Clean up logging in AudioSendStream::SetupSendCodec().

webrtc/logging/BUILD.gn

 # Copyright (c) 2016 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.
 
 import("../build/webrtc.gni")
 import("//third_party/protobuf/proto_library.gni")
@@ skipping 14 matching lines @@
     "rtc_event_log/rtc_event_log.cc",
     "rtc_event_log/rtc_event_log_helper_thread.cc",
     "rtc_event_log/rtc_event_log_helper_thread.h",
   ]
 
   defines = []
 
   deps = [
     ":rtc_event_log_api",
     "..:webrtc_common",
+    "../api:call_api",
     "../modules/rtp_rtcp",
   ]
 
   if (rtc_enable_protobuf) {
     defines += [ "ENABLE_RTC_EVENT_LOG" ]
     deps += [ ":rtc_event_log_proto" ]
   }
   if (!build_with_chromium && is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ skipping 43 matching lines @@
       ":rtc_event_log_proto",
       "..:webrtc_common",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }