diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index aaf77a8a97..1f24b2ca27 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -21,6 +21,8 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
@@ -81,6 +83,8 @@ AudioReceiveStream::AudioReceiveStream(
   RTC_DCHECK(remote_bitrate_estimator);
   RTC_DCHECK(rtp_header_parser_);
 
+  module_process_thread_checker_.DetachFromThread();
+
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetRtcEventLog(event_log);
@@ -125,7 +129,7 @@ AudioReceiveStream::AudioReceiveStream(
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
   if (playing_) {
     Stop();
@@ -138,7 +142,7 @@ AudioReceiveStream::~AudioReceiveStream() {
 }
 
 void AudioReceiveStream::Start() {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   if (playing_) {
     return;
   }
@@ -159,7 +163,7 @@ void AudioReceiveStream::Start() {
 }
 
 void AudioReceiveStream::Stop() {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   if (!playing_) {
     return;
   }
@@ -170,7 +174,7 @@ void AudioReceiveStream::Stop() {
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   webrtc::AudioReceiveStream::Stats stats;
   stats.remote_ssrc = config_.rtp.remote_ssrc;
   ScopedVoEInterface<VoECodec> codec(voice_engine());
@@ -220,22 +224,78 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
 }
 
 void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   channel_proxy_->SetSink(std::move(sink));
 }
 
 void AudioReceiveStream::SetGain(float gain) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   channel_proxy_->SetChannelOutputVolumeScaling(gain);
 }
 
-const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  return config_;
+AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
+    int sample_rate_hz,
+    AudioFrame* audio_frame) {
+  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+}
+
+int AudioReceiveStream::Ssrc() const {
+  return config_.rtp.remote_ssrc;
+}
+
+int AudioReceiveStream::PreferredSampleRate() const {
+  return channel_proxy_->NeededFrequency();
+}
+
+int AudioReceiveStream::id() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return config_.rtp.remote_ssrc;
+}
+
+rtc::Optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
+  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+  Syncable::Info info;
+
+  RtpRtcp* rtp_rtcp = nullptr;
+  RtpReceiver* rtp_receiver = nullptr;
+  channel_proxy_->GetRtpRtcp(&rtp_rtcp, &rtp_receiver);
+  RTC_DCHECK(rtp_rtcp);
+  RTC_DCHECK(rtp_receiver);
+
+  if (!rtp_receiver->Timestamp(&info.latest_received_capture_timestamp)) {
+    return rtc::Optional<Syncable::Info>();
+  }
+  if (!rtp_receiver->LastReceivedTimeMs(&info.latest_receive_time_ms)) {
+    return rtc::Optional<Syncable::Info>();
+  }
+  if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs,
+                          &info.capture_time_ntp_frac,
+                          nullptr,
+                          nullptr,
+                          &info.capture_time_source_clock) != 0) {
+    return rtc::Optional<Syncable::Info>();
+  }
+
+  int jitter_buffer_delay_ms = 0;
+  int playout_buffer_delay_ms = 0;
+  channel_proxy_->GetDelayEstimate(&jitter_buffer_delay_ms,
+                                   &playout_buffer_delay_ms);
+  info.current_delay_ms = jitter_buffer_delay_ms + playout_buffer_delay_ms;
+  return rtc::Optional<Syncable::Info>(info);
+}
+
+uint32_t AudioReceiveStream::GetPlayoutTimestamp() const {
+  // Called on video capture thread.
+  return channel_proxy_->GetPlayoutTimestamp();
+}
+
+void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
+  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+  return channel_proxy_->SetMinimumPlayoutDelay(delay_ms);
 }
 
 void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   if (send_stream) {
     VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
     std::unique_ptr<voe::ChannelProxy> send_channel_proxy =
@@ -247,7 +307,7 @@ void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
 }
 
 void AudioReceiveStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
 }
 
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
@@ -286,24 +346,9 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
-AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
-    int sample_rate_hz,
-    AudioFrame* audio_frame) {
-  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
-}
-
-int AudioReceiveStream::PreferredSampleRate() const {
-  return channel_proxy_->NeededFrequency();
-}
-
-int AudioReceiveStream::Ssrc() const {
-  return config_.rtp.remote_ssrc;
-}
-
-internal::AudioState* AudioReceiveStream::audio_state() const {
-  auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
-  RTC_DCHECK(audio_state);
-  return audio_state;
+const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return config_;
 }
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
@@ -312,6 +357,12 @@ VoiceEngine* AudioReceiveStream::voice_engine() const {
   return voice_engine;
 }
 
