Change NetEq::InsertPacket to take an RTPHeader

webrtc/modules/audio_coding/neteq/include/neteq.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_INCLUDE_NETEQ_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
 
 #include <string.h>  // Provide access to size_t.
 
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Forward declarations.
 class AudioFrame;
-struct WebRtcRTPHeader;
 class AudioDecoderFactory;
 
 struct NetEqNetworkStatistics {
   uint16_t current_buffer_size_ms;  // Current jitter buffer size in ms.
   uint16_t preferred_buffer_size_ms;  // Target buffer size in ms.
   uint16_t jitter_peaks_found;  // 1 if adding extra delay due to peaky
                                 // jitter; 0 otherwise.
   uint16_t packet_loss_rate;  // Loss rate (network + late) in Q14.
   uint16_t packet_discard_rate;  // Late loss rate in Q14.
   uint16_t expand_rate;  // Fraction (of original stream) of synthesized
@@ skipping 94 matching lines @@
   static NetEq* Create(
       const NetEq::Config& config,
       const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
 
   virtual ~NetEq() {}
 
   // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
   // Returns 0 on success, -1 on failure.
-  virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
+  virtual int InsertPacket(const RTPHeader& rtp_header,
                            rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp) = 0;
 
   // Instructs NetEq to deliver 10 ms of audio data. The data is written to
   // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |speech_type_|,
   // |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and
   // |vad_activity_| are updated upon success. If an error is returned, some
   // fields may not have been updated, or may contain inconsistent values.
   // If muted state is enabled (through Config::enable_muted_state), |muted|
   // may be set to true after a prolonged expand period. When this happens, the
@@ skipping 150 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_