Extract simulcast rate allocation outside of video encoder.

webrtc/modules/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("audio_coding/audio_coding.gni")
@@ skipping 443 matching lines @@
       "video_coding/rtp_frame_reference_finder_unittest.cc",
       "video_coding/sequence_number_util_unittest.cc",
       "video_coding/session_info_unittest.cc",
       "video_coding/test/stream_generator.cc",
       "video_coding/test/stream_generator.h",
       "video_coding/timing_unittest.cc",
       "video_coding/utility/frame_dropper_unittest.cc",
       "video_coding/utility/h264_bitstream_parser_unittest.cc",
       "video_coding/utility/ivf_file_writer_unittest.cc",
       "video_coding/utility/quality_scaler_unittest.cc",
+      "video_coding/utility/simulcast_rate_allocator_unittest.cc",
       "video_coding/video_coding_robustness_unittest.cc",
       "video_coding/video_packet_buffer_unittest.cc",
       "video_coding/video_receiver_unittest.cc",
       "video_coding/video_sender_unittest.cc",
       "video_processing/test/denoiser_test.cc",
       "video_processing/test/video_processing_unittest.cc",
       "video_processing/test/video_processing_unittest.h",
     ]
 
     if (apm_debug_dump) {
@@ skipping 188 matching lines @@
       sources += [
         "audio_device/ios/audio_device_unittest_ios.cc",
         "audio_device/ios/objc/RTCAudioSessionTest.mm",
         "video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc",
       ]
 
       ldflags = [ "-ObjC" ]
     }
   }
 }