Change from WebRtcRTPHeader to RTPHeader in NetEq tests and tools

webrtc/modules/audio_coding/neteq/tools/rtp_generator.h

 /*
  *  Copyright (c) 2012 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_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace test {
 
 // Class for generating RTP headers.
 class RtpGenerator {
  public:
   RtpGenerator(int samples_per_ms,
                uint16_t start_seq_number = 0,
                uint32_t start_timestamp = 0,
                uint32_t start_send_time_ms = 0,
                uint32_t ssrc = 0x12345678)
       : seq_number_(start_seq_number),
         timestamp_(start_timestamp),
         next_send_time_ms_(start_send_time_ms),
         ssrc_(ssrc),
         samples_per_ms_(samples_per_ms),
         drift_factor_(0.0) {
   }
 
   virtual ~RtpGenerator() {}
 
   // Writes the next RTP header to |rtp_header|, which will be of type
   // |payload_type|. Returns the send time for this packet (in ms). The value of
   // |payload_length_samples| determines the send time for the next packet.
   virtual uint32_t GetRtpHeader(uint8_t payload_type,
                                 size_t payload_length_samples,
-                                WebRtcRTPHeader* rtp_header);
+                                RTPHeader* rtp_header);
 
   void set_drift_factor(double factor);
 
  protected:
   uint16_t seq_number_;
   uint32_t timestamp_;
   uint32_t next_send_time_ms_;
   const uint32_t ssrc_;
   const int samples_per_ms_;
   double drift_factor_;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
 };
 
 class TimestampJumpRtpGenerator : public RtpGenerator {
  public:
   TimestampJumpRtpGenerator(int samples_per_ms,
                             uint16_t start_seq_number,
                             uint32_t start_timestamp,
                             uint32_t jump_from_timestamp,
                             uint32_t jump_to_timestamp)
       : RtpGenerator(samples_per_ms, start_seq_number, start_timestamp),
         jump_from_timestamp_(jump_from_timestamp),
         jump_to_timestamp_(jump_to_timestamp) {}
 
   uint32_t GetRtpHeader(uint8_t payload_type,
                         size_t payload_length_samples,
-                        WebRtcRTPHeader* rtp_header) override;
+                        RTPHeader* rtp_header) override;
 
  private:
   uint32_t jump_from_timestamp_;
   uint32_t jump_to_timestamp_;
   RTC_DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator);
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_