+internal::AudioState* AudioReceiveStream::audio_state() const {
+  auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
+  RTC_DCHECK(audio_state);
+  return audio_state;
+}
+
 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
@@ -320,6 +371,5 @@ int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
-
 }  // namespace internal
 }  // namespace webrtc
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 7dfc5d6bb8..6721c7ee65 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -18,13 +18,13 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_receive_stream.h"
-#include "webrtc/call/audio_state.h"
+#include "webrtc/call/syncable.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 
 namespace webrtc {
+class PacketRouter;
 class RemoteBitrateEstimator;
 class RtcEventLog;
-class PacketRouter;
 
 namespace voe {
 class ChannelProxy;
@@ -34,7 +34,8 @@ namespace internal {
 class AudioSendStream;
 
 class AudioReceiveStream final : public webrtc::AudioReceiveStream,
-                                 public AudioMixer::Source {
+                                 public AudioMixer::Source,
+                                 public Syncable {
  public:
   AudioReceiveStream(PacketRouter* packet_router,
                      RemoteBitrateEstimator* remote_bitrate_estimator,
@@ -50,6 +51,18 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
   void SetGain(float gain) override;
 
+  // AudioMixer::Source
+  AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
+                                       AudioFrame* audio_frame) override;
+  int Ssrc() const override;
+  int PreferredSampleRate() const override;
+
+  // Syncable
+  int id() const override;
+  rtc::Optional<Syncable::Info> GetInfo() const override;
+  uint32_t GetPlayoutTimestamp() const override;
+  void SetMinimumPlayoutDelay(int delay_ms) override;
+
   void AssociateSendStream(AudioSendStream* send_stream);
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
@@ -58,25 +71,20 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
                   const PacketTime& packet_time);
   const webrtc::AudioReceiveStream::Config& config() const;
 
-  // AudioMixer::Source
-  AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
-                                       AudioFrame* audio_frame) override;
-  int PreferredSampleRate() const override;
-  int Ssrc() const override;
-
  private:
   VoiceEngine* voice_engine() const;
   AudioState* audio_state() const;
   int SetVoiceEnginePlayout(bool playout);
 
-  rtc::ThreadChecker thread_checker_;
+  rtc::ThreadChecker worker_thread_checker_;
+  rtc::ThreadChecker module_process_thread_checker_;
   RemoteBitrateEstimator* const remote_bitrate_estimator_;
   const webrtc::AudioReceiveStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
-  bool playing_ ACCESS_ON(thread_checker_) = false;
+  bool playing_ ACCESS_ON(worker_thread_checker_) = false;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
 };
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index acad72d461..382ae2e66b 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -16,6 +16,8 @@ rtc_source_set("call_interfaces") {
     "audio_state.h",
     "call.h",
     "flexfec_receive_stream.h",
+    "syncable.cc",
+    "syncable.h",
   ]
 }
 
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 88faf871a9..6aa564e95e 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -55,7 +55,6 @@
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 #include "webrtc/video/vie_remb.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace webrtc {
 
@@ -146,15 +145,6 @@ class Call : public webrtc::Call,
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
-  VoiceEngine* voice_engine() {
-    internal::AudioState* audio_state =
-        static_cast<internal::AudioState*>(config_.audio_state.get());
-    if (audio_state)
-      return audio_state->voice_engine();
-    else
-      return nullptr;
-  }
-
   rtc::Optional<RtpPacketReceived> ParseRtpPacket(const uint8_t* packet,
                                                   size_t length,
                                                   const PacketTime& packet_time)
@@ -648,8 +638,8 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
   }
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       num_cpu_cores_, protected_by_flexfec, congestion_controller_.get(),
-      &packet_router_, std::move(configuration), voice_engine(),
-      module_process_thread_.get(), call_stats_.get(), &remb_);
+      &packet_router_, std::move(configuration), module_process_thread_.get(),
+      call_stats_.get(), &remb_);
 
   const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
   {
@@ -1019,7 +1009,7 @@ void Call::OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
 
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
-  if (voice_engine() == nullptr || sync_group.empty())
+  if (sync_group.empty())
     return;
 
