Using AudioOption to enable audio network adaptor.

webrtc/media/engine/webrtcvoiceengine.h

Index: webrtc/media/engine/webrtcvoiceengine.h
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index 27ae593cf15e66cc82fc935696075f80ca04883f..7e8caba658852f6b9fa19e3f0955682a59d5e26a 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -54,6 +54,10 @@ struct SendCodecSpec {
   int red_payload_type = -1;
   int cng_payload_type = -1;
   int cng_plfreq = -1;
+  // max_ptime_ms: maximal frame length in ms.
+  int max_ptime_ms = kPreferredMaxPTime;
+  // min_ptime_ms: minimal frame length in ms.
+  int min_ptime_ms = kPreferredMinPTime;
   webrtc::CodecInst codec_inst;
 };
 
@@ -237,7 +241,8 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool SetOptions(const AudioOptions& options);
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
-  bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters);
+  bool SetSendCodecs(uint32_t ssrc,
+                     const webrtc::RtpParameters& rtp_parameters);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
   bool SetLocalSource(uint32_t ssrc, AudioSource* source);
   bool MuteStream(uint32_t ssrc, bool mute);