+    int overhead_rate_bps = static_cast<int>(
+        (*overhead_bytes_per_packet_ + offset) * 8 * 1000 / frame_length_ms_);
+    bitrate_bps_ = std::max(0, *target_audio_bitrate_bps_ - overhead_rate_bps);
+  }
+  config->bitrate_bps = bitrate_bps_;
+}
+
+}  // namespace audio_network_adaptor
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/bitrate_controller.h

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+namespace audio_network_adaptor {
+
+class BitrateController final : public Controller {
+ public:
+  struct Config {
+    Config(int initial_bitrate_bps,
+           int initial_frame_length_ms,
+           int fl_increase_overhead_offset,
+           int fl_decrease_overhead_offset);
+    ~Config();
+    int initial_bitrate_bps;
+    int initial_frame_length_ms;
+    int fl_increase_overhead_offset;
+    int fl_decrease_overhead_offset;
+  };
+
+  explicit BitrateController(const Config& config);
+
+  ~BitrateController() override;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  const Config config_;
+  int bitrate_bps_;
+  int frame_length_ms_;
+  absl::optional<int> target_audio_bitrate_bps_;
+  absl::optional<size_t> overhead_bytes_per_packet_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(BitrateController);
+};
+
+}  // namespace audio_network_adaptor
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc

@@ -0,0 +1,250 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace audio_network_adaptor {
+
+namespace {
+
+void UpdateNetworkMetrics(
+    BitrateController* controller,
+    const absl::optional<int>& target_audio_bitrate_bps,
+    const absl::optional<size_t>& overhead_bytes_per_packet) {
+  // UpdateNetworkMetrics can accept multiple network metric updates at once.
+  // However, currently, the most used case is to update one metric at a time.
+  // To reflect this fact, we separate the calls.
+  if (target_audio_bitrate_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  if (overhead_bytes_per_packet) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+}
+
+void CheckDecision(BitrateController* controller,
+                   const absl::optional<int>& frame_length_ms,
+                   int expected_bitrate_bps) {
+  AudioEncoderRuntimeConfig config;
+  config.frame_length_ms = frame_length_ms;
+  controller->MakeDecision(&config);
+  EXPECT_EQ(expected_bitrate_bps, config.bitrate_bps);
+}
+
+}  // namespace
+
+// These tests are named AnaBitrateControllerTest to distinguish from
+// BitrateControllerTest in
+// modules/bitrate_controller/bitrate_controller_unittest.cc.
+
+TEST(AnaBitrateControllerTest, OutputInitValueWhenTargetBitrateUnknown) {
+  constexpr int kInitialBitrateBps = 32000;
+  constexpr int kInitialFrameLengthMs = 20;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  BitrateController controller(BitrateController::Config(
+      kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
+  UpdateNetworkMetrics(&controller, absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, OutputInitValueWhenOverheadUnknown) {
+  constexpr int kInitialBitrateBps = 32000;
+  constexpr int kInitialFrameLengthMs = 20;
+  constexpr int kTargetBitrateBps = 48000;
+  BitrateController controller(BitrateController::Config(
+      kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, absl::nullopt);
+  CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, ChangeBitrateOnTargetBitrateChanged) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  constexpr int kInitialFrameLengthMs = 20;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  // Frame length unchanged, bitrate changes in accordance with
+  // |metrics.target_audio_bitrate_bps| and |metrics.overhead_bytes_per_packet|.
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kInitialFrameLengthMs, kBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, UpdateMultipleNetworkMetricsAtOnce) {
+  // This test is similar to ChangeBitrateOnTargetBitrateChanged. But instead of
+  // using ::UpdateNetworkMetrics(...), which calls
+  // BitrateController::UpdateNetworkMetrics(...) multiple times, we
+  // we call it only once. This is to verify that
+  // BitrateController::UpdateNetworkMetrics(...) can handle multiple
+  // network updates at once. This is, however, not a common use case in current
+  // audio_network_adaptor_impl.cc.
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  constexpr int kInitialFrameLengthMs = 20;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.target_audio_bitrate_bps = kTargetBitrateBps;
+  network_metrics.overhead_bytes_per_packet = kOverheadBytesPerPacket;
+  controller.UpdateNetworkMetrics(network_metrics);
+  CheckDecision(&controller, kInitialFrameLengthMs, kBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, TreatUnknownFrameLengthAsFrameLengthUnchanged) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  constexpr int kInitialFrameLengthMs = 20;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, absl::nullopt, kBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, IncreaseBitrateOnFrameLengthIncreased) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  constexpr int kInitialFrameLengthMs = 20;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, absl::nullopt, kBitrateBps);
+
+  constexpr int kFrameLengthMs = 60;
+  constexpr size_t kPacketOverheadRateDiff =
+      kOverheadBytesPerPacket * 8 * 1000 / 20 -
+      kOverheadBytesPerPacket * 8 * 1000 / 60;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kFrameLengthMs,
+                kBitrateBps + kPacketOverheadRateDiff);
+}
+
+TEST(AnaBitrateControllerTest, DecreaseBitrateOnFrameLengthDecreased) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  constexpr int kInitialFrameLengthMs = 60;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, absl::nullopt, kBitrateBps);
+
+  constexpr int kFrameLengthMs = 20;
+  constexpr size_t kPacketOverheadRateDiff =
+      kOverheadBytesPerPacket * 8 * 1000 / 20 -
+      kOverheadBytesPerPacket * 8 * 1000 / 60;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kFrameLengthMs,
+                kBitrateBps - kPacketOverheadRateDiff);
+}
+
+TEST(AnaBitrateControllerTest, BitrateNeverBecomesNegative) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  BitrateController controller(BitrateController::Config(32000, 20, 0, 0));
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kFrameLengthMs = 60;
+  // Set a target rate smaller than overhead rate, the bitrate is bounded by 0.
+  constexpr int kTargetBitrateBps =
+      kOverheadBytesPerPacket * 8 * 1000 / kFrameLengthMs - 1;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kFrameLengthMs, 0);
+}
+
+TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  BitrateController controller(BitrateController::Config(32000, 20, 0, 0));
+
+  // Start from an arbitrary overall bitrate.
+  int overall_bitrate = 34567;
+  size_t overhead_bytes_per_packet = 64;
+  int frame_length_ms = 20;
+  int current_bitrate = rtc::checked_cast<int>(
+      overall_bitrate - overhead_bytes_per_packet * 8 * 1000 / frame_length_ms);
+
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: increase overall bitrate.
+  overall_bitrate += 100;
+  current_bitrate += 100;
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: change frame length.
+  frame_length_ms = 60;
+  current_bitrate +=
+      rtc::checked_cast<int>(overhead_bytes_per_packet * 8 * 1000 / 20 -
+                             overhead_bytes_per_packet * 8 * 1000 / 60);
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: change overhead.
+  overhead_bytes_per_packet -= 30;
+  current_bitrate += 30 * 8 * 1000 / frame_length_ms;
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: change frame length.
+  frame_length_ms = 20;
+  current_bitrate -=
+      rtc::checked_cast<int>(overhead_bytes_per_packet * 8 * 1000 / 20 -
+                             overhead_bytes_per_packet * 8 * 1000 / 60);
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: decrease overall bitrate and frame length.
+  overall_bitrate -= 100;
+  current_bitrate -= 100;
+  frame_length_ms = 60;
+  current_bitrate +=
+      rtc::checked_cast<int>(overhead_bytes_per_packet * 8 * 1000 / 20 -
+                             overhead_bytes_per_packet * 8 * 1000 / 60);
+
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+}
+
+}  // namespace audio_network_adaptor
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/channel_controller.cc

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+ChannelController::Config::Config(size_t num_encoder_channels,
+                                  size_t intial_channels_to_encode,
+                                  int channel_1_to_2_bandwidth_bps,
+                                  int channel_2_to_1_bandwidth_bps)
+    : num_encoder_channels(num_encoder_channels),
+      intial_channels_to_encode(intial_channels_to_encode),
+      channel_1_to_2_bandwidth_bps(channel_1_to_2_bandwidth_bps),
+      channel_2_to_1_bandwidth_bps(channel_2_to_1_bandwidth_bps) {}
+
+ChannelController::ChannelController(const Config& config)
+    : config_(config), channels_to_encode_(config_.intial_channels_to_encode) {
+  RTC_DCHECK_GT(config_.intial_channels_to_encode, 0lu);
+  // Currently, we require |intial_channels_to_encode| to be <= 2.
+  RTC_DCHECK_LE(config_.intial_channels_to_encode, 2lu);
+  RTC_DCHECK_GE(config_.num_encoder_channels,
+                config_.intial_channels_to_encode);
+}
+
+ChannelController::~ChannelController() = default;
+
+void ChannelController::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.uplink_bandwidth_bps)
+    uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+}
+
+void ChannelController::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  // Decision on |num_channels| should not have been made.
+  RTC_DCHECK(!config->num_channels);
+
+  if (uplink_bandwidth_bps_) {
+    if (channels_to_encode_ == 2 &&
+        *uplink_bandwidth_bps_ <= config_.channel_2_to_1_bandwidth_bps) {
+      channels_to_encode_ = 1;
+    } else if (channels_to_encode_ == 1 &&
+               *uplink_bandwidth_bps_ >= config_.channel_1_to_2_bandwidth_bps) {
+      channels_to_encode_ =
+          std::min(static_cast<size_t>(2), config_.num_encoder_channels);
+    }
+  }
+  config->num_channels = channels_to_encode_;
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/channel_controller.h

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class ChannelController final : public Controller {
+ public:
+  struct Config {
+    Config(size_t num_encoder_channels,
+           size_t intial_channels_to_encode,
+           int channel_1_to_2_bandwidth_bps,
+           int channel_2_to_1_bandwidth_bps);
+    size_t num_encoder_channels;
+    size_t intial_channels_to_encode;
+    // Uplink bandwidth above which the number of encoded channels should switch
+    // from 1 to 2.
+    int channel_1_to_2_bandwidth_bps;
+    // Uplink bandwidth below which the number of encoded channels should switch
+    // from 2 to 1.
+    int channel_2_to_1_bandwidth_bps;
+  };
+
+  explicit ChannelController(const Config& config);
+
+  ~ChannelController() override;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  const Config config_;
+  size_t channels_to_encode_;
+  absl::optional<int> uplink_bandwidth_bps_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelController);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc

@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kNumChannels = 2;
+constexpr int kChannel1To2BandwidthBps = 31000;
+constexpr int kChannel2To1BandwidthBps = 29000;
+constexpr int kMediumBandwidthBps =
+    (kChannel1To2BandwidthBps + kChannel2To1BandwidthBps) / 2;
+
+std::unique_ptr<ChannelController> CreateChannelController(int init_channels) {
+  std::unique_ptr<ChannelController> controller(
+      new ChannelController(ChannelController::Config(
+          kNumChannels, init_channels, kChannel1To2BandwidthBps,
+          kChannel2To1BandwidthBps)));
+  return controller;
+}
+
+void CheckDecision(ChannelController* controller,
+                   const absl::optional<int>& uplink_bandwidth_bps,
+                   size_t expected_num_channels) {
+  if (uplink_bandwidth_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  AudioEncoderRuntimeConfig config;
+  controller->MakeDecision(&config);
+  EXPECT_EQ(expected_num_channels, config.num_channels);
+}
+
+}  // namespace
+
+TEST(ChannelControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
+  constexpr int kInitChannels = 2;
+  auto controller = CreateChannelController(kInitChannels);
+  CheckDecision(controller.get(), absl::nullopt, kInitChannels);
+}
+
+TEST(ChannelControllerTest, SwitchTo2ChannelsOnHighUplinkBandwidth) {
+  constexpr int kInitChannels = 1;
+  auto controller = CreateChannelController(kInitChannels);
+  // Use high bandwidth to check output switch to 2.
+  CheckDecision(controller.get(), kChannel1To2BandwidthBps, 2);
+}
+
+TEST(ChannelControllerTest, SwitchTo1ChannelOnLowUplinkBandwidth) {
+  constexpr int kInitChannels = 2;
+  auto controller = CreateChannelController(kInitChannels);
+  // Use low bandwidth to check output switch to 1.
+  CheckDecision(controller.get(), kChannel2To1BandwidthBps, 1);
+}
+
+TEST(ChannelControllerTest, Maintain1ChannelOnMediumUplinkBandwidth) {
+  constexpr int kInitChannels = 1;
+  auto controller = CreateChannelController(kInitChannels);
+  // Use between-thresholds bandwidth to check output remains at 1.
+  CheckDecision(controller.get(), kMediumBandwidthBps, 1);
+}
+
+TEST(ChannelControllerTest, Maintain2ChannelsOnMediumUplinkBandwidth) {
+  constexpr int kInitChannels = 2;
+  auto controller = CreateChannelController(kInitChannels);
+  // Use between-thresholds bandwidth to check output remains at 2.
+  CheckDecision(controller.get(), kMediumBandwidthBps, 2);
+}
+
+TEST(ChannelControllerTest, CheckBehaviorOnChangingUplinkBandwidth) {
+  constexpr int kInitChannels = 1;
+  auto controller = CreateChannelController(kInitChannels);
+
+  // Use between-thresholds bandwidth to check output remains at 1.
+  CheckDecision(controller.get(), kMediumBandwidthBps, 1);
+
+  // Use high bandwidth to check output switch to 2.
+  CheckDecision(controller.get(), kChannel1To2BandwidthBps, 2);
+
+  // Use between-thresholds bandwidth to check output remains at 2.
+  CheckDecision(controller.get(), kMediumBandwidthBps, 2);
+
+  // Use low bandwidth to check output switch to 1.
+  CheckDecision(controller.get(), kChannel2To1BandwidthBps, 1);
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/config.proto

@@ -0,0 +1,180 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+option java_package = "org.webrtc.AudioNetworkAdaptor";
+option java_outer_classname = "Config";
+package webrtc.audio_network_adaptor.config;
+
+message FecController {
+  message Threshold {
+    // Threshold defines a curve in the bandwidth/packet-loss domain. The
+    // curve is characterized by the two conjunction points: A and B.
+    //
+    // packet ^  |
+    //  loss  | A|
+    //        |   \        A: (low_bandwidth_bps, low_bandwidth_packet_loss)
+    //        |    \       B: (high_bandwidth_bps, high_bandwidth_packet_loss)
+    //        |    B\________
+    //        |---------------> bandwidth
+    optional int32 low_bandwidth_bps = 1;
+    optional float low_bandwidth_packet_loss = 2;
+    optional int32 high_bandwidth_bps = 3;
+    optional float high_bandwidth_packet_loss = 4;
+  }
+
+  // |fec_enabling_threshold| defines a curve, above which FEC should be
+  // enabled. |fec_disabling_threshold| defines a curve, under which FEC
+  // should be disabled. See below
+  //
+  // packet-loss ^   |  |
+  //             |   |  |   FEC
+  //             |    \  \   ON
+  //             | FEC \  \_______ fec_enabling_threshold
+  //             | OFF  \_________ fec_disabling_threshold
+  //             |-----------------> bandwidth
+  optional Threshold fec_enabling_threshold = 1;
+  optional Threshold fec_disabling_threshold = 2;
+
+  // |time_constant_ms| is the time constant for an exponential filter, which
+  // is used for smoothing the packet loss fraction.
+  optional int32 time_constant_ms = 3;
+}
+
+message FecControllerRplrBased {
+  message Threshold {
+    // Threshold defines a curve in the bandwidth/recoverable-packet-loss
+    // domain.
+    // The curve is characterized by the two conjunction points: A and B.
+    //
+    // recoverable ^
+    // packet      |  |
+    //  loss       | A|
+    //             |   \        A: (low_bandwidth_bps,
+    //             |    \           low_bandwidth_recoverable_packet_loss)
+    //             |     \       B: (high_bandwidth_bps,
+    //             |      \          high_bandwidth_recoverable_packet_loss)
+    //             |      B\________
+    //             |---------------> bandwidth
+    optional int32 low_bandwidth_bps = 1;
+    optional float low_bandwidth_recoverable_packet_loss = 2;
+    optional int32 high_bandwidth_bps = 3;
+    optional float high_bandwidth_recoverable_packet_loss = 4;
+  }
+
+  // |fec_enabling_threshold| defines a curve, above which FEC should be
+  // enabled. |fec_disabling_threshold| defines a curve, under which FEC
+  // should be disabled. See below
+  //
+  // packet-loss ^   |  |
+  //             |   |  |   FEC
+  //             |    \  \   ON
+  //             | FEC \  \_______ fec_enabling_threshold
+  //             | OFF  \_________ fec_disabling_threshold
+  //             |-----------------> bandwidth
+  optional Threshold fec_enabling_threshold = 1;
+  optional Threshold fec_disabling_threshold = 2;
+}
+
+message FrameLengthController {
+  // Uplink packet loss fraction below which frame length can increase.
+  optional float fl_increasing_packet_loss_fraction = 1;
+
+  // Uplink packet loss fraction above which frame length should decrease.
+  optional float fl_decreasing_packet_loss_fraction = 2;
+
+  // Uplink bandwidth below which frame length can switch from 20ms to 60ms.
+  optional int32 fl_20ms_to_60ms_bandwidth_bps = 3;
+
+  // Uplink bandwidth above which frame length should switch from 60ms to 20ms.
+  optional int32 fl_60ms_to_20ms_bandwidth_bps = 4;
+
+  // Uplink bandwidth below which frame length can switch from 60ms to 120ms.
+  optional int32 fl_60ms_to_120ms_bandwidth_bps = 5;
+
+  // Uplink bandwidth above which frame length should switch from 120ms to 60ms.
+  optional int32 fl_120ms_to_60ms_bandwidth_bps = 6;
+
+  // Offset to apply to the per-packet overhead when increasing frame length.
+  optional int32 fl_increase_overhead_offset = 7;
+
+  // Offset to apply to the per-packet overhead when decreasing frame length.
+  optional int32 fl_decrease_overhead_offset = 8;
+
+  // Uplink bandwidth below which frame length can switch from 20ms to 40ms. In
+  // current implementation, defining this will invalidate
+  // fl_20ms_to_60ms_bandwidth_bps.
+  optional int32 fl_20ms_to_40ms_bandwidth_bps = 9;
+
+  // Uplink bandwidth above which frame length should switch from 40ms to 20ms.
+  optional int32 fl_40ms_to_20ms_bandwidth_bps = 10;
+
+  // Uplink bandwidth below which frame length can switch from 40ms to 60ms.
+  optional int32 fl_40ms_to_60ms_bandwidth_bps = 11;
+
+  // Uplink bandwidth above which frame length should switch from 60ms to 40ms.
+  // In current implementation, defining this will invalidate
+  // fl_60ms_to_20ms_bandwidth_bps.
+  optional int32 fl_60ms_to_40ms_bandwidth_bps = 12;
+}
+
+message ChannelController {
+  // Uplink bandwidth above which the number of encoded channels should switch
+  // from 1 to 2.
+  optional int32 channel_1_to_2_bandwidth_bps = 1;
+
+  // Uplink bandwidth below which the number of encoded channels should switch
+  // from 2 to 1.
+  optional int32 channel_2_to_1_bandwidth_bps = 2;
+}
+
+message DtxController {
+  // Uplink bandwidth below which DTX should be switched on.
+  optional int32 dtx_enabling_bandwidth_bps = 1;
+
+  // Uplink bandwidth above which DTX should be switched off.
+  optional int32 dtx_disabling_bandwidth_bps = 2;
+}
+
+message BitrateController {
+  // Offset to apply to per-packet overhead when the frame length is increased.
+  optional int32 fl_increase_overhead_offset = 1;
+  // Offset to apply to per-packet overhead when the frame length is decreased.
+  optional int32 fl_decrease_overhead_offset = 2;
+}
+
+message Controller {
+  message ScoringPoint {
+    // |ScoringPoint| is a subspace of network condition. It is used for
+    // comparing the significance of controllers.
+    optional int32 uplink_bandwidth_bps = 1;
+    optional float uplink_packet_loss_fraction = 2;
+  }
+
+  // The distance from |scoring_point| to a given network condition defines
+  // the significance of this controller with respect that network condition.
+  // Shorter distance means higher significance. The significances of
+  // controllers determine their order in the processing pipeline. Controllers
+  // without |scoring_point| follow their default order in
+  // |ControllerManager::controllers|.
+  optional ScoringPoint scoring_point = 1;
+
+  oneof controller {
+    FecController fec_controller = 21;
+    FrameLengthController frame_length_controller = 22;
+    ChannelController channel_controller = 23;
+    DtxController dtx_controller = 24;
+    BitrateController bitrate_controller = 25;
+    FecControllerRplrBased fec_controller_rplr_based = 26;
+  }
+}
+
+message ControllerManager {
+  repeated Controller controllers = 1;
+
+  // Least time since last reordering for a new reordering to be made.
+  optional int32 min_reordering_time_ms = 2;
+
+  // Least squared distance from last scoring point for a new reordering to be
+  // made.
+  optional float min_reordering_squared_distance = 3;
+}
+

modules/audio_coding/audio_network_adaptor/controller.cc

@@ -0,0 +1,19 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+
+namespace webrtc {
+
+Controller::NetworkMetrics::NetworkMetrics() = default;
+
+Controller::NetworkMetrics::~NetworkMetrics() = default;
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/controller.h

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+
+namespace webrtc {
+
+class Controller {
+ public:
+  struct NetworkMetrics {
+    NetworkMetrics();
+    ~NetworkMetrics();
+    absl::optional<int> uplink_bandwidth_bps;
+    absl::optional<float> uplink_packet_loss_fraction;
+    absl::optional<int> target_audio_bitrate_bps;
+    absl::optional<int> rtt_ms;
+    absl::optional<size_t> overhead_bytes_per_packet;
+  };
+
+  virtual ~Controller() = default;
+
+  // Informs network metrics update to this controller. Any non-empty field
+  // indicates an update on the corresponding network metric.
+  virtual void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) = 0;
+
+  virtual void MakeDecision(AudioEncoderRuntimeConfig* config) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/controller_manager.cc

@@ -0,0 +1,437 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+
+#include <cmath>
+#include <string>
+#include <utility>
+
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+namespace {
+
+#if WEBRTC_ENABLE_PROTOBUF
+
+std::unique_ptr<FecControllerPlrBased> CreateFecControllerPlrBased(
+    const audio_network_adaptor::config::FecController& config,
+    bool initial_fec_enabled) {
+  RTC_CHECK(config.has_fec_enabling_threshold());
+  RTC_CHECK(config.has_fec_disabling_threshold());
+  RTC_CHECK(config.has_time_constant_ms());
+
+  auto& fec_enabling_threshold = config.fec_enabling_threshold();
+  RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_bps());
+  RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_packet_loss());
+  RTC_CHECK(fec_enabling_threshold.has_high_bandwidth_bps());
+  RTC_CHECK(fec_enabling_threshold.has_high_bandwidth_packet_loss());
+
+  auto& fec_disabling_threshold = config.fec_disabling_threshold();
+  RTC_CHECK(fec_disabling_threshold.has_low_bandwidth_bps());
+  RTC_CHECK(fec_disabling_threshold.has_low_bandwidth_packet_loss());
+  RTC_CHECK(fec_disabling_threshold.has_high_bandwidth_bps());
+  RTC_CHECK(fec_disabling_threshold.has_high_bandwidth_packet_loss());
+
+  return std::unique_ptr<FecControllerPlrBased>(
+      new FecControllerPlrBased(FecControllerPlrBased::Config(
+          initial_fec_enabled,
+          ThresholdCurve(fec_enabling_threshold.low_bandwidth_bps(),
+                         fec_enabling_threshold.low_bandwidth_packet_loss(),
+                         fec_enabling_threshold.high_bandwidth_bps(),
+                         fec_enabling_threshold.high_bandwidth_packet_loss()),
+          ThresholdCurve(fec_disabling_threshold.low_bandwidth_bps(),
+                         fec_disabling_threshold.low_bandwidth_packet_loss(),
+                         fec_disabling_threshold.high_bandwidth_bps(),
+                         fec_disabling_threshold.high_bandwidth_packet_loss()),
+          config.time_constant_ms())));
+}
+
+std::unique_ptr<FrameLengthController> CreateFrameLengthController(
+    const audio_network_adaptor::config::FrameLengthController& config,
+    rtc::ArrayView<const int> encoder_frame_lengths_ms,
+    int initial_frame_length_ms,
+    int min_encoder_bitrate_bps) {
+  RTC_CHECK(config.has_fl_increasing_packet_loss_fraction());
+  RTC_CHECK(config.has_fl_decreasing_packet_loss_fraction());
+
+  std::map<FrameLengthController::Config::FrameLengthChange, int>
+      fl_changing_bandwidths_bps;
+
+  if (config.has_fl_20ms_to_60ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(20, 60),
+                       config.fl_20ms_to_60ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_60ms_to_20ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(60, 20),
+                       config.fl_60ms_to_20ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_20ms_to_40ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(20, 40),
+                       config.fl_20ms_to_40ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_40ms_to_20ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(40, 20),
+                       config.fl_40ms_to_20ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_40ms_to_60ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(40, 60),
+                       config.fl_40ms_to_60ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_60ms_to_40ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(60, 40),
+                       config.fl_60ms_to_40ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_60ms_to_120ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(std::make_pair(
+        FrameLengthController::Config::FrameLengthChange(60, 120),
+        config.fl_60ms_to_120ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_120ms_to_60ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(std::make_pair(
+        FrameLengthController::Config::FrameLengthChange(120, 60),
+        config.fl_120ms_to_60ms_bandwidth_bps()));
+  }
+
+  int fl_increase_overhead_offset = 0;
+  if (config.has_fl_increase_overhead_offset()) {
+    fl_increase_overhead_offset = config.fl_increase_overhead_offset();
+  }
+  int fl_decrease_overhead_offset = 0;
+  if (config.has_fl_decrease_overhead_offset()) {
+    fl_decrease_overhead_offset = config.fl_decrease_overhead_offset();
+  }
+
+  FrameLengthController::Config ctor_config(
+      std::set<int>(), initial_frame_length_ms, min_encoder_bitrate_bps,
+      config.fl_increasing_packet_loss_fraction(),
+      config.fl_decreasing_packet_loss_fraction(), fl_increase_overhead_offset,
+      fl_decrease_overhead_offset, std::move(fl_changing_bandwidths_bps));
+
+  for (auto frame_length : encoder_frame_lengths_ms)
+    ctor_config.encoder_frame_lengths_ms.insert(frame_length);
+
+  return std::unique_ptr<FrameLengthController>(
+      new FrameLengthController(ctor_config));
+}
+
+std::unique_ptr<ChannelController> CreateChannelController(
+    const audio_network_adaptor::config::ChannelController& config,
+    size_t num_encoder_channels,
+    size_t intial_channels_to_encode) {
+  RTC_CHECK(config.has_channel_1_to_2_bandwidth_bps());
+  RTC_CHECK(config.has_channel_2_to_1_bandwidth_bps());
+
+  return std::unique_ptr<ChannelController>(new ChannelController(
+      ChannelController::Config(num_encoder_channels, intial_channels_to_encode,
+                                config.channel_1_to_2_bandwidth_bps(),
+                                config.channel_2_to_1_bandwidth_bps())));
+}
+
+std::unique_ptr<DtxController> CreateDtxController(
+    const audio_network_adaptor::config::DtxController& dtx_config,
+    bool initial_dtx_enabled) {
+  RTC_CHECK(dtx_config.has_dtx_enabling_bandwidth_bps());
+  RTC_CHECK(dtx_config.has_dtx_disabling_bandwidth_bps());
+
+  return std::unique_ptr<DtxController>(new DtxController(DtxController::Config(
+      initial_dtx_enabled, dtx_config.dtx_enabling_bandwidth_bps(),
+      dtx_config.dtx_disabling_bandwidth_bps())));
+}
+
+using audio_network_adaptor::BitrateController;
+std::unique_ptr<BitrateController> CreateBitrateController(
+    const audio_network_adaptor::config::BitrateController& bitrate_config,
+    int initial_bitrate_bps,
+    int initial_frame_length_ms) {
+  int fl_increase_overhead_offset = 0;
+  if (bitrate_config.has_fl_increase_overhead_offset()) {
+    fl_increase_overhead_offset = bitrate_config.fl_increase_overhead_offset();
+  }
+  int fl_decrease_overhead_offset = 0;
+  if (bitrate_config.has_fl_decrease_overhead_offset()) {
+    fl_decrease_overhead_offset = bitrate_config.fl_decrease_overhead_offset();
+  }
+  return std::unique_ptr<BitrateController>(
+      new BitrateController(BitrateController::Config(
+          initial_bitrate_bps, initial_frame_length_ms,
+          fl_increase_overhead_offset, fl_decrease_overhead_offset)));
+}
+#endif  // WEBRTC_ENABLE_PROTOBUF
+
+}  // namespace
+
+ControllerManagerImpl::Config::Config(int min_reordering_time_ms,
+                                      float min_reordering_squared_distance)
+    : min_reordering_time_ms(min_reordering_time_ms),
+      min_reordering_squared_distance(min_reordering_squared_distance) {}
+
+ControllerManagerImpl::Config::~Config() = default;
+
+std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
+    const std::string& config_string,
+    size_t num_encoder_channels,
+    rtc::ArrayView<const int> encoder_frame_lengths_ms,
+    int min_encoder_bitrate_bps,
+    size_t intial_channels_to_encode,
+    int initial_frame_length_ms,
+    int initial_bitrate_bps,
+    bool initial_fec_enabled,
+    bool initial_dtx_enabled) {
+  return Create(config_string, num_encoder_channels, encoder_frame_lengths_ms,
+                min_encoder_bitrate_bps, intial_channels_to_encode,
+                initial_frame_length_ms, initial_bitrate_bps,
+                initial_fec_enabled, initial_dtx_enabled, nullptr);
+}
+
+std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
+    const std::string& config_string,
+    size_t num_encoder_channels,
+    rtc::ArrayView<const int> encoder_frame_lengths_ms,
+    int min_encoder_bitrate_bps,
+    size_t intial_channels_to_encode,
+    int initial_frame_length_ms,
+    int initial_bitrate_bps,
+    bool initial_fec_enabled,
+    bool initial_dtx_enabled,
+    DebugDumpWriter* debug_dump_writer) {
+#if WEBRTC_ENABLE_PROTOBUF
+  audio_network_adaptor::config::ControllerManager controller_manager_config;
+  RTC_CHECK(controller_manager_config.ParseFromString(config_string));
+  if (debug_dump_writer)
+    debug_dump_writer->DumpControllerManagerConfig(controller_manager_config,
+                                                   rtc::TimeMillis());
+
+  std::vector<std::unique_ptr<Controller>> controllers;
+  std::map<const Controller*, std::pair<int, float>> scoring_points;
+
+  for (int i = 0; i < controller_manager_config.controllers_size(); ++i) {
+    auto& controller_config = controller_manager_config.controllers(i);
+    std::unique_ptr<Controller> controller;
+    switch (controller_config.controller_case()) {
+      case audio_network_adaptor::config::Controller::kFecController:
+        controller = CreateFecControllerPlrBased(
+            controller_config.fec_controller(), initial_fec_enabled);
+        break;
+      case audio_network_adaptor::config::Controller::kFecControllerRplrBased:
+        // FecControllerRplrBased has been removed and can't be used anymore.
+        RTC_NOTREACHED();
+        continue;
+      case audio_network_adaptor::config::Controller::kFrameLengthController:
+        controller = CreateFrameLengthController(
+            controller_config.frame_length_controller(),
+            encoder_frame_lengths_ms, initial_frame_length_ms,
+            min_encoder_bitrate_bps);
+        break;
+      case audio_network_adaptor::config::Controller::kChannelController:
+        controller = CreateChannelController(
+            controller_config.channel_controller(), num_encoder_channels,
+            intial_channels_to_encode);
+        break;
+      case audio_network_adaptor::config::Controller::kDtxController:
+        controller = CreateDtxController(controller_config.dtx_controller(),
+                                         initial_dtx_enabled);
+        break;
+      case audio_network_adaptor::config::Controller::kBitrateController:
+        controller = CreateBitrateController(
+            controller_config.bitrate_controller(), initial_bitrate_bps,
+            initial_frame_length_ms);
+        break;
+      default:
+        RTC_NOTREACHED();
+    }
+    if (controller_config.has_scoring_point()) {
+      auto& scoring_point = controller_config.scoring_point();
+      RTC_CHECK(scoring_point.has_uplink_bandwidth_bps());
+      RTC_CHECK(scoring_point.has_uplink_packet_loss_fraction());
+      scoring_points[controller.get()] = std::make_pair<int, float>(
+          scoring_point.uplink_bandwidth_bps(),
+          scoring_point.uplink_packet_loss_fraction());
+    }
+    controllers.push_back(std::move(controller));
+  }
+
+  if (scoring_points.size() == 0) {
+    return std::unique_ptr<ControllerManagerImpl>(
+        new ControllerManagerImpl(ControllerManagerImpl::Config(0, 0),
+                                  std::move(controllers), scoring_points));
+  } else {
+    RTC_CHECK(controller_manager_config.has_min_reordering_time_ms());
+    RTC_CHECK(controller_manager_config.has_min_reordering_squared_distance());
+    return std::unique_ptr<ControllerManagerImpl>(new ControllerManagerImpl(
+        ControllerManagerImpl::Config(
+            controller_manager_config.min_reordering_time_ms(),
+            controller_manager_config.min_reordering_squared_distance()),
+        std::move(controllers), scoring_points));
+  }
+
+#else
+  RTC_NOTREACHED();
+  return nullptr;
+#endif  // WEBRTC_ENABLE_PROTOBUF
+}
+
+ControllerManagerImpl::ControllerManagerImpl(const Config& config)
+    : ControllerManagerImpl(
+          config,
+          std::vector<std::unique_ptr<Controller>>(),
+          std::map<const Controller*, std::pair<int, float>>()) {}
+
+ControllerManagerImpl::ControllerManagerImpl(
+    const Config& config,
+    std::vector<std::unique_ptr<Controller>> controllers,
+    const std::map<const Controller*, std::pair<int, float>>& scoring_points)
+    : config_(config),
+      controllers_(std::move(controllers)),
+      last_reordering_time_ms_(absl::nullopt),
+      last_scoring_point_(0, 0.0) {
+  for (auto& controller : controllers_)
+    default_sorted_controllers_.push_back(controller.get());
+  sorted_controllers_ = default_sorted_controllers_;
+  for (auto& controller_point : scoring_points) {
+    controller_scoring_points_.insert(std::make_pair(
+        controller_point.first, ScoringPoint(controller_point.second.first,
+                                             controller_point.second.second)));
+  }
+}
+
+ControllerManagerImpl::~ControllerManagerImpl() = default;
+
+std::vector<Controller*> ControllerManagerImpl::GetSortedControllers(
+    const Controller::NetworkMetrics& metrics) {
+  if (controller_scoring_points_.size() == 0)
+    return default_sorted_controllers_;
+
+  if (!metrics.uplink_bandwidth_bps || !metrics.uplink_packet_loss_fraction)
+    return sorted_controllers_;
+
+  const int64_t now_ms = rtc::TimeMillis();
+  if (last_reordering_time_ms_ &&
+      now_ms - *last_reordering_time_ms_ < config_.min_reordering_time_ms)
+    return sorted_controllers_;
+
+  ScoringPoint scoring_point(*metrics.uplink_bandwidth_bps,
+                             *metrics.uplink_packet_loss_fraction);
+
+  if (last_reordering_time_ms_ &&
+      last_scoring_point_.SquaredDistanceTo(scoring_point) <
+          config_.min_reordering_squared_distance)
+    return sorted_controllers_;
+
+  // Sort controllers according to the distances of |scoring_point| to the
+  // scoring points of controllers.
+  //
+  // A controller that does not associate with any scoring point
+  // are treated as if
+  // 1) they are less important than any controller that has a scoring point,
+  // 2) they are equally important to any controller that has no scoring point,
+  //    and their relative order will follow |default_sorted_controllers_|.
+  std::vector<Controller*> sorted_controllers(default_sorted_controllers_);
+  std::stable_sort(
+      sorted_controllers.begin(), sorted_controllers.end(),
+      [this, &scoring_point](const Controller* lhs, const Controller* rhs) {
+        auto lhs_scoring_point = controller_scoring_points_.find(lhs);
+        auto rhs_scoring_point = controller_scoring_points_.find(rhs);
+
+        if (lhs_scoring_point == controller_scoring_points_.end())
+          return false;
+
+        if (rhs_scoring_point == controller_scoring_points_.end())
+          return true;
+
+        return lhs_scoring_point->second.SquaredDistanceTo(scoring_point) <
+               rhs_scoring_point->second.SquaredDistanceTo(scoring_point);
+      });
+
+  if (sorted_controllers_ != sorted_controllers) {
+    sorted_controllers_ = sorted_controllers;
+    last_reordering_time_ms_ = now_ms;
+    last_scoring_point_ = scoring_point;
+  }
+  return sorted_controllers_;
+}
+
+std::vector<Controller*> ControllerManagerImpl::GetControllers() const {
+  return default_sorted_controllers_;
+}
+
+ControllerManagerImpl::ScoringPoint::ScoringPoint(
+    int uplink_bandwidth_bps,
+    float uplink_packet_loss_fraction)
+    : uplink_bandwidth_bps(uplink_bandwidth_bps),
+      uplink_packet_loss_fraction(uplink_packet_loss_fraction) {}
+
+namespace {
+
+constexpr int kMinUplinkBandwidthBps = 0;
+constexpr int kMaxUplinkBandwidthBps = 120000;
+
+float NormalizeUplinkBandwidth(int uplink_bandwidth_bps) {
+  uplink_bandwidth_bps =
+      std::min(kMaxUplinkBandwidthBps,
+               std::max(kMinUplinkBandwidthBps, uplink_bandwidth_bps));
+  return static_cast<float>(uplink_bandwidth_bps - kMinUplinkBandwidthBps) /
+         (kMaxUplinkBandwidthBps - kMinUplinkBandwidthBps);
+}
+
+float NormalizePacketLossFraction(float uplink_packet_loss_fraction) {
+  // |uplink_packet_loss_fraction| is seldom larger than 0.3, so we scale it up
+  // by 3.3333f.
+  return std::min(uplink_packet_loss_fraction * 3.3333f, 1.0f);
+}
+
+}  // namespace
+
+float ControllerManagerImpl::ScoringPoint::SquaredDistanceTo(
+    const ScoringPoint& scoring_point) const {
+  float diff_normalized_bitrate_bps =
+      NormalizeUplinkBandwidth(scoring_point.uplink_bandwidth_bps) -
+      NormalizeUplinkBandwidth(uplink_bandwidth_bps);
+  float diff_normalized_packet_loss =
+      NormalizePacketLossFraction(scoring_point.uplink_packet_loss_fraction) -
+      NormalizePacketLossFraction(uplink_packet_loss_fraction);
+  return std::pow(diff_normalized_bitrate_bps, 2) +
+         std::pow(diff_normalized_packet_loss, 2);
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/controller_manager.h

