Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.

This is part of rewriting the ConferenceMixer and OutputMixer.

Calls are instead routed through AudioReceiveStream::Start/Stop.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2206223002
Cr-Commit-Position: refs/heads/master@{#13636}

webrtc/audio/audio_receive_stream.cc

@@ -154,10 +154,17 @@ AudioReceiveStream::~AudioReceiveStream() {
 
 void AudioReceiveStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  ScopedVoEInterface<VoEBase> base(voice_engine());
+  int error = base->StartPlayout(config_.voe_channel_id);
+  if (error != 0) {
+    LOG(LS_ERROR) << "AudioReceiveStream::Start failed with error: " << error;
+  }
 }
 
 void AudioReceiveStream::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  ScopedVoEInterface<VoEBase> base(voice_engine());
+  base->StopPlayout(config_.voe_channel_id);
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {

webrtc/media/base/fakemediaengine.h

@@ -319,10 +319,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
             SetRecvRtpHeaderExtensions(params.extensions));
   }
 
-  virtual bool SetPlayout(bool playout) {
-    set_playout(playout);
-    return true;
-  }
+  virtual void SetPlayout(bool playout) { set_playout(playout); }
   virtual void SetSend(bool send) { set_sending(send); }
   virtual bool SetAudioSend(uint32_t ssrc,
                             bool enable,

webrtc/media/base/mediachannel.h

@@ -908,7 +908,7 @@ class VoiceMediaChannel : public MediaChannel {
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) = 0;
   // Starts or stops playout of received audio.
-  virtual bool SetPlayout(bool playout) = 0;
+  virtual void SetPlayout(bool playout) = 0;
   // Starts or stops sending (and potentially capture) of local audio.
   virtual void SetSend(bool send) = 0;
   // Configure stream for sending.

webrtc/media/engine/fakewebrtccall.h

@@ -79,11 +79,12 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
                   const webrtc::PacketTime& packet_time);
+  bool started() const { return started_; }
 
  private:
   // webrtc::AudioReceiveStream implementation.
-  void Start() override {}
-  void Stop() override {}
+  void Start() override { started_ = true; }
+  void Stop() override { started_ = false; }
 
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
@@ -95,6 +96,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
   float gain_ = 1.0f;
   rtc::Buffer last_packet_;
+  bool started_ = false;
 };
 
 class FakeVideoSendStream final : public webrtc::VideoSendStream,

webrtc/media/engine/fakewebrtcvoiceengine.h

@@ -128,7 +128,6 @@ class FakeWebRtcVoiceEngine
     Channel() {
       memset(&send_codec, 0, sizeof(send_codec));
     }
-    bool playout = false;
     bool vad = false;
     bool codec_fec = false;
     int max_encoding_bandwidth = 0;
@@ -154,9 +153,6 @@ class FakeWebRtcVoiceEngine
   bool IsInited() const { return inited_; }
   int GetLastChannel() const { return last_channel_; }
   int GetNumChannels() const { return static_cast<int>(channels_.size()); }
-  bool GetPlayout(int channel) {
-    return channels_[channel]->playout;
-  }
   bool GetVAD(int channel) {
     return channels_[channel]->vad;
   }
@@ -174,9 +170,6 @@ class FakeWebRtcVoiceEngine
         channels_[channel]->cn16_type :
         channels_[channel]->cn8_type;
   }
-  void set_playout_fail_channel(int channel) {
-    playout_fail_channel_ = channel;
-  }
   void set_fail_create_channel(bool fail_create_channel) {
     fail_create_channel_ = fail_create_channel;
   }
@@ -245,24 +238,10 @@ class FakeWebRtcVoiceEngine
     return 0;
   }
   WEBRTC_STUB(StartReceive, (int channel));
-  WEBRTC_FUNC(StartPlayout, (int channel)) {
-    if (playout_fail_channel_ != channel) {
-      WEBRTC_CHECK_CHANNEL(channel);
-      channels_[channel]->playout = true;
-      return 0;
-    } else {
-      // When playout_fail_channel_ == channel, fail the StartPlayout on this
-      // channel.
-      return -1;
-    }
-  }
+  WEBRTC_STUB(StartPlayout, (int channel));
   WEBRTC_STUB(StartSend, (int channel));
   WEBRTC_STUB(StopReceive, (int channel));
-  WEBRTC_FUNC(StopPlayout, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->playout = false;
-    return 0;
-  }
+  WEBRTC_STUB(StopPlayout, (int channel));
   WEBRTC_STUB(StopSend, (int channel));
   WEBRTC_STUB(GetVersion, (char version[1024]));
   WEBRTC_STUB(LastError, ());
@@ -300,8 +279,6 @@ class FakeWebRtcVoiceEngine
                                   const webrtc::CodecInst& codec)) {
     WEBRTC_CHECK_CHANNEL(channel);
     Channel* ch = channels_[channel];
-    if (ch->playout)
-      return -1;  // Channel is in use.
     // Check if something else already has this slot.
     if (codec.pltype != -1) {
       for (std::vector<webrtc::CodecInst>::iterator it =
@@ -583,7 +560,6 @@ class FakeWebRtcVoiceEngine
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
-  int playout_fail_channel_ = -1;
   FakeAudioProcessing audio_processing_;
 };
 

