Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

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 a05c00ba49b4c3d9dfcedfbec3c53e8e6c734932..68b46d4ee6e6a0de8d289e3ee163c1f76d3220d7 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -111,6 +111,7 @@ struct ConfigHelper {
               .Times(1);  // Destructor resets the event log
           return channel_proxy_;
         }));
+    SetupMockForSetSendCodecs();
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 200;
@@ -133,6 +134,14 @@ struct ConfigHelper {
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
   RtcEventLog* event_log() { return &event_log_; }
 
+  void SetupMockForSetSendCodecs() {
+    EXPECT_CALL(voice_engine_, SetVADStatus(_, _, _, _))
+        .WillRepeatedly(Return(0));
+    EXPECT_CALL(voice_engine_, SetFECStatus(_, _)).WillRepeatedly(Return(0));
+    EXPECT_CALL(voice_engine_, SetSendCodec(_, _)).WillRepeatedly(Return(0));
+    EXPECT_CALL(voice_engine_, GetSendCodec(_, _)).WillOnce(Return(-1));
+  }
+
   void SetupMockForSendTelephoneEvent() {
     EXPECT_TRUE(channel_proxy_);
     EXPECT_CALL(*channel_proxy_,
@@ -287,5 +296,8 @@ TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 }
+
+// TODO(minyue): Add tests on logics involved in SetSendCodecs.
+
 }  // namespace test
 }  // namespace webrtc