@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class DebugDumpWriter;
+
+class ControllerManager {
+ public:
+  virtual ~ControllerManager() = default;
+
+  // Sort controllers based on their significance.
+  virtual std::vector<Controller*> GetSortedControllers(
+      const Controller::NetworkMetrics& metrics) = 0;
+
+  virtual std::vector<Controller*> GetControllers() const = 0;
+};
+
+class ControllerManagerImpl final : public ControllerManager {
+ public:
+  struct Config {
+    Config(int min_reordering_time_ms, float min_reordering_squared_distance);
+    ~Config();
+    // Least time since last reordering for a new reordering to be made.
+    int min_reordering_time_ms;
+    // Least squared distance from last scoring point for a new reordering to be
+    // made.
+    float min_reordering_squared_distance;
+  };
+
+  static std::unique_ptr<ControllerManager> Create(
+      const std::string& config_string,
+      size_t num_encoder_channels,
+      rtc::ArrayView<const int> encoder_frame_lengths_ms,
+      int min_encoder_bitrate_bps,
+      size_t intial_channels_to_encode,
+      int initial_frame_length_ms,
+      int initial_bitrate_bps,
+      bool initial_fec_enabled,
+      bool initial_dtx_enabled);
+
+  static std::unique_ptr<ControllerManager> Create(
+      const std::string& config_string,
+      size_t num_encoder_channels,
+      rtc::ArrayView<const int> encoder_frame_lengths_ms,
+      int min_encoder_bitrate_bps,
+      size_t intial_channels_to_encode,
+      int initial_frame_length_ms,
+      int initial_bitrate_bps,
+      bool initial_fec_enabled,
+      bool initial_dtx_enabled,
+      DebugDumpWriter* debug_dump_writer);
+
+  explicit ControllerManagerImpl(const Config& config);
+
+  // Dependency injection for testing.
+  ControllerManagerImpl(
+      const Config& config,
+      std::vector<std::unique_ptr<Controller>> controllers,
+      const std::map<const Controller*, std::pair<int, float>>&
+          chracteristic_points);
+
+  ~ControllerManagerImpl() override;
+
+  // Sort controllers based on their significance.
+  std::vector<Controller*> GetSortedControllers(
+      const Controller::NetworkMetrics& metrics) override;
+
+  std::vector<Controller*> GetControllers() const override;
+
+ private:
+  // Scoring point is a subset of NetworkMetrics that is used for comparing the
+  // significance of controllers.
+  struct ScoringPoint {
+    // TODO(eladalon): Do we want to experiment with RPLR-based scoring?
+    ScoringPoint(int uplink_bandwidth_bps, float uplink_packet_loss_fraction);
+
+    // Calculate the normalized [0,1] distance between two scoring points.
+    float SquaredDistanceTo(const ScoringPoint& scoring_point) const;
+
+    int uplink_bandwidth_bps;
+    float uplink_packet_loss_fraction;
+  };
+
+  const Config config_;
+
+  std::vector<std::unique_ptr<Controller>> controllers_;
+
+  absl::optional<int64_t> last_reordering_time_ms_;
+  ScoringPoint last_scoring_point_;
+
+  std::vector<Controller*> default_sorted_controllers_;
+
+  std::vector<Controller*> sorted_controllers_;
+
+  // |scoring_points_| saves the scoring points of various
+  // controllers.
+  std::map<const Controller*, ScoringPoint> controller_scoring_points_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ControllerManagerImpl);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_

modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc

@@ -0,0 +1,469 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+
+#include <string>
+#include <utility>
+
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/ignore_wundef.h"
+#include "test/gtest.h"
+
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::NiceMock;
+
+namespace {
+
+constexpr size_t kNumControllers = 4;
+constexpr int kChracteristicBandwithBps[2] = {15000, 0};
+constexpr float kChracteristicPacketLossFraction[2] = {0.2f, 0.0f};
+constexpr int kMinReorderingTimeMs = 200;
+constexpr int kFactor = 100;
+constexpr float kMinReorderingSquareDistance = 1.0f / kFactor / kFactor;
+
+// |kMinUplinkBandwidthBps| and |kMaxUplinkBandwidthBps| are copied from
+// controller_manager.cc
+constexpr int kMinUplinkBandwidthBps = 0;
+constexpr int kMaxUplinkBandwidthBps = 120000;
+constexpr int kMinBandwithChangeBps =
+    (kMaxUplinkBandwidthBps - kMinUplinkBandwidthBps) / kFactor;
+
+struct ControllerManagerStates {
+  std::unique_ptr<ControllerManager> controller_manager;
+  std::vector<MockController*> mock_controllers;
+};
+
+ControllerManagerStates CreateControllerManager() {
+  ControllerManagerStates states;
+  std::vector<std::unique_ptr<Controller>> controllers;
+  std::map<const Controller*, std::pair<int, float>> chracteristic_points;
+  for (size_t i = 0; i < kNumControllers; ++i) {
+    auto controller =
+        std::unique_ptr<MockController>(new NiceMock<MockController>());
+    EXPECT_CALL(*controller, Die());
+    states.mock_controllers.push_back(controller.get());
+    controllers.push_back(std::move(controller));
+  }
+
+  // Assign characteristic points to the last two controllers.
+  chracteristic_points[states.mock_controllers[kNumControllers - 2]] =
+      std::make_pair(kChracteristicBandwithBps[0],
+                     kChracteristicPacketLossFraction[0]);
+  chracteristic_points[states.mock_controllers[kNumControllers - 1]] =
+      std::make_pair(kChracteristicBandwithBps[1],
+                     kChracteristicPacketLossFraction[1]);
+
+  states.controller_manager.reset(new ControllerManagerImpl(
+      ControllerManagerImpl::Config(kMinReorderingTimeMs,
+                                    kMinReorderingSquareDistance),
+      std::move(controllers), chracteristic_points));
+  return states;
+}
+
+// |expected_order| contains the expected indices of all controllers in the
+// vector of controllers returned by GetSortedControllers(). A negative index
+// means that we do not care about its exact place, but we do check that it
+// exists in the vector.
+void CheckControllersOrder(
+    ControllerManagerStates* states,
+    const absl::optional<int>& uplink_bandwidth_bps,
+    const absl::optional<float>& uplink_packet_loss_fraction,
+    const std::vector<int>& expected_order) {
+  RTC_DCHECK_EQ(kNumControllers, expected_order.size());
+  Controller::NetworkMetrics metrics;
+  metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+  metrics.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
+  auto check = states->controller_manager->GetSortedControllers(metrics);
+  EXPECT_EQ(states->mock_controllers.size(), check.size());
+  for (size_t i = 0; i < states->mock_controllers.size(); ++i) {
+    if (expected_order[i] >= 0) {
+      EXPECT_EQ(states->mock_controllers[i], check[expected_order[i]]);
+    } else {
+      EXPECT_NE(check.end(), std::find(check.begin(), check.end(),
+                                       states->mock_controllers[i]));
+    }
+  }
+}
+
+}  // namespace
+
+TEST(ControllerManagerTest, GetControllersReturnAllControllers) {
+  auto states = CreateControllerManager();
+  auto check = states.controller_manager->GetControllers();
+  // Verify that controllers in |check| are one-to-one mapped to those in
+  // |mock_controllers_|.
+  EXPECT_EQ(states.mock_controllers.size(), check.size());
+  for (auto& controller : check)
+    EXPECT_NE(states.mock_controllers.end(),
+              std::find(states.mock_controllers.begin(),
+                        states.mock_controllers.end(), controller));
+}
+
+TEST(ControllerManagerTest, ControllersInDefaultOrderOnEmptyNetworkMetrics) {
+  auto states = CreateControllerManager();
+  // |network_metrics| are empty, and the controllers are supposed to follow the
+  // default order.
+  CheckControllersOrder(&states, absl::nullopt, absl::nullopt, {0, 1, 2, 3});
+}
+
+TEST(ControllerManagerTest, ControllersWithoutCharPointAtEndAndInDefaultOrder) {
+  auto states = CreateControllerManager();
+  CheckControllersOrder(&states, 0, 0.0,
+                        {kNumControllers - 2, kNumControllers - 1, -1, -1});
+}
+
+TEST(ControllerManagerTest, ControllersWithCharPointDependOnNetworkMetrics) {
+  auto states = CreateControllerManager();
+  CheckControllersOrder(&states, kChracteristicBandwithBps[1],
+                        kChracteristicPacketLossFraction[1],
+                        {kNumControllers - 2, kNumControllers - 1, 1, 0});
+}
+
+TEST(ControllerManagerTest, DoNotReorderBeforeMinReordingTime) {
+  rtc::ScopedFakeClock fake_clock;
+  auto states = CreateControllerManager();
+  CheckControllersOrder(&states, kChracteristicBandwithBps[0],
+                        kChracteristicPacketLossFraction[0],
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs - 1));
+  // Move uplink bandwidth and packet loss fraction to the other controller's
+  // characteristic point, which would cause controller manager to reorder the
+  // controllers if time had reached min reordering time.
+  CheckControllersOrder(&states, kChracteristicBandwithBps[1],
+                        kChracteristicPacketLossFraction[1],
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+}
+
+TEST(ControllerManagerTest, ReorderBeyondMinReordingTimeAndMinDistance) {
+  rtc::ScopedFakeClock fake_clock;
+  auto states = CreateControllerManager();
+  constexpr int kBandwidthBps =
+      (kChracteristicBandwithBps[0] + kChracteristicBandwithBps[1]) / 2;
+  constexpr float kPacketLossFraction = (kChracteristicPacketLossFraction[0] +
+                                         kChracteristicPacketLossFraction[1]) /
+                                        2.0f;
+  // Set network metrics to be in the middle between the characteristic points
+  // of two controllers.
+  CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs));
+  // Then let network metrics move a little towards the other controller.
+  CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps - 1,
+                        kPacketLossFraction,
+                        {kNumControllers - 2, kNumControllers - 1, 1, 0});
+}
+
+TEST(ControllerManagerTest, DoNotReorderIfNetworkMetricsChangeTooSmall) {
+  rtc::ScopedFakeClock fake_clock;
+  auto states = CreateControllerManager();
+  constexpr int kBandwidthBps =
+      (kChracteristicBandwithBps[0] + kChracteristicBandwithBps[1]) / 2;
+  constexpr float kPacketLossFraction = (kChracteristicPacketLossFraction[0] +
+                                         kChracteristicPacketLossFraction[1]) /
+                                        2.0f;
+  // Set network metrics to be in the middle between the characteristic points
+  // of two controllers.
+  CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs));
+  // Then let network metrics move a little towards the other controller.
+  CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps + 1,
+                        kPacketLossFraction,
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+}
+
+#if WEBRTC_ENABLE_PROTOBUF
+
+namespace {
+
+void AddBitrateControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  config->add_controllers()->mutable_bitrate_controller();
+}
+
+void AddChannelControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  auto controller_config =
+      config->add_controllers()->mutable_channel_controller();
+  controller_config->set_channel_1_to_2_bandwidth_bps(31000);
+  controller_config->set_channel_2_to_1_bandwidth_bps(29000);
+}
+
+void AddDtxControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  auto controller_config = config->add_controllers()->mutable_dtx_controller();
+  controller_config->set_dtx_enabling_bandwidth_bps(55000);
+  controller_config->set_dtx_disabling_bandwidth_bps(65000);
+}
+
+void AddFecControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  auto controller_config_ext = config->add_controllers();
+  auto controller_config = controller_config_ext->mutable_fec_controller();
+  auto fec_enabling_threshold =
+      controller_config->mutable_fec_enabling_threshold();
+  fec_enabling_threshold->set_low_bandwidth_bps(17000);
+  fec_enabling_threshold->set_low_bandwidth_packet_loss(0.1f);
+  fec_enabling_threshold->set_high_bandwidth_bps(64000);
+  fec_enabling_threshold->set_high_bandwidth_packet_loss(0.05f);
+  auto fec_disabling_threshold =
+      controller_config->mutable_fec_disabling_threshold();
+  fec_disabling_threshold->set_low_bandwidth_bps(15000);
+  fec_disabling_threshold->set_low_bandwidth_packet_loss(0.08f);
+  fec_disabling_threshold->set_high_bandwidth_bps(64000);
+  fec_disabling_threshold->set_high_bandwidth_packet_loss(0.01f);
+  controller_config->set_time_constant_ms(500);
+
+  auto scoring_point = controller_config_ext->mutable_scoring_point();
+  scoring_point->set_uplink_bandwidth_bps(kChracteristicBandwithBps[0]);
+  scoring_point->set_uplink_packet_loss_fraction(
+      kChracteristicPacketLossFraction[0]);
+}
+
+void AddFrameLengthControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  auto controller_config_ext = config->add_controllers();
+  auto controller_config =
+      controller_config_ext->mutable_frame_length_controller();
+  controller_config->set_fl_decreasing_packet_loss_fraction(0.05f);
+  controller_config->set_fl_increasing_packet_loss_fraction(0.04f);
+  controller_config->set_fl_20ms_to_40ms_bandwidth_bps(80000);
+  controller_config->set_fl_40ms_to_20ms_bandwidth_bps(88000);
+  controller_config->set_fl_40ms_to_60ms_bandwidth_bps(72000);
+  controller_config->set_fl_60ms_to_40ms_bandwidth_bps(80000);
+
+  auto scoring_point = controller_config_ext->mutable_scoring_point();
+  scoring_point->set_uplink_bandwidth_bps(kChracteristicBandwithBps[1]);
+  scoring_point->set_uplink_packet_loss_fraction(
+      kChracteristicPacketLossFraction[1]);
+}
+
+constexpr int kInitialBitrateBps = 24000;
+constexpr size_t kIntialChannelsToEncode = 1;
+constexpr bool kInitialDtxEnabled = true;
+constexpr bool kInitialFecEnabled = true;
+constexpr int kInitialFrameLengthMs = 60;
+constexpr int kMinBitrateBps = 6000;
+
+ControllerManagerStates CreateControllerManager(
+    const std::string& config_string) {
+  ControllerManagerStates states;
+  constexpr size_t kNumEncoderChannels = 2;
+  const std::vector<int> encoder_frame_lengths_ms = {20, 60};
+  states.controller_manager = ControllerManagerImpl::Create(
+      config_string, kNumEncoderChannels, encoder_frame_lengths_ms,
+      kMinBitrateBps, kIntialChannelsToEncode, kInitialFrameLengthMs,
+      kInitialBitrateBps, kInitialFecEnabled, kInitialDtxEnabled);
+  return states;
+}
+
+enum class ControllerType : int8_t {
+  FEC,
+  CHANNEL,
+  DTX,
+  FRAME_LENGTH,
+  BIT_RATE
+};
+
+void CheckControllersOrder(const std::vector<Controller*>& controllers,
+                           const std::vector<ControllerType>& expected_types) {
+  ASSERT_EQ(expected_types.size(), controllers.size());
+
+  // We also check that the controllers follow the initial settings.
+  AudioEncoderRuntimeConfig encoder_config;
+
+  for (size_t i = 0; i < controllers.size(); ++i) {
+    AudioEncoderRuntimeConfig encoder_config;
+    // We check the order of |controllers| by judging their decisions.
+    controllers[i]->MakeDecision(&encoder_config);
+
+    // Since controllers are not provided with network metrics, they give the
+    // initial values.
+    switch (expected_types[i]) {
+      case ControllerType::FEC:
+        EXPECT_EQ(kInitialFecEnabled, encoder_config.enable_fec);
+        break;
+      case ControllerType::CHANNEL:
+        EXPECT_EQ(kIntialChannelsToEncode, encoder_config.num_channels);
+        break;
+      case ControllerType::DTX:
+        EXPECT_EQ(kInitialDtxEnabled, encoder_config.enable_dtx);
+        break;
+      case ControllerType::FRAME_LENGTH:
+        EXPECT_EQ(kInitialFrameLengthMs, encoder_config.frame_length_ms);
+        break;
+      case ControllerType::BIT_RATE:
+        EXPECT_EQ(kInitialBitrateBps, encoder_config.bitrate_bps);
+    }
+  }
+}
+
+MATCHER_P(ControllerManagerEqual, value, "") {
+  std::string value_string;
+  std::string arg_string;
+  EXPECT_TRUE(arg.SerializeToString(&arg_string));
+  EXPECT_TRUE(value.SerializeToString(&value_string));
+  return arg_string == value_string;
+}
+
+}  // namespace
+
+TEST(ControllerManagerTest, DebugDumpLoggedWhenCreateFromConfigString) {
+  audio_network_adaptor::config::ControllerManager config;
+  config.set_min_reordering_time_ms(kMinReorderingTimeMs);
+  config.set_min_reordering_squared_distance(kMinReorderingSquareDistance);
+
+  AddFecControllerConfig(&config);
+  AddChannelControllerConfig(&config);
+  AddDtxControllerConfig(&config);
+  AddFrameLengthControllerConfig(&config);
+  AddBitrateControllerConfig(&config);
+
+  std::string config_string;
+  config.SerializeToString(&config_string);
+
+  constexpr size_t kNumEncoderChannels = 2;
+  const std::vector<int> encoder_frame_lengths_ms = {20, 60};
+
+  constexpr int64_t kClockInitialTimeMs = 12345678;
+  rtc::ScopedFakeClock fake_clock;
+  fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
+  auto debug_dump_writer =
+      std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
+  EXPECT_CALL(*debug_dump_writer, Die());
+  EXPECT_CALL(*debug_dump_writer,
+              DumpControllerManagerConfig(ControllerManagerEqual(config),
+                                          kClockInitialTimeMs));
+
+  ControllerManagerImpl::Create(config_string, kNumEncoderChannels,
+                                encoder_frame_lengths_ms, kMinBitrateBps,
+                                kIntialChannelsToEncode, kInitialFrameLengthMs,
+                                kInitialBitrateBps, kInitialFecEnabled,
+                                kInitialDtxEnabled, debug_dump_writer.get());
+}
+
+TEST(ControllerManagerTest, CreateFromConfigStringAndCheckDefaultOrder) {
+  audio_network_adaptor::config::ControllerManager config;
+  config.set_min_reordering_time_ms(kMinReorderingTimeMs);
+  config.set_min_reordering_squared_distance(kMinReorderingSquareDistance);
+
+  AddFecControllerConfig(&config);
+  AddChannelControllerConfig(&config);
+  AddDtxControllerConfig(&config);
+  AddFrameLengthControllerConfig(&config);
+  AddBitrateControllerConfig(&config);
+
+  std::string config_string;
+  config.SerializeToString(&config_string);
+
+  auto states = CreateControllerManager(config_string);
+  Controller::NetworkMetrics metrics;
+
+  auto controllers = states.controller_manager->GetSortedControllers(metrics);
+  CheckControllersOrder(
+      controllers,
+      std::vector<ControllerType>{
+          ControllerType::FEC, ControllerType::CHANNEL, ControllerType::DTX,
+          ControllerType::FRAME_LENGTH, ControllerType::BIT_RATE});
+}
+
+TEST(ControllerManagerTest, CreateCharPointFreeConfigAndCheckDefaultOrder) {
+  audio_network_adaptor::config::ControllerManager config;
+
+  // Following controllers have no characteristic points.
+  AddChannelControllerConfig(&config);
+  AddDtxControllerConfig(&config);
+  AddBitrateControllerConfig(&config);
+
+  std::string config_string;
+  config.SerializeToString(&config_string);
+
+  auto states = CreateControllerManager(config_string);
+  Controller::NetworkMetrics metrics;
+
+  auto controllers = states.controller_manager->GetSortedControllers(metrics);
+  CheckControllersOrder(
+      controllers,
+      std::vector<ControllerType>{ControllerType::CHANNEL, ControllerType::DTX,
+                                  ControllerType::BIT_RATE});
+}
+
+TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
+  rtc::ScopedFakeClock fake_clock;
+  audio_network_adaptor::config::ControllerManager config;
+  config.set_min_reordering_time_ms(kMinReorderingTimeMs);
+  config.set_min_reordering_squared_distance(kMinReorderingSquareDistance);
+
+  AddChannelControllerConfig(&config);
+
+  // Internally associated with characteristic point 0.
+  AddFecControllerConfig(&config);
+
+  AddDtxControllerConfig(&config);
+
+  // Internally associated with characteristic point 1.
+  AddFrameLengthControllerConfig(&config);
+
+  AddBitrateControllerConfig(&config);
+
+  std::string config_string;
+  config.SerializeToString(&config_string);
+
+  auto states = CreateControllerManager(config_string);
+
+  Controller::NetworkMetrics metrics;
+  metrics.uplink_bandwidth_bps = kChracteristicBandwithBps[0];
+  metrics.uplink_packet_loss_fraction = kChracteristicPacketLossFraction[0];
+
+  auto controllers = states.controller_manager->GetSortedControllers(metrics);
+  CheckControllersOrder(controllers,
+                        std::vector<ControllerType>{
+                            ControllerType::FEC, ControllerType::FRAME_LENGTH,
+                            ControllerType::CHANNEL, ControllerType::DTX,
+                            ControllerType::BIT_RATE});
+
+  metrics.uplink_bandwidth_bps = kChracteristicBandwithBps[1];
+  metrics.uplink_packet_loss_fraction = kChracteristicPacketLossFraction[1];
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs - 1));
+  controllers = states.controller_manager->GetSortedControllers(metrics);
+  // Should not reorder since min reordering time is not met.
+  CheckControllersOrder(controllers,
+                        std::vector<ControllerType>{
+                            ControllerType::FEC, ControllerType::FRAME_LENGTH,
+                            ControllerType::CHANNEL, ControllerType::DTX,
+                            ControllerType::BIT_RATE});
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(1));
+  controllers = states.controller_manager->GetSortedControllers(metrics);
+  // Reorder now.
+  CheckControllersOrder(controllers,
+                        std::vector<ControllerType>{
+                            ControllerType::FRAME_LENGTH, ControllerType::FEC,
+                            ControllerType::CHANNEL, ControllerType::DTX,
+                            ControllerType::BIT_RATE});
+}
+#endif  // WEBRTC_ENABLE_PROTOBUF
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/debug_dump.proto

@@ -0,0 +1,42 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.audio_network_adaptor.debug_dump;
+
+import "config.proto";
+
+message NetworkMetrics {
+  optional int32 uplink_bandwidth_bps = 1;
+  optional float uplink_packet_loss_fraction = 2;
+  optional int32 target_audio_bitrate_bps = 3;
+  optional int32 rtt_ms = 4;
+  optional int32 uplink_recoverable_packet_loss_fraction = 5;
+}
+
+message EncoderRuntimeConfig {
+  optional int32 bitrate_bps = 1;
+  optional int32 frame_length_ms = 2;
+  // Note: This is what we tell the encoder. It doesn't have to reflect
+  // the actual NetworkMetrics; it's subject to our decision.
+  optional float uplink_packet_loss_fraction = 3;
+  optional bool enable_fec = 4;
+  optional bool enable_dtx = 5;
+  // Some encoders can encode fewer channels than the actual input to make
+  // better use of the bandwidth. |num_channels| sets the number of channels
+  // to encode.
+  optional uint32 num_channels = 6;
+}
+
+message Event {
+  enum Type {
+    NETWORK_METRICS = 0;
+    ENCODER_RUNTIME_CONFIG = 1;
+    CONTROLLER_MANAGER_CONFIG = 2;
+  }
+  required Type type = 1;
+  required uint32 timestamp = 2;
+  optional NetworkMetrics network_metrics = 3;
+  optional EncoderRuntimeConfig encoder_runtime_config = 4;
+  optional webrtc.audio_network_adaptor.config.ControllerManager
+      controller_manager_config = 5;
+}
+

modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc

@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/system/file_wrapper.h"
+
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+#if WEBRTC_ENABLE_PROTOBUF
+namespace {
+
+using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
+using audio_network_adaptor::debug_dump::Event;
+using audio_network_adaptor::debug_dump::NetworkMetrics;
+
+void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
+  RTC_CHECK(dump_file->is_open());
+  std::string dump_data;
+  event.SerializeToString(&dump_data);
+  int32_t size = rtc::checked_cast<int32_t>(event.ByteSizeLong());
+  dump_file->Write(&size, sizeof(size));
+  dump_file->Write(dump_data.data(), dump_data.length());
+}
+
+}  // namespace
+#endif  // WEBRTC_ENABLE_PROTOBUF
+
+class DebugDumpWriterImpl final : public DebugDumpWriter {
+ public:
+  explicit DebugDumpWriterImpl(FILE* file_handle);
+  ~DebugDumpWriterImpl() override = default;
+
+  void DumpEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& config,
+                                int64_t timestamp) override;
+
+  void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
+                          int64_t timestamp) override;
+
+#if WEBRTC_ENABLE_PROTOBUF
+  void DumpControllerManagerConfig(
+      const audio_network_adaptor::config::ControllerManager&
+          controller_manager_config,
+      int64_t timestamp) override;
+#endif
+
+ private:
+  FileWrapper dump_file_;
+};
+
+DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle) {
+#if WEBRTC_ENABLE_PROTOBUF
+  dump_file_ = FileWrapper(file_handle);
+  RTC_CHECK(dump_file_.is_open());
+#else
+  RTC_NOTREACHED();
+#endif
+}
+
+void DebugDumpWriterImpl::DumpNetworkMetrics(
+    const Controller::NetworkMetrics& metrics,
+    int64_t timestamp) {
+#if WEBRTC_ENABLE_PROTOBUF
+  Event event;
+  event.set_timestamp(timestamp);
+  event.set_type(Event::NETWORK_METRICS);
+  auto dump_metrics = event.mutable_network_metrics();
+
+  if (metrics.uplink_bandwidth_bps)
+    dump_metrics->set_uplink_bandwidth_bps(*metrics.uplink_bandwidth_bps);
+
+  if (metrics.uplink_packet_loss_fraction) {
+    dump_metrics->set_uplink_packet_loss_fraction(
+        *metrics.uplink_packet_loss_fraction);
+  }
+
+  if (metrics.target_audio_bitrate_bps) {
+    dump_metrics->set_target_audio_bitrate_bps(
+        *metrics.target_audio_bitrate_bps);
+  }
+
+  if (metrics.rtt_ms)
+    dump_metrics->set_rtt_ms(*metrics.rtt_ms);
+
+  DumpEventToFile(event, &dump_file_);
+#endif  // WEBRTC_ENABLE_PROTOBUF
+}
+
+void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
+    const AudioEncoderRuntimeConfig& config,
+    int64_t timestamp) {
+#if WEBRTC_ENABLE_PROTOBUF
+  Event event;
+  event.set_timestamp(timestamp);
+  event.set_type(Event::ENCODER_RUNTIME_CONFIG);
+  auto dump_config = event.mutable_encoder_runtime_config();
+
+  if (config.bitrate_bps)
+    dump_config->set_bitrate_bps(*config.bitrate_bps);
+
+  if (config.frame_length_ms)
+    dump_config->set_frame_length_ms(*config.frame_length_ms);
+
+  if (config.uplink_packet_loss_fraction) {
+    dump_config->set_uplink_packet_loss_fraction(
+        *config.uplink_packet_loss_fraction);
+  }
+
+  if (config.enable_fec)
+    dump_config->set_enable_fec(*config.enable_fec);
+
+  if (config.enable_dtx)
+    dump_config->set_enable_dtx(*config.enable_dtx);
+
+  if (config.num_channels)
+    dump_config->set_num_channels(*config.num_channels);
+
+  DumpEventToFile(event, &dump_file_);
+#endif  // WEBRTC_ENABLE_PROTOBUF
+}
+
+#if WEBRTC_ENABLE_PROTOBUF
+void DebugDumpWriterImpl::DumpControllerManagerConfig(
+    const audio_network_adaptor::config::ControllerManager&
+        controller_manager_config,
+    int64_t timestamp) {
+  Event event;
+  event.set_timestamp(timestamp);
+  event.set_type(Event::CONTROLLER_MANAGER_CONFIG);
+  event.mutable_controller_manager_config()->CopyFrom(
+      controller_manager_config);
+  DumpEventToFile(event, &dump_file_);
+}
+#endif  // WEBRTC_ENABLE_PROTOBUF
+
+std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {
+  return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle));
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/debug_dump_writer.h