webrtc/media/engine/webrtcvoiceengine.cc

@@ -1363,6 +1363,18 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
     stream_->SetGain(volume);
   }
 
+  void SetPlayout(bool playout) {
+    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+    RTC_DCHECK(stream_);
+    if (playout) {
+      LOG(LS_INFO) << "Starting playout for channel #" << channel();
+      stream_->Start();
+    } else {
+      LOG(LS_INFO) << "Stopping playout for channel #" << channel();
+      stream_->Stop();
+    }
+  }
+
  private:
   void RecreateAudioReceiveStream(
       uint32_t local_ssrc,
@@ -1642,7 +1654,7 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
   if (playout_) {
     // Receive codecs can not be changed while playing. So we temporarily
     // pause playout.
-    PausePlayout();
+    ChangePlayout(false);
   }
 
   bool result = true;
@@ -1670,7 +1682,7 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
   }
 
   if (desired_playout_ && !playout_) {
-    ResumePlayout();
+    ChangePlayout(desired_playout_);
   }
   return result;
 }
@@ -1925,35 +1937,22 @@ bool WebRtcVoiceMediaChannel::SetSendCodec(
   return true;
 }
 
-bool WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
+void WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
   desired_playout_ = playout;
   return ChangePlayout(desired_playout_);
 }
 
-bool WebRtcVoiceMediaChannel::PausePlayout() {
-  return ChangePlayout(false);
-}
-
-bool WebRtcVoiceMediaChannel::ResumePlayout() {
-  return ChangePlayout(desired_playout_);
-}
-
-bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
+void WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (playout_ == playout) {
-    return true;
+    return;
   }
 
-  for (const auto& ch : recv_streams_) {
-    if (!SetPlayout(ch.second->channel(), playout)) {
-      LOG(LS_ERROR) << "SetPlayout " << playout << " on channel "
-                    << ch.second->channel() << " failed";
-      return false;
-    }
+  for (const auto& kv : recv_streams_) {
+    kv.second->SetPlayout(playout);
   }
   playout_ = playout;
-  return true;
 }
 
 void WebRtcVoiceMediaChannel::SetSend(bool send) {
@@ -2180,7 +2179,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
                                          sp.sync_label, recv_rtp_extensions_,
                                          call_, this,
                                          engine()->decoder_factory_)));
-  SetPlayout(channel, playout_);
+  recv_streams_[ssrc]->SetPlayout(playout_);
 
