Add NetEqInput::PacketData::ToString method

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

 /*
  *  Copyright (c) 2016 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_NETEQ_INPUT_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
 
 #include <algorithm>
 #include <memory>
+#include <string>
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 
 namespace webrtc {
 namespace test {
 
 // Interface class for input to the NetEqTest class.
 class NetEqInput {
  public:
   struct PacketData {
+    std::string ToString() const;
+
     RTPHeader header;
     rtc::Buffer payload;
     double time_ms;
   };
 
   virtual ~NetEqInput() = default;
 
   // Returns at what time (in ms) NetEq::InsertPacket should be called next, or
   // empty if the source is out of packets.
   virtual rtc::Optional<int64_t> NextPacketTime() const = 0;
@@ skipping 31 matching lines @@
   virtual bool ended() const = 0;
 
   // Returns the RTP header for the next packet, i.e., the packet that will be
   // delivered next by PopPacket().
   virtual rtc::Optional<RTPHeader> NextHeader() const = 0;
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_