Always ResetSenderCongestionControlObjects before RegisterEtc...

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 155 matching lines @@
     stream_config_.send_codec_spec =
         rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
             {kIsacPayloadType, kIsacFormat});
     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));
     if (audio_bwe_enabled) {
-      stream_config_.rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
-                       kTransportSequenceNumberId));
-      stream_config_.send_codec_spec->transport_cc_enabled = true;
+      AddBweToConfig(&stream_config_);
     }
     stream_config_.encoder_factory = SetupEncoderFactoryMock();
     stream_config_.min_bitrate_bps = 10000;
     stream_config_.max_bitrate_bps = 65000;
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   MockAudioEncoderFactory& mock_encoder_factory() {
     return *static_cast<MockAudioEncoderFactory*>(
         stream_config_.encoder_factory.get());
   }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
   RtpTransportControllerSendInterface* transport() { return &fake_transport_; }
   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_; }
 
+  static void AddBweToConfig(AudioSendStream::Config* config) {
+    config->rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                     kTransportSequenceNumberId));
+    config->send_codec_spec->transport_cc_enabled = true;
+  }
+
   void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
     using testing::StrEq;
     channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
     EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _))
         .WillRepeatedly(Invoke(
             [this](RtpRtcp** rtp_rtcp_module, RtpReceiver** rtp_receiver) {
               *rtp_rtcp_module = &this->rtp_rtcp_;
               *rtp_receiver = nullptr;  // Not deemed necessary for tests yet.
             }));
     EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
@@ skipping 326 matching lines @@
   stream_config.send_codec_spec =
       rtc::Optional<AudioSendStream::Config::SendCodecSpec>({9, kG722Format});
   stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105);
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
       helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
   send_stream.Reconfigure(stream_config);
 }
 
+TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
+  ConfigHelper helper(false, true);
+  internal::AudioSendStream send_stream(
+      helper.config(), helper.audio_state(), helper.worker_queue(),
+      helper.transport(), helper.bitrate_allocator(), helper.event_log(),
+      helper.rtcp_rtt_stats(), rtc::Optional<RtpState>());
+  auto new_config = helper.config();
+  ConfigHelper::AddBweToConfig(&new_config);
+  EXPECT_CALL(*helper.channel_proxy(),
+              EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
+      .Times(1);
+  {
+    ::testing::InSequence seq;
+    EXPECT_CALL(*helper.channel_proxy(), ResetSenderCongestionControlObjects())
+        .Times(1);
+    EXPECT_CALL(*helper.channel_proxy(), RegisterSenderCongestionControlObjects(
+        helper.transport(), Ne(nullptr)))
+        .Times(1);
+  }
+  send_stream.Reconfigure(new_config);
+}
+
+
 }  // namespace test
 }  // namespace webrtc