   return true;
 }
@@ -2614,20 +2613,6 @@ int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
   }
   return -1;
 }
-
-bool WebRtcVoiceMediaChannel::SetPlayout(int channel, bool playout) {
-  if (playout) {
-    LOG(LS_INFO) << "Starting playout for channel #" << channel;
-    if (engine()->voe()->base()->StartPlayout(channel) == -1) {
-      LOG_RTCERR1(StartPlayout, channel);
-      return false;
-    }
-  } else {
-    LOG(LS_INFO) << "Stopping playout for channel #" << channel;
-    engine()->voe()->base()->StopPlayout(channel);
-  }
-  return true;
-}
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE

webrtc/media/engine/webrtcvoiceengine.h

@@ -177,9 +177,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) override;
 
-  bool SetPlayout(bool playout) override;
-  bool PausePlayout();
-  bool ResumePlayout();
+  void SetPlayout(bool playout) override;
   void SetSend(bool send) override;
   bool SetAudioSend(uint32_t ssrc,
                     bool enable,
@@ -245,8 +243,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   WebRtcVoiceEngine* engine() { return engine_; }
   int GetLastEngineError() { return engine()->GetLastEngineError(); }
   int GetOutputLevel(int channel);
-  bool SetPlayout(int channel, bool playout);
-  bool ChangePlayout(bool playout);
+  void ChangePlayout(bool playout);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
   bool IsDefaultRecvStream(uint32_t ssrc) {

webrtc/media/engine/webrtcvoiceengine_unittest.cc

@@ -753,14 +753,13 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) {
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kCn16000Codec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
-  EXPECT_TRUE(channel_->SetPlayout(true));
+  channel_->SetPlayout(true);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
 
   // Changing the payload type of a codec should fail.
   parameters.codecs[0].id = 127;
   EXPECT_FALSE(channel_->SetRecvParameters(parameters));
-  int channel_num = voe_.GetLastChannel();
-  EXPECT_TRUE(voe_.GetPlayout(channel_num));
+  EXPECT_TRUE(GetRecvStream(kSsrc1).started());
 }
 
 // Test that we can add a codec while playing.
@@ -770,12 +769,11 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) {
   parameters.codecs.push_back(kIsacCodec);
   parameters.codecs.push_back(kCn16000Codec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
-  EXPECT_TRUE(channel_->SetPlayout(true));
+  channel_->SetPlayout(true);
 
   parameters.codecs.push_back(kOpusCodec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
-  int channel_num = voe_.GetLastChannel();
-  EXPECT_TRUE(voe_.GetPlayout(channel_num));
+  EXPECT_TRUE(GetRecvStream(kSsrc1).started());
   webrtc::CodecInst gcodec;
   EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &gcodec));
   EXPECT_EQ(kOpusCodec.id, gcodec.pltype);
@@ -2173,12 +2171,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) {
 // Test that we can create a channel and start playing out on it.
 TEST_F(WebRtcVoiceEngineTestFake, Playout) {
   EXPECT_TRUE(SetupRecvStream());
-  int channel_num = voe_.GetLastChannel();
   EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
-  EXPECT_TRUE(channel_->SetPlayout(true));
-  EXPECT_TRUE(voe_.GetPlayout(channel_num));
-  EXPECT_TRUE(channel_->SetPlayout(false));
-  EXPECT_FALSE(voe_.GetPlayout(channel_num));
+  channel_->SetPlayout(true);
+  EXPECT_TRUE(GetRecvStream(kSsrc1).started());
+  channel_->SetPlayout(false);
+  EXPECT_FALSE(GetRecvStream(kSsrc1).started());
 }
 
 // Test that we can add and remove send streams.
@@ -2332,50 +2329,41 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
 // We can receive on multiple streams while sending one stream.
 TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
   EXPECT_TRUE(SetupSendStream());
-  int channel_num1 = voe_.GetLastChannel();
 
