diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 082506ab56..6cfe464a32 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -110,6 +110,7 @@ AudioCodingModule::Config MakeAcmConfig(
     Clock* clock,
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
   AudioCodingModule::Config config;
+  config.id = 0;
   config.clock = clock;
   config.decoder_factory = std::move(decoder_factory);
   return config;
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index 307c9066b1..d5f196b9a2 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/modules/audio_coding/acm2/acm_send_test.cc
@@ -28,7 +28,7 @@ AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
                                      int source_rate_hz,
                                      int test_duration_ms)
     : clock_(0),
-      acm_(webrtc::AudioCodingModule::Create(&clock_)),
+      acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
       audio_source_(audio_source),
       source_rate_hz_(source_rate_hz),
       input_block_size_samples_(
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index c48fbef329..5997d1269e 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -269,6 +269,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
 
   rtc::CriticalSection acm_crit_sect_;
   rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_crit_sect_);
+  int id_;  // TODO(henrik.lundin) Make const.
   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_);
@@ -455,7 +456,8 @@ void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
 
 AudioCodingModuleImpl::AudioCodingModuleImpl(
     const AudioCodingModule::Config& config)
-    : expected_codec_ts_(0xD87F3F9F),
+    : id_(config.id),
+      expected_codec_ts_(0xD87F3F9F),
       expected_in_ts_(0xD87F3F9F),
       receiver_(config),
       bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
@@ -1118,6 +1120,7 @@ int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
     LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
     return -1;
   }
+  audio_frame->id_ = id_;
   return 0;
 }
 
@@ -1283,7 +1286,7 @@ ANAStats AudioCodingModuleImpl::GetANAStats() const {
 }  // namespace
 
 AudioCodingModule::Config::Config()
-    : neteq_config(), clock(Clock::GetRealTimeClock()) {
+    : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {
   // 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;
@@ -1293,15 +1296,17 @@ AudioCodingModule::Config::Config(const Config&) = default;
 AudioCodingModule::Config::~Config() = default;
 
 // Create module
-AudioCodingModule* AudioCodingModule::Create() {
+AudioCodingModule* AudioCodingModule::Create(int id) {
   Config config;
+  config.id = id;
   config.clock = Clock::GetRealTimeClock();
   config.decoder_factory = CreateBuiltinAudioDecoderFactory();
   return Create(config);
 }
 
-AudioCodingModule* AudioCodingModule::Create(Clock* clock) {
+AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
   Config config;
+  config.id = id;
   config.clock = clock;
   config.decoder_factory = CreateBuiltinAudioDecoderFactory();
   return Create(config);
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index a010619b7e..80fc4d80b7 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -157,7 +157,8 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
 class AudioCodingModuleTestOldApi : public ::testing::Test {
  protected:
   AudioCodingModuleTestOldApi()
-      : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
+      : id_(1),
+        rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
         clock_(Clock::GetRealTimeClock()) {}
 
   ~AudioCodingModuleTestOldApi() {}
@@ -165,7 +166,7 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
   void TearDown() {}
 
   void SetUp() {
-    acm_.reset(AudioCodingModule::Create(clock_));
+    acm_.reset(AudioCodingModule::Create(id_, clock_));
 
     rtp_utility_->Populate(&rtp_header_);
 
@@ -229,6 +230,7 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
     VerifyEncoding();
   }
 
+  const int id_;
   std::unique_ptr<RtpUtility> rtp_utility_;
   std::unique_ptr<AudioCodingModule> acm_;
   PacketizationCallbackStubOldApi packet_cb_;
@@ -312,6 +314,7 @@ TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
   bool muted;
   EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
   ASSERT_FALSE(muted);
+  EXPECT_EQ(id_, audio_frame.id_);
   EXPECT_EQ(0u, audio_frame.timestamp_);
   EXPECT_GT(audio_frame.num_channels_, 0u);
   EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 63af3ab298..944ad604f5 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -70,6 +70,7 @@ class AudioCodingModule {
     Config(const Config&);
     ~Config();
 
+    int id;
     NetEq::Config neteq_config;
     Clock* clock;
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
@@ -82,8 +83,8 @@ class AudioCodingModule {
   // injected into ACM. ACM will take the ownership of the object clock and
   // delete it when destroyed.
   //
-  static AudioCodingModule* Create();
-  static AudioCodingModule* Create(Clock* clock);
+  static AudioCodingModule* Create(int id);
+  static AudioCodingModule* Create(int id, Clock* clock);
   static AudioCodingModule* Create(const Config& config);
   virtual ~AudioCodingModule() = default;
 
diff --git a/modules/audio_coding/test/APITest.cc b/modules/audio_coding/test/APITest.cc
index b29e84e106..5418342714 100644
--- a/modules/audio_coding/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -48,8 +48,8 @@ void APITest::Wait(uint32_t waitLengthMs) {
 }
 
