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

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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.
  */
 
@@ skipping 110 matching lines @@
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
       public webrtc::VoEHardware,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
     Channel() {
       memset(&send_codec, 0, sizeof(send_codec));
     }
-    bool playout = false;
     bool vad = false;
     bool codec_fec = false;
     int max_encoding_bandwidth = 0;
     bool opus_dtx = false;
     int cn8_type = 13;
     int cn16_type = 105;
     int associate_send_channel = -1;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
     int neteq_capacity = -1;
     bool neteq_fast_accelerate = false;
   };
 
   FakeWebRtcVoiceEngine() {
     memset(&agc_config_, 0, sizeof(agc_config_));
   }
   ~FakeWebRtcVoiceEngine() override {
     RTC_CHECK(channels_.empty());
   }
 
   bool ec_metrics_enabled() const { return ec_metrics_enabled_; }
 
   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;
   }
   bool GetOpusDtx(int channel) {
     return channels_[channel]->opus_dtx;
   }
   bool GetCodecFEC(int channel) {
     return channels_[channel]->codec_fec;
   }
   int GetMaxEncodingBandwidth(int channel) {
     return channels_[channel]->max_encoding_bandwidth;
   }
   int GetSendCNPayloadType(int channel, bool wideband) {
     return (wideband) ?
         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;
   }
   int AddChannel(const webrtc::Config& config) {
     if (fail_create_channel_) {
       return -1;
     }
     Channel* ch = new Channel();
     auto db = webrtc::acm2::RentACodec::Database();
     ch->recv_codecs.assign(db.begin(), db.end());
@@ skipping 48 matching lines @@
     for (const auto& ch : channels_) {
       if (ch.second->associate_send_channel == channel) {
         ch.second->associate_send_channel = -1;
       }
     }
     delete channels_[channel];
     channels_.erase(channel);
     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, ());
   WEBRTC_FUNC(AssociateSendChannel, (int channel,
                                      int accociate_send_channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->associate_send_channel = accociate_send_channel;
     return 0;
   }
 
@@ skipping 17 matching lines @@
     WEBRTC_CHECK_CHANNEL(channel);
     codec = channels_[channel]->send_codec;
     return 0;
   }
   WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps));
   WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec));
   WEBRTC_FUNC(SetRecPayloadType, (int channel,
                                   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 =
           ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) {
         if (it->pltype == codec.pltype &&
             _stricmp(it->plname, codec.plname) != 0) {
           return -1;
         }
       }
     }
@@ skipping 261 matching lines @@
   bool ns_enabled_ = false;
   bool agc_enabled_ = false;
   bool highpass_filter_enabled_ = false;
   bool stereo_swapping_enabled_ = false;
   bool typing_detection_enabled_ = false;
   webrtc::EcModes ec_mode_ = webrtc::kEcDefault;
   webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone;
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
-  int playout_fail_channel_ = -1;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_