Unit test out of band H264 SPS,PPS within RtpStreamReceiver.

webrtc/video/BUILD.gn

 # Copyright (c) 2014 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")
 
@@ skipping 180 matching lines @@
     defines = []
     sources = [
       "call_stats_unittest.cc",
       "encoder_rtcp_feedback_unittest.cc",
       "end_to_end_tests.cc",
       "overuse_frame_detector_unittest.cc",
       "payload_router_unittest.cc",
       "quality_threshold_unittest.cc",
       "receive_statistics_proxy_unittest.cc",
       "report_block_stats_unittest.cc",
+      "rtp_stream_receiver_unittest.cc",
       "send_delay_stats_unittest.cc",
       "send_statistics_proxy_unittest.cc",
       "stats_counter_unittest.cc",
       "stream_synchronization_unittest.cc",
       "video_send_stream_tests.cc",
       "vie_encoder_unittest.cc",
       "vie_remb_unittest.cc",
     ]
     deps = [
       ":video",
       "../media:rtc_media_base",
       "//testing/gmock",
       "//testing/gtest",
     ]
     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" ]
     }
     if (rtc_use_h264) {
       defines += [ "WEBRTC_USE_H264" ]
     }
   }
 }