 APITest::APITest()
-    : _acmA(AudioCodingModule::Create()),
-      _acmB(AudioCodingModule::Create()),
+    : _acmA(AudioCodingModule::Create(1)),
+      _acmB(AudioCodingModule::Create(2)),
       _channel_A2B(NULL),
       _channel_B2A(NULL),
       _writeToFile(true),
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 2b6b4acfd0..8257ed9c10 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -281,7 +281,7 @@ void EncodeDecodeTest::Perform() {
   codePars[1] = 0;
   codePars[2] = 0;
 
-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
   struct CodecInst sendCodecTmp;
   numCodecs = acm->NumberOfCodecs();
 
@@ -337,7 +337,7 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
                                            int codeId,
                                            int* codePars,
                                            int testMode) {
-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
   RTPFile rtpFile;
   std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
                                                     "encode_decode_rtp");
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index a6c56fa780..c80615a36d 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -127,7 +127,7 @@ void PacketLossTest::Perform() {
 #ifndef WEBRTC_CODEC_OPUS
   return;
 #else
-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
 
   int codec_id = acm->Codec("opus", 48000, channels_);
 
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index ff28a28807..74319c2dc5 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -104,8 +104,8 @@ void TestPack::reset_payload_size() {
 }
 
 TestAllCodecs::TestAllCodecs(int test_mode)
-    : acm_a_(AudioCodingModule::Create()),
-      acm_b_(AudioCodingModule::Create()),
+    : acm_a_(AudioCodingModule::Create(0)),
+      acm_b_(AudioCodingModule::Create(1)),
       channel_a_to_b_(NULL),
       test_count_(0),
       packet_size_samples_(0),
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index 58561c6fe1..3e88290a1c 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -48,8 +48,8 @@ namespace {
 }
 
 TestRedFec::TestRedFec()
-    : _acmA(AudioCodingModule::Create()),
-      _acmB(AudioCodingModule::Create()),
+    : _acmA(AudioCodingModule::Create(0)),
+      _acmB(AudioCodingModule::Create(1)),
       _channelA2B(NULL),
       _testCntr(0) {
 }
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index eca81f81cf..d598191f81 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -108,8 +108,8 @@ void TestPackStereo::set_lost_packet(bool lost) {
 }
 
 TestStereo::TestStereo(int test_mode)
-    : acm_a_(AudioCodingModule::Create()),
-      acm_b_(AudioCodingModule::Create()),
+    : acm_a_(AudioCodingModule::Create(0)),
+      acm_b_(AudioCodingModule::Create(1)),
       channel_a2b_(NULL),
       test_cntr_(0),
       pack_size_samp_(0),
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index 628582d55c..1aa00b5524 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -62,8 +62,8 @@ void ActivityMonitor::GetStatistics(uint32_t* counter) {
 }
 
 TestVadDtx::TestVadDtx()
-    : acm_send_(AudioCodingModule::Create()),
-      acm_receive_(AudioCodingModule::Create()),
+    : acm_send_(AudioCodingModule::Create(0)),
+      acm_receive_(AudioCodingModule::Create(1)),
       channel_(new Channel),
       monitor_(new ActivityMonitor) {
   EXPECT_EQ(0, acm_send_->RegisterTransportCallback(channel_.get()));
diff --git a/modules/audio_coding/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
index 8049436cd0..addb71764e 100644
--- a/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/test/TwoWayCommunication.cc
@@ -34,14 +34,16 @@ namespace webrtc {
 #define MAX_FILE_NAME_LENGTH_BYTE 500
 
 TwoWayCommunication::TwoWayCommunication(int testMode)
-    : _acmA(AudioCodingModule::Create()),
-      _acmRefA(AudioCodingModule::Create()),
+    : _acmA(AudioCodingModule::Create(1)),
+      _acmRefA(AudioCodingModule::Create(3)),
       _testMode(testMode) {
   AudioCodingModule::Config config;
   // The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
   config.neteq_config.playout_mode = kPlayoutFax;
+  config.id = 2;
   config.decoder_factory = CreateBuiltinAudioDecoderFactory();
   _acmB.reset(AudioCodingModule::Create(config));
+  config.id = 4;
   _acmRefB.reset(AudioCodingModule::Create(config));
 }
 
