Add support for handling reordered SS data on the receive-side for VP9.

webrtc/modules/video_coding/main/source/jitter_buffer.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.
  */
 
@@ skipping 56 matching lines @@
   VCMFrameBuffer* PopFrame(uint32_t timestamp);
   VCMFrameBuffer* Front() const;
   VCMFrameBuffer* Back() const;
   int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
       UnorderedFrameList* free_frames);
   void CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
                                UnorderedFrameList* free_frames);
   void Reset(UnorderedFrameList* free_frames);
 };
 
+class Vp9SsMap {
+ public:
+  Vp9SsMap() {}
+  ~Vp9SsMap() {}

[stefan-webrtc, 2015/10/13 13:49:43]

No need for these constructors, right?

[åsapersson, 2015/10/14 15:13:51]

Done.

+
+  bool Insert(const VCMPacket& packet);
+  void Reset();
+
+  // Removes SS data that are older than |timestamp|.
+  void RemoveOld(uint32_t timestamp);
+
+  bool UpdatePacket(const VCMPacket* packet);
+  void UpdateFrames(FrameList* frames);
+
+  // Public for testing.
+  // Finds the corresponding |ss_timestamp| for the input |timestamp|.
+  bool Find(uint32_t timestamp, uint32_t* ss_timestamp) const;

[stefan-webrtc, 2015/10/13 13:49:43]

Not a big fan of this type of testing. Isn't it possible to verify in some other
way?

+
+ private:
+  bool TimeForPeriodicUpdate(uint32_t timestamp) const;
+  void RemoveOlderThan(uint32_t timestamp);
+  void AdvanceFront(uint32_t timestamp);
+
+  std::map<uint32_t, GofInfoVP9, TimestampLessThan> ss_map_;
+};
+
 class VCMJitterBuffer {
  public:
   VCMJitterBuffer(Clock* clock, rtc::scoped_ptr<EventWrapper> event);
 
   ~VCMJitterBuffer();
 
   // Initializes and starts jitter buffer.
   void Start();
 
   // Signals all internal events and stops the jitter buffer.
@@ skipping 214 matching lines @@
   // Event to signal when we have a frame ready for decoder.
   rtc::scoped_ptr<EventWrapper> frame_event_;
   // Number of allocated frames.
   int max_number_of_frames_;
   UnorderedFrameList free_frames_ GUARDED_BY(crit_sect_);
   FrameList decodable_frames_ GUARDED_BY(crit_sect_);
   FrameList incomplete_frames_ GUARDED_BY(crit_sect_);
   VCMDecodingState last_decoded_state_ GUARDED_BY(crit_sect_);
   bool first_packet_since_reset_;
   // Contains last received frame's temporal information for non-flexible mode.
-  GofInfoVP9 last_gof_;
-  uint32_t last_gof_timestamp_;
-  bool last_gof_valid_;
+  Vp9SsMap vp9_ss_map_;  // GUARDED_BY(crit_sect_)
 
   // Statistics.
   VCMReceiveStatisticsCallback* stats_callback_ GUARDED_BY(crit_sect_);
   // Frame counts for each type (key, delta, ...)
   FrameCounts receive_statistics_;
   // Latest calculated frame rates of incoming stream.
   unsigned int incoming_frame_rate_;
   unsigned int incoming_frame_count_;
   int64_t time_last_incoming_frame_count_;
   unsigned int incoming_bit_count_;
@@ skipping 34 matching lines @@
   // Estimated rolling average of packets per frame
   float average_packets_per_frame_;
   // average_packets_per_frame converges fast if we have fewer than this many
   // frames.
   int frame_counter_;
   RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_