Prepare the AudioSendStream to be hooked up to send-side BWE.

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index eed97c7fa0f2015a5256a7ac2ef05327b32dd44b..17bf80a8093189d313630298615e7c03434f9b26 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -13,8 +13,11 @@
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/video_engine/call_stats.h"

[the sun, 2015/11/30 12:37:20]

Should call_stats be moved to webrtc/call?

[stefan-webrtc, 2015/11/30 15:22:02]

Yes, it definitely should. I plan on doing that separately, is that fine with
you?

[the sun, 2015/12/01 10:25:35]

Acknowledged.

 
 namespace webrtc {
 namespace test {
@@ -38,6 +41,13 @@ const CallStatistics kCallStats = {
 const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, -451, -671};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 
+class NullBitrateObserver : public BitrateObserver {
+ public:
+  virtual void OnNetworkChanged(uint32_t bitrate_bps,
+                                uint8_t fraction_loss,
+                                int64_t rtt_ms) {}
+};
+
 struct ConfigHelper {
   ConfigHelper() : stream_config_(nullptr) {
     using testing::Invoke;
@@ -62,6 +72,11 @@ struct ConfigHelper {
               SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1);

[the sun, 2015/11/30 12:37:20]

You're missing an EXPECT_CALL(*channel_proxy,
SetSendTransportSequenceNumber(kTransportSequenceNumberId)).Times(1);

[stefan-webrtc, 2015/11/30 15:22:02]

Done.

+          EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(_, _, _))

[the sun, 2015/11/30 12:37:20]

By moving the common config stuff in here you could verify the inputs as well.
NullBitrateObserver could even be private to ConfigHelper.

[stefan-webrtc, 2015/11/30 15:22:02]

Done.

+              .Times(1);
+          EXPECT_CALL(*channel_proxy_,
+                      SetCongestionControlObjects(nullptr, nullptr, nullptr))
+              .Times(1);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
@@ -136,12 +151,26 @@ TEST(AudioSendStreamTest, ConfigToString) {
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
   ConfigHelper helper;
-  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  rtc::scoped_ptr<ProcessThread> thread(

[the sun, 2015/11/30 12:37:20]

Put the common configuration stuff in ConfigHelper

[stefan-webrtc, 2015/11/30 15:22:02]

Done.

+      ProcessThread::Create("AudioTestThread"));
+  CallStats call_stats;
+  NullBitrateObserver bitrate_observer;
+  CongestionController congestion_controller(thread.get(), &call_stats,
+                                             &bitrate_observer);
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        &congestion_controller);
 }
 
 TEST(AudioSendStreamTest, GetStats) {
   ConfigHelper helper;
-  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  rtc::scoped_ptr<ProcessThread> thread(
+      ProcessThread::Create("AudioTestThread"));
+  CallStats call_stats;
+  NullBitrateObserver bitrate_observer;
+  CongestionController congestion_controller(thread.get(), &call_stats,
+                                             &bitrate_observer);
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        &congestion_controller);
   helper.SetupMockForGetStats();
   AudioSendStream::Stats stats = send_stream.GetStats();
   EXPECT_EQ(kSsrc, stats.local_ssrc);
@@ -168,7 +197,14 @@ TEST(AudioSendStreamTest, GetStats) {
 
 TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
   ConfigHelper helper;
-  internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
+  rtc::scoped_ptr<ProcessThread> thread(
+      ProcessThread::Create("AudioTestThread"));
+  CallStats call_stats;
+  NullBitrateObserver bitrate_observer;
+  CongestionController congestion_controller(thread.get(), &call_stats,
+                                             &bitrate_observer);
+  internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+                                        &congestion_controller);
   helper.SetupMockForGetStats();
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);