Build WebRTC with data channel only.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 2b0f72aef1ff5ee8786f5a73e0a0e58d4f29147b..99aee186f6f366f7c9c88d1d297f222d2849fb34 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -160,6 +160,7 @@ rtc::Optional<std::string> GetAudioNetworkAdaptorConfig(
   return rtc::Optional<std::string>();
 }
 
+#ifdef HAVE_MEDIA
 webrtc::AudioState::Config MakeAudioStateConfig(
     VoEWrapper* voe_wrapper,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
@@ -172,6 +173,7 @@ webrtc::AudioState::Config MakeAudioStateConfig(
   }
   return config;
 }
+#endif

[pthatcher1, 2017/05/03 18:05:53]

Would it be easier to just not include this file if HAVE_MEDIA is false?

[Zhi Huang, 2017/05/04 01:08:03]

Oh, I just realized that I've already excluded this file. None of these changes
are needed. :)

 
 // |max_send_bitrate_bps| is the bitrate from "b=" in SDP.
 // |rtp_max_bitrate_bps| is the bitrate from RtpSender::SetParameters.
@@ -218,8 +220,10 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
                         decoder_factory,
                         audio_mixer,
                         new VoEWrapper()) {
+#ifdef HAVE_MEDIA
   audio_state_ =
       webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer));
+#endif
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(