@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
+
+#include <memory>
+
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "rtc_base/constructor_magic.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/system/file_wrapper.h"
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+class DebugDumpWriter {
+ public:
+  static std::unique_ptr<DebugDumpWriter> Create(FILE* file_handle);
+
+  virtual ~DebugDumpWriter() = default;
+
+  virtual void DumpEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& config,
+                                        int64_t timestamp) = 0;
+
+  virtual void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
+                                  int64_t timestamp) = 0;
+
+#if WEBRTC_ENABLE_PROTOBUF
+  virtual void DumpControllerManagerConfig(
+      const audio_network_adaptor::config::ControllerManager&
+          controller_manager_config,
+      int64_t timestamp) = 0;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_

modules/audio_coding/audio_network_adaptor/dtx_controller.cc

@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+DtxController::Config::Config(bool initial_dtx_enabled,
+                              int dtx_enabling_bandwidth_bps,
+                              int dtx_disabling_bandwidth_bps)
+    : initial_dtx_enabled(initial_dtx_enabled),
+      dtx_enabling_bandwidth_bps(dtx_enabling_bandwidth_bps),
+      dtx_disabling_bandwidth_bps(dtx_disabling_bandwidth_bps) {}
+
+DtxController::DtxController(const Config& config)
+    : config_(config), dtx_enabled_(config_.initial_dtx_enabled) {}
+
+DtxController::~DtxController() = default;
+
+void DtxController::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.uplink_bandwidth_bps)
+    uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+}
+
+void DtxController::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  // Decision on |enable_dtx| should not have been made.
+  RTC_DCHECK(!config->enable_dtx);
+
+  if (uplink_bandwidth_bps_) {
+    if (dtx_enabled_ &&
+        *uplink_bandwidth_bps_ >= config_.dtx_disabling_bandwidth_bps) {
+      dtx_enabled_ = false;
+    } else if (!dtx_enabled_ &&
+               *uplink_bandwidth_bps_ <= config_.dtx_enabling_bandwidth_bps) {
+      dtx_enabled_ = true;
+    }
+  }
+  config->enable_dtx = dtx_enabled_;
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/dtx_controller.h

@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class DtxController final : public Controller {
+ public:
+  struct Config {
+    Config(bool initial_dtx_enabled,
+           int dtx_enabling_bandwidth_bps,
+           int dtx_disabling_bandwidth_bps);
+    bool initial_dtx_enabled;
+    // Uplink bandwidth below which DTX should be switched on.
+    int dtx_enabling_bandwidth_bps;
+    // Uplink bandwidth above which DTX should be switched off.
+    int dtx_disabling_bandwidth_bps;
+  };
+
+  explicit DtxController(const Config& config);
+
+  ~DtxController() override;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  const Config config_;
+  bool dtx_enabled_;
+  absl::optional<int> uplink_bandwidth_bps_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(DtxController);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc

@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDtxEnablingBandwidthBps = 55000;
+constexpr int kDtxDisablingBandwidthBps = 65000;
+constexpr int kMediumBandwidthBps =
+    (kDtxEnablingBandwidthBps + kDtxDisablingBandwidthBps) / 2;
+
+std::unique_ptr<DtxController> CreateController(int initial_dtx_enabled) {
+  std::unique_ptr<DtxController> controller(new DtxController(
+      DtxController::Config(initial_dtx_enabled, kDtxEnablingBandwidthBps,
+                            kDtxDisablingBandwidthBps)));
+  return controller;
+}
+
+void CheckDecision(DtxController* controller,
+                   const absl::optional<int>& uplink_bandwidth_bps,
+                   bool expected_dtx_enabled) {
+  if (uplink_bandwidth_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  AudioEncoderRuntimeConfig config;
+  controller->MakeDecision(&config);
+  EXPECT_EQ(expected_dtx_enabled, config.enable_dtx);
+}
+
+}  // namespace
+
+TEST(DtxControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
+  constexpr bool kInitialDtxEnabled = true;
+  auto controller = CreateController(kInitialDtxEnabled);
+  CheckDecision(controller.get(), absl::nullopt, kInitialDtxEnabled);
+}
+
+TEST(DtxControllerTest, TurnOnDtxForLowUplinkBandwidth) {
+  auto controller = CreateController(false);
+  CheckDecision(controller.get(), kDtxEnablingBandwidthBps, true);
+}
+
+TEST(DtxControllerTest, TurnOffDtxForHighUplinkBandwidth) {
+  auto controller = CreateController(true);
+  CheckDecision(controller.get(), kDtxDisablingBandwidthBps, false);
+}
+
+TEST(DtxControllerTest, MaintainDtxOffForMediumUplinkBandwidth) {
+  auto controller = CreateController(false);
+  CheckDecision(controller.get(), kMediumBandwidthBps, false);
+}
+
+TEST(DtxControllerTest, MaintainDtxOnForMediumUplinkBandwidth) {
+  auto controller = CreateController(true);
+  CheckDecision(controller.get(), kMediumBandwidthBps, true);
+}
+
+TEST(DtxControllerTest, CheckBehaviorOnChangingUplinkBandwidth) {
+  auto controller = CreateController(false);
+  CheckDecision(controller.get(), kMediumBandwidthBps, false);
+  CheckDecision(controller.get(), kDtxEnablingBandwidthBps, true);
+  CheckDecision(controller.get(), kMediumBandwidthBps, true);
+  CheckDecision(controller.get(), kDtxDisablingBandwidthBps, false);
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/event_log_writer.cc

@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <cstdlib>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+EventLogWriter::EventLogWriter(RtcEventLog* event_log,
+                               int min_bitrate_change_bps,
+                               float min_bitrate_change_fraction,
+                               float min_packet_loss_change_fraction)
+    : event_log_(event_log),
+      min_bitrate_change_bps_(min_bitrate_change_bps),
+      min_bitrate_change_fraction_(min_bitrate_change_fraction),
+      min_packet_loss_change_fraction_(min_packet_loss_change_fraction) {
+  RTC_DCHECK(event_log_);
+}
+
+EventLogWriter::~EventLogWriter() = default;
+
+void EventLogWriter::MaybeLogEncoderConfig(
+    const AudioEncoderRuntimeConfig& config) {
+  if (last_logged_config_.num_channels != config.num_channels)
+    return LogEncoderConfig(config);
+  if (last_logged_config_.enable_dtx != config.enable_dtx)
+    return LogEncoderConfig(config);
+  if (last_logged_config_.enable_fec != config.enable_fec)
+    return LogEncoderConfig(config);
+  if (last_logged_config_.frame_length_ms != config.frame_length_ms)
+    return LogEncoderConfig(config);
+  if ((!last_logged_config_.bitrate_bps && config.bitrate_bps) ||
+      (last_logged_config_.bitrate_bps && config.bitrate_bps &&
+       std::abs(*last_logged_config_.bitrate_bps - *config.bitrate_bps) >=
+           std::min(static_cast<int>(*last_logged_config_.bitrate_bps *
+                                     min_bitrate_change_fraction_),
+                    min_bitrate_change_bps_))) {
+    return LogEncoderConfig(config);
+  }
+  if ((!last_logged_config_.uplink_packet_loss_fraction &&
+       config.uplink_packet_loss_fraction) ||
+      (last_logged_config_.uplink_packet_loss_fraction &&
+       config.uplink_packet_loss_fraction &&
+       fabs(*last_logged_config_.uplink_packet_loss_fraction -
+            *config.uplink_packet_loss_fraction) >=
+           min_packet_loss_change_fraction_ *
+               *last_logged_config_.uplink_packet_loss_fraction)) {
+    return LogEncoderConfig(config);
+  }
+}
+
+void EventLogWriter::LogEncoderConfig(const AudioEncoderRuntimeConfig& config) {
+  auto config_copy = std::make_unique<AudioEncoderRuntimeConfig>(config);
+  event_log_->Log(
+      std::make_unique<RtcEventAudioNetworkAdaptation>(std::move(config_copy)));
+  last_logged_config_ = config;
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/event_log_writer.h

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
+
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+class RtcEventLog;
+
+class EventLogWriter final {
+ public:
+  EventLogWriter(RtcEventLog* event_log,
+                 int min_bitrate_change_bps,
+                 float min_bitrate_change_fraction,
+                 float min_packet_loss_change_fraction);
+  ~EventLogWriter();
+  void MaybeLogEncoderConfig(const AudioEncoderRuntimeConfig& config);
+
+ private:
+  void LogEncoderConfig(const AudioEncoderRuntimeConfig& config);
+
+  RtcEventLog* const event_log_;
+  const int min_bitrate_change_bps_;
+  const float min_bitrate_change_fraction_;
+  const float min_packet_loss_change_fraction_;
+  AudioEncoderRuntimeConfig last_logged_config_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(EventLogWriter);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_

modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc

@@ -0,0 +1,240 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+
+#include <memory>
+
+#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kMinBitrateChangeBps = 5000;
+constexpr float kMinPacketLossChangeFraction = 0.5;
+constexpr float kMinBitrateChangeFraction = 0.25;
+
+constexpr int kHighBitrateBps = 70000;
+constexpr int kLowBitrateBps = 10000;
+constexpr int kFrameLengthMs = 60;
+constexpr bool kEnableFec = true;
+constexpr bool kEnableDtx = true;
+constexpr float kPacketLossFraction = 0.05f;
+constexpr size_t kNumChannels = 1;
+
+MATCHER_P(IsRtcEventAnaConfigEqualTo, config, "") {
+  if (arg->GetType() != RtcEvent::Type::AudioNetworkAdaptation) {
+    return false;
+  }
+  auto ana_event = static_cast<RtcEventAudioNetworkAdaptation*>(arg);
+  return ana_event->config() == config;
+}
+
+struct EventLogWriterStates {
+  std::unique_ptr<EventLogWriter> event_log_writer;
+  std::unique_ptr<testing::StrictMock<MockRtcEventLog>> event_log;
+  AudioEncoderRuntimeConfig runtime_config;
+};
+
+EventLogWriterStates CreateEventLogWriter() {
+  EventLogWriterStates state;
+  state.event_log.reset(new ::testing::StrictMock<MockRtcEventLog>());
+  state.event_log_writer.reset(new EventLogWriter(
+      state.event_log.get(), kMinBitrateChangeBps, kMinBitrateChangeFraction,
+      kMinPacketLossChangeFraction));
+  state.runtime_config.bitrate_bps = kHighBitrateBps;
+  state.runtime_config.frame_length_ms = kFrameLengthMs;
+  state.runtime_config.uplink_packet_loss_fraction = kPacketLossFraction;
+  state.runtime_config.enable_fec = kEnableFec;
+  state.runtime_config.enable_dtx = kEnableDtx;
+  state.runtime_config.num_channels = kNumChannels;
+  return state;
+}
+}  // namespace
+
+TEST(EventLogWriterTest, FirstConfigIsLogged) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, SameConfigIsNotLogged) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogFecStateChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+
+  state.runtime_config.enable_fec = !kEnableFec;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogDtxStateChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+
+  state.runtime_config.enable_dtx = !kEnableDtx;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogChannelChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+
+  state.runtime_config.num_channels = kNumChannels + 1;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogFrameLengthChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+
+  state.runtime_config.frame_length_ms = 20;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, DoNotLogSmallBitrateChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.bitrate_bps = kHighBitrateBps + kMinBitrateChangeBps - 1;
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogLargeBitrateChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  // At high bitrate, the min fraction rule requires a larger change than the
+  // min change rule. We make sure that the min change rule applies.
+  RTC_DCHECK_GT(kHighBitrateBps * kMinBitrateChangeFraction,
+                kMinBitrateChangeBps);
+  state.runtime_config.bitrate_bps = kHighBitrateBps + kMinBitrateChangeBps;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogMinBitrateChangeFractionOnLowBitrateChange) {
+  auto state = CreateEventLogWriter();
+  state.runtime_config.bitrate_bps = kLowBitrateBps;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  // At high bitrate, the min change rule requires a larger change than the min
+  // fraction rule. We make sure that the min fraction rule applies.
+  state.runtime_config.bitrate_bps =
+      kLowBitrateBps + kLowBitrateBps * kMinBitrateChangeFraction;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, DoNotLogSmallPacketLossFractionChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.uplink_packet_loss_fraction =
+      kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction -
+      0.001f;
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogLargePacketLossFractionChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.uplink_packet_loss_fraction =
+      kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogJustOnceOnMultipleChanges) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.uplink_packet_loss_fraction =
+      kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction;
+  state.runtime_config.frame_length_ms = 20;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogAfterGradualChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.bitrate_bps = kHighBitrateBps + kMinBitrateChangeBps;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  for (int bitrate_bps = kHighBitrateBps;
+       bitrate_bps <= kHighBitrateBps + kMinBitrateChangeBps; bitrate_bps++) {
+    state.runtime_config.bitrate_bps = bitrate_bps;
+    state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  }
+}
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc

@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+
+#include <string>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+class NullSmoothingFilter final : public SmoothingFilter {
+ public:
+  void AddSample(float sample) override { last_sample_ = sample; }
+
+  absl::optional<float> GetAverage() override { return last_sample_; }
+
+  bool SetTimeConstantMs(int time_constant_ms) override {
+    RTC_NOTREACHED();
+    return false;
+  }
+
+ private:
+  absl::optional<float> last_sample_;
+};
+}  // namespace
+
+FecControllerPlrBased::Config::Config(
+    bool initial_fec_enabled,
+    const ThresholdCurve& fec_enabling_threshold,
+    const ThresholdCurve& fec_disabling_threshold,
+    int time_constant_ms)
+    : initial_fec_enabled(initial_fec_enabled),
+      fec_enabling_threshold(fec_enabling_threshold),
+      fec_disabling_threshold(fec_disabling_threshold),
+      time_constant_ms(time_constant_ms) {}
+
+FecControllerPlrBased::FecControllerPlrBased(
+    const Config& config,
+    std::unique_ptr<SmoothingFilter> smoothing_filter)
+    : config_(config),
+      fec_enabled_(config.initial_fec_enabled),
+      packet_loss_smoother_(std::move(smoothing_filter)) {
+  RTC_DCHECK(config_.fec_disabling_threshold <= config_.fec_enabling_threshold);
+}
+
+FecControllerPlrBased::FecControllerPlrBased(const Config& config)
+    : FecControllerPlrBased(
+          config,
+          webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"
+              ? std::unique_ptr<NullSmoothingFilter>(new NullSmoothingFilter())
+              : std::unique_ptr<SmoothingFilter>(
+                    new SmoothingFilterImpl(config.time_constant_ms))) {}
+
+FecControllerPlrBased::~FecControllerPlrBased() = default;
+
+void FecControllerPlrBased::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.uplink_bandwidth_bps)
+    uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+  if (network_metrics.uplink_packet_loss_fraction) {
+    packet_loss_smoother_->AddSample(
+        *network_metrics.uplink_packet_loss_fraction);
+  }
+}
+
+void FecControllerPlrBased::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  RTC_DCHECK(!config->enable_fec);
+  RTC_DCHECK(!config->uplink_packet_loss_fraction);
+
+  const auto& packet_loss = packet_loss_smoother_->GetAverage();
+
+  fec_enabled_ = fec_enabled_ ? !FecDisablingDecision(packet_loss)
+                              : FecEnablingDecision(packet_loss);
+
+  config->enable_fec = fec_enabled_;
+
+  config->uplink_packet_loss_fraction = packet_loss ? *packet_loss : 0.0;
+}
+
+bool FecControllerPlrBased::FecEnablingDecision(
+    const absl::optional<float>& packet_loss) const {
+  if (!uplink_bandwidth_bps_ || !packet_loss) {
+    return false;
+  } else {
+    // Enable when above the curve or exactly on it.
+    return !config_.fec_enabling_threshold.IsBelowCurve(
+        {static_cast<float>(*uplink_bandwidth_bps_), *packet_loss});
+  }
+}
+
+bool FecControllerPlrBased::FecDisablingDecision(
+    const absl::optional<float>& packet_loss) const {
+  if (!uplink_bandwidth_bps_ || !packet_loss) {
+    return false;
+  } else {
+    // Disable when below the curve.
+    return config_.fec_disabling_threshold.IsBelowCurve(
+        {static_cast<float>(*uplink_bandwidth_bps_), *packet_loss});
+  }
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h

@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "common_audio/smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class FecControllerPlrBased final : public Controller {
+ public:
+  struct Config {
+    // |fec_enabling_threshold| defines a curve, above which FEC should be
+    // enabled. |fec_disabling_threshold| defines a curve, under which FEC
+    // should be disabled. See below
+    //
+    // packet-loss ^   |  |
+    //             |   |  |   FEC
+    //             |    \  \   ON
+    //             | FEC \  \_______ fec_enabling_threshold
+    //             | OFF  \_________ fec_disabling_threshold
+    //             |-----------------> bandwidth
+    Config(bool initial_fec_enabled,
+           const ThresholdCurve& fec_enabling_threshold,
+           const ThresholdCurve& fec_disabling_threshold,
+           int time_constant_ms);
+    bool initial_fec_enabled;
+    ThresholdCurve fec_enabling_threshold;
+    ThresholdCurve fec_disabling_threshold;
+    int time_constant_ms;
+  };
+
+  // Dependency injection for testing.
+  FecControllerPlrBased(const Config& config,
+                        std::unique_ptr<SmoothingFilter> smoothing_filter);
+
+  explicit FecControllerPlrBased(const Config& config);
+
+  ~FecControllerPlrBased() override;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  bool FecEnablingDecision(const absl::optional<float>& packet_loss) const;
+  bool FecDisablingDecision(const absl::optional<float>& packet_loss) const;
+
+  const Config config_;
+  bool fec_enabled_;
+  absl::optional<int> uplink_bandwidth_bps_;
+  const std::unique_ptr<SmoothingFilter> packet_loss_smoother_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerPlrBased);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_

modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc

@@ -0,0 +1,489 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+
+#include <utility>
+
+#include "common_audio/mocks/mock_smoothing_filter.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace {
+
+// The test uses the following settings:
+//
+// packet-loss ^   |  |
+//             |  A| C|   FEC
+//             |    \  \   ON
+//             | FEC \ D\_______
+//             | OFF B\_________
+//             |-----------------> bandwidth
+//
+// A : (kDisablingBandwidthLow, kDisablingPacketLossAtLowBw)
+// B : (kDisablingBandwidthHigh, kDisablingPacketLossAtHighBw)
+// C : (kEnablingBandwidthLow, kEnablingPacketLossAtLowBw)
+// D : (kEnablingBandwidthHigh, kEnablingPacketLossAtHighBw)
+
+constexpr int kDisablingBandwidthLow = 15000;
+constexpr float kDisablingPacketLossAtLowBw = 0.08f;
+constexpr int kDisablingBandwidthHigh = 64000;
+constexpr float kDisablingPacketLossAtHighBw = 0.01f;
+constexpr int kEnablingBandwidthLow = 17000;
+constexpr float kEnablingPacketLossAtLowBw = 0.1f;
+constexpr int kEnablingBandwidthHigh = 64000;
+constexpr float kEnablingPacketLossAtHighBw = 0.05f;
+
+constexpr float kEpsilon = 1e-5f;
+
+struct FecControllerPlrBasedTestStates {
+  std::unique_ptr<FecControllerPlrBased> controller;
+  MockSmoothingFilter* packet_loss_smoother;
+};
+
+FecControllerPlrBasedTestStates CreateFecControllerPlrBased(
+    bool initial_fec_enabled,
+    const ThresholdCurve& enabling_curve,
+    const ThresholdCurve& disabling_curve) {
+  FecControllerPlrBasedTestStates states;
+  std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
+      new NiceMock<MockSmoothingFilter>());
+  states.packet_loss_smoother = mock_smoothing_filter.get();
+  states.controller.reset(new FecControllerPlrBased(
+      FecControllerPlrBased::Config(initial_fec_enabled, enabling_curve,
+                                    disabling_curve, 0),
+      std::move(mock_smoothing_filter)));
+  return states;
+}
+
+FecControllerPlrBasedTestStates CreateFecControllerPlrBased(
+    bool initial_fec_enabled) {
+  return CreateFecControllerPlrBased(
+      initial_fec_enabled,
+      ThresholdCurve(kEnablingBandwidthLow, kEnablingPacketLossAtLowBw,
+                     kEnablingBandwidthHigh, kEnablingPacketLossAtHighBw),
+      ThresholdCurve(kDisablingBandwidthLow, kDisablingPacketLossAtLowBw,
+                     kDisablingBandwidthHigh, kDisablingPacketLossAtHighBw));
+}
+
+void UpdateNetworkMetrics(FecControllerPlrBasedTestStates* states,
+                          const absl::optional<int>& uplink_bandwidth_bps,
+                          const absl::optional<float>& uplink_packet_loss) {
+  // UpdateNetworkMetrics can accept multiple network metric updates at once.
+  // However, currently, the most used case is to update one metric at a time.
+  // To reflect this fact, we separate the calls.
+  if (uplink_bandwidth_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+    states->controller->UpdateNetworkMetrics(network_metrics);
+  }
+  if (uplink_packet_loss) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_packet_loss_fraction = uplink_packet_loss;
+    EXPECT_CALL(*states->packet_loss_smoother, AddSample(*uplink_packet_loss));
+    states->controller->UpdateNetworkMetrics(network_metrics);
+    // This is called during CheckDecision().
+    EXPECT_CALL(*states->packet_loss_smoother, GetAverage())
+        .WillOnce(Return(*uplink_packet_loss));
+  }
+}
+
+// Checks that the FEC decision and |uplink_packet_loss_fraction| given by
+// |states->controller->MakeDecision| matches |expected_enable_fec| and
+// |expected_uplink_packet_loss_fraction|, respectively.
+void CheckDecision(FecControllerPlrBasedTestStates* states,
+                   bool expected_enable_fec,
+                   float expected_uplink_packet_loss_fraction) {
+  AudioEncoderRuntimeConfig config;
+  states->controller->MakeDecision(&config);
+  EXPECT_EQ(expected_enable_fec, config.enable_fec);
+  EXPECT_EQ(expected_uplink_packet_loss_fraction,
+            config.uplink_packet_loss_fraction);
+}
+
+}  // namespace
+
+TEST(FecControllerPlrBasedTest, OutputInitValueBeforeAnyInputsAreReceived) {
+  for (bool initial_fec_enabled : {false, true}) {
+    auto states = CreateFecControllerPlrBased(initial_fec_enabled);
+    CheckDecision(&states, initial_fec_enabled, 0);
+  }
+}
+
+TEST(FecControllerPlrBasedTest, OutputInitValueWhenUplinkBandwidthUnknown) {
+  // Regardless of the initial FEC state and the packet-loss rate,
+  // the initial FEC state is maintained as long as the BWE is unknown.
+  for (bool initial_fec_enabled : {false, true}) {
+    for (float packet_loss :
+         {kDisablingPacketLossAtLowBw - kEpsilon, kDisablingPacketLossAtLowBw,
+          kDisablingPacketLossAtLowBw + kEpsilon,
+          kEnablingPacketLossAtLowBw - kEpsilon, kEnablingPacketLossAtLowBw,
+          kEnablingPacketLossAtLowBw + kEpsilon}) {
+      auto states = CreateFecControllerPlrBased(initial_fec_enabled);
+      UpdateNetworkMetrics(&states, absl::nullopt, packet_loss);
+      CheckDecision(&states, initial_fec_enabled, packet_loss);
+    }
+  }
+}
+
+TEST(FecControllerPlrBasedTest,
+     OutputInitValueWhenUplinkPacketLossFractionUnknown) {
+  // Regardless of the initial FEC state and the BWE, the initial FEC state
+  // is maintained as long as the packet-loss rate is unknown.
+  for (bool initial_fec_enabled : {false, true}) {
+    for (int bandwidth : {kDisablingBandwidthLow - 1, kDisablingBandwidthLow,
+                          kDisablingBandwidthLow + 1, kEnablingBandwidthLow - 1,
+                          kEnablingBandwidthLow, kEnablingBandwidthLow + 1}) {
+      auto states = CreateFecControllerPlrBased(initial_fec_enabled);
+      UpdateNetworkMetrics(&states, bandwidth, absl::nullopt);
+      CheckDecision(&states, initial_fec_enabled, 0.0);
+    }
+  }
+}
+
+TEST(FecControllerPlrBasedTest, EnableFecForHighBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  UpdateNetworkMetrics(&states, kEnablingBandwidthHigh,
+                       kEnablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kEnablingPacketLossAtHighBw);
+}
+
+TEST(FecControllerPlrBasedTest, UpdateMultipleNetworkMetricsAtOnce) {
+  // This test is similar to EnableFecForHighBandwidth. But instead of
+  // using ::UpdateNetworkMetrics(...), which calls
+  // FecControllerPlrBased::UpdateNetworkMetrics(...) multiple times, we
+  // we call it only once. This is to verify that
+  // FecControllerPlrBased::UpdateNetworkMetrics(...) can handle multiple
+  // network updates at once. This is, however, not a common use case in current
+  // audio_network_adaptor_impl.cc.
+  auto states = CreateFecControllerPlrBased(false);
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.uplink_bandwidth_bps = kEnablingBandwidthHigh;
+  network_metrics.uplink_packet_loss_fraction = kEnablingPacketLossAtHighBw;
+  EXPECT_CALL(*states.packet_loss_smoother, GetAverage())
+      .WillOnce(Return(kEnablingPacketLossAtHighBw));
+  states.controller->UpdateNetworkMetrics(network_metrics);
+  CheckDecision(&states, true, kEnablingPacketLossAtHighBw);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOffForHighBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  constexpr float kPacketLoss = kEnablingPacketLossAtHighBw * 0.99f;
+  UpdateNetworkMetrics(&states, kEnablingBandwidthHigh, kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, EnableFecForMediumBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  constexpr float kPacketLoss =
+      (kEnablingPacketLossAtLowBw + kEnablingPacketLossAtHighBw) / 2.0;
+  UpdateNetworkMetrics(&states,
+                       (kEnablingBandwidthHigh + kEnablingBandwidthLow) / 2,
+                       kPacketLoss);
+  CheckDecision(&states, true, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOffForMediumBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  constexpr float kPacketLoss =
+      kEnablingPacketLossAtLowBw * 0.49f + kEnablingPacketLossAtHighBw * 0.51f;
+  UpdateNetworkMetrics(&states,
+                       (kEnablingBandwidthHigh + kEnablingBandwidthLow) / 2,
+                       kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, EnableFecForLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow,
+                       kEnablingPacketLossAtLowBw);
+  CheckDecision(&states, true, kEnablingPacketLossAtLowBw);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOffForLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  constexpr float kPacketLoss = kEnablingPacketLossAtLowBw * 0.99f;
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow, kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOffForVeryLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  // Below |kEnablingBandwidthLow|, no packet loss fraction can cause FEC to
+  // turn on.
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow - 1, 1.0);
+  CheckDecision(&states, false, 1.0);
+}
+
+TEST(FecControllerPlrBasedTest, DisableFecForHighBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  constexpr float kPacketLoss = kDisablingPacketLossAtHighBw - kEpsilon;
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh, kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOnForHighBandwidth) {
+  // Note: Disabling happens when the value is strictly below the threshold.
+  auto states = CreateFecControllerPlrBased(true);
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh,
+                       kDisablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kDisablingPacketLossAtHighBw);
+}
+
+TEST(FecControllerPlrBasedTest, DisableFecOnMediumBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  constexpr float kPacketLoss =
+      (kDisablingPacketLossAtLowBw + kDisablingPacketLossAtHighBw) / 2.0f -
+      kEpsilon;
+  UpdateNetworkMetrics(&states,
+                       (kDisablingBandwidthHigh + kDisablingBandwidthLow) / 2,
+                       kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOnForMediumBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  constexpr float kPacketLoss = kDisablingPacketLossAtLowBw * 0.51f +
+                                kDisablingPacketLossAtHighBw * 0.49f - kEpsilon;
+  UpdateNetworkMetrics(&states,
+                       (kEnablingBandwidthHigh + kDisablingBandwidthLow) / 2,
+                       kPacketLoss);
+  CheckDecision(&states, true, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, DisableFecForLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  constexpr float kPacketLoss = kDisablingPacketLossAtLowBw - kEpsilon;
+  UpdateNetworkMetrics(&states, kDisablingBandwidthLow, kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, DisableFecForVeryLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  // Below |kEnablingBandwidthLow|, any packet loss fraction can cause FEC to
+  // turn off.
+  UpdateNetworkMetrics(&states, kDisablingBandwidthLow - 1, 1.0);
+  CheckDecision(&states, false, 1.0);
+}
+
+TEST(FecControllerPlrBasedTest, CheckBehaviorOnChangingNetworkMetrics) {
+  // In this test, we let the network metrics to traverse from 1 to 5.
+  // packet-loss ^ 1 |  |
+  //             |   | 2|
+  //             |    \  \ 3
+  //             |     \4 \_______
+  //             |      \_________
+  //             |---------5-------> bandwidth
+
+  auto states = CreateFecControllerPlrBased(true);
+  UpdateNetworkMetrics(&states, kDisablingBandwidthLow - 1, 1.0);
+  CheckDecision(&states, false, 1.0);
+
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow,
+                       kEnablingPacketLossAtLowBw * 0.99f);
+  CheckDecision(&states, false, kEnablingPacketLossAtLowBw * 0.99f);
+
+  UpdateNetworkMetrics(&states, kEnablingBandwidthHigh,
+                       kEnablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kEnablingPacketLossAtHighBw);
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh,
+                       kDisablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kDisablingPacketLossAtHighBw);
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh + 1, 0.0);
+  CheckDecision(&states, false, 0.0);
+}
+
+TEST(FecControllerPlrBasedTest, CheckBehaviorOnSpecialCurves) {
+  // We test a special configuration, where the points to define the FEC
+  // enabling/disabling curves are placed like the following, otherwise the test
+  // is the same as CheckBehaviorOnChangingNetworkMetrics.
+  //
+  // packet-loss ^   |  |
+  //             |   | C|
+  //             |   |  |
+  //             |   | D|_______
+  //             |  A|___B______
+  //             |-----------------> bandwidth
+
+  constexpr int kEnablingBandwidthHigh = kEnablingBandwidthLow;
+  constexpr float kDisablingPacketLossAtLowBw = kDisablingPacketLossAtHighBw;
+  FecControllerPlrBasedTestStates states;
+  std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
+      new NiceMock<MockSmoothingFilter>());
+  states.packet_loss_smoother = mock_smoothing_filter.get();
+  states.controller.reset(new FecControllerPlrBased(
+      FecControllerPlrBased::Config(
+          true,
+          ThresholdCurve(kEnablingBandwidthLow, kEnablingPacketLossAtLowBw,
+                         kEnablingBandwidthHigh, kEnablingPacketLossAtHighBw),
+          ThresholdCurve(kDisablingBandwidthLow, kDisablingPacketLossAtLowBw,
+                         kDisablingBandwidthHigh, kDisablingPacketLossAtHighBw),
+          0),
+      std::move(mock_smoothing_filter)));
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthLow - 1, 1.0);
+  CheckDecision(&states, false, 1.0);
+
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow,
+                       kEnablingPacketLossAtHighBw * 0.99f);
+  CheckDecision(&states, false, kEnablingPacketLossAtHighBw * 0.99f);
+
+  UpdateNetworkMetrics(&states, kEnablingBandwidthHigh,
+                       kEnablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kEnablingPacketLossAtHighBw);
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh,
+                       kDisablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kDisablingPacketLossAtHighBw);
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh + 1, 0.0);
+  CheckDecision(&states, false, 0.0);
+}
+
+TEST(FecControllerPlrBasedTest, SingleThresholdCurveForEnablingAndDisabling) {
+  // Note: To avoid numerical errors, keep kPacketLossAtLowBw and
+  // kPacketLossAthighBw as (negative) integer powers of 2.
+  // This is mostly relevant for the O3 case.
+  constexpr int kBandwidthLow = 10000;
+  constexpr float kPacketLossAtLowBw = 0.25f;
+  constexpr int kBandwidthHigh = 20000;
+  constexpr float kPacketLossAtHighBw = 0.125f;
+  auto curve = ThresholdCurve(kBandwidthLow, kPacketLossAtLowBw, kBandwidthHigh,
+                              kPacketLossAtHighBw);
+
+  // B* stands for "below-curve", O* for "on-curve", and A* for "above-curve".
+  //
+  //                                            //
+  // packet-loss ^                              //
+  //             |    |                         //
+  //             | B1 O1                        //
+  //             |    |                         //
+  //             |    O2                        //
+  //             |     \ A1                     //
+  //             |      \                       //
+  //             |       O3   A2                //
+  //             |     B2 \                     //
+  //             |         \                    //
+  //             |          O4--O5----          //
+  //             |                              //
+  //             |            B3                //
+  //             |-----------------> bandwidth  //
+
+  struct NetworkState {
+    int bandwidth;
+    float packet_loss;
+  };
+
+  std::vector<NetworkState> below{
+      {kBandwidthLow - 1, kPacketLossAtLowBw + 0.1f},  // B1
+      {(kBandwidthLow + kBandwidthHigh) / 2,
+       (kPacketLossAtLowBw + kPacketLossAtHighBw) / 2 - kEpsilon},  // B2
+      {kBandwidthHigh + 1, kPacketLossAtHighBw - kEpsilon}          // B3
+  };
+
+  std::vector<NetworkState> on{
+      {kBandwidthLow, kPacketLossAtLowBw + 0.1f},  // O1
+      {kBandwidthLow, kPacketLossAtLowBw},         // O2
+      {(kBandwidthLow + kBandwidthHigh) / 2,
+       (kPacketLossAtLowBw + kPacketLossAtHighBw) / 2},  // O3
+      {kBandwidthHigh, kPacketLossAtHighBw},             // O4
+      {kBandwidthHigh + 1, kPacketLossAtHighBw},         // O5
+  };
+
+  std::vector<NetworkState> above{
+      {(kBandwidthLow + kBandwidthHigh) / 2,
+       (kPacketLossAtLowBw + kPacketLossAtHighBw) / 2 + kEpsilon},  // A1
+      {kBandwidthHigh + 1, kPacketLossAtHighBw + kEpsilon},         // A2
+  };
+
+  // Test that FEC is turned off whenever we're below the curve, independent
+  // of the starting FEC state.
+  for (NetworkState net_state : below) {
+    for (bool initial_fec_enabled : {false, true}) {
+      auto states =
+          CreateFecControllerPlrBased(initial_fec_enabled, curve, curve);
+      UpdateNetworkMetrics(&states, net_state.bandwidth, net_state.packet_loss);
+      CheckDecision(&states, false, net_state.packet_loss);
+    }
+  }
+
+  // Test that FEC is turned on whenever we're on the curve or above it,
+  // independent of the starting FEC state.
+  for (const std::vector<NetworkState>& states_list : {on, above}) {
+    for (NetworkState net_state : states_list) {
+      for (bool initial_fec_enabled : {false, true}) {
+        auto states =
+            CreateFecControllerPlrBased(initial_fec_enabled, curve, curve);
+        UpdateNetworkMetrics(&states, net_state.bandwidth,
+                             net_state.packet_loss);
+        CheckDecision(&states, true, net_state.packet_loss);
+      }
+    }
+  }
+}
+
+TEST(FecControllerPlrBasedTest, FecAlwaysOff) {
+  ThresholdCurve always_off_curve(0, 1.0f + kEpsilon, 0, 1.0f + kEpsilon);
+  for (bool initial_fec_enabled : {false, true}) {
+    for (int bandwidth : {0, 10000}) {
+      for (float packet_loss : {0.0f, 0.5f, 1.0f}) {
+        auto states = CreateFecControllerPlrBased(
+            initial_fec_enabled, always_off_curve, always_off_curve);
+        UpdateNetworkMetrics(&states, bandwidth, packet_loss);
+        CheckDecision(&states, false, packet_loss);
+      }
+    }
+  }
+}
+
+TEST(FecControllerPlrBasedTest, FecAlwaysOn) {
+  ThresholdCurve always_on_curve(0, 0.0f, 0, 0.0f);
+  for (bool initial_fec_enabled : {false, true}) {
+    for (int bandwidth : {0, 10000}) {
+      for (float packet_loss : {0.0f, 0.5f, 1.0f}) {
+        auto states = CreateFecControllerPlrBased(
+            initial_fec_enabled, always_on_curve, always_on_curve);
+        UpdateNetworkMetrics(&states, bandwidth, packet_loss);
+        CheckDecision(&states, true, packet_loss);
+      }
+    }
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(FecControllerPlrBasedDeathTest, InvalidConfig) {
+  FecControllerPlrBasedTestStates states;
+  std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
+      new NiceMock<MockSmoothingFilter>());
+  states.packet_loss_smoother = mock_smoothing_filter.get();
+  EXPECT_DEATH(
+      states.controller.reset(new FecControllerPlrBased(
+          FecControllerPlrBased::Config(
+              true,
+              ThresholdCurve(kDisablingBandwidthLow - 1,
+                             kEnablingPacketLossAtLowBw, kEnablingBandwidthHigh,
+                             kEnablingPacketLossAtHighBw),
+              ThresholdCurve(
+                  kDisablingBandwidthLow, kDisablingPacketLossAtLowBw,
+                  kDisablingBandwidthHigh, kDisablingPacketLossAtHighBw),
+              0),
+          std::move(mock_smoothing_filter))),
+      "Check failed");
+}
+#endif
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/frame_length_controller.cc