@@ -60,7 +62,7 @@ TwoWayCommunication::~TwoWayCommunication() {
 
 void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
                                       uint8_t* codecID_B) {
-  std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create());
+  std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
   uint8_t noCodec = tmpACM->NumberOfCodecs();
   CodecInst codecInst;
   printf("List of Supported Codecs\n");
diff --git a/modules/audio_coding/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc
index 407f7095cb..3f78ea68e3 100644
--- a/modules/audio_coding/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -64,8 +64,8 @@ struct TestSettings {
 class DelayTest {
  public:
   DelayTest()
-      : acm_a_(AudioCodingModule::Create()),
-        acm_b_(AudioCodingModule::Create()),
+      : acm_a_(AudioCodingModule::Create(0)),
+        acm_b_(AudioCodingModule::Create(1)),
         channel_a2b_(new Channel),
         test_cntr_(0),
         encoding_sample_rate_hz_(8000) {}
diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
index a44259fbbe..a14f795b9b 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -67,8 +67,8 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
 }
 
 ISACTest::ISACTest(int testMode)
-    : _acmA(AudioCodingModule::Create()),
-      _acmB(AudioCodingModule::Create()),
+    : _acmA(AudioCodingModule::Create(1)),
+      _acmB(AudioCodingModule::Create(2)),
       _testMode(testMode) {}
 
 ISACTest::~ISACTest() {}
diff --git a/modules/audio_coding/test/insert_packet_with_timing.cc b/modules/audio_coding/test/insert_packet_with_timing.cc
index 2c0e54b0cb..500375c8da 100644
--- a/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -61,8 +61,8 @@ class InsertPacketWithTiming {
   InsertPacketWithTiming()
       : sender_clock_(new SimulatedClock(0)),
         receiver_clock_(new SimulatedClock(0)),
-        send_acm_(AudioCodingModule::Create(sender_clock_)),
-        receive_acm_(AudioCodingModule::Create(receiver_clock_)),
+        send_acm_(AudioCodingModule::Create(0, sender_clock_)),
+        receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
         channel_(new Channel),
         seq_num_fid_(fopen(FLAG_seq_num, "rt")),
         send_ts_fid_(fopen(FLAG_send_ts, "rt")),
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index b7acc0f7c6..7b54668bae 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -27,7 +27,7 @@
 namespace webrtc {
 
 OpusTest::OpusTest()
-    : acm_receiver_(AudioCodingModule::Create()),
+    : acm_receiver_(AudioCodingModule::Create(0)),
       channel_a2b_(NULL),
       counter_(0),
       payload_type_(255),
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 03135da6c4..2a757069e8 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -22,7 +22,7 @@ namespace webrtc {
 
 class TargetDelayTest : public ::testing::Test {
  protected:
-  TargetDelayTest() : acm_(AudioCodingModule::Create()) {}
+  TargetDelayTest() : acm_(AudioCodingModule::Create(0)) {}
 
   ~TargetDelayTest() {}
 
diff --git a/modules/audio_mixer/audio_mixer_impl_unittest.cc b/modules/audio_mixer/audio_mixer_impl_unittest.cc
index 4461e6e038..678f625eac 100644
--- a/modules/audio_mixer/audio_mixer_impl_unittest.cc
+++ b/modules/audio_mixer/audio_mixer_impl_unittest.cc
@@ -35,10 +35,12 @@ namespace webrtc {
 namespace {
 
 constexpr int kDefaultSampleRateHz = 48000;
+constexpr int kId = 1;
 
 // Utility function that resets the frame member variables with
 // sensible defaults.
 void ResetFrame(AudioFrame* frame) {
+  frame->id_ = kId;
   frame->sample_rate_hz_ = kDefaultSampleRateHz;
   frame->num_channels_ = 1;
 
diff --git a/modules/audio_mixer/frame_combiner.cc b/modules/audio_mixer/frame_combiner.cc
index 8000904529..9e9bcfafb0 100644
--- a/modules/audio_mixer/frame_combiner.cc
+++ b/modules/audio_mixer/frame_combiner.cc
@@ -193,7 +193,7 @@ void FrameCombiner::Combine(const std::vector<AudioFrame*>& mix_list,
   // value '0', because it is only supported in the one channel case and
   // is then updated in the helper functions.
   audio_frame_for_mixing->UpdateFrame(
-      0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined,
+      -1, 0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined,
       AudioFrame::kVadUnknown, number_of_channels);
 
   const bool use_limiter_this_round = use_apm_limiter_ && number_of_streams > 1;
diff --git a/modules/audio_mixer/frame_combiner_unittest.cc b/modules/audio_mixer/frame_combiner_unittest.cc
index 490e99e495..568143608c 100644
--- a/modules/audio_mixer/frame_combiner_unittest.cc
+++ b/modules/audio_mixer/frame_combiner_unittest.cc
@@ -53,7 +53,8 @@ AudioFrame audio_frame_for_mixing;
 
