Using AudioOption to enable audio network adaptor.

webrtc/audio/audio_send_stream_unittest.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 51 matching lines @@
   ConfigHelper()
       : simulated_clock_(123456),
         stream_config_(nullptr),
         congestion_controller_(&simulated_clock_,
                                &bitrate_observer_,
                                &remote_bitrate_observer_,
                                &event_log_),
         bitrate_allocator_(&limit_observer_),
         worker_queue_("ConfigHelper_worker_queue") {
     using testing::Invoke;
-    using testing::StrEq;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);
 
+    SetupDefaultChannelProxy();
+
     EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
         .WillOnce(Invoke([this](int channel_id) {
-          EXPECT_FALSE(channel_proxy_);
-          channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
-          EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
-          EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
-          EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
-          EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
-          EXPECT_CALL(*channel_proxy_,
-              SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
-          EXPECT_CALL(*channel_proxy_,
-              SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1);
-          EXPECT_CALL(*channel_proxy_, EnableSendTransportSequenceNumber(
-                                           kTransportSequenceNumberId))
-              .Times(1);
-          EXPECT_CALL(*channel_proxy_,
-                      RegisterSenderCongestionControlObjects(
-                          congestion_controller_.pacer(),
-                          congestion_controller_.GetTransportFeedbackObserver(),
-                          congestion_controller_.packet_router()))
-              .Times(1);
-          EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
-              .Times(1);
-          EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
-              .Times(1);
-          EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
-              .Times(1);
-          EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull()))
-              .Times(1);
-          EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
-              .Times(1);  // Destructor resets the event log
           return channel_proxy_;
         }));
+
     SetupMockForSetupSendCodec();
+
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 200;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
     // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
     // calls from the default ctor behavior.
     stream_config_.send_codec_spec.codec_inst = kIsacCodec;
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
   CongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
   RtcEventLog* event_log() { return &event_log_; }
   MockVoiceEngine* voice_engine() { return &voice_engine_; }
 
+  void SetupDefaultChannelProxy() {
+    using testing::StrEq;
+    channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
+    EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
+    EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
+    EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
+    EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
+    EXPECT_CALL(*channel_proxy_,
+                SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
+        .Times(1);
+    EXPECT_CALL(*channel_proxy_,
+                SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
+        .Times(1);
+    EXPECT_CALL(*channel_proxy_,
+                EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
+        .Times(1);
+    EXPECT_CALL(*channel_proxy_,
+                RegisterSenderCongestionControlObjects(
+                    congestion_controller_.pacer(),
+                    congestion_controller_.GetTransportFeedbackObserver(),
+                    congestion_controller_.packet_router()))
+        .Times(1);
+    EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()).Times(1);
+    EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)).Times(1);
+    EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()).Times(1);
+    EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1);
+    EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
+        .Times(1);  // Destructor resets the event log
+  }
+
   void SetupMockForSetupSendCodec() {
     EXPECT_CALL(voice_engine_, SetVADStatus(kChannelId, false, _, _))
         .WillOnce(Return(0));
     EXPECT_CALL(voice_engine_, SetFECStatus(kChannelId, false))
         .WillOnce(Return(0));
     // Let |GetSendCodec| return -1 for the first time to indicate that no send
     // codec has been set.
     EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
         .WillOnce(Return(-1));
     EXPECT_CALL(voice_engine_, SetSendCodec(kChannelId, _)).WillOnce(Return(0));
@@ skipping 69 matching lines @@
   config.voe_channel_id = kChannelId;
   config.min_bitrate_kbps = 12;
   config.max_bitrate_kbps = 34;
   config.send_codec_spec.nack_enabled = true;
   config.send_codec_spec.transport_cc_enabled = false;
   config.send_codec_spec.enable_codec_fec = true;
   config.send_codec_spec.enable_opus_dtx = false;
   config.send_codec_spec.opus_max_playback_rate = 32000;
   config.send_codec_spec.cng_payload_type = 42;
   config.send_codec_spec.cng_plfreq = 56;
+  config.send_codec_spec.min_ptime_ms = 20;
+  config.send_codec_spec.max_ptime_ms = 60;
   config.send_codec_spec.codec_inst = kIsacCodec;
   EXPECT_EQ(
       "{rtp: {ssrc: 1234, extensions: [{uri: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
       "nack: {rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
       "voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, "
       "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
       "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "
-      "32000, cng_payload_type: 42, cng_plfreq: 56, codec_inst: {pltype: "
-      "103, plname: \"isac\", plfreq: 16000, pacsize: 320, channels: 1, rate: "
-      "32000}}}",
+      "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: "
+      "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: "
+      "320, channels: 1, rate: 32000}}}",
       config.ToString());
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
 }
@@ skipping 72 matching lines @@
 TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) {
   ConfigHelper helper;
   auto stream_config = helper.config();
   const CodecInst kOpusCodec = {111, "opus", 48000, 960, 2, 64000};
   stream_config.send_codec_spec.codec_inst = kOpusCodec;
   stream_config.send_codec_spec.enable_codec_fec = true;
   stream_config.send_codec_spec.enable_opus_dtx = true;
   stream_config.send_codec_spec.opus_max_playback_rate = 12345;
   stream_config.send_codec_spec.cng_plfreq = 16000;
   stream_config.send_codec_spec.cng_payload_type = 105;
+  stream_config.send_codec_spec.min_ptime_ms = 10;
+  stream_config.send_codec_spec.max_ptime_ms = 60;
+  stream_config.audio_network_adaptor_config =
+      rtc::Optional<std::string>("abced");
   EXPECT_CALL(*helper.voice_engine(), SetFECStatus(kChannelId, true))
       .WillOnce(Return(0));
   EXPECT_CALL(
       *helper.voice_engine(),
       SetOpusDtx(kChannelId, stream_config.send_codec_spec.enable_opus_dtx))
       .WillOnce(Return(0));
   EXPECT_CALL(
       *helper.voice_engine(),
       SetOpusMaxPlaybackRate(
           kChannelId, stream_config.send_codec_spec.opus_max_playback_rate))
       .WillOnce(Return(0));
   EXPECT_CALL(*helper.voice_engine(),
               SetSendCNPayloadType(
                   kChannelId, stream_config.send_codec_spec.cng_payload_type,
                   webrtc::kFreq16000Hz))
       .WillOnce(Return(0));
+  EXPECT_CALL(
+      *helper.channel_proxy(),
+      SetReceiverFrameLengthRange(stream_config.send_codec_spec.min_ptime_ms,
+                                  stream_config.send_codec_spec.max_ptime_ms));
+  EXPECT_CALL(
+      *helper.channel_proxy(),
+      EnableAudioNetworkAdaptor(*stream_config.audio_network_adaptor_config));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
 }
 
 // VAD is applied when codec is mono and the CNG frequency matches the codec
 // sample rate.
 TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
   ConfigHelper helper;
   auto stream_config = helper.config();
   const CodecInst kG722Codec = {9, "g722", 8000, 160, 1, 16000};
   stream_config.send_codec_spec.codec_inst = kG722Codec;
   stream_config.send_codec_spec.cng_plfreq = 8000;
   stream_config.send_codec_spec.cng_payload_type = 105;
   EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _))
       .WillOnce(Return(0));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
 }
 
 }  // namespace test
 }  // namespace webrtc