Fix video_replay tool to respect RTX stream and fix default parameters.

webrtc/test/call_test.h

 /*
  *  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.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
@@ skipping 40 matching lines @@
   static const uint8_t kFlexfecPayloadType;
   static const uint8_t kAudioSendPayloadType;
   static const uint32_t kSendRtxSsrcs[kNumSsrcs];
   static const uint32_t kVideoSendSsrcs[kNumSsrcs];
   static const uint32_t kAudioSendSsrc;
   static const uint32_t kFlexfecSendSsrc;
   static const uint32_t kReceiverLocalVideoSsrc;
   static const uint32_t kReceiverLocalAudioSsrc;
   static const int kNackRtpHistoryMs;
   static const std::map<uint8_t, MediaType> payload_type_map_;
+  static const int kPayloadTypeH264;
+  static const int kPayloadTypeVP8;
+  static const int kPayloadTypeVP9;
 
  protected:
   // RunBaseTest overwrites the audio_state and the voice_engine of the send and
   // receive Call configs to simplify test code and avoid having old VoiceEngine
   // APIs in the tests.
   void RunBaseTest(BaseTest* test);
 
   void CreateCalls(const Call::Config& sender_config,
                    const Call::Config& receiver_config);
   void CreateSenderCall(const Call::Config& config);
@@ skipping 151 matching lines @@
   EndToEndTest();
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_