Refactor NACK bitrate allocation

webrtc/video/payload_router_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 168 matching lines @@
   // The modules return a value lower than default.
   const size_t kTestMinPayloadLength = 1001;
   EXPECT_CALL(rtp_1, MaxDataPayloadLength())
       .Times(1)
       .WillOnce(Return(kTestMinPayloadLength + 10));
   EXPECT_CALL(rtp_2, MaxDataPayloadLength())
       .Times(1)
       .WillOnce(Return(kTestMinPayloadLength));
   EXPECT_EQ(kTestMinPayloadLength, payload_router.MaxPayloadLength());
 }
-
-TEST(PayloadRouterTest, SetTargetSendBitrates) {
-  NiceMock<MockRtpRtcp> rtp_1;
-  NiceMock<MockRtpRtcp> rtp_2;
-  std::vector<RtpRtcp*> modules;
-  modules.push_back(&rtp_1);
-  modules.push_back(&rtp_2);
-  PayloadRouter payload_router(modules, 42);
-  std::vector<VideoStream> streams(2);
-  streams[0].max_bitrate_bps = 10000;
-  streams[1].max_bitrate_bps = 100000;
-  payload_router.SetSendStreams(streams);
-
-  const uint32_t bitrate_1 = 10000;
-  const uint32_t bitrate_2 = 76543;
-  EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
-      .Times(1);
-  EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
-      .Times(1);
-  payload_router.SetTargetSendBitrate(bitrate_1 + bitrate_2);
-}
 }  // namespace webrtc