   AudioReceiveStream* sync_audio_stream = nullptr;
@@ -1056,11 +1046,11 @@ void Call::ConfigureSync(const std::string& sync_group) {
                          "the current implementation.";
     }
     // Only sync the first A/V pair within this sync group.
-    if (sync_audio_stream != nullptr && num_synced_streams == 1) {
-      video_stream->SetSyncChannel(voice_engine(),
-                                   sync_audio_stream->config().voe_channel_id);
+    if (num_synced_streams == 1) {
+      // sync_audio_stream may be null and that's ok.
+      video_stream->SetSync(sync_audio_stream);
     } else {
-      video_stream->SetSyncChannel(voice_engine(), -1);
+      video_stream->SetSync(nullptr);
     }
   }
 }
diff --git a/webrtc/call/syncable.cc b/webrtc/call/syncable.cc
new file mode 100644
index 0000000000..c254e4f755
--- /dev/null
+++ b/webrtc/call/syncable.cc
@@ -0,0 +1,17 @@
+/*
+ *  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 "webrtc/call/syncable.h"
+
+namespace webrtc {
+
+Syncable::~Syncable() = default;
+
+}  // namespace webrtc
diff --git a/webrtc/call/syncable.h b/webrtc/call/syncable.h
new file mode 100644
index 0000000000..3459f9eb81
--- /dev/null
+++ b/webrtc/call/syncable.h
@@ -0,0 +1,43 @@
+/*
+ *  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.
+ */
+
+// Syncable is used by RtpStreamsSynchronizer in VideoReceiveStream, and
+// implemented by AudioReceiveStream.
+
+#ifndef WEBRTC_CALL_SYNCABLE_H_
+#define WEBRTC_CALL_SYNCABLE_H_
+
+#include <stdint.h>
+
+#include "webrtc/base/optional.h"
+
+namespace webrtc {
+
+class Syncable {
+ public:
+  struct Info {
+    int64_t latest_receive_time_ms = 0;
+    uint32_t latest_received_capture_timestamp = 0;
+    uint32_t capture_time_ntp_secs = 0;
+    uint32_t capture_time_ntp_frac = 0;
+    uint32_t capture_time_source_clock = 0;
+    int current_delay_ms = 0;
+  };
+
+  virtual ~Syncable();
+
+  virtual int id() const = 0;
+  virtual rtc::Optional<Info> GetInfo() const = 0;
+  virtual uint32_t GetPlayoutTimestamp() const = 0;
+  virtual void SetMinimumPlayoutDelay(int delay_ms) = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_SYNCABLE_H_
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index 44f3c7cbf5..dc2c532b1a 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -73,6 +73,12 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   MOCK_METHOD1(AssociateSendChannel,
                void(const ChannelProxy& send_channel_proxy));
   MOCK_METHOD0(DisassociateSendChannel, void());
+  MOCK_CONST_METHOD2(GetRtpRtcp, void(RtpRtcp** rtp_rtcp,
+                                      RtpReceiver** rtp_receiver));
+  MOCK_CONST_METHOD2(GetDelayEstimate, void(int* jitter_buffer_delay_ms,
+                                            int* playout_buffer_delay_ms));
+  MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
+  MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/webrtc/video/rtp_streams_synchronizer.cc b/webrtc/video/rtp_streams_synchronizer.cc
index 0d026b310a..1edb9b8e4f 100644
--- a/webrtc/video/rtp_streams_synchronizer.cc
+++ b/webrtc/video/rtp_streams_synchronizer.cc
@@ -14,83 +14,48 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/call/syncable.h"
 #include "webrtc/modules/video_coding/video_coding_impl.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/video/stream_synchronization.h"
-#include "webrtc/video_frame.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
 namespace {
 bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
-                        RtpRtcp* rtp_rtcp,
-                        RtpReceiver* receiver) {
-  if (!receiver->Timestamp(&stream->latest_timestamp))
-    return false;
-  if (!receiver->LastReceivedTimeMs(&stream->latest_receive_time_ms))
-    return false;
-
-  uint32_t ntp_secs = 0;
-  uint32_t ntp_frac = 0;
-  uint32_t rtp_timestamp = 0;
-  if (rtp_rtcp->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
-                          &rtp_timestamp) != 0) {
-    return false;
-  }
-
+                        const Syncable::Info& info) {
+  RTC_DCHECK(stream);
+  stream->latest_timestamp = info.latest_received_capture_timestamp;
+  stream->latest_receive_time_ms = info.latest_receive_time_ms;
   bool new_rtcp_sr = false;
-  if (!stream->rtp_to_ntp.UpdateMeasurements(ntp_secs, ntp_frac, rtp_timestamp,
+  if (!stream->rtp_to_ntp.UpdateMeasurements(info.capture_time_ntp_secs,
+                                             info.capture_time_ntp_frac,
+                                             info.capture_time_source_clock,
                                              &new_rtcp_sr)) {
     return false;
   }
-
   return true;
 }
 }  // namespace
 