@@ -0,0 +1,201 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+
+#include <algorithm>
+#include <iterator>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kPreventOveruseMarginBps = 5000;
+
+int OverheadRateBps(size_t overhead_bytes_per_packet, int frame_length_ms) {
+  return static_cast<int>(overhead_bytes_per_packet * 8 * 1000 /
+                          frame_length_ms);
+}
+}  // namespace
+
+FrameLengthController::Config::Config(
+    const std::set<int>& encoder_frame_lengths_ms,
+    int initial_frame_length_ms,
+    int min_encoder_bitrate_bps,
+    float fl_increasing_packet_loss_fraction,
+    float fl_decreasing_packet_loss_fraction,
+    int fl_increase_overhead_offset,
+    int fl_decrease_overhead_offset,
+    std::map<FrameLengthChange, int> fl_changing_bandwidths_bps)
+    : encoder_frame_lengths_ms(encoder_frame_lengths_ms),
+      initial_frame_length_ms(initial_frame_length_ms),
+      min_encoder_bitrate_bps(min_encoder_bitrate_bps),
+      fl_increasing_packet_loss_fraction(fl_increasing_packet_loss_fraction),
+      fl_decreasing_packet_loss_fraction(fl_decreasing_packet_loss_fraction),
+      fl_increase_overhead_offset(fl_increase_overhead_offset),
+      fl_decrease_overhead_offset(fl_decrease_overhead_offset),
+      fl_changing_bandwidths_bps(std::move(fl_changing_bandwidths_bps)) {}
+
+FrameLengthController::Config::Config(const Config& other) = default;
+
+FrameLengthController::Config::~Config() = default;
+
+FrameLengthController::FrameLengthController(const Config& config)
+    : config_(config) {
+  frame_length_ms_ = std::find(config_.encoder_frame_lengths_ms.begin(),
+                               config_.encoder_frame_lengths_ms.end(),
+                               config_.initial_frame_length_ms);
+  // |encoder_frame_lengths_ms| must contain |initial_frame_length_ms|.
+  RTC_DCHECK(frame_length_ms_ != config_.encoder_frame_lengths_ms.end());
+}
+
+FrameLengthController::~FrameLengthController() = default;
+
+void FrameLengthController::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.uplink_bandwidth_bps)
+    uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+  if (network_metrics.uplink_packet_loss_fraction)
+    uplink_packet_loss_fraction_ = network_metrics.uplink_packet_loss_fraction;
+  if (network_metrics.overhead_bytes_per_packet)
+    overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet;
+}
+
+void FrameLengthController::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  // Decision on |frame_length_ms| should not have been made.
+  RTC_DCHECK(!config->frame_length_ms);
+
+  if (FrameLengthIncreasingDecision(*config)) {
+    prev_decision_increase_ = true;
+  } else if (FrameLengthDecreasingDecision(*config)) {
+    prev_decision_increase_ = false;
+  }
+  config->last_fl_change_increase = prev_decision_increase_;
+  config->frame_length_ms = *frame_length_ms_;
+}
+
+FrameLengthController::Config::FrameLengthChange::FrameLengthChange(
+    int from_frame_length_ms,
+    int to_frame_length_ms)
+    : from_frame_length_ms(from_frame_length_ms),
+      to_frame_length_ms(to_frame_length_ms) {}
+
+bool FrameLengthController::Config::FrameLengthChange::operator<(
+    const FrameLengthChange& rhs) const {
+  return from_frame_length_ms < rhs.from_frame_length_ms ||
+         (from_frame_length_ms == rhs.from_frame_length_ms &&
+          to_frame_length_ms < rhs.to_frame_length_ms);
+}
+
+bool FrameLengthController::FrameLengthIncreasingDecision(
+    const AudioEncoderRuntimeConfig& config) {
+  // Increase frame length if
+  // 1. |uplink_bandwidth_bps| is known to be smaller or equal than
+  //    |min_encoder_bitrate_bps| plus |prevent_overuse_margin_bps| plus the
+  //    current overhead rate OR all the following:
+  // 2. longer frame length is available AND
+  // 3. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
+  // 4. |uplink_packet_loss_fraction| is known to be smaller than a threshold.
+
+  // Find next frame length to which a criterion is defined to shift from
+  // current frame length.
+  auto longer_frame_length_ms = std::next(frame_length_ms_);
+  auto increase_threshold = config_.fl_changing_bandwidths_bps.end();
+  while (longer_frame_length_ms != config_.encoder_frame_lengths_ms.end()) {
+    increase_threshold = config_.fl_changing_bandwidths_bps.find(
+        Config::FrameLengthChange(*frame_length_ms_, *longer_frame_length_ms));
+    if (increase_threshold != config_.fl_changing_bandwidths_bps.end())
+      break;
+    longer_frame_length_ms = std::next(longer_frame_length_ms);
+  }
+
+  if (increase_threshold == config_.fl_changing_bandwidths_bps.end())
+    return false;
+
+  // Check that
+  // -(*overhead_bytes_per_packet_) <= offset <= (*overhead_bytes_per_packet_)
+  RTC_DCHECK(
+      !overhead_bytes_per_packet_ ||
+      (overhead_bytes_per_packet_ &&
+       static_cast<size_t>(std::max(0, -config_.fl_increase_overhead_offset)) <=
+           *overhead_bytes_per_packet_ &&
+       static_cast<size_t>(std::max(0, config_.fl_increase_overhead_offset)) <=
+           *overhead_bytes_per_packet_));
+
+  if (uplink_bandwidth_bps_ && overhead_bytes_per_packet_ &&
+      *uplink_bandwidth_bps_ <=
+          config_.min_encoder_bitrate_bps + kPreventOveruseMarginBps +
+              OverheadRateBps(*overhead_bytes_per_packet_ +
+                                  config_.fl_increase_overhead_offset,
+                              *frame_length_ms_)) {
+    frame_length_ms_ = longer_frame_length_ms;
+    return true;
+  }
+
+  if ((uplink_bandwidth_bps_ &&
+       *uplink_bandwidth_bps_ <= increase_threshold->second) &&
+      (uplink_packet_loss_fraction_ &&
+       *uplink_packet_loss_fraction_ <=
+           config_.fl_increasing_packet_loss_fraction)) {
+    frame_length_ms_ = longer_frame_length_ms;
+    return true;
+  }
+  return false;
+}
+
+bool FrameLengthController::FrameLengthDecreasingDecision(
+    const AudioEncoderRuntimeConfig& config) {
+  // Decrease frame length if
+  // 1. shorter frame length is available AND
+  // 2. |uplink_bandwidth_bps| is known to be bigger than
+  // |min_encoder_bitrate_bps| plus |prevent_overuse_margin_bps| plus the
+  // overhead which would be produced with the shorter frame length AND
+  // one or more of the followings:
+  // 3. |uplink_bandwidth_bps| is known to be larger than a threshold,
+  // 4. |uplink_packet_loss_fraction| is known to be larger than a threshold,
+
+  // Find next frame length to which a criterion is defined to shift from
+  // current frame length.
+  auto shorter_frame_length_ms = frame_length_ms_;
+  auto decrease_threshold = config_.fl_changing_bandwidths_bps.end();
+  while (shorter_frame_length_ms != config_.encoder_frame_lengths_ms.begin()) {
+    shorter_frame_length_ms = std::prev(shorter_frame_length_ms);
+    decrease_threshold = config_.fl_changing_bandwidths_bps.find(
+        Config::FrameLengthChange(*frame_length_ms_, *shorter_frame_length_ms));
+    if (decrease_threshold != config_.fl_changing_bandwidths_bps.end())
+      break;
+  }
+
+  if (decrease_threshold == config_.fl_changing_bandwidths_bps.end())
+    return false;
+
+  if (uplink_bandwidth_bps_ && overhead_bytes_per_packet_ &&
+      *uplink_bandwidth_bps_ <=
+          config_.min_encoder_bitrate_bps + kPreventOveruseMarginBps +
+              OverheadRateBps(*overhead_bytes_per_packet_ +
+                                  config_.fl_decrease_overhead_offset,
+                              *shorter_frame_length_ms)) {
+    return false;
+  }
+
+  if ((uplink_bandwidth_bps_ &&
+       *uplink_bandwidth_bps_ >= decrease_threshold->second) ||
+      (uplink_packet_loss_fraction_ &&
+       *uplink_packet_loss_fraction_ >=
+           config_.fl_decreasing_packet_loss_fraction)) {
+    frame_length_ms_ = shorter_frame_length_ms;
+    return true;
+  }
+  return false;
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/frame_length_controller.h

@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
+
+#include <stddef.h>
+
+#include <map>
+#include <set>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+// Determines target frame length based on the network metrics and the decision
+// of FEC controller.
+class FrameLengthController final : public Controller {
+ public:
+  struct Config {
+    struct FrameLengthChange {
+      FrameLengthChange(int from_frame_length_ms, int to_frame_length_ms);
+      bool operator<(const FrameLengthChange& rhs) const;
+      int from_frame_length_ms;
+      int to_frame_length_ms;
+    };
+    Config(const std::set<int>& encoder_frame_lengths_ms,
+           int initial_frame_length_ms,
+           int min_encoder_bitrate_bps,
+           float fl_increasing_packet_loss_fraction,
+           float fl_decreasing_packet_loss_fraction,
+           int fl_increase_overhead_offset,
+           int fl_decrease_overhead_offset,
+           std::map<FrameLengthChange, int> fl_changing_bandwidths_bps);
+    Config(const Config& other);
+    ~Config();
+    std::set<int> encoder_frame_lengths_ms;
+    int initial_frame_length_ms;
+    int min_encoder_bitrate_bps;
+    // Uplink packet loss fraction below which frame length can increase.
+    float fl_increasing_packet_loss_fraction;
+    // Uplink packet loss fraction below which frame length should decrease.
+    float fl_decreasing_packet_loss_fraction;
+    // Offset to apply to overhead calculation when increasing frame length.
+    int fl_increase_overhead_offset;
+    // Offset to apply to overhead calculation when decreasing frame length.
+    int fl_decrease_overhead_offset;
+    std::map<FrameLengthChange, int> fl_changing_bandwidths_bps;
+  };
+
+  explicit FrameLengthController(const Config& config);
+
+  ~FrameLengthController() override;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  bool FrameLengthIncreasingDecision(const AudioEncoderRuntimeConfig& config);
+
+  bool FrameLengthDecreasingDecision(const AudioEncoderRuntimeConfig& config);
+
+  const Config config_;
+
+  std::set<int>::const_iterator frame_length_ms_;
+
+  absl::optional<int> uplink_bandwidth_bps_;
+
+  absl::optional<float> uplink_packet_loss_fraction_;
+
+  absl::optional<size_t> overhead_bytes_per_packet_;
+
+  // True if the previous frame length decision was an increase, otherwise
+  // false.
+  bool prev_decision_increase_ = false;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(FrameLengthController);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc

@@ -0,0 +1,444 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+
+#include <memory>
+#include <utility>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr float kFlIncreasingPacketLossFraction = 0.04f;
+constexpr float kFlDecreasingPacketLossFraction = 0.05f;
+constexpr int kFlIncreaseOverheadOffset = 0;
+constexpr int kFlDecreaseOverheadOffset = 0;
+constexpr int kMinEncoderBitrateBps = 6000;
+constexpr int kPreventOveruseMarginBps = 5000;
+constexpr size_t kOverheadBytesPerPacket = 20;
+constexpr int kFl20msTo60msBandwidthBps = 40000;
+constexpr int kFl60msTo20msBandwidthBps = 50000;
+constexpr int kFl60msTo120msBandwidthBps = 30000;
+constexpr int kFl120msTo60msBandwidthBps = 40000;
+constexpr int kFl20msTo40msBandwidthBps = 45000;
+constexpr int kFl40msTo20msBandwidthBps = 50000;
+constexpr int kFl40msTo60msBandwidthBps = 40000;
+constexpr int kFl60msTo40msBandwidthBps = 45000;
+
+constexpr int kMediumBandwidthBps =
+    (kFl40msTo20msBandwidthBps + kFl20msTo40msBandwidthBps) / 2;
+constexpr float kMediumPacketLossFraction =
+    (kFlDecreasingPacketLossFraction + kFlIncreasingPacketLossFraction) / 2;
+const std::set<int> kDefaultEncoderFrameLengthsMs = {20, 40, 60, 120};
+
+int VeryLowBitrate(int frame_length_ms) {
+  return kMinEncoderBitrateBps + kPreventOveruseMarginBps +
+         (kOverheadBytesPerPacket * 8 * 1000 / frame_length_ms);
+}
+
+std::unique_ptr<FrameLengthController> CreateController(
+    const std::map<FrameLengthController::Config::FrameLengthChange, int>&
+        frame_length_change_criteria,
+    const std::set<int>& encoder_frame_lengths_ms,
+    int initial_frame_length_ms) {
+  std::unique_ptr<FrameLengthController> controller(
+      new FrameLengthController(FrameLengthController::Config(
+          encoder_frame_lengths_ms, initial_frame_length_ms,
+          kMinEncoderBitrateBps, kFlIncreasingPacketLossFraction,
+          kFlDecreasingPacketLossFraction, kFlIncreaseOverheadOffset,
+          kFlDecreaseOverheadOffset, frame_length_change_criteria)));
+
+  return controller;
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor20msAnd60ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(20, 60),
+       kFl20msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 20),
+       kFl60msTo20msBandwidthBps}};
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor20msAnd40ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(20, 40),
+       kFl20msTo40msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(40, 20),
+       kFl40msTo20msBandwidthBps}};
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor20ms60msAnd120ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(20, 60),
+       kFl20msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 20),
+       kFl60msTo20msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 120),
+       kFl60msTo120msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(120, 60),
+       kFl120msTo60msBandwidthBps}};
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor20ms40ms60msAnd120ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(20, 60),
+       kFl20msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 20),
+       kFl60msTo20msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(20, 40),
+       kFl20msTo40msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(40, 20),
+       kFl40msTo20msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(40, 60),
+       kFl40msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 40),
+       kFl60msTo40msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 120),
+       kFl60msTo120msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(120, 60),
+       kFl120msTo60msBandwidthBps}};
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor40msAnd60ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(40, 60),
+       kFl40msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 40),
+       kFl60msTo40msBandwidthBps}};
+}
+
+void UpdateNetworkMetrics(
+    FrameLengthController* controller,
+    const absl::optional<int>& uplink_bandwidth_bps,
+    const absl::optional<float>& uplink_packet_loss_fraction,
+    const absl::optional<size_t>& overhead_bytes_per_packet) {
+  // UpdateNetworkMetrics can accept multiple network metric updates at once.
+  // However, currently, the most used case is to update one metric at a time.
+  // To reflect this fact, we separate the calls.
+  if (uplink_bandwidth_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  if (uplink_packet_loss_fraction) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  if (overhead_bytes_per_packet) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+}
+
+void CheckDecision(FrameLengthController* controller,
+                   int expected_frame_length_ms) {
+  AudioEncoderRuntimeConfig config;
+  controller->MakeDecision(&config);
+  EXPECT_EQ(expected_frame_length_ms, config.frame_length_ms);
+}
+
+}  // namespace
+
+TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
+                       absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkPacketLossFraction) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  UpdateNetworkMetrics(controller.get(), absl::nullopt,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest,
+     Maintain60MsIf20MsNotInReceiverFrameLengthRange) {
+  auto controller =
+      CreateController(CreateChangeCriteriaFor20msAnd60ms(), {60}, 60);
+  // Set FEC on that would cause frame length to decrease if receiver frame
+  // length range included 20ms.
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, IncreaseTo40MsOnMultipleConditions) {
+  // Increase to 40ms frame length if
+  // 1. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
+  // 2. |uplink_packet_loss_fraction| is known to be smaller than a threshold
+  //    AND
+  // 3. FEC is not decided or OFF.
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd40ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  UpdateNetworkMetrics(controller.get(), kFl20msTo40msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+}
+
+TEST(FrameLengthControllerTest, DecreaseTo40MsOnHighUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor40msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 40);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo40msBandwidthBps,
+                       absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+}
+
+TEST(FrameLengthControllerTest, Maintain60MsOnMultipleConditions) {
+  // Maintain 60ms frame length if
+  // 1. |uplink_bandwidth_bps| is at medium level,
+  // 2. |uplink_packet_loss_fraction| is at medium,
+  // 3. FEC is not decided ON.
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
+                       kMediumPacketLossFraction, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, IncreaseTo60MsOnMultipleConditions) {
+  // Increase to 60ms frame length if
+  // 1. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
+  // 2. |uplink_packet_loss_fraction| is known to be smaller than a threshold
+  //    AND
+  // 3. FEC is not decided or OFF.
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, IncreaseTo60MsOnVeryLowUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  // We set packet loss fraction to kFlDecreasingPacketLossFraction, which
+  // should have prevented frame length to increase, if the uplink bandwidth
+  // was not this low.
+  UpdateNetworkMetrics(controller.get(), VeryLowBitrate(20),
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, Maintain60MsOnVeryLowUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  // We set packet loss fraction to FlDecreasingPacketLossFraction, which should
+  // have caused the frame length to decrease, if the uplink bandwidth was not
+  // this low.
+  UpdateNetworkMetrics(controller.get(), VeryLowBitrate(20),
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, UpdateMultipleNetworkMetricsAtOnce) {
+  // This test is similar to IncreaseTo60MsOnMultipleConditions. But instead of
+  // using ::UpdateNetworkMetrics(...), which calls
+  // FrameLengthController::UpdateNetworkMetrics(...) multiple times, we
+  // we call it only once. This is to verify that
+  // FrameLengthController::UpdateNetworkMetrics(...) can handle multiple
+  // network updates at once. This is, however, not a common use case in current
+  // audio_network_adaptor_impl.cc.
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.uplink_bandwidth_bps = kFl20msTo60msBandwidthBps;
+  network_metrics.uplink_packet_loss_fraction = kFlIncreasingPacketLossFraction;
+  controller->UpdateNetworkMetrics(network_metrics);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest,
+     Maintain20MsIf60MsNotInReceiverFrameLengthRange) {
+  auto controller =
+      CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20}, 20);
+  // Use a low uplink bandwidth and a low uplink packet loss fraction that would
+  // cause frame length to increase if receiver frame length included 60ms.
+  UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, Maintain20MsOnMediumUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, Maintain20MsOnMediumUplinkPacketLossFraction) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  // Use a low uplink bandwidth that would cause frame length to increase if
+  // uplink packet loss fraction was low.
+  UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+                       kMediumPacketLossFraction, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, Maintain60MsWhenNo120msCriteriaIsSet) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, From120MsTo20MsOnHighUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 120);
+  // It takes two steps for frame length to go from 120ms to 20ms.
+  UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
+                       absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
+                       absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, From120MsTo20MsOnHighUplinkPacketLossFraction) {
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 120);
+  // It takes two steps for frame length to go from 120ms to 20ms.
+  UpdateNetworkMetrics(controller.get(), absl::nullopt,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), absl::nullopt,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, Maintain120MsOnVeryLowUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 120);
+  // We set packet loss fraction to FlDecreasingPacketLossFraction, which should
+  // have caused the frame length to decrease, if the uplink bandwidth was not
+  // this low.
+  UpdateNetworkMetrics(controller.get(), VeryLowBitrate(60),
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 120);
+}
+
+TEST(FrameLengthControllerTest, From60MsTo120MsOnVeryLowUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  // We set packet loss fraction to FlDecreasingPacketLossFraction, which should
+  // have prevented frame length to increase, if the uplink bandwidth was not
+  // this low.
+  UpdateNetworkMetrics(controller.get(), VeryLowBitrate(60),
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 120);
+}
+
+TEST(FrameLengthControllerTest, From20MsTo120MsOnMultipleConditions) {
+  // Increase to 120ms frame length if
+  // 1. |uplink_bandwidth_bps| is known to be smaller than a threshold AND
+  // 2. |uplink_packet_loss_fraction| is known to be smaller than a threshold.
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  // It takes two steps for frame length to go from 20ms to 120ms.
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 120);
+}
+
+TEST(FrameLengthControllerTest, Stall60MsIf120MsNotInReceiverFrameLengthRange) {
+  auto controller =
+      CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(), {20, 60}, 20);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, CheckBehaviorOnChangingNetworkMetrics) {
+  auto controller =
+      CreateController(CreateChangeCriteriaFor20ms40ms60msAnd120ms(),
+                       kDefaultEncoderFrameLengthsMs, 20);
+  UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+
+  UpdateNetworkMetrics(controller.get(), kFl20msTo40msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo40msBandwidthBps,
+                       kMediumPacketLossFraction, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+
+  UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kMediumPacketLossFraction, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 120);
+
+  UpdateNetworkMetrics(controller.get(), kFl120msTo60msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo40msBandwidthBps,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+
+  UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+}  // namespace webrtc

modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h

@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+
+// An AudioNetworkAdaptor optimizes the audio experience by suggesting a
+// suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
+// encoder based on network metrics.
+class AudioNetworkAdaptor {
+ public:
+  virtual ~AudioNetworkAdaptor() = default;
+
+  virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
+
+  virtual void SetUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) = 0;
+
+  virtual void SetRtt(int rtt_ms) = 0;
+
+  virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0;
+
+  virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0;
+
+  virtual AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
+
+  virtual void StartDebugDump(FILE* file_handle) = 0;
+
+  virtual void StopDebugDump() = 0;
+
+  virtual ANAStats GetStats() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_

modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h

@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2017 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+struct AudioEncoderRuntimeConfig {
+  AudioEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
+  ~AudioEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig& operator=(const AudioEncoderRuntimeConfig& other);
+  bool operator==(const AudioEncoderRuntimeConfig& other) const;
+  absl::optional<int> bitrate_bps;
+  absl::optional<int> frame_length_ms;
+  // Note: This is what we tell the encoder. It doesn't have to reflect
+  // the actual NetworkMetrics; it's subject to our decision.
+  absl::optional<float> uplink_packet_loss_fraction;
+  absl::optional<bool> enable_fec;
+  absl::optional<bool> enable_dtx;
+
+  // Some encoders can encode fewer channels than the actual input to make
+  // better use of the bandwidth. |num_channels| sets the number of channels
+  // to encode.
+  absl::optional<size_t> num_channels;
+
+  // This is true if the last frame length change was an increase, and otherwise
+  // false.
+  // The value of this boolean is used to apply a different offset to the
+  // per-packet overhead that is reported by the BWE. The exact offset value
+  // is most important right after a frame length change, because the frame
+  // length change affects the overhead. In the steady state, the exact value is
+  // not important because the BWE will compensate.
+  bool last_fl_change_increase = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_

modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h

@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
+
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioNetworkAdaptor : public AudioNetworkAdaptor {
+ public:
+  virtual ~MockAudioNetworkAdaptor() { Die(); }
+  MOCK_METHOD0(Die, void());
+
+  MOCK_METHOD1(SetUplinkBandwidth, void(int uplink_bandwidth_bps));
+
+  MOCK_METHOD1(SetUplinkPacketLossFraction,
+               void(float uplink_packet_loss_fraction));
+
+  MOCK_METHOD1(SetRtt, void(int rtt_ms));
+
+  MOCK_METHOD1(SetTargetAudioBitrate, void(int target_audio_bitrate_bps));
+
+  MOCK_METHOD1(SetOverhead, void(size_t overhead_bytes_per_packet));
+
+  MOCK_METHOD0(GetEncoderRuntimeConfig, AudioEncoderRuntimeConfig());
+
+  MOCK_METHOD1(StartDebugDump, void(FILE* file_handle));
+
+  MOCK_METHOD0(StopDebugDump, void());
+
+  MOCK_CONST_METHOD0(GetStats, ANAStats());
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_

modules/audio_coding/audio_network_adaptor/mock/mock_controller.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
+
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockController : public Controller {
+ public:
+  virtual ~MockController() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD1(UpdateNetworkMetrics,
+               void(const NetworkMetrics& network_metrics));
+  MOCK_METHOD1(MakeDecision, void(AudioEncoderRuntimeConfig* config));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h

@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
+
+#include <vector>
+
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockControllerManager : public ControllerManager {
+ public:
+  virtual ~MockControllerManager() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD1(
+      GetSortedControllers,
+      std::vector<Controller*>(const Controller::NetworkMetrics& metrics));
+  MOCK_CONST_METHOD0(GetControllers, std::vector<Controller*>());
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_

modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h

@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2016 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
+
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDebugDumpWriter : public DebugDumpWriter {
+ public:
+  virtual ~MockDebugDumpWriter() { Die(); }
+  MOCK_METHOD0(Die, void());
+
+  MOCK_METHOD2(DumpEncoderRuntimeConfig,
+               void(const AudioEncoderRuntimeConfig& config,
+                    int64_t timestamp));
+  MOCK_METHOD2(DumpNetworkMetrics,
+               void(const Controller::NetworkMetrics& metrics,
+                    int64_t timestamp));
+#if WEBRTC_ENABLE_PROTOBUF
+  MOCK_METHOD2(DumpControllerManagerConfig,
+               void(const audio_network_adaptor::config::ControllerManager&
+                        controller_manager_config,
+                    int64_t timestamp));
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_

modules/audio_coding/audio_network_adaptor/parse_ana_dump.py

@@ -0,0 +1,147 @@
+#!/usr/bin/python2
+#  Copyright (c) 2017 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.
+
+#  To run this script please copy "out/<build_name>//pyproto/webrtc/modules/
+#  audio_coding/audio_network_adaptor/debug_dump_pb2.py" to this folder.
+#  The you can run this script with:
+#  "python parse_ana_dump.py -m uplink_bandwidth_bps -f dump_file.dat"
+#  You can add as may metrics or decisions to the plot as you like.
+#  form more information call:
+#  "python parse_ana_dump.py --help"
+
+import struct
+from optparse import OptionParser
+
+import matplotlib.pyplot as plt
+
+import debug_dump_pb2
+
+
+def GetNextMessageSize(file_to_parse):
+  data = file_to_parse.read(4)
+  if data == '':
+    return 0
+  return struct.unpack('<I', data)[0]
+
+
+def GetNextMessageFromFile(file_to_parse):
+  message_size = GetNextMessageSize(file_to_parse)
+  if message_size == 0:
+    return None
+  try:
+    event = debug_dump_pb2.Event()
+    event.ParseFromString(file_to_parse.read(message_size))
+  except IOError:
+    print 'Invalid message in file'
+    return None
+  return event
+
+
+def InitMetrics():
+  metrics = {}
+  event = debug_dump_pb2.Event()
+  for metric in event.network_metrics.DESCRIPTOR.fields:
+    metrics[metric.name] = {'time': [], 'value': []}
+  return metrics
+
+
+def InitDecisions():
+  decisions = {}
+  event = debug_dump_pb2.Event()
+  for decision in event.encoder_runtime_config.DESCRIPTOR.fields:
+    decisions[decision.name] = {'time': [], 'value': []}
+  return decisions
+
+
+def ParseAnaDump(dump_file_to_parse):
+  with open(dump_file_to_parse, 'rb') as file_to_parse:
+    metrics = InitMetrics()
+    decisions = InitDecisions()
+    first_time_stamp = None
+    while True:
+      event = GetNextMessageFromFile(file_to_parse)
+      if event is None:
+        break
+      if first_time_stamp is None:
+        first_time_stamp = event.timestamp
+      if event.type == debug_dump_pb2.Event.ENCODER_RUNTIME_CONFIG:
+        for decision in event.encoder_runtime_config.DESCRIPTOR.fields:
+          if event.encoder_runtime_config.HasField(decision.name):
+            decisions[decision.name]['time'].append(event.timestamp -
+                                                    first_time_stamp)
+            decisions[decision.name]['value'].append(
+                getattr(event.encoder_runtime_config, decision.name))
+      if event.type == debug_dump_pb2.Event.NETWORK_METRICS:
+        for metric in event.network_metrics.DESCRIPTOR.fields:
+          if event.network_metrics.HasField(metric.name):
+            metrics[metric.name]['time'].append(event.timestamp -
+                                                first_time_stamp)
+            metrics[metric.name]['value'].append(
+                getattr(event.network_metrics, metric.name))
+  return (metrics, decisions)
+
+
+def main():
+  parser = OptionParser()
+  parser.add_option(
+      "-f", "--dump_file", dest="dump_file_to_parse", help="dump file to parse")
+  parser.add_option(
+      '-m',
+      '--metric_plot',
+      default=[],
+      type=str,
+      help='metric key (name of the metric) to plot',
+      dest='metric_keys',
+      action='append')
+
+  parser.add_option(
+      '-d',
+      '--decision_plot',
+      default=[],
+      type=str,
+      help='decision key (name of the decision) to plot',
+      dest='decision_keys',
+      action='append')
+
+  options = parser.parse_args()[0]
+  if options.dump_file_to_parse is None:
+    print "No dump file to parse is set.\n"
+    parser.print_help()
+    exit()
+  (metrics, decisions) = ParseAnaDump(options.dump_file_to_parse)
+  metric_keys = options.metric_keys
+  decision_keys = options.decision_keys
+  plot_count = len(metric_keys) + len(decision_keys)
+  if plot_count == 0:
+    print "You have to set at least one metric or decision to plot.\n"
+    parser.print_help()
+    exit()
+  plots = []
+  if plot_count == 1:
+    f, mp_plot = plt.subplots()
+    plots.append(mp_plot)
+  else:
+    f, mp_plots = plt.subplots(plot_count, sharex=True)
+    plots.extend(mp_plots.tolist())
+
+  for key in metric_keys:
+    plot = plots.pop()
+    plot.grid(True)
+    plot.set_title(key + " (metric)")
+    plot.plot(metrics[key]['time'], metrics[key]['value'])
+  for key in decision_keys:
+    plot = plots.pop()
+    plot.grid(True)
+    plot.set_title(key + " (decision)")
+    plot.plot(decisions[key]['time'], decisions[key]['value'])
+  f.subplots_adjust(hspace=0.3)
+  plt.show()
+
+if __name__ == "__main__":
+  main()

