Removed RTPHeader from NetEq's Packet struct.

webrtc/modules/audio_coding/neteq/decision_logic.h

 /*
  *  Copyright (c) 2013 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_DECISION_LOGIC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/neteq/defines.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tick_timer.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Forward declarations.
 class BufferLevelFilter;
 class DecoderDatabase;
 class DelayManager;
 class Expand;
 class PacketBuffer;
 class SyncBuffer;
-struct RTPHeader;
+struct Packet;
 
 // This is the base class for the decision tree implementations. Derived classes
 // must implement the method GetDecisionSpecialized().
 class DecisionLogic {
  public:
   // Static factory function which creates different types of objects depending
   // on the |playout_mode|.
   static DecisionLogic* Create(int fs_hz,
                                size_t output_size_samples,
                                NetEqPlayoutMode playout_mode,
@@ skipping 20 matching lines @@
 
   // Resets parts of the state. Typically done when switching codecs.
   void SoftReset();
 
   // Sets the sample rate and the output block size.
   void SetSampleRate(int fs_hz, size_t output_size_samples);
 
   // Returns the operation that should be done next. |sync_buffer| and |expand|
   // are provided for reference. |decoder_frame_length| is the number of samples
   // obtained from the last decoded frame. If there is a packet available, the
   // packet header should be supplied in |packet_header|; otherwise it should

[hlundin-webrtc, 2016/10/12 20:39:57]

Fix the comment, still mentioning packet_header.

[ossu, 2016/10/12 21:46:02]

Done.

   // be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
   // supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
   // should be set to true. The output variable |reset_decoder| will be set to
   // true if a reset is required; otherwise it is left unchanged (i.e., it can
   // remain true if it was true before the call).
   // This method end with calling GetDecisionSpecialized to get the actual
   // return value.
   Operations GetDecision(const SyncBuffer& sync_buffer,
                          const Expand& expand,
                          size_t decoder_frame_length,
-                         const RTPHeader* packet_header,
+                         const Packet* next_packet,
                          Modes prev_mode,
                          bool play_dtmf,
                          size_t generated_noise_samples,
                          bool* reset_decoder);
 
   // These methods test the |cng_state_| for different conditions.
   bool CngRfc3389On() const { return cng_state_ == kCngRfc3389On; }
   bool CngOff() const { return cng_state_ == kCngOff; }
 
   // Resets the |cng_state_| to kCngOff.
@@ skipping 26 matching lines @@
 
   enum CngState {
     kCngOff,
     kCngRfc3389On,
     kCngInternalOn
   };
 
   // Returns the operation that should be done next. |sync_buffer| and |expand|
   // are provided for reference. |decoder_frame_length| is the number of samples
   // obtained from the last decoded frame. If there is a packet available, the
   // packet header should be supplied in |packet_header|; otherwise it should

[hlundin-webrtc, 2016/10/12 20:39:57]

And here.

[ossu, 2016/10/12 21:46:02]

Done.

   // be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
   // supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
   // should be set to true. The output variable |reset_decoder| will be set to
   // true if a reset is required; otherwise it is left unchanged (i.e., it can
   // remain true if it was true before the call).
   // Should be implemented by derived classes.
   virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
                                             const Expand& expand,
                                             size_t decoder_frame_length,
-                                            const RTPHeader* packet_header,
+                                            const Packet* next_packet,
                                             Modes prev_mode,
                                             bool play_dtmf,
                                             bool* reset_decoder,
                                             size_t generated_noise_samples) = 0;
 
   // Updates the |buffer_level_filter_| with the current buffer level
   // |buffer_size_packets|.
   void FilterBufferLevel(size_t buffer_size_packets, Modes prev_mode);
 
   DecoderDatabase* decoder_database_;
@@ skipping 12 matching lines @@
   std::unique_ptr<TickTimer::Countdown> timescale_countdown_;
   int num_consecutive_expands_;
   const NetEqPlayoutMode playout_mode_;
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_