-RtpStreamsSynchronizer::RtpStreamsSynchronizer(
-    vcm::VideoReceiver* video_receiver,
-    RtpStreamReceiver* rtp_stream_receiver)
-    : clock_(Clock::GetRealTimeClock()),
-      video_receiver_(video_receiver),
-      video_rtp_receiver_(rtp_stream_receiver->GetRtpReceiver()),
-      video_rtp_rtcp_(rtp_stream_receiver->rtp_rtcp()),
-      voe_channel_id_(-1),
-      voe_sync_interface_(nullptr),
-      audio_rtp_receiver_(nullptr),
-      audio_rtp_rtcp_(nullptr),
+RtpStreamsSynchronizer::RtpStreamsSynchronizer(Syncable* syncable_video)
+    : syncable_video_(syncable_video),
+      syncable_audio_(nullptr),
       sync_(),
       last_sync_time_(rtc::TimeNanos()) {
+  RTC_DCHECK(syncable_video);
   process_thread_checker_.DetachFromThread();
 }
 
-void RtpStreamsSynchronizer::ConfigureSync(int voe_channel_id,
-                                           VoEVideoSync* voe_sync_interface) {
-  if (voe_channel_id != -1)
-    RTC_DCHECK(voe_sync_interface);
-
+void RtpStreamsSynchronizer::ConfigureSync(Syncable* syncable_audio) {
   rtc::CritScope lock(&crit_);
-  if (voe_channel_id_ == voe_channel_id &&
-      voe_sync_interface_ == voe_sync_interface) {
+  if (syncable_audio == syncable_audio_) {
     // This prevents expensive no-ops.
     return;
   }
-  voe_channel_id_ = voe_channel_id;
-  voe_sync_interface_ = voe_sync_interface;
 
-  audio_rtp_rtcp_ = nullptr;
-  audio_rtp_receiver_ = nullptr;
+  syncable_audio_ = syncable_audio;
   sync_.reset(nullptr);
-
-  if (voe_channel_id_ != -1) {
-    voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &audio_rtp_rtcp_,
-                                    &audio_rtp_receiver_);
-    RTC_DCHECK(audio_rtp_rtcp_);
-    RTC_DCHECK(audio_rtp_receiver_);
-    sync_.reset(new StreamSynchronization(video_rtp_rtcp_->SSRC(),
-                                          voe_channel_id_));
+  if (syncable_audio_) {
+    sync_.reset(new StreamSynchronization(syncable_video_->id(),
+                                          syncable_audio_->id()));
   }
 }
 
@@ -103,35 +68,22 @@ int64_t RtpStreamsSynchronizer::TimeUntilNextProcess() {
 
 void RtpStreamsSynchronizer::Process() {
   RTC_DCHECK_RUN_ON(&process_thread_checker_);
-
-  const int current_video_delay_ms = video_receiver_->Delay();
   last_sync_time_ = rtc::TimeNanos();
 
   rtc::CritScope lock(&crit_);
-  if (voe_channel_id_ == -1) {
+  if (!syncable_audio_) {
     return;
   }
-  RTC_DCHECK(voe_sync_interface_);
   RTC_DCHECK(sync_.get());
 
-  int audio_jitter_buffer_delay_ms = 0;
-  int playout_buffer_delay_ms = 0;
-  if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
-                                            &audio_jitter_buffer_delay_ms,
-                                            &playout_buffer_delay_ms) != 0) {
+  rtc::Optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
+  if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
     return;
   }
-  const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
-      playout_buffer_delay_ms;
 
   int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
-  if (!UpdateMeasurements(&video_measurement_, video_rtp_rtcp_,
-                          video_rtp_receiver_)) {
-    return;
-  }
-
-  if (!UpdateMeasurements(&audio_measurement_, audio_rtp_rtcp_,
-                          audio_rtp_receiver_)) {
+  rtc::Optional<Syncable::Info> video_info = syncable_video_->GetInfo();
+  if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
     return;
   }
 
@@ -147,41 +99,38 @@ void RtpStreamsSynchronizer::Process() {
     return;
   }
 
-  TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay", current_video_delay_ms);
-  TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay", current_audio_delay_ms);
+  TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay",
+      video_info->current_delay_ms);
+  TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay",
+      audio_info->current_delay_ms);
   TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
   int target_audio_delay_ms = 0;
-  int target_video_delay_ms = current_video_delay_ms;
+  int target_video_delay_ms = video_info->current_delay_ms;
   // Calculate the necessary extra audio delay and desired total video
   // delay to get the streams in sync.
   if (!sync_->ComputeDelays(relative_delay_ms,
-                            current_audio_delay_ms,
+                            audio_info->current_delay_ms,
                             &target_audio_delay_ms,
                             &target_video_delay_ms)) {
     return;
   }
 