modules/audio_coding/audio_network_adaptor/util/threshold_curve.h

@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2017 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 MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class ThresholdCurve {
+ public:
+  struct Point {
+    constexpr Point(float x, float y) : x(x), y(y) {}
+    float x;
+    float y;
+  };
+
+  // ThresholdCurve defines a curve. The curve is characterized by the two
+  // conjunction points: A and B. The curve segments the metric space into
+  // three domains - above the curve, on it and below it.
+  //
+  // y-axis ^  |
+  //        | A|
+  //        |   \    A: (a.x, a.y)
+  //        |    \   B: (b.x, b.y)
+  //        |    B\________
+  //        |---------------> bandwidth
+  //
+  // If either a.x == b.x or a.y == b.y, the curve can be defined
+  // by a single point. (We merge the two points into one - either the lower or
+  // the leftmost one - for easier treatment.)
+  //
+  // y-axis ^  |
+  //        |  |
+  //        |  |
+  //        |  |
+  //        | P|__________
+  //        |---------------> bandwidth
+  ThresholdCurve(const Point& left, const Point& right)
+      : a(GetPoint(left, right, true)),
+        b(GetPoint(left, right, false)),
+        slope(b.x - a.x == 0.0f ? 0.0f : (b.y - a.y) / (b.x - a.x)),
+        offset(a.y - slope * a.x) {
+    // TODO(eladalon): We might want to introduce some numerical validations.
+  }
+
+  ThresholdCurve(float a_x, float a_y, float b_x, float b_y)
+      : ThresholdCurve(Point{a_x, a_y}, Point{b_x, b_y}) {}
+
+  // Checks if a point is strictly below the curve.
+  bool IsBelowCurve(const Point& p) const {
+    if (p.x < a.x) {
+      return true;
+    } else if (p.x == a.x) {
+      // In principle, we could merge this into the next else, but to avoid
+      // numerical errors, we treat it separately.
+      return p.y < a.y;
+    } else if (a.x < p.x && p.x < b.x) {
+      return p.y < offset + slope * p.x;
+    } else {  // if (b.x <= p.x)
+      return p.y < b.y;
+    }
+  }
+
+  // Checks if a point is strictly above the curve.
+  bool IsAboveCurve(const Point& p) const {
+    if (p.x <= a.x) {
+      return false;
+    } else if (a.x < p.x && p.x < b.x) {
+      return p.y > offset + slope * p.x;
+    } else {  // if (b.x <= p.x)
+      return p.y > b.y;
+    }
+  }
+
+  bool operator<=(const ThresholdCurve& rhs) const {
+    // This curve is <= the rhs curve if no point from this curve is
+    // above a corresponding point from the rhs curve.
+    return !IsBelowCurve(rhs.a) && !IsBelowCurve(rhs.b) &&
+           !rhs.IsAboveCurve(a) && !rhs.IsAboveCurve(b);
+  }
+
+ private:
+  static const Point& GetPoint(const Point& left,
+                               const Point& right,
+                               bool is_for_left) {
+    RTC_DCHECK_LE(left.x, right.x);
+    RTC_DCHECK_GE(left.y, right.y);
+
+    // Same X-value or Y-value triggers merging both points to the
+    // lower and/or left of the two points, respectively.
+    if (left.x == right.x) {
+      return right;
+    } else if (left.y == right.y) {
+      return left;
+    }
+
+    // If unmerged, boolean flag determines which of the points is desired.
+    return is_for_left ? left : right;
+  }
+
+  const Point a;
+  const Point b;
+  const float slope;
+  const float offset;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_

modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc

@@ -0,0 +1,632 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+// A threshold curve divides 2D space into three domains - below, on and above
+// the threshold curve.
+// The curve is defined by two points. Those points, P1 and P2, are ordered so
+// that (P1.x <= P2.x && P1.y >= P2.y).
+// The part of the curve which is between the two points is hereon referred
+// to as the "segment".
+// A "ray" extends from P1 directly upwards into infinity; that's the "vertical
+// ray". Likewise, a "horizontal ray" extends from P2 directly rightwards.
+//
+//  ^   |                         //
+//  |   | vertical ray            //
+//  |   |                         //
+//  |   |                         //
+//  | P1|                         //
+//  |    \                        //
+//  |     \ segment               //
+//  |      \                      //
+//  |       \    horizontal ray   //
+//  |     P2 ------------------   //
+//  *---------------------------> //
+
+namespace webrtc {
+
+namespace {
+enum RelativePosition { kBelow, kOn, kAbove };
+
+void CheckRelativePosition(const ThresholdCurve& curve,
+                           ThresholdCurve::Point point,
+                           RelativePosition pos) {
+  RTC_CHECK(pos == kBelow || pos == kOn || pos == kAbove);
+
+  EXPECT_EQ(pos == kBelow, curve.IsBelowCurve(point));
+  EXPECT_EQ(pos == kAbove, curve.IsAboveCurve(point));
+}
+}  // namespace
+
+// Test that the curve correctly reports the below/above position of points,
+// when the curve is a "normal" one - P1 and P2 are different in both their
+// X and Y values.
+TEST(ThresholdCurveTest, PointPositionToCommonCurve) {
+  // The points (P1-P2) define the curve.           //
+  // All other points are above/below/on the curve. //
+  //                                                //
+  //  ^                                             //
+  //  |     |                                       //
+  //  |  A  F    J  R   V                           //
+  //  |     |                                       //
+  //  |  B  P1   K  S   W                           //
+  //  |      \                                      //
+  //  |       \                                     //
+  //  |        \ L                                  //
+  //  |         \                                   //
+  //  |  C  G    M  T   X                           //
+  //  |           \                                 //
+  //  |          N \                                //
+  //  |             \                               //
+  //  |  D  H    O  P2--Y----------------           //
+  //  |  E  I    Q  U   Z                           //
+  //  *---------------------------------->          //
+  constexpr ThresholdCurve::Point p1{1000, 2000};
+  constexpr ThresholdCurve::Point p2{2000, 1000};
+
+  RTC_CHECK_GT((p1.x + p2.x) / 2, p1.x);
+  RTC_CHECK_LT((p1.x + p2.x) / 2, p2.x);
+  RTC_CHECK_LT((p1.y + p2.y) / 2, p1.y);
+  RTC_CHECK_GT((p1.y + p2.y) / 2, p2.y);
+
+  const ThresholdCurve curve(p1, p2);
+
+  {
+    // All cases where the point lies to the left of P1.
+    constexpr float x = p1.x - 1;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kBelow);           // A
+    CheckRelativePosition(curve, {x, p1.y + 0}, kBelow);           // B
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kBelow);  // C
+    CheckRelativePosition(curve, {x, p2.y + 0}, kBelow);           // D
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);           // E
+  }
+
+  {
+    // All cases where the point has the same x-value as P1.
+    constexpr float x = p1.x;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kOn);              // F
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);              // P1
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kBelow);  // G
+    CheckRelativePosition(curve, {x, p2.y + 0}, kBelow);           // H
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);           // I
+  }
+
+  {
+    // To make sure we're really covering all of the cases, make sure that P1
+    // and P2 were chosen so that L would really be below K, and O would really
+    // be below N. (This would not hold if the Y values are too close together.)
+    RTC_CHECK_LT(((p1.y + p2.y) / 2) + 1, p1.y);
+    RTC_CHECK_LT(p2.y, ((p1.y + p2.y) / 2) - 1);
+
+    // All cases where the point's x-value is between P1 and P2.
+    constexpr float x = (p1.x + p2.x) / 2;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);                 // J
+    CheckRelativePosition(curve, {x, p1.y + 0}, kAbove);                 // K
+    CheckRelativePosition(curve, {x, ((p1.y + p2.y) / 2) + 1}, kAbove);  // L
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kOn);           // M
+    CheckRelativePosition(curve, {x, ((p1.y + p2.y) / 2) - 1}, kBelow);  // N
+    CheckRelativePosition(curve, {x, p2.y + 0}, kBelow);                 // O
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);                 // Q
+  }
+
+  {
+    // All cases where the point has the same x-value as P2.
+    constexpr float x = p2.x;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);           // R
+    CheckRelativePosition(curve, {x, p1.y + 0}, kAbove);           // S
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kAbove);  // T
+    CheckRelativePosition(curve, {x, p2.y + 0}, kOn);              // P2
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);           // U
+  }
+
+  {
+    // All cases where the point lies to the right of P2.
+    constexpr float x = p2.x + 1;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);           // V
+    CheckRelativePosition(curve, {x, p1.y + 0}, kAbove);           // W
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kAbove);  // X
+    CheckRelativePosition(curve, {x, p2.y + 0}, kOn);              // Y
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);           // Z
+  }
+}
+
+// Test that the curve correctly reports the below/above position of points,
+// when the curve is defined by two points with the same Y value.
+TEST(ThresholdCurveTest, PointPositionToCurveWithHorizaontalSegment) {
+  // The points (P1-P2) define the curve.
+  // All other points are above/below/on the curve.
+  //
+  //  ^
+  //  |    |
+  //  |    |
+  //  | A  D   F  I  K
+  //  |    |
+  //  |    |
+  //  | B  P1--G--P2-L--
+  //  | C  E   H  J  M
+  //  *------------------>
+
+  constexpr ThresholdCurve::Point p1{100, 200};
+  constexpr ThresholdCurve::Point p2{p1.x + 1, p1.y};
+
+  RTC_CHECK_GT((p1.x + p2.x) / 2, p1.x);
+  RTC_CHECK_LT((p1.x + p2.x) / 2, p2.x);
+
+  const ThresholdCurve curve(p1, p2);
+
+  {
+    // All cases where the point lies to the left of P1.
+    constexpr float x = p1.x - 1;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kBelow);  // A
+    CheckRelativePosition(curve, {x, p1.y + 0}, kBelow);  // B
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // C
+  }
+
+  {
+    // All cases where the point has the same x-value as P1.
+    constexpr float x = p1.x;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kOn);     // D
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);     // P1
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // E
+  }
+
+  {
+    // All cases where the point's x-value is between P1 and P2.
+    constexpr float x = (p1.x + p2.x) / 2;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);  // F
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);     // G
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // H
+  }
+
+  {
+    // All cases where the point has the same x-value as P2.
+    constexpr float x = p2.x;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);  // I
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);     // P2
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // J
+  }
+
+  {
+    // All cases where the point lies to the right of P2.
+    constexpr float x = p2.x + 1;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);  // K
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);     // L
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // M
+  }
+}
+
+// Test that the curve correctly reports the below/above position of points,
+// when the curve is defined by two points with the same X value.
+TEST(ThresholdCurveTest, PointPositionToCurveWithVerticalSegment) {
+  // The points (P1-P2) define the curve.
+  // All other points are above/below/on the curve.
+  //
+  //  ^
+  //  |    |
+  //  | A  B   C
+  //  |    |
+  //  | D  P1  E
+  //  |    |
+  //  | F  G   H
+  //  |    |
+  //  | I  P2--J------
+  //  | K  L   M
+  //  *------------------>
+
+  constexpr ThresholdCurve::Point p1{100, 200};
+  constexpr ThresholdCurve::Point p2{p1.x, p1.y - 1};
+
+  constexpr float left = p1.x - 1;
+  constexpr float on = p1.x;
+  constexpr float right = p1.x + 1;
+
+  RTC_CHECK_LT((p1.y + p2.y) / 2, p1.y);
+  RTC_CHECK_GT((p1.y + p2.y) / 2, p2.y);
+
+  const ThresholdCurve curve(p1, p2);
+
+  {
+    // All cases where the point lies above P1.
+    constexpr float y = p1.y + 1;
+    CheckRelativePosition(curve, {left, y}, kBelow);   // A
+    CheckRelativePosition(curve, {on, y}, kOn);        // B
+    CheckRelativePosition(curve, {right, y}, kAbove);  // C
+  }
+
+  {
+    // All cases where the point has the same y-value as P1.
+    constexpr float y = p1.y;
+    CheckRelativePosition(curve, {left, y}, kBelow);   // D
+    CheckRelativePosition(curve, {on, y}, kOn);        // P1
+    CheckRelativePosition(curve, {right, y}, kAbove);  // E
+  }
+
+  {
+    // All cases where the point's y-value is between P1 and P2.
+    constexpr float y = (p1.y + p2.y) / 2;
+    CheckRelativePosition(curve, {left, y}, kBelow);   // F
+    CheckRelativePosition(curve, {on, y}, kOn);        // G
+    CheckRelativePosition(curve, {right, y}, kAbove);  // H
+  }
+
+  {
+    // All cases where the point has the same y-value as P2.
+    constexpr float y = p2.y;
+    CheckRelativePosition(curve, {left, y}, kBelow);  // I
+    CheckRelativePosition(curve, {on, y}, kOn);       // P2
+    CheckRelativePosition(curve, {right, y}, kOn);    // J
+  }
+
+  {
+    // All cases where the point lies below P2.
+    constexpr float y = p2.y - 1;
+    CheckRelativePosition(curve, {left, y}, kBelow);   // K
+    CheckRelativePosition(curve, {on, y}, kBelow);     // L
+    CheckRelativePosition(curve, {right, y}, kBelow);  // M
+  }
+}
+
+// Test that the curve correctly reports the below/above position of points,
+// when the curve is defined by two points which are identical.
+TEST(ThresholdCurveTest, PointPositionCurveWithNullSegment) {
+  // The points (P1-P2) define the curve.
+  // All other points are above/below/on the curve.
+  //
+  //  ^
+  //  |    |
+  //  | A  D   F
+  //  |    |
+  //  | B  P---G------
+  //  | C  E   H
+  //  *------------------>
+
+  constexpr ThresholdCurve::Point p{100, 200};
+
+  const ThresholdCurve curve(p, p);
+
+  {
+    // All cases where the point lies to the left of P.
+    constexpr float x = p.x - 1;
+    CheckRelativePosition(curve, {x, p.y + 1}, kBelow);  // A
+    CheckRelativePosition(curve, {x, p.y + 0}, kBelow);  // B
+    CheckRelativePosition(curve, {x, p.y - 1}, kBelow);  // C
+  }
+
+  {
+    // All cases where the point has the same x-value as P.
+    constexpr float x = p.x + 0;
+    CheckRelativePosition(curve, {x, p.y + 1}, kOn);     // D
+    CheckRelativePosition(curve, {x, p.y + 0}, kOn);     // P
+    CheckRelativePosition(curve, {x, p.y - 1}, kBelow);  // E
+  }
+
+  {
+    // All cases where the point lies to the right of P.
+    constexpr float x = p.x + 1;
+    CheckRelativePosition(curve, {x, p.y + 1}, kAbove);  // F
+    CheckRelativePosition(curve, {x, p.y + 0}, kOn);     // G
+    CheckRelativePosition(curve, {x, p.y - 1}, kBelow);  // H
+  }
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the two curves have the same projection on the X-axis.
+TEST(ThresholdCurveTest, TwoCurvesSegmentHasSameProjectionAxisX) {
+  //  ^                        //
+  //  | C1 + C2                //
+  //  |  |                     //
+  //  |  |\                    //
+  //  |  | \                   //
+  //  |   \ \                  //
+  //  |    \ \                 //
+  //  |     \ \                //
+  //  |      \ -------- C2     //
+  //  |       --------- C1     //
+  //  *--------------------->  //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  // Same x-values, but higher on Y. (Can be parallel, but doesn't have to be.)
+  constexpr ThresholdCurve::Point c2_left{c1_left.x, c1_left.y + 20};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x, c1_right.y + 10};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the higher curve's projection on the X-axis is a strict subset of the
+// lower curve's projection on the X-axis (on both ends).
+TEST(ThresholdCurveTest, TwoCurvesSegmentOfHigherSubsetProjectionAxisX) {
+  //  ^                       //
+  //  | C1    C2              //
+  //  |  |    |               //
+  //  |  |    |               //
+  //  |   \   |               //
+  //  |    \  |               //
+  //  |     \ \               //
+  //  |      \ \              //
+  //  |       \ --------- C2  //
+  //  |        \              //
+  //  |         \             //
+  //  |          ---------C1  //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{6, 11};
+  constexpr ThresholdCurve::Point c2_right{9, 7};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the higher curve's right point is above lower curve's horizontal ray (meaning
+// the higher curve's projection on the X-axis extends further right than
+// the lower curve's).
+TEST(ThresholdCurveTest,
+     TwoCurvesRightPointOfHigherCurveAboveHorizontalRayOfLower) {
+  //  ^                        //
+  //  | C1 + C2                //
+  //  |  |                     //
+  //  |  |\                    //
+  //  |  | \                   //
+  //  |  |  \                  //
+  //  |  |   \                 //
+  //  |  |    \                //
+  //  |   \    \               //
+  //  |    \    \              //
+  //  |     \    \             //
+  //  |      \    ----- C2     //
+  //  |       --------- C1     //
+  //  *--------------------->  //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{c1_left.x, c1_left.y + 1};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x + 1, c1_right.y + 1};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the higher curve's points are on the lower curve's rays (left point on the
+// veritcal ray, right point on the horizontal ray).
+TEST(ThresholdCurveTest, TwoCurvesPointsOfHigherOnRaysOfLower) {
+  //  ^
+  //  | C1 + C2               //
+  //  |  |                    //
+  //  |  |\                   //
+  //  |  | \                  //
+  //  |   \ \                 //
+  //  |    \ \                //
+  //  |     \ \               //
+  //  |      \ \              //
+  //  |       ----- C1 + C2   //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  // Same x-values, but one of the points is higher on Y (the other isn't).
+  constexpr ThresholdCurve::Point c2_left{c1_left.x, c1_left.y + 2};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x + 3, c1_right.y};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the second curve's segment intersects the first curve's vertical ray.
+TEST(ThresholdCurveTest, SecondCurveCrossesVerticalRayOfFirstCurve) {
+  //  ^                       //
+  //  | C2 C1                 //
+  //  |  | |                  //
+  //  |   \|                  //
+  //  |    |                  //
+  //  |    |\                 //
+  //  |    | \                //
+  //  |     \ \               //
+  //  |      \ \              //
+  //  |       \ \             //
+  //  |        \ ------- C2   //
+  //  |         -------- C1   //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{c1_left.x - 1, c1_left.y + 1};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x, c1_right.y + 1};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the second curve's segment intersects the first curve's horizontal ray.
+TEST(ThresholdCurveTest, SecondCurveCrossesHorizontalRayOfFirstCurve) {
+  //  ^                      //
+  //  | C1 +  C2             //
+  //  |  |                   //
+  //  |  |\                  //
+  //  |  \ \                 //
+  //  |   \ \                //
+  //  |    \ \               //
+  //  |     \ \              //
+  //  |      ----------- C1  //
+  //  |         \            //
+  //  |          ------- C2  //
+  //  *--------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{c1_left.x, c1_left.y + 1};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x + 2, c1_right.y - 1};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the second curve's segment intersects the first curve's segment.
+TEST(ThresholdCurveTest, TwoCurvesWithCrossingSegments) {
+  //  ^                           //
+  //  | C2 C1                     //
+  //  | |  |                      //
+  //  | |  |                      //
+  //  | |  \                      //
+  //  | |   \                     //
+  //  |  -_  \                    //
+  //  |    -_ \                   //
+  //  |      -_\                  //
+  //  |        -_                 //
+  //  |          \-_              //
+  //  |           \ ---------- C2 //
+  //  |            ----------- C1 //
+  //  |                           //
+  //  |                           //
+  //  *-------------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{4, 9};
+  constexpr ThresholdCurve::Point c2_right{10, 6};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  // The test is structured so that the two curves intersect at (8, 7).
+  RTC_CHECK(!c1_curve.IsAboveCurve({8, 7}));
+  RTC_CHECK(!c1_curve.IsBelowCurve({8, 7}));
+  RTC_CHECK(!c2_curve.IsAboveCurve({8, 7}));
+  RTC_CHECK(!c2_curve.IsBelowCurve({8, 7}));
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// both curves are identical.
+TEST(ThresholdCurveTest, IdenticalCurves) {
+  //  ^                       //
+  //  |  C1 + C2              //
+  //  |  |                    //
+  //  |  |                    //
+  //  |   \                   //
+  //  |    \                  //
+  //  |     \                 //
+  //  |      ------- C1 + C2  //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point left{5, 10};
+  constexpr ThresholdCurve::Point right{10, 5};
+
+  const ThresholdCurve c1_curve(left, right);
+  const ThresholdCurve c2_curve(left, right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_TRUE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// they are "nearly identical" - the first curve's segment is contained within
+// the second curve's segment, but the second curve's segment extends further
+// to the left (which also produces separate vertical rays for the curves).
+TEST(ThresholdCurveTest, NearlyIdenticalCurvesSecondContinuesOnOtherLeftSide) {
+  //  ^                       //
+  //  | C2 C1                 //
+  //  |  | |                  //
+  //  |  | |                  //
+  //  |   \|                  //
+  //  |    |                  //
+  //  |     \                 //
+  //  |      \                //
+  //  |       \               //
+  //  |        ----- C1 + C2  //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_left);
+
+  constexpr ThresholdCurve::Point c2_left{c1_left.x - 1, c1_left.y + 1};
+  constexpr ThresholdCurve::Point c2_right = c1_right;
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_TRUE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// they are "nearly identical" - the first curve's segment is contained within
+// the second curve's segment, but the second curve's segment extends further
+// to the right (which also produces separate horizontal rays for the curves).
+TEST(ThresholdCurveTest, NearlyIdenticalCurvesSecondContinuesOnOtherRightSide) {
+  //  ^                       //
+  //  | C1 + C2               //
+  //  |  |                    //
+  //  |  |                    //
+  //  |   \                   //
+  //  |    \                  //
+  //  |     \                 //
+  //  |      \----------- C1  //
+  //  |       \               //
+  //  |        ---------- C2  //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_left);
+
+  constexpr ThresholdCurve::Point c2_left = c1_left;
+  constexpr ThresholdCurve::Point c2_right{c1_right.x + 1, c1_right.y - 1};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_TRUE(c2_curve <= c1_curve);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// The higher-left point must be given as the first point, and the lower-right
+// point must be given as the second.
+// This necessarily produces a non-positive slope.
+TEST(ThresholdCurveTest, WrongOrderPoints) {
+  std::unique_ptr<ThresholdCurve> curve;
+  constexpr ThresholdCurve::Point left{5, 10};
+  constexpr ThresholdCurve::Point right{10, 5};
+  EXPECT_DEATH(curve.reset(new ThresholdCurve(right, left)), "");
+}
+#endif
+
+}  // namespace webrtc

modules/audio_coding/codecs/audio_decoder.h

@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/audio_decoder.h instead!
+// TODO(kwiberg): Remove it.
+
+#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+#define MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+
+#include "api/audio_codecs/audio_decoder.h"
+
+#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_

modules/audio_coding/codecs/audio_encoder.h

@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/audio_encoder.h instead!
+// TODO(ossu): Remove it.
+
+#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+#define MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+
+#include "api/audio_codecs/audio_encoder.h"
+
+#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_

modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc

@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(AudioDecoderFactoryTest, CreateUnknownDecoder) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreatePcmu) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // PCMu supports 8 kHz, and any number of channels.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreatePcma) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // PCMa supports 8 kHz, and any number of channels.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreateIlbc) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // iLBC supports 8 kHz, 1 channel.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0), absl::nullopt));
+#ifdef WEBRTC_CODEC_ILBC
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1), absl::nullopt));
+#endif
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreateIsac) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // iSAC supports 16 kHz, 1 channel. The float implementation additionally
+  // supports 32 kHz, 1 channel.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 1), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 2), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 8000, 1), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 48000, 1), absl::nullopt));
+#ifdef WEBRTC_ARCH_ARM
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), absl::nullopt));
+#else
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), absl::nullopt));
+#endif
+}
+
+TEST(AudioDecoderFactoryTest, CreateL16) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // L16 supports any clock rate, any number of channels.
+  const int clockrates[] = {8000, 16000, 32000, 48000};
+  const int num_channels[] = {1, 2, 3, 4711};
+  for (int clockrate : clockrates) {
+    EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0),
+                                       absl::nullopt));
+    for (int channels : num_channels) {
+      EXPECT_TRUE(adf->MakeAudioDecoder(
+          SdpAudioFormat("l16", clockrate, channels), absl::nullopt));
+    }
+  }
+}
+
+TEST(AudioDecoderFactoryTest, CreateG722) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // g722 supports 8 kHz, 1-2 channels.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1), absl::nullopt));
+
+  // g722 actually uses a 16 kHz sample rate instead of the nominal 8 kHz.
+  std::unique_ptr<AudioDecoder> dec =
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt);
+  EXPECT_EQ(16000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTest, CreateOpus) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // Opus supports 48 kHz, 2 channels, and wants a "stereo" parameter whose
+  // value is either "0" or "1".
+  for (int hz : {8000, 16000, 32000, 48000}) {
+    for (int channels : {0, 1, 2, 3}) {
+      for (std::string stereo : {"XX", "0", "1", "2"}) {
+        std::map<std::string, std::string> params;
+        if (stereo != "XX") {
+          params["stereo"] = stereo;
+        }
+        const bool good = (hz == 48000 && channels == 2 &&
+                           (stereo == "XX" || stereo == "0" || stereo == "1"));
+        EXPECT_EQ(good,
+                  static_cast<bool>(adf->MakeAudioDecoder(
+                      SdpAudioFormat("opus", hz, channels, std::move(params)),
+                      absl::nullopt)));
+      }
+    }
+  }
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc

@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+
+#include <limits>
+#include <memory>
+#include <vector>
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class AudioEncoderFactoryTest
+    : public ::testing::TestWithParam<rtc::scoped_refptr<AudioEncoderFactory>> {
+};
+
+TEST_P(AudioEncoderFactoryTest, SupportsAtLeastOneFormat) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  EXPECT_FALSE(supported_encoders.empty());
+}
+
+TEST_P(AudioEncoderFactoryTest, CanQueryAllSupportedFormats) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto info = factory->QueryAudioEncoder(spec.format);
+    EXPECT_TRUE(info);
+  }
+}
+
+TEST_P(AudioEncoderFactoryTest, CanConstructAllSupportedEncoders) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto info = factory->QueryAudioEncoder(spec.format);
+    auto encoder = factory->MakeAudioEncoder(127, spec.format, absl::nullopt);
+    EXPECT_TRUE(encoder);
+    EXPECT_EQ(encoder->SampleRateHz(), info->sample_rate_hz);
+    EXPECT_EQ(encoder->NumChannels(), info->num_channels);
+    EXPECT_EQ(encoder->RtpTimestampRateHz(), spec.format.clockrate_hz);
+  }
+}
+
+TEST_P(AudioEncoderFactoryTest, CanRunAllSupportedEncoders) {
+  constexpr int kTestPayloadType = 127;
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto encoder =
+        factory->MakeAudioEncoder(kTestPayloadType, spec.format, absl::nullopt);
+    EXPECT_TRUE(encoder);
+    encoder->Reset();
+    const int num_samples = rtc::checked_cast<int>(
+        encoder->SampleRateHz() * encoder->NumChannels() / 100);
+    rtc::Buffer out;
+    rtc::BufferT<int16_t> audio;
+    audio.SetData(num_samples, [](rtc::ArrayView<int16_t> audio) {
+      for (size_t i = 0; i != audio.size(); ++i) {
+        // Just put some numbers in there, ensure they're within range.
+        audio[i] =
+            static_cast<int16_t>(i & std::numeric_limits<int16_t>::max());
+      }
+      return audio.size();
+    });
+    // This is here to stop the test going forever with a broken encoder.
+    constexpr int kMaxEncodeCalls = 100;
+    int blocks = 0;
+    for (; blocks < kMaxEncodeCalls; ++blocks) {
+      AudioEncoder::EncodedInfo info = encoder->Encode(
+          blocks * encoder->RtpTimestampRateHz() / 100, audio, &out);
+      EXPECT_EQ(info.encoded_bytes, out.size());
+      if (info.encoded_bytes > 0) {
+        EXPECT_EQ(0u, info.encoded_timestamp);
+        EXPECT_EQ(kTestPayloadType, info.payload_type);
+        break;
+      }
+    }
+    ASSERT_LT(blocks, kMaxEncodeCalls);
+    const unsigned int next_timestamp =
+        blocks * encoder->RtpTimestampRateHz() / 100;
+    out.Clear();
+    for (; blocks < kMaxEncodeCalls; ++blocks) {
+      AudioEncoder::EncodedInfo info = encoder->Encode(
+          blocks * encoder->RtpTimestampRateHz() / 100, audio, &out);
+      EXPECT_EQ(info.encoded_bytes, out.size());
+      if (info.encoded_bytes > 0) {
+        EXPECT_EQ(next_timestamp, info.encoded_timestamp);
+        EXPECT_EQ(kTestPayloadType, info.payload_type);
+        break;
+      }
+    }
+    ASSERT_LT(blocks, kMaxEncodeCalls);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(BuiltinAudioEncoderFactoryTest,
+                         AudioEncoderFactoryTest,
+                         ::testing::Values(CreateBuiltinAudioEncoderFactory()));
+
+TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) {
+  using ::testing::ElementsAreArray;
+  // Check that we claim to support the formats we expect from build flags, and
+  // we've ordered them correctly.
+  auto factory = CreateBuiltinAudioEncoderFactory();
+  auto specs = factory->GetSupportedEncoders();
+
+  const std::vector<SdpAudioFormat> supported_formats = [&specs] {
+    std::vector<SdpAudioFormat> formats;
+    formats.reserve(specs.size());
+    for (const auto& spec : specs) {
+      formats.push_back(spec.format);
+    }
+    return formats;
+  }();
+
+  const std::vector<SdpAudioFormat> expected_formats = {
+#ifdef WEBRTC_CODEC_OPUS
+    {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+    {"isac", 16000, 1},
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+    {"isac", 32000, 1},
+#endif
+    {"G722", 8000, 1},
+#ifdef WEBRTC_CODEC_ILBC
+    {"ilbc", 8000, 1},
+#endif
+    {"pcmu", 8000, 1},
+    {"pcma", 8000, 1}
+  };
+
+  ASSERT_THAT(supported_formats, ElementsAreArray(expected_formats));
+}
+}  // namespace webrtc