 void SetUpFrames(int sample_rate_hz, int number_of_channels) {
   for (auto* frame : {&frame1, &frame2}) {
-    frame->UpdateFrame(0, nullptr, rtc::CheckedDivExact(sample_rate_hz, 100),
+    frame->UpdateFrame(-1, 0, nullptr,
+                       rtc::CheckedDivExact(sample_rate_hz, 100),
                        sample_rate_hz, AudioFrame::kNormalSpeech,
                        AudioFrame::kVadActive, number_of_channels);
   }
diff --git a/modules/include/module_common_types.h b/modules/include/module_common_types.h
index c3ad993555..bc5c347a4a 100644
--- a/modules/include/module_common_types.h
+++ b/modules/include/module_common_types.h
@@ -330,18 +330,7 @@ class AudioFrame {
   // ResetWithoutMuting() to skip this wasteful zeroing.
   void ResetWithoutMuting();
 
-  // TODO(solenberg): Remove once downstream users of AudioFrame have updated.
-  RTC_DEPRECATED
-      void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
-                       size_t samples_per_channel, int sample_rate_hz,
-                       SpeechType speech_type, VADActivity vad_activity,
-                       size_t num_channels = 1) {
-    RTC_UNUSED(id);
-    UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
-                speech_type, vad_activity, num_channels);
-  }
-
-  void UpdateFrame(uint32_t timestamp, const int16_t* data,
+  void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
                    size_t samples_per_channel, int sample_rate_hz,
                    SpeechType speech_type, VADActivity vad_activity,
                    size_t num_channels = 1);
@@ -377,6 +366,7 @@ class AudioFrame {
   RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
   RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);
 
+  int id_;
   // RTP timestamp of the first sample in the AudioFrame.
   uint32_t timestamp_ = 0;
   // Time since the first frame in milliseconds.
@@ -424,6 +414,7 @@ inline void AudioFrame::Reset() {
 }
 
 inline void AudioFrame::ResetWithoutMuting() {
+  id_ = -1;
   // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
   // to an invalid value, or add a new member to indicate invalidity.
   timestamp_ = 0;
@@ -437,13 +428,15 @@ inline void AudioFrame::ResetWithoutMuting() {
   profile_timestamp_ms_ = 0;
 }
 
-inline void AudioFrame::UpdateFrame(uint32_t timestamp,
+inline void AudioFrame::UpdateFrame(int id,
+                                    uint32_t timestamp,
                                     const int16_t* data,
                                     size_t samples_per_channel,
                                     int sample_rate_hz,
                                     SpeechType speech_type,
                                     VADActivity vad_activity,
                                     size_t num_channels) {
+  id_ = id;
   timestamp_ = timestamp;
   samples_per_channel_ = samples_per_channel;
   sample_rate_hz_ = sample_rate_hz;
@@ -464,6 +457,7 @@ inline void AudioFrame::UpdateFrame(uint32_t timestamp,
 inline void AudioFrame::CopyFrom(const AudioFrame& src) {
   if (this == &src) return;
 
+  id_ = src.id_;
   timestamp_ = src.timestamp_;
   elapsed_time_ms_ = src.elapsed_time_ms_;
   ntp_time_ms_ = src.ntp_time_ms_;
diff --git a/modules/module_common_types_unittest.cc b/modules/module_common_types_unittest.cc
index c8bb5f9850..f601a59051 100644
--- a/modules/module_common_types_unittest.cc
+++ b/modules/module_common_types_unittest.cc
@@ -28,6 +28,7 @@ bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
   return true;
 }
 
+constexpr int kId = 16;
 constexpr uint32_t kTimestamp = 27;
 constexpr int kSampleRateHz = 16000;
 constexpr size_t kNumChannels = 1;
@@ -63,9 +64,10 @@ TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
 TEST(AudioFrameTest, UpdateFrame) {
   AudioFrame frame;
   int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
-  frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
+  frame.UpdateFrame(kId, kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
                     AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
 
+  EXPECT_EQ(kId, frame.id_);
   EXPECT_EQ(kTimestamp, frame.timestamp_);
   EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel_);
   EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz_);
@@ -76,7 +78,7 @@ TEST(AudioFrameTest, UpdateFrame) {
   EXPECT_FALSE(frame.muted());
   EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples)));
 