-  if (voe_sync_interface_->SetMinimumPlayoutDelay(
-      voe_channel_id_, target_audio_delay_ms) == -1) {
-    LOG(LS_ERROR) << "Error setting voice delay.";
-  }
-  video_receiver_->SetMinimumPlayoutDelay(target_video_delay_ms);
+  syncable_audio_->SetMinimumPlayoutDelay(target_audio_delay_ms);
+  syncable_video_->SetMinimumPlayoutDelay(target_video_delay_ms);
 }
 
 bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(
-    const VideoFrame& frame,
+    uint32_t timestamp,
+    int64_t render_time_ms,
     int64_t* stream_offset_ms,
     double* estimated_freq_khz) const {
   rtc::CritScope lock(&crit_);
-  if (voe_channel_id_ == -1)
-    return false;
-
-  uint32_t playout_timestamp = 0;
-  if (voe_sync_interface_->GetPlayoutTimestamp(voe_channel_id_,
-                                               playout_timestamp) != 0) {
+  if (!syncable_audio_) {
     return false;
   }
 
+  uint32_t playout_timestamp = syncable_audio_->GetPlayoutTimestamp();
+
   int64_t latest_audio_ntp;
   if (!audio_measurement_.rtp_to_ntp.Estimate(playout_timestamp,
                                               &latest_audio_ntp)) {
@@ -189,13 +138,11 @@ bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(
   }
 
   int64_t latest_video_ntp;
-  if (!video_measurement_.rtp_to_ntp.Estimate(frame.timestamp(),
-                                              &latest_video_ntp)) {
+  if (!video_measurement_.rtp_to_ntp.Estimate(timestamp, &latest_video_ntp)) {
     return false;
   }
 
-  int64_t time_to_render_ms =
-      frame.render_time_ms() - clock_->TimeInMilliseconds();
+  int64_t time_to_render_ms = render_time_ms - rtc::TimeMillis();
   if (time_to_render_ms > 0)
     latest_video_ntp += time_to_render_ms;
 
diff --git a/webrtc/video/rtp_streams_synchronizer.h b/webrtc/video/rtp_streams_synchronizer.h
index bc24d6f807..afa15015d4 100644
--- a/webrtc/video/rtp_streams_synchronizer.h
+++ b/webrtc/video/rtp_streams_synchronizer.h
@@ -19,14 +19,11 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/include/module.h"
-#include "webrtc/video/rtp_stream_receiver.h"
 #include "webrtc/video/stream_synchronization.h"
 
 namespace webrtc {
 
-class Clock;
-class VideoFrame;
-class VoEVideoSync;
+class Syncable;
 
 namespace vcm {
 class VideoReceiver;
@@ -34,11 +31,9 @@ class VideoReceiver;
 
 class RtpStreamsSynchronizer : public Module {
  public:
-  RtpStreamsSynchronizer(vcm::VideoReceiver* vcm,
-                         RtpStreamReceiver* rtp_stream_receiver);
+  explicit RtpStreamsSynchronizer(Syncable* syncable_video);
 
-  void ConfigureSync(int voe_channel_id,
-                     VoEVideoSync* voe_sync_interface);
+  void ConfigureSync(Syncable* syncable_audio);
 
   // Implements Module.
   int64_t TimeUntilNextProcess() override;
@@ -48,21 +43,16 @@ class RtpStreamsSynchronizer : public Module {
   // video |frame|. Returns true on success, false otherwise.
   // The estimated frequency is the frequency used in the RTP to NTP timestamp
   // conversion.
-  bool GetStreamSyncOffsetInMs(const VideoFrame& frame,
+  bool GetStreamSyncOffsetInMs(uint32_t timestamp,
+                               int64_t render_time_ms,
                                int64_t* stream_offset_ms,
                                double* estimated_freq_khz) const;
 
  private:
-  Clock* const clock_;
-  vcm::VideoReceiver* const video_receiver_;
-  RtpReceiver* const video_rtp_receiver_;
-  RtpRtcp* const video_rtp_rtcp_;
+  Syncable* syncable_video_;
 
   rtc::CriticalSection crit_;
-  int voe_channel_id_ GUARDED_BY(crit_);
-  VoEVideoSync* voe_sync_interface_ GUARDED_BY(crit_);
-  RtpReceiver* audio_rtp_receiver_ GUARDED_BY(crit_);
-  RtpRtcp* audio_rtp_rtcp_ GUARDED_BY(crit_);
+  Syncable* syncable_audio_ GUARDED_BY(crit_);
   std::unique_ptr<StreamSynchronization> sync_ GUARDED_BY(crit_);
   StreamSynchronization::Measurements audio_measurement_ GUARDED_BY(crit_);
   StreamSynchronization::Measurements video_measurement_ GUARDED_BY(crit_);
diff --git a/webrtc/video/stream_synchronization.cc b/webrtc/video/stream_synchronization.cc
index 145922c3a0..32a2d722f5 100644
--- a/webrtc/video/stream_synchronization.cc
+++ b/webrtc/video/stream_synchronization.cc
@@ -26,10 +26,10 @@ static const int kFilterLength = 4;
 // Minimum difference between audio and video to warrant a change.
 static const int kMinDeltaMs = 30;
 