modules/audio_coding/codecs/cng/audio_encoder_cng.cc

@@ -0,0 +1,323 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kMaxFrameSizeMs = 60;
+
+class AudioEncoderCng final : public AudioEncoder {
+ public:
+  explicit AudioEncoderCng(AudioEncoderCngConfig&& config);
+  ~AudioEncoderCng() override;
+
+  // Not copyable or moveable.
+  AudioEncoderCng(const AudioEncoderCng&) = delete;
+  AudioEncoderCng(AudioEncoderCng&&) = delete;
+  AudioEncoderCng& operator=(const AudioEncoderCng&) = delete;
+  AudioEncoderCng& operator=(AudioEncoderCng&&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+  void Reset() override;
+  bool SetFec(bool enable) override;
+  bool SetDtx(bool enable) override;
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
+      override;
+  void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) override;
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms) override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ private:
+  EncodedInfo EncodePassive(size_t frames_to_encode, rtc::Buffer* encoded);
+  EncodedInfo EncodeActive(size_t frames_to_encode, rtc::Buffer* encoded);
+  size_t SamplesPer10msFrame() const;
+
+  std::unique_ptr<AudioEncoder> speech_encoder_;
+  const int cng_payload_type_;
+  const int num_cng_coefficients_;
+  const int sid_frame_interval_ms_;
+  std::vector<int16_t> speech_buffer_;
+  std::vector<uint32_t> rtp_timestamps_;
+  bool last_frame_active_;
+  std::unique_ptr<Vad> vad_;
+  std::unique_ptr<ComfortNoiseEncoder> cng_encoder_;
+};
+
+AudioEncoderCng::AudioEncoderCng(AudioEncoderCngConfig&& config)
+    : speech_encoder_((static_cast<void>([&] {
+                         RTC_CHECK(config.IsOk()) << "Invalid configuration.";
+                       }()),
+                       std::move(config.speech_encoder))),
+      cng_payload_type_(config.payload_type),
+      num_cng_coefficients_(config.num_cng_coefficients),
+      sid_frame_interval_ms_(config.sid_frame_interval_ms),
+      last_frame_active_(true),
+      vad_(config.vad ? std::unique_ptr<Vad>(config.vad)
+                      : CreateVad(config.vad_mode)),
+      cng_encoder_(new ComfortNoiseEncoder(SampleRateHz(),
+                                           sid_frame_interval_ms_,
+                                           num_cng_coefficients_)) {}
+
+AudioEncoderCng::~AudioEncoderCng() = default;
+
+int AudioEncoderCng::SampleRateHz() const {
+  return speech_encoder_->SampleRateHz();
+}
+
+size_t AudioEncoderCng::NumChannels() const {
+  return 1;
+}
+
+int AudioEncoderCng::RtpTimestampRateHz() const {
+  return speech_encoder_->RtpTimestampRateHz();
+}
+
+size_t AudioEncoderCng::Num10MsFramesInNextPacket() const {
+  return speech_encoder_->Num10MsFramesInNextPacket();
+}
+
+size_t AudioEncoderCng::Max10MsFramesInAPacket() const {
+  return speech_encoder_->Max10MsFramesInAPacket();
+}
+
+int AudioEncoderCng::GetTargetBitrate() const {
+  return speech_encoder_->GetTargetBitrate();
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  RTC_CHECK_EQ(speech_buffer_.size(),
+               rtp_timestamps_.size() * samples_per_10ms_frame);
+  rtp_timestamps_.push_back(rtp_timestamp);
+  RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
+  speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
+  const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
+  if (rtp_timestamps_.size() < frames_to_encode) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_LE(frames_to_encode * 10, kMaxFrameSizeMs)
+      << "Frame size cannot be larger than " << kMaxFrameSizeMs
+      << " ms when using VAD/CNG.";
+
+  // Group several 10 ms blocks per VAD call. Call VAD once or twice using the
+  // following split sizes:
+  // 10 ms = 10 + 0 ms; 20 ms = 20 + 0 ms; 30 ms = 30 + 0 ms;
+  // 40 ms = 20 + 20 ms; 50 ms = 30 + 20 ms; 60 ms = 30 + 30 ms.
+  size_t blocks_in_first_vad_call =
+      (frames_to_encode > 3 ? 3 : frames_to_encode);
+  if (frames_to_encode == 4)
+    blocks_in_first_vad_call = 2;
+  RTC_CHECK_GE(frames_to_encode, blocks_in_first_vad_call);
+  const size_t blocks_in_second_vad_call =
+      frames_to_encode - blocks_in_first_vad_call;
+
+  // Check if all of the buffer is passive speech. Start with checking the first
+  // block.
+  Vad::Activity activity = vad_->VoiceActivity(
+      &speech_buffer_[0], samples_per_10ms_frame * blocks_in_first_vad_call,
+      SampleRateHz());
+  if (activity == Vad::kPassive && blocks_in_second_vad_call > 0) {
+    // Only check the second block if the first was passive.
+    activity = vad_->VoiceActivity(
+        &speech_buffer_[samples_per_10ms_frame * blocks_in_first_vad_call],
+        samples_per_10ms_frame * blocks_in_second_vad_call, SampleRateHz());
+  }
+
+  EncodedInfo info;
+  switch (activity) {
+    case Vad::kPassive: {
+      info = EncodePassive(frames_to_encode, encoded);
+      last_frame_active_ = false;
+      break;
+    }
+    case Vad::kActive: {
+      info = EncodeActive(frames_to_encode, encoded);
+      last_frame_active_ = true;
+      break;
+    }
+    case Vad::kError: {
+      FATAL();  // Fails only if fed invalid data.
+      break;
+    }
+  }
+
+  speech_buffer_.erase(
+      speech_buffer_.begin(),
+      speech_buffer_.begin() + frames_to_encode * samples_per_10ms_frame);
+  rtp_timestamps_.erase(rtp_timestamps_.begin(),
+                        rtp_timestamps_.begin() + frames_to_encode);
+  return info;
+}
+
+void AudioEncoderCng::Reset() {
+  speech_encoder_->Reset();
+  speech_buffer_.clear();
+  rtp_timestamps_.clear();
+  last_frame_active_ = true;
+  vad_->Reset();
+  cng_encoder_.reset(new ComfortNoiseEncoder(
+      SampleRateHz(), sid_frame_interval_ms_, num_cng_coefficients_));
+}
+
+bool AudioEncoderCng::SetFec(bool enable) {
+  return speech_encoder_->SetFec(enable);
+}
+
+bool AudioEncoderCng::SetDtx(bool enable) {
+  return speech_encoder_->SetDtx(enable);
+}
+
+bool AudioEncoderCng::SetApplication(Application application) {
+  return speech_encoder_->SetApplication(application);
+}
+
+void AudioEncoderCng::SetMaxPlaybackRate(int frequency_hz) {
+  speech_encoder_->SetMaxPlaybackRate(frequency_hz);
+}
+
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoderCng::ReclaimContainedEncoders() {
+  return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+}
+
+void AudioEncoderCng::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  speech_encoder_->OnReceivedUplinkPacketLossFraction(
+      uplink_packet_loss_fraction);
+}
+
+void AudioEncoderCng::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {
+  speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
+                                             bwe_period_ms);
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderCng::GetFrameLengthRange() const {
+  return speech_encoder_->GetFrameLengthRange();
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
+    size_t frames_to_encode,
+    rtc::Buffer* encoded) {
+  bool force_sid = last_frame_active_;
+  bool output_produced = false;
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  AudioEncoder::EncodedInfo info;
+
+  for (size_t i = 0; i < frames_to_encode; ++i) {
+    // It's important not to pass &info.encoded_bytes directly to
+    // WebRtcCng_Encode(), since later loop iterations may return zero in
+    // that value, in which case we don't want to overwrite any value from
+    // an earlier iteration.
+    size_t encoded_bytes_tmp =
+        cng_encoder_->Encode(rtc::ArrayView<const int16_t>(
+                                 &speech_buffer_[i * samples_per_10ms_frame],
+                                 samples_per_10ms_frame),
+                             force_sid, encoded);
+
+    if (encoded_bytes_tmp > 0) {
+      RTC_CHECK(!output_produced);
+      info.encoded_bytes = encoded_bytes_tmp;
+      output_produced = true;
+      force_sid = false;
+    }
+  }
+
+  info.encoded_timestamp = rtp_timestamps_.front();
+  info.payload_type = cng_payload_type_;
+  info.send_even_if_empty = true;
+  info.speech = false;
+  return info;
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(size_t frames_to_encode,
+                                                        rtc::Buffer* encoded) {
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  AudioEncoder::EncodedInfo info;
+  for (size_t i = 0; i < frames_to_encode; ++i) {
+    info =
+        speech_encoder_->Encode(rtp_timestamps_.front(),
+                                rtc::ArrayView<const int16_t>(
+                                    &speech_buffer_[i * samples_per_10ms_frame],
+                                    samples_per_10ms_frame),
+                                encoded);
+    if (i + 1 == frames_to_encode) {
+      RTC_CHECK_GT(info.encoded_bytes, 0) << "Encoder didn't deliver data.";
+    } else {
+      RTC_CHECK_EQ(info.encoded_bytes, 0)
+          << "Encoder delivered data too early.";
+    }
+  }
+  return info;
+}
+
+size_t AudioEncoderCng::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
+}
+
+}  // namespace
+
+AudioEncoderCngConfig::AudioEncoderCngConfig() = default;
+AudioEncoderCngConfig::AudioEncoderCngConfig(AudioEncoderCngConfig&&) = default;
+AudioEncoderCngConfig::~AudioEncoderCngConfig() = default;
+
+bool AudioEncoderCngConfig::IsOk() const {
+  if (num_channels != 1)
+    return false;
+  if (!speech_encoder)
+    return false;
+  if (num_channels != speech_encoder->NumChannels())
+    return false;
+  if (sid_frame_interval_ms <
+      static_cast<int>(speech_encoder->Max10MsFramesInAPacket() * 10))
+    return false;
+  if (num_cng_coefficients > WEBRTC_CNG_MAX_LPC_ORDER ||
+      num_cng_coefficients <= 0)
+    return false;
+  return true;
+}
+
+std::unique_ptr<AudioEncoder> CreateComfortNoiseEncoder(
+    AudioEncoderCngConfig&& config) {
+  return std::make_unique<AudioEncoderCng>(std::move(config));
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/cng/audio_encoder_cng.h

@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2014 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 MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+#define MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+
+#include <stddef.h>
+
+#include <memory>
+
+#include "api/audio_codecs/audio_encoder.h"
+#include "common_audio/vad/include/vad.h"
+
+namespace webrtc {
+
+struct AudioEncoderCngConfig {
+  // Moveable, not copyable.
+  AudioEncoderCngConfig();
+  AudioEncoderCngConfig(AudioEncoderCngConfig&&);
+  ~AudioEncoderCngConfig();
+
+  bool IsOk() const;
+
+  size_t num_channels = 1;
+  int payload_type = 13;
+  std::unique_ptr<AudioEncoder> speech_encoder;
+  Vad::Aggressiveness vad_mode = Vad::kVadNormal;
+  int sid_frame_interval_ms = 100;
+  int num_cng_coefficients = 8;
+  // The Vad pointer is mainly for testing. If a NULL pointer is passed, the
+  // AudioEncoderCng creates (and destroys) a Vad object internally. If an
+  // object is passed, the AudioEncoderCng assumes ownership of the Vad
+  // object.
+  Vad* vad = nullptr;
+};
+
+std::unique_ptr<AudioEncoder> CreateComfortNoiseEncoder(
+    AudioEncoderCngConfig&& config);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_

modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc

@@ -0,0 +1,520 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+
+#include <memory>
+#include <vector>
+
+#include "common_audio/vad/mock/mock_vad.h"
+#include "rtc_base/constructor_magic.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+using ::testing::_;
+using ::testing::Eq;
+using ::testing::InSequence;
+using ::testing::Invoke;
+using ::testing::Not;
+using ::testing::Optional;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+namespace webrtc {
+
+namespace {
+static const size_t kMaxNumSamples = 48 * 10 * 2;  // 10 ms @ 48 kHz stereo.
+static const size_t kMockReturnEncodedBytes = 17;
+static const int kCngPayloadType = 18;
+}  // namespace
+
+class AudioEncoderCngTest : public ::testing::Test {
+ protected:
+  AudioEncoderCngTest()
+      : mock_encoder_owner_(new MockAudioEncoder),
+        mock_encoder_(mock_encoder_owner_.get()),
+        mock_vad_(new MockVad),
+        timestamp_(4711),
+        num_audio_samples_10ms_(0),
+        sample_rate_hz_(8000) {
+    memset(audio_, 0, kMaxNumSamples * 2);
+    EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(1));
+  }
+
+  void TearDown() override {
+    EXPECT_CALL(*mock_vad_, Die()).Times(1);
+    cng_.reset();
+  }
+
+  AudioEncoderCngConfig MakeCngConfig() {
+    AudioEncoderCngConfig config;
+    config.speech_encoder = std::move(mock_encoder_owner_);
+    EXPECT_TRUE(config.speech_encoder);
+
+    // Let the AudioEncoderCng object use a MockVad instead of its internally
+    // created Vad object.
+    config.vad = mock_vad_;
+    config.payload_type = kCngPayloadType;
+
+    return config;
+  }
+
+  void CreateCng(AudioEncoderCngConfig&& config) {
+    num_audio_samples_10ms_ = static_cast<size_t>(10 * sample_rate_hz_ / 1000);
+    ASSERT_LE(num_audio_samples_10ms_, kMaxNumSamples);
+    if (config.speech_encoder) {
+      EXPECT_CALL(*mock_encoder_, SampleRateHz())
+          .WillRepeatedly(Return(sample_rate_hz_));
+      // Max10MsFramesInAPacket() is just used to verify that the SID frame
+      // period is not too small. The return value does not matter that much,
+      // as long as it is smaller than 10.
+      EXPECT_CALL(*mock_encoder_, Max10MsFramesInAPacket())
+          .WillOnce(Return(1u));
+    }
+    cng_ = CreateComfortNoiseEncoder(std::move(config));
+  }
+
+  void Encode() {
+    ASSERT_TRUE(cng_) << "Must call CreateCng() first.";
+    encoded_info_ = cng_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms_),
+        &encoded_);
+    timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
+  }
+
+  // Expect |num_calls| calls to the encoder, all successful. The last call
+  // claims to have encoded |kMockReturnEncodedBytes| bytes, and all the
+  // preceding ones 0 bytes.
+  void ExpectEncodeCalls(size_t num_calls) {
+    InSequence s;
+    AudioEncoder::EncodedInfo info;
+    for (size_t j = 0; j < num_calls - 1; ++j) {
+      EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).WillOnce(Return(info));
+    }
+    info.encoded_bytes = kMockReturnEncodedBytes;
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(
+            Invoke(MockAudioEncoder::FakeEncoding(kMockReturnEncodedBytes)));
+  }
+
+  // Verifies that the cng_ object waits until it has collected
+  // |blocks_per_frame| blocks of audio, and then dispatches all of them to
+  // the underlying codec (speech or cng).
+  void CheckBlockGrouping(size_t blocks_per_frame, bool active_speech) {
+    EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+        .WillRepeatedly(Return(blocks_per_frame));
+    auto config = MakeCngConfig();
+    const int num_cng_coefficients = config.num_cng_coefficients;
+    CreateCng(std::move(config));
+    EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+        .WillRepeatedly(Return(active_speech ? Vad::kActive : Vad::kPassive));
+
+    // Don't expect any calls to the encoder yet.
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).Times(0);
+    for (size_t i = 0; i < blocks_per_frame - 1; ++i) {
+      Encode();
+      EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+    }
+    if (active_speech)
+      ExpectEncodeCalls(blocks_per_frame);
+    Encode();
+    if (active_speech) {
+      EXPECT_EQ(kMockReturnEncodedBytes, encoded_info_.encoded_bytes);
+    } else {
+      EXPECT_EQ(static_cast<size_t>(num_cng_coefficients + 1),
+                encoded_info_.encoded_bytes);
+    }
+  }
+
+  // Verifies that the audio is partitioned into larger blocks before calling
+  // the VAD.
+  void CheckVadInputSize(int input_frame_size_ms,
+                         int expected_first_block_size_ms,
+                         int expected_second_block_size_ms) {
+    const size_t blocks_per_frame =
+        static_cast<size_t>(input_frame_size_ms / 10);
+
+    EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+        .WillRepeatedly(Return(blocks_per_frame));
+
+    // Expect nothing to happen before the last block is sent to cng_.
+    EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)).Times(0);
+    for (size_t i = 0; i < blocks_per_frame - 1; ++i) {
+      Encode();
+    }
+
+    // Let the VAD decision be passive, since an active decision may lead to
+    // early termination of the decision loop.
+    InSequence s;
+    EXPECT_CALL(
+        *mock_vad_,
+        VoiceActivity(_, expected_first_block_size_ms * sample_rate_hz_ / 1000,
+                      sample_rate_hz_))
+        .WillOnce(Return(Vad::kPassive));
+    if (expected_second_block_size_ms > 0) {
+      EXPECT_CALL(*mock_vad_,
+                  VoiceActivity(
+                      _, expected_second_block_size_ms * sample_rate_hz_ / 1000,
+                      sample_rate_hz_))
+          .WillOnce(Return(Vad::kPassive));
+    }
+
+    // With this call to Encode(), |mock_vad_| should be called according to the
+    // above expectations.
+    Encode();
+  }
+
+  // Tests a frame with both active and passive speech. Returns true if the
+  // decision was active speech, false if it was passive.
+  bool CheckMixedActivePassive(Vad::Activity first_type,
+                               Vad::Activity second_type) {
+    // Set the speech encoder frame size to 60 ms, to ensure that the VAD will
+    // be called twice.
+    const size_t blocks_per_frame = 6;
+    EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+        .WillRepeatedly(Return(blocks_per_frame));
+    InSequence s;
+    EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+        .WillOnce(Return(first_type));
+    if (first_type == Vad::kPassive) {
+      // Expect a second call to the VAD only if the first frame was passive.
+      EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+          .WillOnce(Return(second_type));
+    }
+    encoded_info_.payload_type = 0;
+    for (size_t i = 0; i < blocks_per_frame; ++i) {
+      Encode();
+    }
+    return encoded_info_.payload_type != kCngPayloadType;
+  }
+
+  std::unique_ptr<AudioEncoder> cng_;
+  std::unique_ptr<MockAudioEncoder> mock_encoder_owner_;
+  MockAudioEncoder* mock_encoder_;
+  MockVad* mock_vad_;  // Ownership is transferred to |cng_|.
+  uint32_t timestamp_;
+  int16_t audio_[kMaxNumSamples];
+  size_t num_audio_samples_10ms_;
+  rtc::Buffer encoded_;
+  AudioEncoder::EncodedInfo encoded_info_;
+  int sample_rate_hz_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCngTest);
+};
+
+TEST_F(AudioEncoderCngTest, CreateAndDestroy) {
+  CreateCng(MakeCngConfig());
+}
+
+TEST_F(AudioEncoderCngTest, CheckFrameSizePropagation) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillOnce(Return(17U));
+  EXPECT_EQ(17U, cng_->Num10MsFramesInNextPacket());
+}
+
+TEST_F(AudioEncoderCngTest, CheckTargetAudioBitratePropagation) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_,
+              OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
+  cng_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
+}
+
+TEST_F(AudioEncoderCngTest, CheckPacketLossFractionPropagation) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5));
+  cng_->OnReceivedUplinkPacketLossFraction(0.5);
+}
+
+TEST_F(AudioEncoderCngTest, CheckGetFrameLengthRangePropagation) {
+  CreateCng(MakeCngConfig());
+  auto expected_range =
+      std::make_pair(TimeDelta::Millis(20), TimeDelta::Millis(20));
+  EXPECT_CALL(*mock_encoder_, GetFrameLengthRange())
+      .WillRepeatedly(Return(absl::make_optional(expected_range)));
+  EXPECT_THAT(cng_->GetFrameLengthRange(), Optional(Eq(expected_range)));
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCallsVad) {
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(1U));
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kPassive));
+  Encode();
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects1BlockPassiveSpeech) {
+  CheckBlockGrouping(1, false);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects2BlocksPassiveSpeech) {
+  CheckBlockGrouping(2, false);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects3BlocksPassiveSpeech) {
+  CheckBlockGrouping(3, false);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects1BlockActiveSpeech) {
+  CheckBlockGrouping(1, true);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects2BlocksActiveSpeech) {
+  CheckBlockGrouping(2, true);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects3BlocksActiveSpeech) {
+  CheckBlockGrouping(3, true);
+}
+
+TEST_F(AudioEncoderCngTest, EncodePassive) {
+  const size_t kBlocksPerFrame = 3;
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(kBlocksPerFrame));
+  auto config = MakeCngConfig();
+  const auto sid_frame_interval_ms = config.sid_frame_interval_ms;
+  const auto num_cng_coefficients = config.num_cng_coefficients;
+  CreateCng(std::move(config));
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillRepeatedly(Return(Vad::kPassive));
+  // Expect no calls at all to the speech encoder mock.
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).Times(0);
+  uint32_t expected_timestamp = timestamp_;
+  for (size_t i = 0; i < 100; ++i) {
+    Encode();
+    // Check if it was time to call the cng encoder. This is done once every
+    // |kBlocksPerFrame| calls.
+    if ((i + 1) % kBlocksPerFrame == 0) {
+      // Now check if a SID interval has elapsed.
+      if ((i % (sid_frame_interval_ms / 10)) < kBlocksPerFrame) {
+        // If so, verify that we got a CNG encoding.
+        EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+        EXPECT_FALSE(encoded_info_.speech);
+        EXPECT_EQ(static_cast<size_t>(num_cng_coefficients) + 1,
+                  encoded_info_.encoded_bytes);
+        EXPECT_EQ(expected_timestamp, encoded_info_.encoded_timestamp);
+      }
+      expected_timestamp += rtc::checked_cast<uint32_t>(
+          kBlocksPerFrame * num_audio_samples_10ms_);
+    } else {
+      // Otherwise, expect no output.
+      EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+    }
+  }
+}
+
+// Verifies that the correct action is taken for frames with both active and
+// passive speech.
+TEST_F(AudioEncoderCngTest, MixedActivePassive) {
+  CreateCng(MakeCngConfig());
+
+  // All of the frame is active speech.
+  ExpectEncodeCalls(6);
+  EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kActive));
+  EXPECT_TRUE(encoded_info_.speech);
+
+  // First half of the frame is active speech.
+  ExpectEncodeCalls(6);
+  EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kPassive));
+  EXPECT_TRUE(encoded_info_.speech);
+
+  // Second half of the frame is active speech.
+  ExpectEncodeCalls(6);
+  EXPECT_TRUE(CheckMixedActivePassive(Vad::kPassive, Vad::kActive));
+  EXPECT_TRUE(encoded_info_.speech);
+
+  // All of the frame is passive speech. Expect no calls to |mock_encoder_|.
+  EXPECT_FALSE(CheckMixedActivePassive(Vad::kPassive, Vad::kPassive));
+  EXPECT_FALSE(encoded_info_.speech);
+}
+
+// These tests verify that the audio is partitioned into larger blocks before
+// calling the VAD.
+// The parameters for CheckVadInputSize are:
+// CheckVadInputSize(frame_size, expected_first_block_size,
+//                   expected_second_block_size);
+TEST_F(AudioEncoderCngTest, VadInputSize10Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(10, 10, 0);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize20Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(20, 20, 0);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize30Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(30, 30, 0);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize40Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(40, 20, 20);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize50Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(50, 30, 20);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize60Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(60, 30, 30);
+}
+
+// Verifies that the correct payload type is set when CNG is encoded.
+TEST_F(AudioEncoderCngTest, VerifyCngPayloadType) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).Times(0);
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(1U));
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kPassive));
+  encoded_info_.payload_type = 0;
+  Encode();
+  EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+}
+
+// Verifies that a SID frame is encoded immediately as the signal changes from
+// active speech to passive.
+TEST_F(AudioEncoderCngTest, VerifySidFrameAfterSpeech) {
+  auto config = MakeCngConfig();
+  const auto num_cng_coefficients = config.num_cng_coefficients;
+  CreateCng(std::move(config));
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(1U));
+  // Start with encoding noise.
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .Times(2)
+      .WillRepeatedly(Return(Vad::kPassive));
+  Encode();
+  EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+  EXPECT_EQ(static_cast<size_t>(num_cng_coefficients) + 1,
+            encoded_info_.encoded_bytes);
+  // Encode again, and make sure we got no frame at all (since the SID frame
+  // period is 100 ms by default).
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+
+  // Now encode active speech.
+  encoded_info_.payload_type = 0;
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kActive));
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(
+          Invoke(MockAudioEncoder::FakeEncoding(kMockReturnEncodedBytes)));
+  Encode();
+  EXPECT_EQ(kMockReturnEncodedBytes, encoded_info_.encoded_bytes);
+
+  // Go back to noise again, and verify that a SID frame is emitted.
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kPassive));
+  Encode();
+  EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+  EXPECT_EQ(static_cast<size_t>(num_cng_coefficients) + 1,
+            encoded_info_.encoded_bytes);
+}
+
+// Resetting the CNG should reset both the VAD and the encoder.
+TEST_F(AudioEncoderCngTest, Reset) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, Reset()).Times(1);
+  EXPECT_CALL(*mock_vad_, Reset()).Times(1);
+  cng_->Reset();
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// This test fixture tests various error conditions that makes the
+// AudioEncoderCng die via CHECKs.
+class AudioEncoderCngDeathTest : public AudioEncoderCngTest {
+ protected:
+  AudioEncoderCngDeathTest() : AudioEncoderCngTest() {
+    EXPECT_CALL(*mock_vad_, Die()).Times(1);
+    delete mock_vad_;
+    mock_vad_ = nullptr;
+  }
+
+  // Override AudioEncoderCngTest::TearDown, since that one expects a call to
+  // the destructor of |mock_vad_|. In this case, that object is already
+  // deleted.
+  void TearDown() override { cng_.reset(); }
+
+  AudioEncoderCngConfig MakeCngConfig() {
+    // Don't provide a Vad mock object, since it would leak when the test dies.
+    auto config = AudioEncoderCngTest::MakeCngConfig();
+    config.vad = nullptr;
+    return config;
+  }
+
+  void TryWrongNumCoefficients(int num) {
+    RTC_EXPECT_DEATH(
+        [&] {
+          auto config = MakeCngConfig();
+          config.num_cng_coefficients = num;
+          CreateCng(std::move(config));
+        }(),
+        "Invalid configuration");
+  }
+};
+
+TEST_F(AudioEncoderCngDeathTest, WrongFrameSize) {
+  CreateCng(MakeCngConfig());
+  num_audio_samples_10ms_ *= 2;  // 20 ms frame.
+  RTC_EXPECT_DEATH(Encode(), "");
+  num_audio_samples_10ms_ = 0;  // Zero samples.
+  RTC_EXPECT_DEATH(Encode(), "");
+}
+
+TEST_F(AudioEncoderCngDeathTest, WrongNumCoefficientsA) {
+  TryWrongNumCoefficients(-1);
+}
+
+TEST_F(AudioEncoderCngDeathTest, WrongNumCoefficientsB) {
+  TryWrongNumCoefficients(0);
+}
+
+TEST_F(AudioEncoderCngDeathTest, WrongNumCoefficientsC) {
+  TryWrongNumCoefficients(13);
+}
+
+TEST_F(AudioEncoderCngDeathTest, NullSpeechEncoder) {
+  auto config = MakeCngConfig();
+  config.speech_encoder = nullptr;
+  RTC_EXPECT_DEATH(CreateCng(std::move(config)), "");
+}
+
+TEST_F(AudioEncoderCngDeathTest, StereoEncoder) {
+  EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(2));
+  RTC_EXPECT_DEATH(CreateCng(MakeCngConfig()), "Invalid configuration");
+}
+
+TEST_F(AudioEncoderCngDeathTest, StereoConfig) {
+  RTC_EXPECT_DEATH(
+      [&] {
+        auto config = MakeCngConfig();
+        config.num_channels = 2;
+        CreateCng(std::move(config));
+      }(),
+      "Invalid configuration");
+}
+
+TEST_F(AudioEncoderCngDeathTest, EncoderFrameSizeTooLarge) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(7U));
+  for (int i = 0; i < 6; ++i)
+    Encode();
+  RTC_EXPECT_DEATH(
+      Encode(), "Frame size cannot be larger than 60 ms when using VAD/CNG.");
+}
+
+#endif  // GTEST_HAS_DEATH_TEST
+
+}  // namespace webrtc

modules/audio_coding/codecs/cng/cng_unittest.cc

@@ -0,0 +1,250 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+enum {
+  kSidShortIntervalUpdate = 1,
+  kSidNormalIntervalUpdate = 100,
+  kSidLongIntervalUpdate = 10000
+};
+
+enum : size_t {
+  kCNGNumParamsLow = 0,
+  kCNGNumParamsNormal = 8,
+  kCNGNumParamsHigh = WEBRTC_CNG_MAX_LPC_ORDER,
+  kCNGNumParamsTooHigh = WEBRTC_CNG_MAX_LPC_ORDER + 1
+};
+
+enum { kNoSid, kForceSid };
+
+class CngTest : public ::testing::Test {
+ protected:
+  virtual void SetUp();
+
+  void TestCngEncode(int sample_rate_hz, int quality);
+
+  int16_t speech_data_[640];  // Max size of CNG internal buffers.
+};
+
+void CngTest::SetUp() {
+  FILE* input_file;
+  const std::string file_name =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  input_file = fopen(file_name.c_str(), "rb");
+  ASSERT_TRUE(input_file != NULL);
+  ASSERT_EQ(640, static_cast<int32_t>(
+                     fread(speech_data_, sizeof(int16_t), 640, input_file)));
+  fclose(input_file);
+  input_file = NULL;
+}
+
+void CngTest::TestCngEncode(int sample_rate_hz, int quality) {
+  const size_t num_samples_10ms = rtc::CheckedDivExact(sample_rate_hz, 100);
+  rtc::Buffer sid_data;
+
+  ComfortNoiseEncoder cng_encoder(sample_rate_hz, kSidNormalIntervalUpdate,
+                                  quality);
+  EXPECT_EQ(0U, cng_encoder.Encode(rtc::ArrayView<const int16_t>(
+                                       speech_data_, num_samples_10ms),
+                                   kNoSid, &sid_data));
+  EXPECT_EQ(static_cast<size_t>(quality + 1),
+            cng_encoder.Encode(
+                rtc::ArrayView<const int16_t>(speech_data_, num_samples_10ms),
+                kForceSid, &sid_data));
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Create CNG encoder, init with faulty values, free CNG encoder.
+TEST_F(CngTest, CngInitFail) {
+  // Call with too few parameters.
+  EXPECT_DEATH(
+      {
+        ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate, kCNGNumParamsLow);
+      },
+      "");
+  // Call with too many parameters.
+  EXPECT_DEATH(
+      {
+        ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate,
+                            kCNGNumParamsTooHigh);
+      },
+      "");
+}
+
+// Encode Cng with too long input vector.
+TEST_F(CngTest, CngEncodeTooLong) {
+  rtc::Buffer sid_data;
+
+  // Create encoder.
+  ComfortNoiseEncoder cng_encoder(8000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  // Run encoder with too much data.
+  EXPECT_DEATH(
+      cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 641),
+                         kNoSid, &sid_data),
+      "");
+}
+#endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+TEST_F(CngTest, CngEncode8000) {
+  TestCngEncode(8000, kCNGNumParamsNormal);
+}
+
+TEST_F(CngTest, CngEncode16000) {
+  TestCngEncode(16000, kCNGNumParamsNormal);
+}
+
+TEST_F(CngTest, CngEncode32000) {
+  TestCngEncode(32000, kCNGNumParamsHigh);
+}
+
+TEST_F(CngTest, CngEncode48000) {
+  TestCngEncode(48000, kCNGNumParamsNormal);
+}
+
+TEST_F(CngTest, CngEncode64000) {
+  TestCngEncode(64000, kCNGNumParamsNormal);
+}
+
+// Update SID parameters, for both 9 and 16 parameters.
+TEST_F(CngTest, CngUpdateSid) {
+  rtc::Buffer sid_data;
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // Run normal Encode and UpdateSid.
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+  cng_decoder.UpdateSid(sid_data);
+
+  // Reinit with new length.
+  cng_encoder.Reset(16000, kSidNormalIntervalUpdate, kCNGNumParamsHigh);
+  cng_decoder.Reset();
+
+  // Expect 0 because of unstable parameters after switching length.
+  EXPECT_EQ(0U,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+  EXPECT_EQ(
+      kCNGNumParamsHigh + 1,
+      cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_ + 160, 160),
+                         kForceSid, &sid_data));
+  cng_decoder.UpdateSid(
+      rtc::ArrayView<const uint8_t>(sid_data.data(), kCNGNumParamsNormal + 1));
+}
+
+// Update SID parameters, with wrong parameters or without calling decode.
+TEST_F(CngTest, CngUpdateSidErroneous) {
+  rtc::Buffer sid_data;
+
+  // Encode.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+
+  // First run with valid parameters, then with too many CNG parameters.
+  // The function will operate correctly by only reading the maximum number of
+  // parameters, skipping the extra.
+  EXPECT_EQ(kCNGNumParamsNormal + 1, sid_data.size());
+  cng_decoder.UpdateSid(sid_data);
+
+  // Make sure the input buffer is large enough. Since Encode() appends data, we
+  // need to set the size manually only afterwards, or the buffer will be bigger
+  // than anticipated.
+  sid_data.SetSize(kCNGNumParamsTooHigh + 1);
+  cng_decoder.UpdateSid(sid_data);
+}
+
+// Test to generate cng data, by forcing SID. Both normal and faulty condition.
+TEST_F(CngTest, CngGenerate) {
+  rtc::Buffer sid_data;
+  int16_t out_data[640];
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // Normal Encode.
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+
+  // Normal UpdateSid.
+  cng_decoder.UpdateSid(sid_data);
+
+  // Two normal Generate, one with new_period.
+  EXPECT_TRUE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 640), 1));
+  EXPECT_TRUE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 640), 0));
+
+  // Call Genereate with too much data.
+  EXPECT_FALSE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 641), 0));
+}
+
+// Test automatic SID.
+TEST_F(CngTest, CngAutoSid) {
+  rtc::Buffer sid_data;
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // Normal Encode, 100 msec, where no SID data should be generated.
+  for (int i = 0; i < 10; i++) {
+    EXPECT_EQ(
+        0U, cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kNoSid, &sid_data));
+  }
+
+  // We have reached 100 msec, and SID data should be generated.
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kNoSid, &sid_data));
+}
+
+// Test automatic SID, with very short interval.
+TEST_F(CngTest, CngAutoSidShort) {
+  rtc::Buffer sid_data;
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidShortIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // First call will never generate SID, unless forced to.
+  EXPECT_EQ(0U,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kNoSid, &sid_data));
+
+  // Normal Encode, 100 msec, SID data should be generated all the time.
+  for (int i = 0; i < 10; i++) {
+    EXPECT_EQ(
+        kCNGNumParamsNormal + 1,
+        cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                           kNoSid, &sid_data));
+  }
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/cng/webrtc_cng.cc

