Build WebRTC with data channel only.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 142 matching lines @@
     const AudioOptions& options) {
   if (options.audio_network_adaptor && *options.audio_network_adaptor &&
       options.audio_network_adaptor_config) {
     // Turn on audio network adaptor only when |options_.audio_network_adaptor|
     // equals true and |options_.audio_network_adaptor_config| has a value.
     return options.audio_network_adaptor_config;
   }
   return rtc::Optional<std::string>();
 }
 
+#ifdef HAVE_MEDIA
 webrtc::AudioState::Config MakeAudioStateConfig(
     VoEWrapper* voe_wrapper,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
   webrtc::AudioState::Config config;
   config.voice_engine = voe_wrapper->engine();
   if (audio_mixer) {
     config.audio_mixer = audio_mixer;
   } else {
     config.audio_mixer = webrtc::AudioMixerImpl::Create();
   }
   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.
 rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
                                       rtc::Optional<int> rtp_max_bitrate_bps,
                                       const webrtc::AudioCodecSpec& spec) {
   // If application-configured bitrate is set, take minimum of that and SDP
   // bitrate.
   const int bps = rtp_max_bitrate_bps
                       ? MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps)
@@ skipping 26 matching lines @@
 WebRtcVoiceEngine::WebRtcVoiceEngine(
     webrtc::AudioDeviceModule* adm,
     const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer)
     : WebRtcVoiceEngine(adm,
                         encoder_factory,
                         decoder_factory,
                         audio_mixer,
                         new VoEWrapper()) {
+#ifdef HAVE_MEDIA
   audio_state_ =
       webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer));
+#endif
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(
     webrtc::AudioDeviceModule* adm,
     const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
     VoEWrapper* voe_wrapper)
     : adm_(adm),
       encoder_factory_(encoder_factory),
@@ skipping 2094 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE