Propagate base minimum delay to audio_receiver_stream

Bug: webrtc:10287
Change-Id: Id7914976ef5b7eb708802119932b554d9ce4879e
Reviewed-on: https://webrtc-review.googlesource.com/c/121563
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26563}

audio/audio_receive_stream.cc

@@ -239,6 +239,16 @@ void AudioReceiveStream::SetGain(float gain) {
   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();

audio/audio_receive_stream.h

@@ -64,6 +64,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   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

audio/channel_receive.cc

@@ -144,6 +144,10 @@ class ChannelReceive : public ChannelReceiveInterface,
   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 @@ uint32_t ChannelReceive::GetPlayoutTimestamp() const {
   }
 }
 
+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;

audio/channel_receive.h

@@ -102,6 +102,12 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface {
   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;
 

audio/mock_voe_channel_proxy.h

@@ -53,6 +53,8 @@ class MockChannelReceive : public voe::ChannelReceiveInterface {
   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,

call/audio_receive_stream.h

@@ -164,6 +164,15 @@ class AudioReceiveStream {
   // 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:

media/engine/fake_webrtc_call.h

@@ -104,6 +104,13 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   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 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   float gain_ = 1.0f;
   rtc::Buffer last_packet_;
   bool started_ = false;
+  int base_minimum_playout_delay_ms_ = 0;
 };
 
 class FakeVideoSendStream final

modules/audio_coding/acm2/audio_coding_module.cc

@@ -100,6 +100,10 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // 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 @@ int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
   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,

modules/audio_coding/include/audio_coding_module.h

@@ -275,6 +275,15 @@ class AudioCodingModule {
   //
   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