@@ -0,0 +1,436 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+
+#include <algorithm>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kCngMaxOutsizeOrder = 640;
+
+// TODO(ossu): Rename the left-over WebRtcCng according to style guide.
+void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a);
+
+const int32_t WebRtcCng_kDbov[94] = {
+    1081109975, 858756178, 682134279, 541838517, 430397633, 341876992,
+    271562548,  215709799, 171344384, 136103682, 108110997, 85875618,
+    68213428,   54183852,  43039763,  34187699,  27156255,  21570980,
+    17134438,   13610368,  10811100,  8587562,   6821343,   5418385,
+    4303976,    3418770,   2715625,   2157098,   1713444,   1361037,
+    1081110,    858756,    682134,    541839,    430398,    341877,
+    271563,     215710,    171344,    136104,    108111,    85876,
+    68213,      54184,     43040,     34188,     27156,     21571,
+    17134,      13610,     10811,     8588,      6821,      5418,
+    4304,       3419,      2716,      2157,      1713,      1361,
+    1081,       859,       682,       542,       430,       342,
+    272,        216,       171,       136,       108,       86,
+    68,         54,        43,        34,        27,        22,
+    17,         14,        11,        9,         7,         5,
+    4,          3,         3,         2,         2,         1,
+    1,          1,         1,         1};
+
+const int16_t WebRtcCng_kCorrWindow[WEBRTC_CNG_MAX_LPC_ORDER] = {
+    32702, 32636, 32570, 32505, 32439, 32374,
+    32309, 32244, 32179, 32114, 32049, 31985};
+
+}  // namespace
+
+ComfortNoiseDecoder::ComfortNoiseDecoder() {
+  /* Needed to get the right function pointers in SPLIB. */
+  Reset();
+}
+
+void ComfortNoiseDecoder::Reset() {
+  dec_seed_ = 7777; /* For debugging only. */
+  dec_target_energy_ = 0;
+  dec_used_energy_ = 0;
+  for (auto& c : dec_target_reflCoefs_)
+    c = 0;
+  for (auto& c : dec_used_reflCoefs_)
+    c = 0;
+  for (auto& c : dec_filtstate_)
+    c = 0;
+  for (auto& c : dec_filtstateLow_)
+    c = 0;
+  dec_order_ = 5;
+  dec_target_scale_factor_ = 0;
+  dec_used_scale_factor_ = 0;
+}
+
+void ComfortNoiseDecoder::UpdateSid(rtc::ArrayView<const uint8_t> sid) {
+  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER];
+  int32_t targetEnergy;
+  size_t length = sid.size();
+  /* Throw away reflection coefficients of higher order than we can handle. */
+  if (length > (WEBRTC_CNG_MAX_LPC_ORDER + 1))
+    length = WEBRTC_CNG_MAX_LPC_ORDER + 1;
+
+  dec_order_ = static_cast<uint16_t>(length - 1);
+
+  uint8_t sid0 = std::min<uint8_t>(sid[0], 93);
+  targetEnergy = WebRtcCng_kDbov[sid0];
+  /* Take down target energy to 75%. */
+  targetEnergy = targetEnergy >> 1;
+  targetEnergy += targetEnergy >> 2;
+
+  dec_target_energy_ = targetEnergy;
+
+  /* Reconstruct coeffs with tweak for WebRtc implementation of RFC3389. */
+  if (dec_order_ == WEBRTC_CNG_MAX_LPC_ORDER) {
+    for (size_t i = 0; i < (dec_order_); i++) {
+      refCs[i] = sid[i + 1] << 8; /* Q7 to Q15*/
+      dec_target_reflCoefs_[i] = refCs[i];
+    }
+  } else {
+    for (size_t i = 0; i < (dec_order_); i++) {
+      refCs[i] = (sid[i + 1] - 127) * (1 << 8); /* Q7 to Q15. */
+      dec_target_reflCoefs_[i] = refCs[i];
+    }
+  }
+
+  for (size_t i = (dec_order_); i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
+    refCs[i] = 0;
+    dec_target_reflCoefs_[i] = refCs[i];
+  }
+}
+
+bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
+                                   bool new_period) {
+  int16_t excitation[kCngMaxOutsizeOrder];
+  int16_t low[kCngMaxOutsizeOrder];
+  int16_t lpPoly[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t ReflBetaStd = 26214;      /* 0.8 in q15. */
+  int16_t ReflBetaCompStd = 6553;   /* 0.2 in q15. */
+  int16_t ReflBetaNewP = 19661;     /* 0.6 in q15. */
+  int16_t ReflBetaCompNewP = 13107; /* 0.4 in q15. */
+  int16_t Beta, BetaC;              /* These are in Q15. */
+  int32_t targetEnergy;
+  int16_t En;
+  int16_t temp16;
+  const size_t num_samples = out_data.size();
+
+  if (num_samples > kCngMaxOutsizeOrder) {
+    return false;
+  }
+
+  if (new_period) {
+    dec_used_scale_factor_ = dec_target_scale_factor_;
+    Beta = ReflBetaNewP;
+    BetaC = ReflBetaCompNewP;
+  } else {
+    Beta = ReflBetaStd;
+    BetaC = ReflBetaCompStd;
+  }
+
+  /* Calculate new scale factor in Q13 */
+  dec_used_scale_factor_ = rtc::checked_cast<int16_t>(
+      WEBRTC_SPL_MUL_16_16_RSFT(dec_used_scale_factor_, Beta >> 2, 13) +
+      WEBRTC_SPL_MUL_16_16_RSFT(dec_target_scale_factor_, BetaC >> 2, 13));
+
+  dec_used_energy_ = dec_used_energy_ >> 1;
+  dec_used_energy_ += dec_target_energy_ >> 1;
+
+  /* Do the same for the reflection coeffs, albeit in Q15. */
+  for (size_t i = 0; i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
+    dec_used_reflCoefs_[i] =
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_used_reflCoefs_[i], Beta, 15);
+    dec_used_reflCoefs_[i] +=
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_target_reflCoefs_[i], BetaC, 15);
+  }
+
+  /* Compute the polynomial coefficients. */
+  WebRtcCng_K2a16(dec_used_reflCoefs_, WEBRTC_CNG_MAX_LPC_ORDER, lpPoly);
+
+  targetEnergy = dec_used_energy_;
+
+  /* Calculate scaling factor based on filter energy. */
+  En = 8192; /* 1.0 in Q13. */
+  for (size_t i = 0; i < (WEBRTC_CNG_MAX_LPC_ORDER); i++) {
+    /* Floating point value for reference.
+       E *= 1.0 - (dec_used_reflCoefs_[i] / 32768.0) *
+       (dec_used_reflCoefs_[i] / 32768.0);
+     */
+
+    /* Same in fixed point. */
+    /* K(i).^2 in Q15. */
+    temp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_used_reflCoefs_[i],
+                                                dec_used_reflCoefs_[i], 15);
+    /* 1 - K(i).^2 in Q15. */
+    temp16 = 0x7fff - temp16;
+    En = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(En, temp16, 15);
+  }
+
+  /* float scaling= sqrt(E * dec_target_energy_ / (1 << 24)); */
+
+  /* Calculate sqrt(En * target_energy / excitation energy) */
+  targetEnergy = WebRtcSpl_Sqrt(dec_used_energy_);
+
+  En = (int16_t)WebRtcSpl_Sqrt(En) << 6;
+  En = (En * 3) >> 1; /* 1.5 estimates sqrt(2). */
+  dec_used_scale_factor_ = (int16_t)((En * targetEnergy) >> 12);
+
+  /* Generate excitation. */
+  /* Excitation energy per sample is 2.^24 - Q13 N(0,1). */
+  for (size_t i = 0; i < num_samples; i++) {
+    excitation[i] = WebRtcSpl_RandN(&dec_seed_) >> 1;
+  }
+
+  /* Scale to correct energy. */
+  WebRtcSpl_ScaleVector(excitation, excitation, dec_used_scale_factor_,
+                        num_samples, 13);
+
+  /* |lpPoly| - Coefficients in Q12.
+   * |excitation| - Speech samples.
+   * |nst->dec_filtstate| - State preservation.
+   * |out_data| - Filtered speech samples. */
+  WebRtcSpl_FilterAR(lpPoly, WEBRTC_CNG_MAX_LPC_ORDER + 1, excitation,
+                     num_samples, dec_filtstate_, WEBRTC_CNG_MAX_LPC_ORDER,
+                     dec_filtstateLow_, WEBRTC_CNG_MAX_LPC_ORDER,
+                     out_data.data(), low, num_samples);
+
+  return true;
+}
+
+ComfortNoiseEncoder::ComfortNoiseEncoder(int fs, int interval, int quality)
+    : enc_nrOfCoefs_(quality),
+      enc_sampfreq_(fs),
+      enc_interval_(interval),
+      enc_msSinceSid_(0),
+      enc_Energy_(0),
+      enc_reflCoefs_{0},
+      enc_corrVector_{0},
+      enc_seed_(7777) /* For debugging only. */ {
+  RTC_CHECK_GT(quality, 0);
+  RTC_CHECK_LE(quality, WEBRTC_CNG_MAX_LPC_ORDER);
+}
+
+void ComfortNoiseEncoder::Reset(int fs, int interval, int quality) {
+  RTC_CHECK_GT(quality, 0);
+  RTC_CHECK_LE(quality, WEBRTC_CNG_MAX_LPC_ORDER);
+  enc_nrOfCoefs_ = quality;
+  enc_sampfreq_ = fs;
+  enc_interval_ = interval;
+  enc_msSinceSid_ = 0;
+  enc_Energy_ = 0;
+  for (auto& c : enc_reflCoefs_)
+    c = 0;
+  for (auto& c : enc_corrVector_)
+    c = 0;
+  enc_seed_ = 7777; /* For debugging only. */
+}
+
+size_t ComfortNoiseEncoder::Encode(rtc::ArrayView<const int16_t> speech,
+                                   bool force_sid,
+                                   rtc::Buffer* output) {
+  int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int32_t corrVector[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t hanningW[kCngMaxOutsizeOrder];
+  int16_t ReflBeta = 19661;     /* 0.6 in q15. */
+  int16_t ReflBetaComp = 13107; /* 0.4 in q15. */
+  int32_t outEnergy;
+  int outShifts;
+  size_t i;
+  int stab;
+  int acorrScale;
+  size_t index;
+  size_t ind, factor;
+  int32_t* bptr;
+  int32_t blo, bhi;
+  int16_t negate;
+  const int16_t* aptr;
+  int16_t speechBuf[kCngMaxOutsizeOrder];
+
+  const size_t num_samples = speech.size();
+  RTC_CHECK_LE(num_samples, kCngMaxOutsizeOrder);
+
+  for (i = 0; i < num_samples; i++) {
+    speechBuf[i] = speech[i];
+  }
+
+  factor = num_samples;
+
+  /* Calculate energy and a coefficients. */
+  outEnergy = WebRtcSpl_Energy(speechBuf, num_samples, &outShifts);
+  while (outShifts > 0) {
+    /* We can only do 5 shifts without destroying accuracy in
+     * division factor. */
+    if (outShifts > 5) {
+      outEnergy <<= (outShifts - 5);
+      outShifts = 5;
+    } else {
+      factor /= 2;
+      outShifts--;
+    }
+  }
+  outEnergy = WebRtcSpl_DivW32W16(outEnergy, (int16_t)factor);
+
+  if (outEnergy > 1) {
+    /* Create Hanning Window. */
+    WebRtcSpl_GetHanningWindow(hanningW, num_samples / 2);
+    for (i = 0; i < (num_samples / 2); i++)
+      hanningW[num_samples - i - 1] = hanningW[i];
+
+    WebRtcSpl_ElementwiseVectorMult(speechBuf, hanningW, speechBuf, num_samples,
+                                    14);
+
+    WebRtcSpl_AutoCorrelation(speechBuf, num_samples, enc_nrOfCoefs_,
+                              corrVector, &acorrScale);
+
+    if (*corrVector == 0)
+      *corrVector = WEBRTC_SPL_WORD16_MAX;
+
+    /* Adds the bandwidth expansion. */
+    aptr = WebRtcCng_kCorrWindow;
+    bptr = corrVector;
+
+    /* (zzz) lpc16_1 = 17+1+820+2+2 = 842 (ordo2=700). */
+    for (ind = 0; ind < enc_nrOfCoefs_; ind++) {
+      /* The below code multiplies the 16 b corrWindow values (Q15) with
+       * the 32 b corrvector (Q0) and shifts the result down 15 steps. */
+      negate = *bptr < 0;
+      if (negate)
+        *bptr = -*bptr;
+
+      blo = (int32_t)*aptr * (*bptr & 0xffff);
+      bhi = ((blo >> 16) & 0xffff) +
+            ((int32_t)(*aptr++) * ((*bptr >> 16) & 0xffff));
+      blo = (blo & 0xffff) | ((bhi & 0xffff) << 16);
+
+      *bptr = (((bhi >> 16) & 0x7fff) << 17) | ((uint32_t)blo >> 15);
+      if (negate)
+        *bptr = -*bptr;
+      bptr++;
+    }
+    /* End of bandwidth expansion. */
+
+    stab = WebRtcSpl_LevinsonDurbin(corrVector, arCoefs, refCs, enc_nrOfCoefs_);
+
+    if (!stab) {
+      /* Disregard from this frame */
+      return 0;
+    }
+
+  } else {
+    for (i = 0; i < enc_nrOfCoefs_; i++)
+      refCs[i] = 0;
+  }
+
+  if (force_sid) {
+    /* Read instantaneous values instead of averaged. */
+    for (i = 0; i < enc_nrOfCoefs_; i++)
+      enc_reflCoefs_[i] = refCs[i];
+    enc_Energy_ = outEnergy;
+  } else {
+    /* Average history with new values. */
+    for (i = 0; i < enc_nrOfCoefs_; i++) {
+      enc_reflCoefs_[i] =
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(enc_reflCoefs_[i], ReflBeta, 15);
+      enc_reflCoefs_[i] +=
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(refCs[i], ReflBetaComp, 15);
+    }
+    enc_Energy_ = (outEnergy >> 2) + (enc_Energy_ >> 1) + (enc_Energy_ >> 2);
+  }
+
+  if (enc_Energy_ < 1) {
+    enc_Energy_ = 1;
+  }
+
+  if ((enc_msSinceSid_ > (enc_interval_ - 1)) || force_sid) {
+    /* Search for best dbov value. */
+    index = 0;
+    for (i = 1; i < 93; i++) {
+      /* Always round downwards. */
+      if ((enc_Energy_ - WebRtcCng_kDbov[i]) > 0) {
+        index = i;
+        break;
+      }
+    }
+    if ((i == 93) && (index == 0))
+      index = 94;
+
+    const size_t output_coefs = enc_nrOfCoefs_ + 1;
+    output->AppendData(output_coefs, [&](rtc::ArrayView<uint8_t> output) {
+      output[0] = (uint8_t)index;
+
+      /* Quantize coefficients with tweak for WebRtc implementation of
+       * RFC3389. */
+      if (enc_nrOfCoefs_ == WEBRTC_CNG_MAX_LPC_ORDER) {
+        for (i = 0; i < enc_nrOfCoefs_; i++) {
+          /* Q15 to Q7 with rounding. */
+          output[i + 1] = ((enc_reflCoefs_[i] + 128) >> 8);
+        }
+      } else {
+        for (i = 0; i < enc_nrOfCoefs_; i++) {
+          /* Q15 to Q7 with rounding. */
+          output[i + 1] = (127 + ((enc_reflCoefs_[i] + 128) >> 8));
+        }
+      }
+
+      return output_coefs;
+    });
+
+    enc_msSinceSid_ =
+        static_cast<int16_t>((1000 * num_samples) / enc_sampfreq_);
+    return output_coefs;
+  } else {
+    enc_msSinceSid_ +=
+        static_cast<int16_t>((1000 * num_samples) / enc_sampfreq_);
+    return 0;
+  }
+}
+
+namespace {
+/* Values in |k| are Q15, and |a| Q12. */
+void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a) {
+  int16_t any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
+  int16_t* aptr;
+  int16_t* aptr2;
+  int16_t* anyptr;
+  const int16_t* kptr;
+  int m, i;
+
+  kptr = k;
+  *a = 4096; /* i.e., (Word16_MAX >> 3) + 1 */
+  *any = *a;
+  a[1] = (*k + 4) >> 3;
+  for (m = 1; m < useOrder; m++) {
+    kptr++;
+    aptr = a;
+    aptr++;
+    aptr2 = &a[m];
+    anyptr = any;
+    anyptr++;
+
+    any[m + 1] = (*kptr + 4) >> 3;
+    for (i = 0; i < m; i++) {
+      *anyptr++ =
+          (*aptr++) +
+          (int16_t)((((int32_t)(*aptr2--) * (int32_t)*kptr) + 16384) >> 15);
+    }
+
+    aptr = a;
+    anyptr = any;
+    for (i = 0; i < (m + 2); i++) {
+      *aptr++ = *anyptr++;
+    }
+  }
+}
+
+}  // namespace
+
+}  // namespace webrtc

modules/audio_coding/codecs/cng/webrtc_cng.h

@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2012 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 MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+#define MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+
+#include <stdint.h>
+
+#include <cstddef>
+
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+
+#define WEBRTC_CNG_MAX_LPC_ORDER 12
+
+namespace webrtc {
+
+class ComfortNoiseDecoder {
+ public:
+  ComfortNoiseDecoder();
+  ~ComfortNoiseDecoder() = default;
+
+  ComfortNoiseDecoder(const ComfortNoiseDecoder&) = delete;
+  ComfortNoiseDecoder& operator=(const ComfortNoiseDecoder&) = delete;
+
+  void Reset();
+
+  // Updates the CN state when a new SID packet arrives.
+  // |sid| is a view of the SID packet without the headers.
+  void UpdateSid(rtc::ArrayView<const uint8_t> sid);
+
+  // Generates comfort noise.
+  // |out_data| will be filled with samples - its size determines the number of
+  // samples generated. When |new_period| is true, CNG history will be reset
+  // before any audio is generated.  Returns |false| if outData is too large -
+  // currently 640 bytes (equalling 10ms at 64kHz).
+  // TODO(ossu): Specify better limits for the size of out_data. Either let it
+  //             be unbounded or limit to 10ms in the current sample rate.
+  bool Generate(rtc::ArrayView<int16_t> out_data, bool new_period);
+
+ private:
+  uint32_t dec_seed_;
+  int32_t dec_target_energy_;
+  int32_t dec_used_energy_;
+  int16_t dec_target_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_used_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_filtstate_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_filtstateLow_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  uint16_t dec_order_;
+  int16_t dec_target_scale_factor_; /* Q29 */
+  int16_t dec_used_scale_factor_;   /* Q29 */
+};
+
+class ComfortNoiseEncoder {
+ public:
+  // Creates a comfort noise encoder.
+  // |fs| selects sample rate: 8000 for narrowband or 16000 for wideband.
+  // |interval| sets the interval at which to generate SID data (in ms).
+  // |quality| selects the number of refl. coeffs. Maximum allowed is 12.
+  ComfortNoiseEncoder(int fs, int interval, int quality);
+  ~ComfortNoiseEncoder() = default;
+
+  ComfortNoiseEncoder(const ComfortNoiseEncoder&) = delete;
+  ComfortNoiseEncoder& operator=(const ComfortNoiseEncoder&) = delete;
+
+  // Resets the comfort noise encoder to its initial state.
+  // Parameters are set as during construction.
+  void Reset(int fs, int interval, int quality);
+
+  // Analyzes background noise from |speech| and appends coefficients to
+  // |output|.  Returns the number of coefficients generated.  If |force_sid| is
+  // true, a SID frame is forced and the internal sid interval counter is reset.
+  // Will fail if the input size is too large (> 640 samples, see
+  // ComfortNoiseDecoder::Generate).
+  size_t Encode(rtc::ArrayView<const int16_t> speech,
+                bool force_sid,
+                rtc::Buffer* output);
+
+ private:
+  size_t enc_nrOfCoefs_;
+  int enc_sampfreq_;
+  int16_t enc_interval_;
+  int16_t enc_msSinceSid_;
+  int32_t enc_Energy_;
+  int16_t enc_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int32_t enc_corrVector_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  uint32_t enc_seed_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_

modules/audio_coding/codecs/g711/audio_decoder_pcm.cc

@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+namespace webrtc {
+
+void AudioDecoderPcmU::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmU::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmU::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+void AudioDecoderPcmA::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmA::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmA::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/g711/audio_decoder_pcm.h

@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2015 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 MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class AudioDecoderPcmU final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1);
+  }
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const size_t num_channels_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmU);
+};
+
+class AudioDecoderPcmA final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1);
+  }
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const size_t num_channels_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcmA);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_

modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+bool AudioEncoderPcm::Config::IsOk() const {
+  return (frame_size_ms % 10 == 0) && (num_channels >= 1);
+}
+
+AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
+    : sample_rate_hz_(sample_rate_hz),
+      num_channels_(config.num_channels),
+      payload_type_(config.payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      full_frame_samples_(config.num_channels * config.frame_size_ms *
+                          sample_rate_hz / 1000),
+      first_timestamp_in_buffer_(0) {
+  RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
+  RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
+      << "Frame size must be an integer multiple of 10 ms.";
+  speech_buffer_.reserve(full_frame_samples_);
+}
+
+AudioEncoderPcm::~AudioEncoderPcm() = default;
+
+int AudioEncoderPcm::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioEncoderPcm::NumChannels() const {
+  return num_channels_;
+}
+
+size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderPcm::GetTargetBitrate() const {
+  return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
+                          NumChannels());
+}
+
+AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (speech_buffer_.empty()) {
+    first_timestamp_in_buffer_ = rtp_timestamp;
+  }
+  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
+  if (speech_buffer_.size() < full_frame_samples_) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
+  EncodedInfo info;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoded_bytes = encoded->AppendData(
+      full_frame_samples_ * BytesPerSample(),
+      [&](rtc::ArrayView<uint8_t> encoded) {
+        return EncodeCall(&speech_buffer_[0], full_frame_samples_,
+                          encoded.data());
+      });
+  speech_buffer_.clear();
+  info.encoder_type = GetCodecType();
+  return info;
+}
+
+void AudioEncoderPcm::Reset() {
+  speech_buffer_.clear();
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderPcm::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeA(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmA::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmA;
+}
+
+size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeU(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmU::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmU;
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/g711/audio_encoder_pcm.h

@@ -0,0 +1,125 @@
+/*
+ *  Copyright (c) 2014 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 MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class AudioEncoderPcm : public AudioEncoder {
+ public:
+  struct Config {
+   public:
+    bool IsOk() const;
+
+    int frame_size_ms;
+    size_t num_channels;
+    int payload_type;
+
+   protected:
+    explicit Config(int pt)
+        : frame_size_ms(20), num_channels(1), payload_type(pt) {}
+  };
+
+  ~AudioEncoderPcm() override;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  AudioEncoderPcm(const Config& config, int sample_rate_hz);
+
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+  virtual size_t EncodeCall(const int16_t* audio,
+                            size_t input_len,
+                            uint8_t* encoded) = 0;
+
+  virtual size_t BytesPerSample() const = 0;
+
+  // Used to set EncodedInfoLeaf::encoder_type in
+  // AudioEncoderPcm::EncodeImpl
+  virtual AudioEncoder::CodecType GetCodecType() const = 0;
+
+ private:
+  const int sample_rate_hz_;
+  const size_t num_channels_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  const size_t full_frame_samples_;
+  std::vector<int16_t> speech_buffer_;
+  uint32_t first_timestamp_in_buffer_;
+};
+
+class AudioEncoderPcmA final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+    Config() : AudioEncoderPcm::Config(8) {}
+  };
+
+  explicit AudioEncoderPcmA(const Config& config)
+      : AudioEncoderPcm(config, kSampleRateHz) {}
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+
+ private:
+  static const int kSampleRateHz = 8000;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmA);
+};
+
+class AudioEncoderPcmU final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+    Config() : AudioEncoderPcm::Config(0) {}
+  };
+
+  explicit AudioEncoderPcmU(const Config& config)
+      : AudioEncoderPcm(config, kSampleRateHz) {}
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+
+ private:
+  static const int kSampleRateHz = 8000;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_

modules/audio_coding/codecs/g711/g711_interface.c

@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+#include <string.h>
+
+#include "modules/third_party/g711/g711.h"
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    encoded[n] = linear_to_alaw(speechIn[n]);
+  return len;
+}
+
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    encoded[n] = linear_to_ulaw(speechIn[n]);
+  return len;
+}
+
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    decoded[n] = alaw_to_linear(encoded[n]);
+  *speechType = 1;
+  return len;
+}
+
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    decoded[n] = ulaw_to_linear(encoded[n]);
+  *speechType = 1;
+  return len;
+}
+
+int16_t WebRtcG711_Version(char* version, int16_t lenBytes) {
+  strncpy(version, "2.0.0", lenBytes);
+  return 0;
+}

modules/audio_coding/codecs/g711/g711_interface.h

@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2011 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 MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+
+#include <stdint.h>
+
+// Comfort noise constants
+#define G711_WEBRTC_SPEECH 1
+#define G711_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcG711_EncodeA(...)
+ *
+ * This function encodes a G711 A-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ *      - speechIn           : Input speech vector
+ *      - len                : Samples in speechIn
+ *
+ * Output:
+ *      - encoded            : The encoded data vector
+ *
+ * Return value              : Length (in bytes) of coded data.
+ *                             Always equal to len input parameter.
+ */
+
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG711_EncodeU(...)
+ *
+ * This function encodes a G711 U-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ *      - speechIn           : Input speech vector
+ *      - len                : Samples in speechIn
+ *
+ * Output:
+ *      - encoded            : The encoded data vector
+ *
+ * Return value              : Length (in bytes) of coded data.
+ *                             Always equal to len input parameter.
+ */
+
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG711_DecodeA(...)
+ *
+ * This function decodes a packet G711 A-law frame.
+ *
+ * Input:
+ *      - encoded            : Encoded data
+ *      - len                : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - speechType         : 1 normal, 2 CNG (for G711 it should
+ *                             always return 1 since G711 does not have a
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples in decoded vector
+ *                             -1 - Error
+ */
+
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcG711_DecodeU(...)
+ *
+ * This function decodes a packet G711 U-law frame.
+ *
+ * Input:
+ *      - encoded            : Encoded data
+ *      - len                : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - speechType         : 1 normal, 2 CNG (for G711 it should
+ *                             always return 1 since G711 does not have a
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples in decoded vector
+ *                             -1 - Error
+ */
+
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
+
+/**********************************************************************
+ * WebRtcG711_Version(...)
+ *
+ * This function gives the version string of the G.711 codec.
+ *
+ * Input:
+ *      - lenBytes:     the size of Allocated space (in Bytes) where
+ *                      the version number is written to (in string format).
+ *
+ * Output:
+ *      - version:      Pointer to a buffer where the version number is
+ *                      written to.
+ *
+ */
+
+int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_

modules/audio_coding/codecs/g711/test/testG711.cc

@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+/*
+ * testG711.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+/* include API */
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G711 1000
+
+/* function for reading audio data from PCM file */
+bool readframe(int16_t* data, FILE* inp, size_t length) {
+  size_t rlen = fread(data, sizeof(int16_t), length, inp);
+  if (rlen >= length)
+    return false;
+  memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+  return true;
+}
+
+int main(int argc, char* argv[]) {
+  char inname[80], outname[40], bitname[40];
+  FILE* inp;
+  FILE* outp;
+  FILE* bitp = NULL;
+  int framecnt;
+  bool endfile;
+
+  size_t framelength = 80;
+
+  /* Runtime statistics */
+  double starttime;
+  double runtime;
+  double length_file;
+
+  size_t stream_len = 0;
+  int16_t shortdata[480];
+  int16_t decoded[480];
+  uint8_t streamdata[1000];
+  int16_t speechType[1];
+  char law[2];
+  char versionNumber[40];
+
+  /* handling wrong input arguments in the command line */
+  if ((argc != 5) && (argc != 6)) {
+    printf("\n\nWrong number of arguments or flag values.\n\n");
+
+    printf("\n");
+    printf("\nG.711 test application\n\n");
+    printf("Usage:\n\n");
+    printf("./testG711.exe framelength law infile outfile \n\n");
+    printf("framelength: Framelength in samples.\n");
+    printf("law        : Coding law, A och u.\n");
+    printf("infile     : Normal speech input file\n");
+    printf("outfile    : Speech output file\n\n");
+    printf("outbits    : Output bitstream file [optional]\n\n");
+    exit(0);
+  }
+
+  /* Get version and print */
+  WebRtcG711_Version(versionNumber, 40);
+
+  printf("-----------------------------------\n");
+  printf("G.711 version: %s\n\n", versionNumber);
+  /* Get frame length */
+  int framelength_int = atoi(argv[1]);
+  if (framelength_int < 0) {
+    printf("  G.722: Invalid framelength %d.\n", framelength_int);
+    exit(1);
+  }
+  framelength = static_cast<size_t>(framelength_int);
+
+  /* Get compression law */
+  strcpy(law, argv[2]);
+
+  /* Get Input and Output files */
+  sscanf(argv[3], "%s", inname);
+  sscanf(argv[4], "%s", outname);
+  if (argc == 6) {
+    sscanf(argv[5], "%s", bitname);
+    if ((bitp = fopen(bitname, "wb")) == NULL) {
+      printf("  G.711: Cannot read file %s.\n", bitname);
+      exit(1);
+    }
+  }
+
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  G.711: Cannot read file %s.\n", inname);
+    exit(1);
+  }
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  G.711: Cannot write file %s.\n", outname);
+    exit(1);
+  }
+  printf("\nInput:  %s\nOutput: %s\n", inname, outname);
+  if (argc == 6) {
+    printf("\nBitfile:  %s\n", bitname);
+  }
+
+  starttime = clock() / (double)CLOCKS_PER_SEC_G711; /* Runtime statistics */
+
+  /* Initialize encoder and decoder */
+  framecnt = 0;
+  endfile = false;
+  while (!endfile) {
+    framecnt++;
+    /* Read speech block */
+    endfile = readframe(shortdata, inp, framelength);
+
+    /* G.711 encoding */
+    if (!strcmp(law, "A")) {
+      /* A-law encoding */
+      stream_len = WebRtcG711_EncodeA(shortdata, framelength, streamdata);
+      if (argc == 6) {
+        /* Write bits to file */
+        if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
+            stream_len) {
+          return -1;
+        }
+      }
+      WebRtcG711_DecodeA(streamdata, stream_len, decoded, speechType);
+    } else if (!strcmp(law, "u")) {
+      /* u-law encoding */
+      stream_len = WebRtcG711_EncodeU(shortdata, framelength, streamdata);
+      if (argc == 6) {
+        /* Write bits to file */
+        if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
+            stream_len) {
+          return -1;
+        }
+      }
+      WebRtcG711_DecodeU(streamdata, stream_len, decoded, speechType);
+    } else {
+      printf("Wrong law mode\n");
+      exit(1);
+    }
+    /* Write coded speech to file */
+    if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) {
+      return -1;
+    }
+  }
+
+  runtime = (double)(clock() / (double)CLOCKS_PER_SEC_G711 - starttime);
+  length_file = ((double)framecnt * (double)framelength / 8000);
+  printf("\n\nLength of speech file: %.1f s\n", length_file);
+  printf("Time to run G.711:      %.2f s (%.2f %% of realtime)\n\n", runtime,
+         (100 * runtime / length_file));
+  printf("---------------------END----------------------\n");
+
+  fclose(inp);
+  fclose(outp);
+
+  return 0;
+}

modules/audio_coding/codecs/g722/audio_decoder_g722.cc

@@ -0,0 +1,164 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+
+#include <string.h>
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderG722Impl::AudioDecoderG722Impl() {
+  WebRtcG722_CreateDecoder(&dec_state_);
+  WebRtcG722_DecoderInit(dec_state_);
+}
+
+AudioDecoderG722Impl::~AudioDecoderG722Impl() {
+  WebRtcG722_FreeDecoder(dec_state_);
+}
+
+bool AudioDecoderG722Impl::HasDecodePlc() const {
+  return false;
+}
+
+int AudioDecoderG722Impl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+void AudioDecoderG722Impl::Reset() {
+  WebRtcG722_DecoderInit(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722Impl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
+                                                 timestamp, 8, 16);
+}
+
+int AudioDecoderG722Impl::PacketDuration(const uint8_t* encoded,
+                                         size_t encoded_len) const {
+  // 1/2 encoded byte per sample per channel.
+  return static_cast<int>(2 * encoded_len / Channels());
+}
+
+int AudioDecoderG722Impl::SampleRateHz() const {
+  return 16000;
+}
+
+size_t AudioDecoderG722Impl::Channels() const {
+  return 1;
+}
+
+AudioDecoderG722StereoImpl::AudioDecoderG722StereoImpl() {
+  WebRtcG722_CreateDecoder(&dec_state_left_);
+  WebRtcG722_CreateDecoder(&dec_state_right_);
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+AudioDecoderG722StereoImpl::~AudioDecoderG722StereoImpl() {
+  WebRtcG722_FreeDecoder(dec_state_left_);
+  WebRtcG722_FreeDecoder(dec_state_right_);
+}
+
+int AudioDecoderG722StereoImpl::DecodeInternal(const uint8_t* encoded,
+                                               size_t encoded_len,
+                                               int sample_rate_hz,
+                                               int16_t* decoded,
+                                               SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  int16_t temp_type = 1;  // Default is speech.
+  // De-interleave the bit-stream into two separate payloads.
+  uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
+  SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
+  // Decode left and right.
+  size_t decoded_len = WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
+                                         encoded_len / 2, decoded, &temp_type);
+  size_t ret = WebRtcG722_Decode(
+      dec_state_right_, &encoded_deinterleaved[encoded_len / 2],
+      encoded_len / 2, &decoded[decoded_len], &temp_type);
+  if (ret == decoded_len) {
+    ret += decoded_len;  // Return total number of samples.
+    // Interleave output.
+    for (size_t k = ret / 2; k < ret; k++) {
+      int16_t temp = decoded[k];
+      memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
+              (ret - k - 1) * sizeof(int16_t));
+      decoded[2 * k - ret + 1] = temp;
+    }
+  }
+  *speech_type = ConvertSpeechType(temp_type);
+  delete[] encoded_deinterleaved;
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderG722StereoImpl::SampleRateHz() const {
+  return 16000;
+}
+
+size_t AudioDecoderG722StereoImpl::Channels() const {
+  return 2;
+}
+
+void AudioDecoderG722StereoImpl::Reset() {
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722StereoImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
+                                                 timestamp, 2 * 8, 16);
+}
+
+// Split the stereo packet and place left and right channel after each other
+// in the output array.
+void AudioDecoderG722StereoImpl::SplitStereoPacket(
+    const uint8_t* encoded,
+    size_t encoded_len,
+    uint8_t* encoded_deinterleaved) {
+  // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
+  // where "lx" is 4 bits representing left sample number x, and "rx" right
+  // sample. Two samples fit in one byte, represented with |...|.
+  for (size_t i = 0; i + 1 < encoded_len; i += 2) {
+    uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
+    encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
+    encoded_deinterleaved[i + 1] = right_byte;
+  }
+
+  // Move one byte representing right channel each loop, and place it at the
+  // end of the bytestream vector. After looping the data is reordered to:
+  // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
+  // where N is the total number of samples.
+  for (size_t i = 0; i < encoded_len / 2; i++) {
+    uint8_t right_byte = encoded_deinterleaved[i + 1];
+    memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
+            encoded_len - i - 2);
+    encoded_deinterleaved[encoded_len - 1] = right_byte;
+  }
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/g722/audio_decoder_g722.h

