Prevent a voice channel from sending data before a renderer is set.

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 112 matching lines @@
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
       public webrtc::VoEHardware,
       public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
     explicit Channel()
         : external_transport(false),
-          send(false),
           playout(false),
           volume_scale(1.0),
           vad(false),
           codec_fec(false),
           max_encoding_bandwidth(0),
           opus_dtx(false),
           red(false),
           nack(false),
           cn8_type(13),
           cn16_type(105),
           red_type(117),
           nack_max_packets(0),
           send_ssrc(0),
           associate_send_channel(-1),
           recv_codecs(),
           neteq_capacity(-1),
           neteq_fast_accelerate(false) {
       memset(&send_codec, 0, sizeof(send_codec));
     }
     bool external_transport;
-    bool send;
     bool playout;
     float volume_scale;
     bool vad;
     bool codec_fec;
     int max_encoding_bandwidth;
     bool opus_dtx;
     bool red;
     bool nack;
     int cn8_type;
     int cn16_type;
@@ skipping 21 matching lines @@
         agc_enabled_(false),
         highpass_filter_enabled_(false),
         stereo_swapping_enabled_(false),
         typing_detection_enabled_(false),
         ec_mode_(webrtc::kEcDefault),
         aecm_mode_(webrtc::kAecmSpeakerphone),
         ns_mode_(webrtc::kNsDefault),
         agc_mode_(webrtc::kAgcDefault),
         observer_(NULL),
         playout_fail_channel_(-1),
-        send_fail_channel_(-1),
         recording_sample_rate_(-1),
         playout_sample_rate_(-1) {
     memset(&agc_config_, 0, sizeof(agc_config_));
   }
   ~FakeWebRtcVoiceEngine() {
     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()); }
   uint32_t GetLocalSSRC(int channel) {
     return channels_[channel]->send_ssrc;
   }
   bool GetPlayout(int channel) {
     return channels_[channel]->playout;
   }
-  bool GetSend(int channel) {
-    return channels_[channel]->send;
-  }
   bool GetVAD(int channel) {
     return channels_[channel]->vad;
   }
   bool GetOpusDtx(int channel) {
     return channels_[channel]->opus_dtx;
   }
   bool GetRED(int channel) {
     return channels_[channel]->red;
   }
   bool GetCodecFEC(int channel) {
@@ skipping 32 matching lines @@
   bool CheckNoPacket(int channel) {
     return channels_[channel]->packets.empty();
   }
   void TriggerCallbackOnError(int channel_num, int err_code) {
     RTC_DCHECK(observer_ != NULL);
     observer_->CallbackOnError(channel_num, err_code);
   }
   void set_playout_fail_channel(int channel) {
     playout_fail_channel_ = channel;
   }
-  void set_send_fail_channel(int channel) {
-    send_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 56 matching lines @@
     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_FUNC(StartSend, (int channel)) {
-    if (send_fail_channel_ != channel) {
-      WEBRTC_CHECK_CHANNEL(channel);
-      channels_[channel]->send = true;
-      return 0;
-    } else {
-      // When send_fail_channel_ == channel, fail the StartSend on this
-      // channel.
-      return -1;
-    }
-  }
+  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_FUNC(StopSend, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->send = false;
-    return 0;
-  }
+  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;
   }
   webrtc::RtcEventLog* GetEventLog() { return nullptr; }
 
@@ skipping 408 matching lines @@
   bool highpass_filter_enabled_;
   bool stereo_swapping_enabled_;
   bool typing_detection_enabled_;
   webrtc::EcModes ec_mode_;
   webrtc::AecmModes aecm_mode_;
   webrtc::NsModes ns_mode_;
   webrtc::AgcModes agc_mode_;
   webrtc::AgcConfig agc_config_;
   webrtc::VoiceEngineObserver* observer_;
   int playout_fail_channel_;
-  int send_fail_channel_;
   int recording_sample_rate_;
   int playout_sample_rate_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_