Delete unused rtpdump code in media/base.

webrtc/media/base/testutils.h

 /*
  *  Copyright (c) 2004 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 61 matching lines @@
   void WriteToByteBuffer(rtc::ByteBufferWriter* buf) const;
   bool ReadFromByteBuffer(rtc::ByteBufferReader* buf);
   bool EqualsTo(const RawRtcpPacket& packet) const;
 
   uint8_t ver_to_count;
   uint8_t type;
   uint16_t length;
   char payload[16];
 };
 
-class RtpTestUtility {
- public:
-  static size_t GetTestPacketCount();
-
-  // Write the first count number of kTestRawRtcpPackets or kTestRawRtpPackets,
-  // depending on the flag rtcp. If it is RTP, use the specified SSRC. Return
-  // true if successful.
-  static bool WriteTestPackets(size_t count,
-                               bool rtcp,
-                               uint32_t rtp_ssrc,
-                               RtpDumpWriter* writer);
-
-  // Loop read the first count number of packets from the specified stream.
-  // Verify the elapsed time of the dump packets increase monotonically. If the
-  // stream is a RTP stream, verify the RTP sequence number, timestamp, and
-  // payload. If the stream is a RTCP stream, verify the RTCP header and
-  // payload.
-  static bool VerifyTestPacketsFromStream(size_t count,
-                                          rtc::StreamInterface* stream,
-                                          uint32_t ssrc);
-
-  // Verify the dump packet is the same as the raw RTP packet.
-  static bool VerifyPacket(const RtpDumpPacket* dump,
-                           const RawRtpPacket* raw,
-                           bool header_only);
-
-  static const uint32_t kDefaultSsrc = 1;
-  static const uint32_t kRtpTimestampIncrease = 90;
-  static const uint32_t kDefaultTimeIncrease = 30;
-  static const uint32_t kElapsedTimeInterval = 10;
-  static const RawRtpPacket kTestRawRtpPackets[];
-  static const RawRtcpPacket kTestRawRtcpPackets[];
-
- private:
-  RtpTestUtility() {}
-};
-
 // Test helper for testing VideoCapturer implementations.
 class VideoCapturerListener
     : public sigslot::has_slots<>,
       public rtc::VideoSinkInterface<webrtc::VideoFrame> {
  public:
   explicit VideoCapturerListener(VideoCapturer* cap);
   ~VideoCapturerListener();
 
   CaptureState last_capture_state() const { return last_capture_state_; }
   int frame_count() const { return frame_count_; }
@@ skipping 65 matching lines @@
 
 // Create StreamParams with single primary SSRC and corresponding FlexFEC SSRC.
 cricket::StreamParams CreatePrimaryWithFecFrStreamParams(
     const std::string& cname,
     uint32_t primary_ssrc,
     uint32_t flexfec_ssrc);
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_TESTUTILS_H_