Using AudioOption to enable audio network adaptor.

webrtc/media/engine/webrtcvoiceengine_unittest.cc

 /*
  *  Copyright (c) 2008 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 302 matching lines @@
   }
 
   int GetCodecBitrate(int32_t ssrc) {
     return GetSendStreamConfig(ssrc).send_codec_spec.codec_inst.rate;
   }
 
   int GetCodecPacSize(int32_t ssrc) {
     return GetSendStreamConfig(ssrc).send_codec_spec.codec_inst.pacsize;
   }
 
+  const rtc::Optional<std::string>& GetAudioNetworkAdaptorConfig(int32_t ssrc) {
+    return GetSendStreamConfig(ssrc).audio_network_adaptor_config;
+  }
+
   void SetAndExpectMaxBitrate(const cricket::AudioCodec& codec,
                               int global_max,
                               int stream_max,
                               bool expected_result,
                               int expected_codec_bitrate) {
     // Clear the bitrate limit from the previous test case.
     EXPECT_TRUE(SetMaxBitrateForStream(kSsrc1, -1));
 
     // Attempt to set the requested bitrate limits.
     SetGlobalMaxBitrate(codec, global_max);
@@ skipping 2027 matching lines @@
 TEST_F(WebRtcVoiceEngineTestFake, SampleRatesViaOptions) {
   EXPECT_TRUE(SetupSendStream());
   EXPECT_CALL(adm_, SetRecordingSampleRate(48000)).WillOnce(Return(0));
   EXPECT_CALL(adm_, SetPlayoutSampleRate(44100)).WillOnce(Return(0));
   send_parameters_.options.recording_sample_rate =
       rtc::Optional<uint32_t>(48000);
   send_parameters_.options.playout_sample_rate = rtc::Optional<uint32_t>(44100);
   SetSendParameters(send_parameters_);
 }
 
+TEST_F(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) {
+  EXPECT_TRUE(SetupSendStream());
+  send_parameters_.options.audio_network_adaptor = rtc::Optional<bool>(true);
+  send_parameters_.options.audio_network_adaptor_config =
+      rtc::Optional<std::string>("1234");
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));

[the sun, 2016/10/28 18:21:52]

You may need to use the test utility "SetSendParameters" if the mock complains.

[minyue-webrtc, 2016/10/28 18:41:57]

I see, SetSendParameters is recently added. Will do.

+  EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
+            GetAudioNetworkAdaptorConfig(kSsrc1));
+}
+
+TEST_F(WebRtcVoiceEngineTestFake, AudioSendResetAudioNetworkAdaptor) {
+  EXPECT_TRUE(SetupSendStream());
+  send_parameters_.options.audio_network_adaptor = rtc::Optional<bool>(true);
+  send_parameters_.options.audio_network_adaptor_config =
+      rtc::Optional<std::string>("1234");
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
+            GetAudioNetworkAdaptorConfig(kSsrc1));
+  const int initial_num = call_.GetNumCreatedSendStreams();
+  cricket::AudioOptions options;
+  options.audio_network_adaptor = rtc::Optional<bool>(false);
+  EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, &options, nullptr));

[the sun, 2016/10/28 18:21:52]

There's a utility for SetAudioSend as well

[minyue-webrtc, 2016/10/28 18:41:57]

will do

+  // AudioSendStream expected to be recreated.
+  EXPECT_EQ(initial_num + 1, call_.GetNumCreatedSendStreams());
+  EXPECT_EQ(rtc::Optional<std::string>(), GetAudioNetworkAdaptorConfig(kSsrc1));
+}
+
+TEST_F(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) {
+  EXPECT_TRUE(SetupSendStream());
+  send_parameters_.options.audio_network_adaptor = rtc::Optional<bool>(true);
+  send_parameters_.options.audio_network_adaptor_config =
+      rtc::Optional<std::string>("1234");
+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+  EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
+            GetAudioNetworkAdaptorConfig(kSsrc1));
+  const int initial_num = call_.GetNumCreatedSendStreams();
+  cricket::AudioOptions options;
+  options.audio_network_adaptor = rtc::Optional<bool>();
+  // Unvalued |options.audio_network_adaptor|.should not reset audio network
+  // adaptor.
+  EXPECT_TRUE(channel_->SetAudioSend(kSsrc1, true, &options, nullptr));
+  // AudioSendStream not expected to be recreated.
+  EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
+  EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
+            GetAudioNetworkAdaptorConfig(kSsrc1));
+}
+
 // Test that we can set the outgoing SSRC properly.
 // SSRC is set in SetupSendStream() by calling AddSendStream.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrc) {
   EXPECT_TRUE(SetupSendStream());
   EXPECT_TRUE(call_.GetAudioSendStream(kSsrc1));
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, GetStats) {
   // Setup. We need send codec to be set to get all stats.
   EXPECT_TRUE(SetupSendStream());
@@ skipping 1081 matching lines @@
       nullptr, webrtc::CreateBuiltinAudioDecoderFactory());
   webrtc::RtcEventLogNullImpl event_log;
   std::unique_ptr<webrtc::Call> call(
       webrtc::Call::Create(webrtc::Call::Config(&event_log)));
   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
                                            cricket::AudioOptions(), call.get());
   cricket::AudioRecvParameters parameters;
   parameters.codecs = engine.recv_codecs();
   EXPECT_TRUE(channel.SetRecvParameters(parameters));
 }