-  frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel,
+  frame.UpdateFrame(kId, kTimestamp, nullptr /* data*/, kSamplesPerChannel,
                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
                     kNumChannels);
   EXPECT_TRUE(frame.muted());
@@ -88,11 +90,12 @@ TEST(AudioFrameTest, CopyFrom) {
   AudioFrame frame2;
 
   int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
-  frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel,
+  frame2.UpdateFrame(kId, kTimestamp, samples, kSamplesPerChannel,
                      kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
                      kNumChannels);
   frame1.CopyFrom(frame2);
 
+  EXPECT_EQ(frame2.id_, frame1.id_);
   EXPECT_EQ(frame2.timestamp_, frame1.timestamp_);
   EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_);
   EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_);
@@ -103,7 +106,7 @@ TEST(AudioFrameTest, CopyFrom) {
   EXPECT_EQ(frame2.muted(), frame1.muted());
   EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
 
-  frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
+  frame2.UpdateFrame(kId, kTimestamp, nullptr /* data */, kSamplesPerChannel,
                      kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
                      kNumChannels);
   frame1.CopyFrom(frame2);
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 875454a17d..4dff3de585 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -646,6 +646,8 @@ AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
     AudioFrameOperations::Mute(audio_frame);
   }
 
+  // Convert module ID to internal VoE channel ID
+  audio_frame->id_ = VoEChannelId(audio_frame->id_);
   // Store speech type for dead-or-alive detection
   _outputSpeechType = audio_frame->speech_type_;
 
@@ -794,6 +796,7 @@ Channel::Channel(int32_t channelId,
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
   AudioCodingModule::Config acm_config(config.acm_config);
+  acm_config.id = VoEModuleId(instanceId, channelId);
   acm_config.neteq_config.enable_muted_state = true;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
@@ -1639,6 +1642,7 @@ void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) {
   // TODO(henrika): try to avoid copying by moving ownership of audio frame
   // either into pool of frames or into the task itself.
   audio_frame->CopyFrom(audio_input);
+  audio_frame->id_ = ChannelId();
   encoder_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
       new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
 }
@@ -1655,6 +1659,7 @@ void Channel::ProcessAndEncodeAudio(const int16_t* audio_data,
   CodecInst codec;
   const int result = GetSendCodec(codec);
   std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
+  audio_frame->id_ = ChannelId();
   // TODO(ossu): Investigate how this could happen. b/62909493
   if (result == 0) {
     audio_frame->sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
@@ -1675,6 +1680,7 @@ void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
   RTC_DCHECK_RUN_ON(encoder_queue_);
   RTC_DCHECK_GT(audio_input->samples_per_channel_, 0);
   RTC_DCHECK_LE(audio_input->num_channels_, 2);
+  RTC_DCHECK_EQ(audio_input->id_, ChannelId());
 
   bool is_muted = InputMute();
   AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);
diff --git a/voice_engine/voice_engine_defines.h b/voice_engine/voice_engine_defines.h
index 4397662057..bc5eb1b81f 100644
--- a/voice_engine/voice_engine_defines.h
+++ b/voice_engine/voice_engine_defines.h
@@ -16,13 +16,23 @@
 #ifndef VOICE_ENGINE_VOICE_ENGINE_DEFINES_H_
 #define VOICE_ENGINE_VOICE_ENGINE_DEFINES_H_
 
+#include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_processing/include/audio_processing.h"
+#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 
 // VolumeControl
 enum { kMinVolumeLevel = 0 };
 enum { kMaxVolumeLevel = 255 };
+// Min scale factor for per-channel volume scaling
+const float kMinOutputVolumeScaling = 0.0f;
+// Max scale factor for per-channel volume scaling
+const float kMaxOutputVolumeScaling = 10.0f;
+// Min scale factor for output volume panning
+const float kMinOutputVolumePanning = 0.0f;
+// Max scale factor for output volume panning
+const float kMaxOutputVolumePanning = 1.0f;
 
 // Audio processing
 const NoiseSuppression::Level kDefaultNsMode = NoiseSuppression::kModerate;
@@ -38,6 +48,7 @@ const bool kDefaultAgcState =
 #else
     true;
 #endif
+const GainControl::Mode kDefaultRxAgcMode = GainControl::kAdaptiveDigital;
 
 // VideoSync
 // Lowest minimum playout delay
@@ -57,6 +68,15 @@ inline int VoEId(int veId, int chId) {
   return (int)((veId << 16) + chId);
 }
 
+inline int VoEModuleId(int veId, int chId) {
+  return (int)((veId << 16) + chId);
+}
+
+// Convert module ID to internal VoE channel ID
+inline int VoEChannelId(int moduleId) {
+  return (int)(moduleId & 0xffff);
+}
+
 }  // namespace webrtc
 
 #if defined(_WIN32)