@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2015 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 MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/constructor_magic.h"
+
+typedef struct WebRtcG722DecInst G722DecInst;
+
+namespace webrtc {
+
+class AudioDecoderG722Impl final : public AudioDecoder {
+ public:
+  AudioDecoderG722Impl();
+  ~AudioDecoderG722Impl() override;
+  bool HasDecodePlc() const override;
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  G722DecInst* dec_state_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722Impl);
+};
+
+class AudioDecoderG722StereoImpl final : public AudioDecoder {
+ public:
+  AudioDecoderG722StereoImpl();
+  ~AudioDecoderG722StereoImpl() override;
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  // Splits the stereo-interleaved payload in |encoded| into separate payloads
+  // for left and right channels. The separated payloads are written to
+  // |encoded_deinterleaved|, which must hold at least |encoded_len| samples.
+  // The left channel starts at offset 0, while the right channel starts at
+  // offset encoded_len / 2 into |encoded_deinterleaved|.
+  void SplitStereoPacket(const uint8_t* encoded,
+                         size_t encoded_len,
+                         uint8_t* encoded_deinterleaved);
+
+  G722DecInst* dec_state_left_;
+  G722DecInst* dec_state_right_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderG722StereoImpl);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_

modules/audio_coding/codecs/g722/audio_encoder_g722.cc

@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kSampleRateHz = 16000;
+
+}  // namespace
+
+AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config,
+                                           int payload_type)
+    : num_channels_(config.num_channels),
+      payload_type_(payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      num_10ms_frames_buffered_(0),
+      first_timestamp_in_buffer_(0),
+      encoders_(new EncoderState[num_channels_]),
+      interleave_buffer_(2 * num_channels_) {
+  RTC_CHECK(config.IsOk());
+  const size_t samples_per_channel =
+      kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+  for (size_t i = 0; i < num_channels_; ++i) {
+    encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
+    encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
+  }
+  Reset();
+}
+
+AudioEncoderG722Impl::~AudioEncoderG722Impl() = default;
+
+int AudioEncoderG722Impl::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+size_t AudioEncoderG722Impl::NumChannels() const {
+  return num_channels_;
+}
+
+int AudioEncoderG722Impl::RtpTimestampRateHz() const {
+  // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
+  // codec.
+  return kSampleRateHz / 2;
+}
+
+size_t AudioEncoderG722Impl::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderG722Impl::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderG722Impl::GetTargetBitrate() const {
+  // 4 bits/sample, 16000 samples/s/channel.
+  return static_cast<int>(64000 * NumChannels());
+}
+
+void AudioEncoderG722Impl::Reset() {
+  num_10ms_frames_buffered_ = 0;
+  for (size_t i = 0; i < num_channels_; ++i)
+    RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderG722Impl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+AudioEncoder::EncodedInfo AudioEncoderG722Impl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (num_10ms_frames_buffered_ == 0)
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  // Deinterleave samples and save them in each channel's buffer.
+  const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
+  for (size_t i = 0; i < kSampleRateHz / 100; ++i)
+    for (size_t j = 0; j < num_channels_; ++j)
+      encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
+
+  // If we don't yet have enough samples for a packet, we're done for now.
+  if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+    return EncodedInfo();
+  }
+
+  // Encode each channel separately.
+  RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  num_10ms_frames_buffered_ = 0;
+  const size_t samples_per_channel = SamplesPerChannel();
+  for (size_t i = 0; i < num_channels_; ++i) {
+    const size_t bytes_encoded = WebRtcG722_Encode(
+        encoders_[i].encoder, encoders_[i].speech_buffer.get(),
+        samples_per_channel, encoders_[i].encoded_buffer.data());
+    RTC_CHECK_EQ(bytes_encoded, samples_per_channel / 2);
+  }
+
+  const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;
+  EncodedInfo info;
+  info.encoded_bytes = encoded->AppendData(
+      bytes_to_encode, [&](rtc::ArrayView<uint8_t> encoded) {
+        // Interleave the encoded bytes of the different channels. Each separate
+        // channel and the interleaved stream encodes two samples per byte, most
+        // significant half first.
+        for (size_t i = 0; i < samples_per_channel / 2; ++i) {
+          for (size_t j = 0; j < num_channels_; ++j) {
+            uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
+            interleave_buffer_.data()[j] = two_samples >> 4;
+            interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
+          }
+          for (size_t j = 0; j < num_channels_; ++j)
+            encoded[i * num_channels_ + j] =
+                interleave_buffer_.data()[2 * j] << 4 |
+                interleave_buffer_.data()[2 * j + 1];
+        }
+
+        return bytes_to_encode;
+      });
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kG722;
+  return info;
+}
+
+AudioEncoderG722Impl::EncoderState::EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
+}
+
+AudioEncoderG722Impl::EncoderState::~EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
+}
+
+size_t AudioEncoderG722Impl::SamplesPerChannel() const {
+  return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/g722/audio_encoder_g722.h

@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2014 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 MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class AudioEncoderG722Impl final : public AudioEncoder {
+ public:
+  AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
+  ~AudioEncoderG722Impl() override;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  // The encoder state for one channel.
+  struct EncoderState {
+    G722EncInst* encoder;
+    std::unique_ptr<int16_t[]> speech_buffer;  // Queued up for encoding.
+    rtc::Buffer encoded_buffer;                // Already encoded.
+    EncoderState();
+    ~EncoderState();
+  };
+
+  size_t SamplesPerChannel() const;
+
+  const size_t num_channels_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
+  uint32_t first_timestamp_in_buffer_;
+  const std::unique_ptr<EncoderState[]> encoders_;
+  rtc::Buffer interleave_buffer_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722Impl);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_

modules/audio_coding/codecs/g722/g722_interface.c

@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/third_party/g722/g722_enc_dec.h"
+
+int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
+{
+    *G722enc_inst=(G722EncInst*)malloc(sizeof(G722EncoderState));
+    if (*G722enc_inst!=NULL) {
+      return(0);
+    } else {
+      return(-1);
+    }
+}
+
+int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
+{
+    // Create and/or reset the G.722 encoder
+    // Bitrate 64 kbps and wideband mode (2)
+    G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
+        (G722EncoderState*) G722enc_inst, 64000, 2);
+    if (G722enc_inst == NULL) {
+        return -1;
+    } else {
+        return 0;
+    }
+}
+
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+{
+    // Free encoder memory
+    return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
+}
+
+size_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded)
+{
+    unsigned char *codechar = (unsigned char*) encoded;
+    // Encode the input speech vector
+    return WebRtc_g722_encode((G722EncoderState*) G722enc_inst, codechar,
+                              speechIn, len);
+}
+
+int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
+{
+    *G722dec_inst=(G722DecInst*)malloc(sizeof(G722DecoderState));
+    if (*G722dec_inst!=NULL) {
+      return(0);
+    } else {
+      return(-1);
+    }
+}
+
+void WebRtcG722_DecoderInit(G722DecInst* inst) {
+  // Create and/or reset the G.722 decoder
+  // Bitrate 64 kbps and wideband mode (2)
+  WebRtc_g722_decode_init((G722DecoderState*)inst, 64000, 2);
+}
+
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+{
+    // Free encoder memory
+    return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
+}
+
+size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+                         const uint8_t *encoded,
+                         size_t len,
+                         int16_t *decoded,
+                         int16_t *speechType)
+{
+    // Decode the G.722 encoder stream
+    *speechType=G722_WEBRTC_SPEECH;
+    return WebRtc_g722_decode((G722DecoderState*) G722dec_inst, decoded,
+                              encoded, len);
+}
+
+int16_t WebRtcG722_Version(char *versionStr, short len)
+{
+    // Get version string
+    char version[30] = "2.0.0\n";
+    if (strlen(version) < (unsigned int)len)
+    {
+        strcpy(versionStr, version);
+        return 0;
+    }
+    else
+    {
+        return -1;
+    }
+}

modules/audio_coding/codecs/g722/g722_interface.h

@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2011 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 MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+
+#include <stdint.h>
+
+/*
+ * Solution to support multiple instances
+ */
+
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/*
+ * Comfort noise constants
+ */
+
+#define G722_WEBRTC_SPEECH 1
+#define G722_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcG722_CreateEncoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance for encoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+int16_t WebRtcG722_CreateEncoder(G722EncInst** G722enc_inst);
+
+/****************************************************************************
+ * WebRtcG722_EncoderInit(...)
+ *
+ * This function initializes a G722 instance
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance, i.e. the user that should receive
+ *                             be initialized
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+
+int16_t WebRtcG722_EncoderInit(G722EncInst* G722enc_inst);
+
+/****************************************************************************
+ * WebRtcG722_FreeEncoder(...)
+ *
+ * Free the memory used for G722 encoder
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance for encoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+int WebRtcG722_FreeEncoder(G722EncInst* G722enc_inst);
+
+/****************************************************************************
+ * WebRtcG722_Encode(...)
+ *
+ * This function encodes G722 encoded data.
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance, i.e. the user that should encode
+ *                              a packet
+ *     - speechIn             : Input speech vector
+ *     - len                  : Samples in speechIn
+ *
+ * Output:
+ *        - encoded           : The encoded data vector
+ *
+ * Return value               : Length (in bytes) of coded data
+ */
+
+size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG722_CreateDecoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ *     - G722dec_inst         : G722 instance for decoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+int16_t WebRtcG722_CreateDecoder(G722DecInst** G722dec_inst);
+
+/****************************************************************************
+ * WebRtcG722_DecoderInit(...)
+ *
+ * This function initializes a G722 instance
+ *
+ * Input:
+ *     - inst      : G722 instance
+ */
+
+void WebRtcG722_DecoderInit(G722DecInst* inst);
+
+/****************************************************************************
+ * WebRtcG722_FreeDecoder(...)
+ *
+ * Free the memory used for G722 decoder
+ *
+ * Input:
+ *     - G722dec_inst         : G722 instance for decoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+
+int WebRtcG722_FreeDecoder(G722DecInst* G722dec_inst);
+
+/****************************************************************************
+ * WebRtcG722_Decode(...)
+ *
+ * This function decodes a packet with G729 frame(s). Output speech length
+ * will be a multiple of 80 samples (80*frames/packet).
+ *
+ * Input:
+ *     - G722dec_inst       : G722 instance, i.e. the user that should decode
+ *                            a packet
+ *     - encoded            : Encoded G722 frame(s)
+ *     - len                : Bytes in encoded vector
+ *
+ * Output:
+ *        - decoded         : The decoded vector
+ *      - speechType        : 1 normal, 2 CNG (Since G722 does not have its own
+ *                            DTX/CNG scheme it should always return 1)
+ *
+ * Return value             : Samples in decoded vector
+ */
+
+size_t WebRtcG722_Decode(G722DecInst* G722dec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcG722_Version(...)
+ *
+ * Get a string with the current version of the codec
+ */
+
+int16_t WebRtcG722_Version(char* versionStr, short len);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */

modules/audio_coding/codecs/g722/test/testG722.cc

@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2012 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.
+ */
+
+/*
+ * testG722.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+/* include API */
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G722 100000
+
+// Forward declaration
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/* function for reading audio data from PCM file */
+bool readframe(int16_t* data, FILE* inp, size_t length) {
+  size_t rlen = fread(data, sizeof(int16_t), length, inp);
+  if (rlen >= length)
+    return false;
+  memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+  return true;
+}
+
+int main(int argc, char* argv[]) {
+  char inname[60], outbit[40], outname[40];
+  FILE *inp, *outbitp, *outp;
+
+  int framecnt;
+  bool endfile;
+  size_t framelength = 160;
+  G722EncInst* G722enc_inst;
+  G722DecInst* G722dec_inst;
+
+  /* Runtime statistics */
+  double starttime;
+  double runtime = 0;
+  double length_file;
+
+  size_t stream_len = 0;
+  int16_t shortdata[960];
+  int16_t decoded[960];
+  uint8_t streamdata[80 * 6];
+  int16_t speechType[1];
+
+  /* handling wrong input arguments in the command line */
+  if (argc != 5) {
+    printf("\n\nWrong number of arguments or flag values.\n\n");
+
+    printf("\n");
+    printf("Usage:\n\n");
+    printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
+    printf("with:\n");
+    printf("framelength  :    Framelength in samples.\n\n");
+    printf("infile       :    Normal speech input file\n\n");
+    printf("outbitfile   :    Bitstream output file\n\n");
+    printf("outspeechfile:    Speech output file\n\n");
+    exit(0);
+  }
+
+  /* Get frame length */
+  int framelength_int = atoi(argv[1]);
+  if (framelength_int < 0) {
+    printf("  G.722: Invalid framelength %d.\n", framelength_int);
+    exit(1);
+  }
+  framelength = static_cast<size_t>(framelength_int);
+
+  /* Get Input and Output files */
+  sscanf(argv[2], "%s", inname);
+  sscanf(argv[3], "%s", outbit);
+  sscanf(argv[4], "%s", outname);
+
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  G.722: Cannot read file %s.\n", inname);
+    exit(1);
+  }
+  if ((outbitp = fopen(outbit, "wb")) == NULL) {
+    printf("  G.722: Cannot write file %s.\n", outbit);
+    exit(1);
+  }
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  G.722: Cannot write file %s.\n", outname);
+    exit(1);
+  }
+  printf("\nInput:%s\nOutput bitstream:%s\nOutput:%s\n", inname, outbit,
+         outname);
+
+  /* Create and init */
+  WebRtcG722_CreateEncoder((G722EncInst**)&G722enc_inst);
+  WebRtcG722_CreateDecoder((G722DecInst**)&G722dec_inst);
+  WebRtcG722_EncoderInit((G722EncInst*)G722enc_inst);
+  WebRtcG722_DecoderInit((G722DecInst*)G722dec_inst);
+
+  /* Initialize encoder and decoder */
+  framecnt = 0;
+  endfile = false;
+  while (!endfile) {
+    framecnt++;
+
+    /* Read speech block */
+    endfile = readframe(shortdata, inp, framelength);
+
+    /* Start clock before call to encoder and decoder */
+    starttime = clock() / (double)CLOCKS_PER_SEC_G722;
+
+    /* G.722 encoding + decoding */
+    stream_len = WebRtcG722_Encode((G722EncInst*)G722enc_inst, shortdata,
+                                   framelength, streamdata);
+    WebRtcG722_Decode(G722dec_inst, streamdata, stream_len, decoded,
+                      speechType);
+
+    /* Stop clock after call to encoder and decoder */
+    runtime += (double)((clock() / (double)CLOCKS_PER_SEC_G722) - starttime);
+
+    /* Write coded bits to file */
+    if (fwrite(streamdata, sizeof(short), stream_len / 2, outbitp) !=
+        stream_len / 2) {
+      return -1;
+    }
+    /* Write coded speech to file */
+    if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) {
+      return -1;
+    }
+  }
+
+  WebRtcG722_FreeEncoder((G722EncInst*)G722enc_inst);
+  WebRtcG722_FreeDecoder((G722DecInst*)G722dec_inst);
+
+  length_file = ((double)framecnt * (double)framelength / 16000);
+  printf("\n\nLength of speech file: %.1f s\n", length_file);
+  printf("Time to run G.722:      %.2f s (%.2f %% of realtime)\n\n", runtime,
+         (100 * runtime / length_file));
+  printf("---------------------END----------------------\n");
+
+  fclose(inp);
+  fclose(outbitp);
+  fclose(outp);
+
+  return 0;
+}

modules/audio_coding/codecs/ilbc/abs_quant.c

@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/abs_quant_loop.h"
+
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+                                   and idxVec, uses state_first as
+                                   input) */
+    int16_t *in,     /* (i) vector to encode */
+    int16_t *weightDenum   /* (i) denominator of synthesis filter */
+                            ) {
+  int16_t *syntOut;
+  size_t quantLen[2];
+
+  /* Stack based */
+  int16_t syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
+  int16_t in_weightedVec[STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t *in_weighted = &in_weightedVec[LPC_FILTERORDER];
+
+  /* Initialize the buffers */
+  WebRtcSpl_MemSetW16(syntOutBuf, 0, LPC_FILTERORDER+STATE_SHORT_LEN_30MS);
+  syntOut = &syntOutBuf[LPC_FILTERORDER];
+  /* Start with zero state */
+  WebRtcSpl_MemSetW16(in_weightedVec, 0, LPC_FILTERORDER);
+
+  /* Perform the quantization loop in two sections of length quantLen[i],
+     where the perceptual weighting filter is updated at the subframe
+     border */
+
+  if (iLBC_encbits->state_first) {
+    quantLen[0]=SUBL;
+    quantLen[1]=iLBCenc_inst->state_short_len-SUBL;
+  } else {
+    quantLen[0]=iLBCenc_inst->state_short_len-SUBL;
+    quantLen[1]=SUBL;
+  }
+
+  /* Calculate the weighted residual, switch perceptual weighting
+     filter at the subframe border */
+  WebRtcSpl_FilterARFastQ12(
+      in, in_weighted,
+      weightDenum, LPC_FILTERORDER+1, quantLen[0]);
+  WebRtcSpl_FilterARFastQ12(
+      &in[quantLen[0]], &in_weighted[quantLen[0]],
+      &weightDenum[LPC_FILTERORDER+1], LPC_FILTERORDER+1, quantLen[1]);
+
+  WebRtcIlbcfix_AbsQuantLoop(
+      syntOut,
+      in_weighted,
+      weightDenum,
+      quantLen,
+      iLBC_encbits->idxVec);
+
+}

modules/audio_coding/codecs/ilbc/abs_quant.h

@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits* iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+                                   and idxVec, uses state_first as
+                                   input) */
+    int16_t* in,             /* (i) vector to encode */
+    int16_t* weightDenum     /* (i) denominator of synthesis filter */
+    );
+
+#endif

modules/audio_coding/codecs/ilbc/abs_quant_loop.c

@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/sort_sq.h"
+
+void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
+                                int16_t *weightDenumIN, size_t *quantLenIN,
+                                int16_t *idxVecIN ) {
+  size_t k1, k2;
+  int16_t index;
+  int32_t toQW32;
+  int32_t toQ32;
+  int16_t tmp16a;
+  int16_t xq;
+
+  int16_t *syntOut   = syntOutIN;
+  int16_t *in_weighted  = in_weightedIN;
+  int16_t *weightDenum  = weightDenumIN;
+  size_t *quantLen  = quantLenIN;
+  int16_t *idxVec   = idxVecIN;
+
+  for(k1=0;k1<2;k1++) {
+    for(k2=0;k2<quantLen[k1];k2++){
+
+      /* Filter to get the predicted value */
+      WebRtcSpl_FilterARFastQ12(
+          syntOut, syntOut,
+          weightDenum, LPC_FILTERORDER+1, 1);
+
+      /* the quantizer */
+      toQW32 = (int32_t)(*in_weighted) - (int32_t)(*syntOut);
+
+      toQ32 = (((int32_t)toQW32)<<2);
+
+      if (toQ32 > 32767) {
+        toQ32 = (int32_t) 32767;
+      } else if (toQ32 < -32768) {
+        toQ32 = (int32_t) -32768;
+      }
+
+      /* Quantize the state */
+      if (toQW32<(-7577)) {
+        /* To prevent negative overflow */
+        index=0;
+      } else if (toQW32>8151) {
+        /* To prevent positive overflow */
+        index=7;
+      } else {
+        /* Find the best quantization index
+           (state_sq3Tbl is in Q13 and toQ is in Q11)
+        */
+        WebRtcIlbcfix_SortSq(&xq, &index,
+                             (int16_t)toQ32,
+                             WebRtcIlbcfix_kStateSq3, 8);
+      }
+
+      /* Store selected index */
+      (*idxVec++) = index;
+
+      /* Compute decoded sample and update of the prediction filter */
+      tmp16a = ((WebRtcIlbcfix_kStateSq3[index] + 2 ) >> 2);
+
+      *syntOut     = (int16_t) (tmp16a + (int32_t)(*in_weighted) - toQW32);
+
+      syntOut++; in_weighted++;
+    }
+    /* Update perceptual weighting filter at subframe border */
+    weightDenum += 11;
+  }
+}

modules/audio_coding/codecs/ilbc/abs_quant_loop.h

@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuantLoop(int16_t* syntOutIN,
+                                int16_t* in_weightedIN,
+                                int16_t* weightDenumIN,
+                                size_t* quantLenIN,
+                                int16_t* idxVecIN);
+
+#endif

modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc

@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+AudioDecoderIlbcImpl::AudioDecoderIlbcImpl() {
+  WebRtcIlbcfix_DecoderCreate(&dec_state_);
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+AudioDecoderIlbcImpl::~AudioDecoderIlbcImpl() {
+  WebRtcIlbcfix_DecoderFree(dec_state_);
+}
+
+bool AudioDecoderIlbcImpl::HasDecodePlc() const {
+  return true;
+}
+
+int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+size_t AudioDecoderIlbcImpl::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
+}
+
+void AudioDecoderIlbcImpl::Reset() {
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+  size_t bytes_per_frame;
+  int timestamps_per_frame;
+  if (payload.size() >= 950) {
+    RTC_LOG(LS_WARNING)
+        << "AudioDecoderIlbcImpl::ParsePayload: Payload too large";
+    return results;
+  }
+  if (payload.size() % 38 == 0) {
+    // 20 ms frames.
+    bytes_per_frame = 38;
+    timestamps_per_frame = 160;
+  } else if (payload.size() % 50 == 0) {
+    // 30 ms frames.
+    bytes_per_frame = 50;
+    timestamps_per_frame = 240;
+  } else {
+    RTC_LOG(LS_WARNING)
+        << "AudioDecoderIlbcImpl::ParsePayload: Invalid payload";
+    return results;
+  }
+
+  RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame);
+  if (payload.size() == bytes_per_frame) {
+    std::unique_ptr<EncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(this, std::move(payload)));
+    results.emplace_back(timestamp, 0, std::move(frame));
+  } else {
+    size_t byte_offset;
+    uint32_t timestamp_offset;
+    for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
+         byte_offset += bytes_per_frame,
+        timestamp_offset += timestamps_per_frame) {
+      std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
+          this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
+      results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
+    }
+  }
+
+  return results;
+}
+
+int AudioDecoderIlbcImpl::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderIlbcImpl::Channels() const {
+  return 1;
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h

@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2015 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 MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/constructor_magic.h"
+
+typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
+
+namespace webrtc {
+
+class AudioDecoderIlbcImpl final : public AudioDecoder {
+ public:
+  AudioDecoderIlbcImpl();
+  ~AudioDecoderIlbcImpl() override;
+  bool HasDecodePlc() const override;
+  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  IlbcDecoderInstance* dec_state_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIlbcImpl);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_

modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc

@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+
+#include <algorithm>
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kSampleRateHz = 8000;
+
+int GetIlbcBitrate(int ptime) {
+  switch (ptime) {
+    case 20:
+    case 40:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 30:
+    case 60:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      FATAL();
+  }
+}
+
+}  // namespace
+
+AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
+                                           int payload_type)
+    : frame_size_ms_(config.frame_size_ms),
+      payload_type_(payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      encoder_(nullptr) {
+  RTC_CHECK(config.IsOk());
+  Reset();
+}
+
+AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+}
+
+int AudioEncoderIlbcImpl::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+size_t AudioEncoderIlbcImpl::NumChannels() const {
+  return 1;
+}
+
+size_t AudioEncoderIlbcImpl::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderIlbcImpl::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderIlbcImpl::GetTargetBitrate() const {
+  return GetIlbcBitrate(rtc::dchecked_cast<int>(num_10ms_frames_per_packet_) *
+                        10);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  // Save timestamp if starting a new packet.
+  if (num_10ms_frames_buffered_ == 0)
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  // Buffer input.
+  std::copy(audio.cbegin(), audio.cend(),
+            input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
+
+  // If we don't yet have enough buffered input for a whole packet, we're done
+  // for now.
+  if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+    return EncodedInfo();
+  }
+
+  // Encode buffered input.
+  RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  num_10ms_frames_buffered_ = 0;
+  size_t encoded_bytes = encoded->AppendData(
+      RequiredOutputSizeBytes(), [&](rtc::ArrayView<uint8_t> encoded) {
+        const int r = WebRtcIlbcfix_Encode(
+            encoder_, input_buffer_,
+            kSampleRateHz / 100 * num_10ms_frames_per_packet_, encoded.data());
+        RTC_CHECK_GE(r, 0);
+
+        return static_cast<size_t>(r);
+      });
+
+  RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
+
+  EncodedInfo info;
+  info.encoded_bytes = encoded_bytes;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kIlbc;
+  return info;
+}
+
+void AudioEncoderIlbcImpl::Reset() {
+  if (encoder_)
+    RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
+  const int encoder_frame_size_ms =
+      frame_size_ms_ > 30 ? frame_size_ms_ / 2 : frame_size_ms_;
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+  num_10ms_frames_buffered_ = 0;
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderIlbcImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
+  switch (num_10ms_frames_per_packet_) {
+    case 2:
+      return 38;
+    case 3:
+      return 50;
+    case 4:
+      return 2 * 38;
+    case 6:
+      return 2 * 50;
+    default:
+      FATAL();
+  }
+}
+
+}  // namespace webrtc

modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2014 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 MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class AudioEncoderIlbcImpl final : public AudioEncoder {
+ public:
+  AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
+  ~AudioEncoderIlbcImpl() override;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ private:
+  size_t RequiredOutputSizeBytes() const;
+
+  static constexpr size_t kMaxSamplesPerPacket = 480;
+  const int frame_size_ms_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
+  uint32_t first_timestamp_in_buffer_;
+  int16_t input_buffer_[kMaxSamplesPerPacket];
+  IlbcEncoderInstance* encoder_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbcImpl);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_

modules/audio_coding/codecs/ilbc/augmented_cb_corr.c

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/augmented_cb_corr.h"
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+    int16_t *target,   /* (i) Target vector */
+    int16_t *buffer,   /* (i) Memory buffer */
+    int16_t *interpSamples, /* (i) buffer with
+                                     interpolated samples */
+    int32_t *crossDot,  /* (o) The cross correlation between
+                                 the target and the Augmented
+                                 vector */
+    size_t low,    /* (i) Lag to start from (typically
+                             20) */
+    size_t high,   /* (i) Lag to end at (typically 39) */
+    int scale)   /* (i) Scale factor to use for
+                              the crossDot */
+{
+  size_t lagcount;
+  size_t ilow;
+  int16_t *targetPtr;
+  int32_t *crossDotPtr;
+  int16_t *iSPtr=interpSamples;
+
+  /* Calculate the correlation between the target and the
+     interpolated codebook. The correlation is calculated in
+     3 sections with the interpolated part in the middle */
+  crossDotPtr=crossDot;
+  for (lagcount=low; lagcount<=high; lagcount++) {
+
+    ilow = lagcount - 4;
+
+    /* Compute dot product for the first (lagcount-4) samples */
+    (*crossDotPtr) = WebRtcSpl_DotProductWithScale(target, buffer-lagcount, ilow, scale);
+
+    /* Compute dot product on the interpolated samples */
+    (*crossDotPtr) += WebRtcSpl_DotProductWithScale(target+ilow, iSPtr, 4, scale);
+    targetPtr = target + lagcount;
+    iSPtr += lagcount-ilow;
+
+    /* Compute dot product for the remaining samples */
+    (*crossDotPtr) += WebRtcSpl_DotProductWithScale(targetPtr, buffer-lagcount, SUBL-lagcount, scale);
+    crossDotPtr++;
+  }
+}

modules/audio_coding/codecs/ilbc/augmented_cb_corr.h

@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Calculate correlation between target and Augmented codebooks
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+    int16_t* target,        /* (i) Target vector */
+    int16_t* buffer,        /* (i) Memory buffer */
+    int16_t* interpSamples, /* (i) buffer with
+                                           interpolated samples */
+    int32_t* crossDot,      /* (o) The cross correlation between
+                                               the target and the Augmented
+                                               vector */
+    size_t low,             /* (i) Lag to start from (typically
+                                                            20) */
+    size_t high,            /* (i) Lag to end at (typically 39 */
+    int scale);             /* (i) Scale factor to use for the crossDot */
+
+#endif

modules/audio_coding/codecs/ilbc/bw_expand.c

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+/* The output is in the same domain as the input */
+void WebRtcIlbcfix_BwExpand(
+    int16_t *out, /* (o) the bandwidth expanded lpc coefficients */
+    int16_t *in,  /* (i) the lpc coefficients before bandwidth
+                                   expansion */
+    int16_t *coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
+                            ) {
+  int i;
+
+  out[0] = in[0];
+  for (i = 1; i < length; i++) {
+    /* out[i] = coef[i] * in[i] with rounding.
+       in[] and out[] are in Q12 and coef[] is in Q15
+    */
+    out[i] = (int16_t)((coef[i] * in[i] + 16384) >> 15);
+  }
+}

modules/audio_coding/codecs/ilbc/bw_expand.h

@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_BwExpand(
+    int16_t* out,  /* (o) the bandwidth expanded lpc coefficients */
+    int16_t* in,   /* (i) the lpc coefficients before bandwidth
+                                    expansion */
+    int16_t* coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
+    );
+
+#endif

modules/audio_coding/codecs/ilbc/cb_construct.c

@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/gain_dequant.h"
+#include "modules/audio_coding/codecs/ilbc/get_cd_vec.h"
+#include "rtc_base/sanitizer.h"
+
+// An arithmetic operation that is allowed to overflow. (It's still undefined
+// behavior, so not a good idea; this just makes UBSan ignore the violation, so
+// that our old code can continue to do what it's always been doing.)
+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
+    OverflowingAddS32S32ToS32(int32_t a, int32_t b) {
+  return a + b;
+}
+
+/*----------------------------------------------------------------*
+ *  Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_CbConstruct(
+    int16_t* decvector,        /* (o) Decoded vector */
+    const int16_t* index,      /* (i) Codebook indices */
+    const int16_t* gain_index, /* (i) Gain quantization indices */
+    int16_t* mem,              /* (i) Buffer for codevector construction */
+    size_t lMem,               /* (i) Length of buffer */
+    size_t veclen) {           /* (i) Length of vector */
+  size_t j;
+  int16_t gain[CB_NSTAGES];
+  /* Stack based */
+  int16_t cbvec0[SUBL];
+  int16_t cbvec1[SUBL];
+  int16_t cbvec2[SUBL];
+  int32_t a32;
+  int16_t *gainPtr;
+
+  /* gain de-quantization */
+
+  gain[0] = WebRtcIlbcfix_GainDequant(gain_index[0], 16384, 0);
+  gain[1] = WebRtcIlbcfix_GainDequant(gain_index[1], gain[0], 1);
+  gain[2] = WebRtcIlbcfix_GainDequant(gain_index[2], gain[1], 2);
+
+  /* codebook vector construction and construction of total vector */
+
+  /* Stack based */
+  if (!WebRtcIlbcfix_GetCbVec(cbvec0, mem, (size_t)index[0], lMem, veclen))
+    return false;  // Failure.
+  if (!WebRtcIlbcfix_GetCbVec(cbvec1, mem, (size_t)index[1], lMem, veclen))
+    return false;  // Failure.
+  if (!WebRtcIlbcfix_GetCbVec(cbvec2, mem, (size_t)index[2], lMem, veclen))
+    return false;  // Failure.
+
+  gainPtr = &gain[0];
+  for (j=0;j<veclen;j++) {
+    a32 = (*gainPtr++) * cbvec0[j];
+    a32 += (*gainPtr++) * cbvec1[j];
+    a32 = OverflowingAddS32S32ToS32(a32, (*gainPtr) * cbvec2[j]);
+    gainPtr -= 2;
+    decvector[j] = (int16_t)((a32 + 8192) >> 14);
+  }
+
+  return true;  // Success.
+}

modules/audio_coding/codecs/ilbc/cb_construct.h

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+
+#include <stdbool.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "rtc_base/system/unused.h"
+
+/*----------------------------------------------------------------*
+ *  Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+// Returns true on success, false on failure.
+bool WebRtcIlbcfix_CbConstruct(
+    int16_t* decvector,        /* (o) Decoded vector */
+    const int16_t* index,      /* (i) Codebook indices */
+    const int16_t* gain_index, /* (i) Gain quantization indices */
+    int16_t* mem,              /* (i) Buffer for codevector construction */
+    size_t lMem,               /* (i) Length of buffer */
+    size_t veclen              /* (i) Length of vector */
+    ) RTC_WARN_UNUSED_RESULT;
+
+#endif

modules/audio_coding/codecs/ilbc/cb_mem_energy.c

@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergy.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h"
+
+/*----------------------------------------------------------------*
+ *  Function WebRtcIlbcfix_CbMemEnergy computes the energy of all
+ * the vectors in the codebook memory that will be used in the
+ * following search for the best match.
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbMemEnergy(
+    size_t range,
+    int16_t *CB,   /* (i) The CB memory (1:st section) */
+    int16_t *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
+    size_t lMem,   /* (i) Length of the CB memory */
+    size_t lTarget,   /* (i) Length of the target vector */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
+                               ) {
+  int16_t *ppi, *ppo, *pp;
+  int32_t energy, tmp32;
+
+  /* Compute the energy and store it in a vector. Also the
+   * corresponding shift values are stored. The energy values
+   * are reused in all three stages. */
+
+  /* Calculate the energy in the first block of 'lTarget' sampels. */
+  ppi = CB+lMem-lTarget-1;
+  ppo = CB+lMem-1;
+
+  pp=CB+lMem-lTarget;
+  energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+  /* Normalize the energy and store the number of shifts */
+  energyShifts[0] = (int16_t)WebRtcSpl_NormW32(energy);
+  tmp32 = energy << energyShifts[0];
+  energyW16[0] = (int16_t)(tmp32 >> 16);
+
+  /* Compute the energy of the rest of the cb memory
+   * by step wise adding and subtracting the next
+   * sample and the last sample respectively. */
+  WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, 0);
+
+  /* Next, precompute the energy values for the filtered cb section */
+  energy=0;
+  pp=filteredCB+lMem-lTarget;
+
+  energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+  /* Normalize the energy and store the number of shifts */
+  energyShifts[base_size] = (int16_t)WebRtcSpl_NormW32(energy);
+  tmp32 = energy << energyShifts[base_size];
+  energyW16[base_size] = (int16_t)(tmp32 >> 16);
+
+  ppi = filteredCB + lMem - 1 - lTarget;
+  ppo = filteredCB + lMem - 1;
+
+  WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, base_size);
+}