diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 58eca4f..a6c81fc 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -239,6 +239,16 @@
   channel_receive_->SetChannelOutputVolumeScaling(gain);
 }
 
+bool AudioReceiveStream::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->SetBaseMinimumPlayoutDelayMs(delay_ms);
+}
+
+int AudioReceiveStream::GetBaseMinimumPlayoutDelayMs() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->GetBaseMinimumPlayoutDelayMs();
+}
+
 std::vector<RtpSource> AudioReceiveStream::GetSources() const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   return channel_receive_->GetSources();
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index 6596cb4..b6d0aa5 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -64,6 +64,8 @@
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(AudioSinkInterface* sink) override;
   void SetGain(float gain) override;
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
+  int GetBaseMinimumPlayoutDelayMs() const override;
   std::vector<webrtc::RtpSource> GetSources() const override;
 
   // TODO(nisse): We don't formally implement RtpPacketSinkInterface, and this
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 9c8aaa6..b899dcf 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -144,6 +144,10 @@
   void SetMinimumPlayoutDelay(int delayMs) override;
   uint32_t GetPlayoutTimestamp() const override;
 
+  // Audio quality.
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
+  int GetBaseMinimumPlayoutDelayMs() const override;
+
   // Produces the transport-related timestamps; current_delay_ms is left unset.
   absl::optional<Syncable::Info> GetSyncInfo() const override;
 
@@ -865,6 +869,14 @@
   }
 }
 
+bool ChannelReceive::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
+  return audio_coding_->SetBaseMinimumPlayoutDelayMs(delay_ms);
+}
+
+int ChannelReceive::GetBaseMinimumPlayoutDelayMs() const {
+  return audio_coding_->GetBaseMinimumPlayoutDelayMs();
+}
+
 absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
   RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
   Syncable::Info info;
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index d93ddad..28c7c06 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -102,6 +102,12 @@
   virtual void SetMinimumPlayoutDelay(int delay_ms) = 0;
   virtual uint32_t GetPlayoutTimestamp() const = 0;
 
+  // Audio quality.
+  // Base minimum delay sets lower bound on minimum delay value which
+  // determines minimum delay until audio playout.
+  virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
+  virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
+
   // Produces the transport-related timestamps; current_delay_ms is left unset.
   virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0;
 
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index 3aa9bf1..4aea131 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -53,6 +53,8 @@
   MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
   MOCK_CONST_METHOD0(GetSyncInfo, absl::optional<Syncable::Info>());
   MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
+  MOCK_METHOD1(SetBaseMinimumPlayoutDelayMs, bool(int delay_ms));
+  MOCK_CONST_METHOD0(GetBaseMinimumPlayoutDelayMs, int());
   MOCK_CONST_METHOD0(GetReceiveCodec,
                      absl::optional<std::pair<int, SdpAudioFormat>>());
   MOCK_METHOD1(SetReceiveCodecs,
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 88a16ed..64934b4 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -164,6 +164,15 @@
   // is potentially forwarded to any attached AudioSinkInterface implementation.
   virtual void SetGain(float gain) = 0;
 
+  // Sets a base minimum for the playout delay. Base minimum delay sets lower
+  // bound on minimum delay value determining lower bound on playout delay.
+  //
+  // Returns true if value was successfully set, false overwise.
+  virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
+
+  // Returns current value of base minimum delay in milliseconds.
+  virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
+
   virtual std::vector<RtpSource> GetSources() const = 0;
 
  protected:
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 4e8aa03..a0147e9 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -104,6 +104,13 @@
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(webrtc::AudioSinkInterface* sink) override;
   void SetGain(float gain) override;
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override {
+    base_minimum_playout_delay_ms_ = delay_ms;
+    return true;
+  }
+  int GetBaseMinimumPlayoutDelayMs() const override {
+    return base_minimum_playout_delay_ms_;
+  }
   std::vector<webrtc::RtpSource> GetSources() const override {
     return std::vector<webrtc::RtpSource>();
   }
@@ -116,6 +123,7 @@
   float gain_ = 1.0f;
   rtc::Buffer last_packet_;
   bool started_ = false;
+  int base_minimum_playout_delay_ms_ = 0;
 };
 
 class FakeVideoSendStream final
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 2ce768b..9e4ac89 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -100,6 +100,10 @@
   // Maximum playout delay.
   int SetMaximumPlayoutDelay(int time_ms) override;
 
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
+
+  int GetBaseMinimumPlayoutDelayMs() const override;
+
   absl::optional<uint32_t> PlayoutTimestamp() override;
 
   int FilteredCurrentDelayMs() const override;
@@ -708,6 +712,15 @@
   return receiver_.SetMaximumDelay(time_ms);
 }
 
+bool AudioCodingModuleImpl::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
+  // All necessary validation happens on NetEq level.
+  return receiver_.SetBaseMinimumDelayMs(delay_ms);
+}
+
+int AudioCodingModuleImpl::GetBaseMinimumPlayoutDelayMs() const {
+  return receiver_.GetBaseMinimumDelayMs();
+}
+
 // Get 10 milliseconds of raw audio data to play out.
 // Automatic resample to the requested frequency.
 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 89c3b01..7d11d70 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -275,6 +275,15 @@
   //
   virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
 
+  // Sets a base minimum for the playout delay. Base minimum delay sets lower
+  // bound minimum delay value which is set via SetMinimumPlayoutDelay.
+  //
+  // Returns true if value was successfully set, false overwise.
+  virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
+
+  // Returns current value of base minimum delay in milliseconds.
+  virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
+
   ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutTimestamp()
   // The send timestamp of an RTP packet is associated with the decoded