   // Start playout without a receive stream.
   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-  EXPECT_TRUE(channel_->SetPlayout(true));
-  EXPECT_FALSE(voe_.GetPlayout(channel_num1));
+  channel_->SetPlayout(true);
 
   // Adding another stream should enable playout on the new stream only.
   EXPECT_TRUE(AddRecvStream(kSsrc2));
-  int channel_num2 = voe_.GetLastChannel();
   SetSend(channel_, true);
   EXPECT_TRUE(GetSendStream(kSsrc1).IsSending());
 
   // Make sure only the new stream is played out.
-  EXPECT_FALSE(voe_.GetPlayout(channel_num1));
-  EXPECT_TRUE(voe_.GetPlayout(channel_num2));
+  EXPECT_TRUE(GetRecvStream(kSsrc2).started());
 
   // Adding yet another stream should have stream 2 and 3 enabled for playout.
   EXPECT_TRUE(AddRecvStream(kSsrc3));
-  int channel_num3 = voe_.GetLastChannel();
-  EXPECT_FALSE(voe_.GetPlayout(channel_num1));
-  EXPECT_TRUE(voe_.GetPlayout(channel_num2));
-  EXPECT_TRUE(voe_.GetPlayout(channel_num3));
+  EXPECT_TRUE(GetRecvStream(kSsrc2).started());
+  EXPECT_TRUE(GetRecvStream(kSsrc3).started());
 
   // Stop sending.
   SetSend(channel_, false);
   EXPECT_FALSE(GetSendStream(kSsrc1).IsSending());
 
   // Stop playout.
-  EXPECT_TRUE(channel_->SetPlayout(false));
-  EXPECT_FALSE(voe_.GetPlayout(channel_num1));
-  EXPECT_FALSE(voe_.GetPlayout(channel_num2));
-  EXPECT_FALSE(voe_.GetPlayout(channel_num3));
+  channel_->SetPlayout(false);
+  EXPECT_FALSE(GetRecvStream(kSsrc2).started());
+  EXPECT_FALSE(GetRecvStream(kSsrc3).started());
 
-  // Restart playout and make sure only recv streams are played out.
-  EXPECT_TRUE(channel_->SetPlayout(true));
-  EXPECT_FALSE(voe_.GetPlayout(channel_num1));
-  EXPECT_TRUE(voe_.GetPlayout(channel_num2));
-  EXPECT_TRUE(voe_.GetPlayout(channel_num3));
+  // Restart playout and make sure recv streams are played out.
+  channel_->SetPlayout(true);
+  EXPECT_TRUE(GetRecvStream(kSsrc2).started());
+  EXPECT_TRUE(GetRecvStream(kSsrc3).started());
 
-  // Now remove the recv streams and verify that the send stream doesn't play.
+  // Now remove the recv streams.
   EXPECT_TRUE(channel_->RemoveRecvStream(3));
   EXPECT_TRUE(channel_->RemoveRecvStream(2));
-  EXPECT_FALSE(voe_.GetPlayout(channel_num1));
 }
 
 // Test that we can create a channel configured for Codian bridges,
@@ -2741,18 +2729,6 @@ TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) {
   TestInsertDtmf(kSsrc1, false);
 }
 
-TEST_F(WebRtcVoiceEngineTestFake, TestSetPlayoutError) {
-  EXPECT_TRUE(SetupSendStream());
-  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-  SetSend(channel_, true);
-  EXPECT_TRUE(AddRecvStream(2));
-  EXPECT_TRUE(AddRecvStream(3));
-  EXPECT_TRUE(channel_->SetPlayout(true));
-  voe_.set_playout_fail_channel(voe_.GetLastChannel() - 1);
-  EXPECT_TRUE(channel_->SetPlayout(false));
-  EXPECT_FALSE(channel_->SetPlayout(true));
-}
-
 TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
   EXPECT_TRUE(SetupSendStream());
   EXPECT_CALL(adm_,