Using AudioOption to enable audio network adaptor.

webrtc/api/call/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 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 16 matching lines @@
 }
 }  // namespace
 
 namespace webrtc {
 
 AudioSendStream::Stats::Stats() = default;
 
 AudioSendStream::Config::Config(Transport* send_transport)
     : send_transport(send_transport) {}
 
+AudioSendStream::Config::~Config() = default;
+
 std::string AudioSendStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
   ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
   ss << ", voe_channel_id: " << voe_channel_id;
   ss << ", min_bitrate_kbps: " << min_bitrate_kbps;
   ss << ", max_bitrate_kbps: " << max_bitrate_kbps;
   ss << ", send_codec_spec: " << send_codec_spec.ToString();
   ss << '}';
   return ss.str();
@@ skipping 28 matching lines @@
 
 std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
   std::stringstream ss;
   ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
   ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false");
   ss << ", enable_codec_fec: " << (enable_codec_fec ? "true" : "false");
   ss << ", enable_opus_dtx: " << (enable_opus_dtx ? "true" : "false");
   ss << ", opus_max_playback_rate: " << opus_max_playback_rate;
   ss << ", cng_payload_type: " << cng_payload_type;
   ss << ", cng_plfreq: " << cng_plfreq;
+  ss << ", min_ptime: " << min_ptime_ms;
+  ss << ", max_ptime: " << max_ptime_ms;
   ss << ", codec_inst: " << ::ToString(codec_inst);
   ss << '}';
   return ss.str();
 }
 
 bool AudioSendStream::Config::SendCodecSpec::operator==(
     const AudioSendStream::Config::SendCodecSpec& rhs) const {
-  if (nack_enabled != rhs.nack_enabled) {
-    return false;
+  if (nack_enabled == rhs.nack_enabled &&
+      transport_cc_enabled == rhs.transport_cc_enabled &&
+      enable_codec_fec == rhs.enable_codec_fec &&
+      enable_opus_dtx == rhs.enable_opus_dtx &&
+      opus_max_playback_rate == rhs.opus_max_playback_rate &&
+      cng_payload_type == rhs.cng_payload_type &&
+      cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms &&
+      min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) {
+    return true;
   }
-  if (transport_cc_enabled != rhs.transport_cc_enabled) {
-    return false;
-  }
-  if (enable_codec_fec != rhs.enable_codec_fec) {
-    return false;
-  }
-  if (enable_opus_dtx != rhs.enable_opus_dtx) {
-    return false;
-  }
-  if (opus_max_playback_rate != rhs.opus_max_playback_rate) {
-    return false;
-  }
-  if (cng_payload_type != rhs.cng_payload_type) {
-    return false;
-  }
-  if (cng_plfreq != rhs.cng_plfreq) {
-    return false;
-  }
-  if (codec_inst != rhs.codec_inst) {
-    return false;
-  }
-  return true;
+  return false;
 }
 }  // namespace webrtc