-StreamSynchronization::StreamSynchronization(uint32_t video_primary_ssrc,
-                                             int audio_channel_id)
-    : video_primary_ssrc_(video_primary_ssrc),
-      audio_channel_id_(audio_channel_id),
+StreamSynchronization::StreamSynchronization(int video_stream_id,
+                                             int audio_stream_id)
+    : video_stream_id_(video_stream_id),
+      audio_stream_id_(audio_stream_id),
       base_target_delay_ms_(0),
       avg_diff_ms_(0) {
 }
@@ -72,7 +72,7 @@ bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
   int current_video_delay_ms = *total_video_delay_target_ms;
   LOG(LS_VERBOSE) << "Audio delay: " << current_audio_delay_ms
                   << " current diff: " << relative_delay_ms
-                  << " for channel " << audio_channel_id_;
+                  << " for stream " << audio_stream_id_;
   // Calculate the difference between the lowest possible video delay and
   // the current audio delay.
   int current_diff_ms = current_video_delay_ms - current_audio_delay_ms +
@@ -166,9 +166,9 @@ bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
   channel_delay_.last_audio_delay_ms = new_audio_delay_ms;
 
   LOG(LS_VERBOSE) << "Sync video delay " << new_video_delay_ms
-                  << " for video primary SSRC " << video_primary_ssrc_
+                  << " for video stream " << video_stream_id_
                   << " and audio delay " << channel_delay_.extra_audio_delay_ms
-                  << " for audio channel " << audio_channel_id_;
+                  << " for audio stream " << audio_stream_id_;
 
   // Return values.
   *total_video_delay_target_ms = new_video_delay_ms;
diff --git a/webrtc/video/stream_synchronization.h b/webrtc/video/stream_synchronization.h
index 0b1ad4fd3a..d32966bc53 100644
--- a/webrtc/video/stream_synchronization.h
+++ b/webrtc/video/stream_synchronization.h
@@ -27,7 +27,7 @@ class StreamSynchronization {
     uint32_t latest_timestamp;
   };
 
-  StreamSynchronization(uint32_t video_primary_ssrc, int audio_channel_id);
+  StreamSynchronization(int video_stream_id, int audio_stream_id);
 
   bool ComputeDelays(int relative_delay_ms,
                      int current_audio_delay_ms,
@@ -53,8 +53,8 @@ class StreamSynchronization {
   };
 
   SynchronizationDelays channel_delay_;
-  const uint32_t video_primary_ssrc_;
-  const int audio_channel_id_;
+  const int video_stream_id_;
+  const int audio_stream_id_;
   int base_target_delay_ms_;
   int avg_diff_ms_;
 };
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 5fd436d6a4..25418c4d8f 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -22,6 +22,8 @@
 #include "webrtc/common_video/h264/profile_level_id.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_coding/frame_object.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
@@ -33,7 +35,6 @@
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/receive_statistics_proxy.h"
 #include "webrtc/video_receive_stream.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
 
@@ -190,7 +191,6 @@ VideoReceiveStream::VideoReceiveStream(
     CongestionController* congestion_controller,
     PacketRouter* packet_router,
     VideoReceiveStream::Config config,
-    webrtc::VoiceEngine* voice_engine,
     ProcessThread* process_thread,
     CallStats* call_stats,
     VieRemb* remb)
@@ -220,7 +220,7 @@ VideoReceiveStream::VideoReceiveStream(
                            this,  // KeyFrameRequestSender
                            this,  // OnCompleteFrameCallback
                            timing_.get()),
-      rtp_stream_sync_(&video_receiver_, &rtp_stream_receiver_),
+      rtp_stream_sync_(this),
       jitter_buffer_experiment_(
           field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") ==
           "Enabled") {
@@ -230,6 +230,8 @@ VideoReceiveStream::VideoReceiveStream(
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(call_stats_);
 
+  module_process_thread_checker_.DetachFromThread();
+
   RTC_DCHECK(!config_.decoders.empty());
   std::set<int> decoder_payload_types;
   for (const Decoder& decoder : config_.decoders) {
@@ -254,6 +256,7 @@ VideoReceiveStream::VideoReceiveStream(
 }
 
 VideoReceiveStream::~VideoReceiveStream() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
   Stop();
 
@@ -265,6 +268,7 @@ VideoReceiveStream::~VideoReceiveStream() {
 }
 
 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   rtp_stream_receiver_.SignalNetworkState(state);
 }
 
@@ -281,21 +285,17 @@ bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
 
 bool VideoReceiveStream::OnRecoveredPacket(const uint8_t* packet,
                                            size_t length) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   return rtp_stream_receiver_.OnRecoveredPacket(packet, length);
 }
 
-void VideoReceiveStream::SetSyncChannel(VoiceEngine* voice_engine,
-                                        int audio_channel_id) {
-  if (voice_engine && audio_channel_id != -1) {
-    VoEVideoSync* voe_sync_interface = VoEVideoSync::GetInterface(voice_engine);
-    rtp_stream_sync_.ConfigureSync(audio_channel_id, voe_sync_interface);
-    voe_sync_interface->Release();
-  } else {
-    rtp_stream_sync_.ConfigureSync(-1, nullptr);
-  }
+void VideoReceiveStream::SetSync(Syncable* audio_syncable) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  rtp_stream_sync_.ConfigureSync(audio_syncable);
 }
 
 void VideoReceiveStream::Start() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   if (decode_thread_.IsRunning())
     return;
 
@@ -346,6 +346,7 @@ void VideoReceiveStream::Start() {
 }
 
 void VideoReceiveStream::Stop() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   rtp_stream_receiver_.StopReceive();
   // TriggerDecoderShutdown will release any waiting decoder thread and make it
   // stop immediately, instead of waiting for a timeout. Needs to be called
@@ -407,7 +408,9 @@ void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) {
   // function itself, another in GetChannel() and a third in
   // GetPlayoutTimestamp.  Seems excessive.  Anyhow, I'm assuming the function
   // succeeds most of the time, which leads to grabbing a fourth lock.
-  if (rtp_stream_sync_.GetStreamSyncOffsetInMs(video_frame, &sync_offset_ms,
+  if (rtp_stream_sync_.GetStreamSyncOffsetInMs(video_frame.timestamp(),
+                                               video_frame.render_time_ms(),
+                                               &sync_offset_ms,
                                                &estimated_freq_khz)) {
     // TODO(tommi): OnSyncOffsetUpdated grabs a lock.
     stats_proxy_.OnSyncOffsetUpdated(sync_offset_ms, estimated_freq_khz);
@@ -461,6 +464,46 @@ void VideoReceiveStream::OnCompleteFrame(
     rtp_stream_receiver_.FrameContinuous(last_continuous_pid);
 }
 
+int VideoReceiveStream::id() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return config_.rtp.remote_ssrc;
+}
+
+rtc::Optional<Syncable::Info> VideoReceiveStream::GetInfo() const {
+  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+  Syncable::Info info;
+
+  RtpReceiver* rtp_receiver = rtp_stream_receiver_.GetRtpReceiver();
+  RTC_DCHECK(rtp_receiver);
+  if (!rtp_receiver->Timestamp(&info.latest_received_capture_timestamp))
+    return rtc::Optional<Syncable::Info>();
+  if (!rtp_receiver->LastReceivedTimeMs(&info.latest_receive_time_ms))
+    return rtc::Optional<Syncable::Info>();
+
+  RtpRtcp* rtp_rtcp = rtp_stream_receiver_.rtp_rtcp();
+  RTC_DCHECK(rtp_rtcp);
+  if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs,
+                          &info.capture_time_ntp_frac,
+                          nullptr,
+                          nullptr,
+                          &info.capture_time_source_clock) != 0) {
+    return rtc::Optional<Syncable::Info>();
+  }
+
+  info.current_delay_ms = video_receiver_.Delay();
+  return rtc::Optional<Syncable::Info>(info);
+}
+
+uint32_t VideoReceiveStream::GetPlayoutTimestamp() const {
+  RTC_NOTREACHED();
+  return 0;
+}
+
+void VideoReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
+  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+  video_receiver_.SetMinimumPlayoutDelay(delay_ms);
+}
+
 bool VideoReceiveStream::DecodeThreadFunction(void* ptr) {
   static_cast<VideoReceiveStream*>(ptr)->Decode();
   return true;
@@ -489,6 +532,5 @@ void VideoReceiveStream::Decode() {
     video_receiver_.Decode(kMaxDecodeWaitTimeMs);
   }
 }
-
 }  // namespace internal
 }  // namespace webrtc
diff --git a/webrtc/video/video_receive_stream.h b/webrtc/video/video_receive_stream.h
index 9960dc9589..063a4e405f 100644
--- a/webrtc/video/video_receive_stream.h
+++ b/webrtc/video/video_receive_stream.h
@@ -14,6 +14,8 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/call/syncable.h"
 #include "webrtc/common_video/include/incoming_video_stream.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
@@ -34,7 +36,6 @@ class CongestionController;
 class IvfFileWriter;
 class ProcessThread;
 class RTPFragmentationHeader;
-class VoiceEngine;
 class VieRemb;
 class VCMTiming;
 class VCMJitterEstimator;
@@ -46,14 +47,14 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
                            public EncodedImageCallback,
                            public NackSender,
                            public KeyFrameRequestSender,
-                           public video_coding::OnCompleteFrameCallback {
+                           public video_coding::OnCompleteFrameCallback,
+                           public Syncable {
  public:
   VideoReceiveStream(int num_cpu_cores,
                      bool protected_by_flexfec,
                      CongestionController* congestion_controller,
                      PacketRouter* packet_router,
                      VideoReceiveStream::Config config,
-                     webrtc::VoiceEngine* voice_engine,
                      ProcessThread* process_thread,
                      CallStats* call_stats,
                      VieRemb* remb);
@@ -69,7 +70,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
 
   bool OnRecoveredPacket(const uint8_t* packet, size_t length);
 
-  void SetSyncChannel(VoiceEngine* voice_engine, int audio_channel_id);
+  void SetSync(Syncable* audio_syncable);
 
   // Implements webrtc::VideoReceiveStream.
   void Start() override;
@@ -104,10 +105,19 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
   void OnCompleteFrame(
       std::unique_ptr<video_coding::FrameObject> frame) override;
 
+  // Implements Syncable.
+  int id() const override;
+  rtc::Optional<Syncable::Info> GetInfo() const override;
+  uint32_t GetPlayoutTimestamp() const override;
+  void SetMinimumPlayoutDelay(int delay_ms) override;
+
  private:
   static bool DecodeThreadFunction(void* ptr);
   void Decode();
 
+  rtc::ThreadChecker worker_thread_checker_;
+  rtc::ThreadChecker module_process_thread_checker_;
+
   TransportAdapter transport_adapter_;
   const VideoReceiveStream::Config config_;
   const int num_cpu_cores_;
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 59b920f8e1..b3c3a98507 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -256,6 +256,42 @@ void ChannelProxy::DisassociateSendChannel() {
   channel()->set_associate_send_channel(ChannelOwner(nullptr));
 }
 
+void ChannelProxy::GetRtpRtcp(RtpRtcp** rtp_rtcp,
+                              RtpReceiver** rtp_receiver) const {
+  // Called on Call's module_process_thread_.
+  RTC_DCHECK(rtp_rtcp);
+  RTC_DCHECK(rtp_receiver);
+  int error = channel()->GetRtpRtcp(rtp_rtcp, rtp_receiver);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::GetDelayEstimate(int* jitter_buffer_delay_ms,
+                                    int* playout_buffer_delay_ms) const {
+  // Called on Call's module_process_thread_.
+  RTC_DCHECK(jitter_buffer_delay_ms);
+  RTC_DCHECK(playout_buffer_delay_ms);
+  bool error = channel()->GetDelayEstimate(jitter_buffer_delay_ms,
+                                           playout_buffer_delay_ms);
+  RTC_DCHECK(error);
+}
+
+uint32_t ChannelProxy::GetPlayoutTimestamp() const {
+  // Called on video capture thread.
+  unsigned int timestamp = 0;
+  int error = channel()->GetPlayoutTimestamp(timestamp);
+  RTC_DCHECK(!error || timestamp == 0);
+  return timestamp;
+}
+
+void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) {
+  // Called on Call's module_process_thread_.
+  // Limit to range accepted by both VoE and ACM, so we're at least getting as
+  // close as possible, instead of failing.
+  delay_ms = std::max(0, std::min(delay_ms, 10000));
+  int error = channel()->SetMinimumPlayoutDelay(delay_ms);
+  RTC_DCHECK_EQ(0, error);
+}
+
 void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel()->SetRtcpRttStats(rtcp_rtt_stats);
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 4bd76a4bed..7d18a74d57 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -29,6 +29,8 @@ class PacketRouter;
 class RtcEventLog;
 class RtcpRttStats;
 class RtpPacketSender;
+class RtpReceiver;
+class RtpRtcp;
 class Transport;
 class TransportFeedbackObserver;
 
@@ -99,6 +101,12 @@ class ChannelProxy {
   virtual void SetTransportOverhead(int transport_overhead_per_packet);
   virtual void AssociateSendChannel(const ChannelProxy& send_channel_proxy);
   virtual void DisassociateSendChannel();
+  virtual void GetRtpRtcp(RtpRtcp** rtp_rtcp,
+                          RtpReceiver** rtp_receiver) const;
+  virtual void GetDelayEstimate(int* jitter_buffer_delay_ms,
+                                int* playout_buffer_delay_ms) const;
+  virtual uint32_t GetPlayoutTimestamp() const;
+  virtual void SetMinimumPlayoutDelay(int delay_ms);
 
   virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);