Integration of VP9 packetization.

webrtc/modules/video_coding/main/source/jitter_buffer.cc

 /*
  *  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.
  */
 #include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
@@ skipping 107 matching lines @@
     : clock_(clock),
       running_(false),
       crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       frame_event_(event.Pass()),
       max_number_of_frames_(kStartNumberOfFrames),
       free_frames_(),
       decodable_frames_(),
       incomplete_frames_(),
       last_decoded_state_(),
       first_packet_since_reset_(true),
+      last_gof_timestamp_(0),
+      last_gof_valid_(false),
       stats_callback_(NULL),
       incoming_frame_rate_(0),
       incoming_frame_count_(0),
       time_last_incoming_frame_count_(0),
       incoming_bit_count_(0),
       incoming_bit_rate_(0),
       num_consecutive_old_packets_(0),
       num_packets_(0),
       num_duplicated_packets_(0),
       num_discarded_packets_(0),
@@ skipping 75 matching lines @@
   num_discarded_packets_ = 0;
   time_first_packet_ms_ = 0;
 
   // Start in a non-signaled state.
   waiting_for_completion_.frame_size = 0;
   waiting_for_completion_.timestamp = 0;
   waiting_for_completion_.latest_packet_time = -1;
   first_packet_since_reset_ = true;
   rtt_ms_ = kDefaultRtt;
   last_decoded_state_.Reset();
+  last_gof_valid_ = false;
 }
 
 void VCMJitterBuffer::Stop() {
   crit_sect_->Enter();
   UpdateHistograms();
   running_ = false;
   last_decoded_state_.Reset();
+  last_gof_valid_ = false;
+
   // Make sure all frames are free and reset.
   for (FrameList::iterator it = decodable_frames_.begin();
        it != decodable_frames_.end(); ++it) {
     free_frames_.push_back(it->second);
   }
   for (FrameList::iterator it = incomplete_frames_.begin();
        it != incomplete_frames_.end(); ++it) {
     free_frames_.push_back(it->second);
   }
   for (UnorderedFrameList::iterator it = free_frames_.begin();
@@ skipping 10 matching lines @@
 bool VCMJitterBuffer::Running() const {
   CriticalSectionScoped cs(crit_sect_);
   return running_;
 }
 
 void VCMJitterBuffer::Flush() {
   CriticalSectionScoped cs(crit_sect_);
   decodable_frames_.Reset(&free_frames_);
   incomplete_frames_.Reset(&free_frames_);
   last_decoded_state_.Reset();  // TODO(mikhal): sync reset.
+  last_gof_valid_ = false;
   num_consecutive_old_packets_ = 0;
   // Also reset the jitter and delay estimates
   jitter_estimate_.Reset();
   inter_frame_delay_.Reset(clock_->TimeInMilliseconds());
   waiting_for_completion_.frame_size = 0;
   waiting_for_completion_.timestamp = 0;
   waiting_for_completion_.latest_packet_time = -1;
   first_packet_since_reset_ = true;
   missing_sequence_numbers_.clear();
 }
@@ skipping 309 matching lines @@
     if (num_consecutive_old_packets_ > kMaxConsecutiveOldPackets) {
       LOG(LS_WARNING)
           << num_consecutive_old_packets_
           << " consecutive old packets received. Flushing the jitter buffer.";
       Flush();
       return kFlushIndicator;
     }
     return kOldPacket;
   }
 
+  if (packet.codec == kVideoCodecVP9) {
+    // TODO(asapersson): Move this code to appropriate place.
+    // TODO(asapersson): Handle out of order GOF.
+    if (packet.codecSpecificHeader.codecHeader.VP9.flexible_mode) {
+      // TODO(asapersson): Add support for flexible mode.
+      return kGeneralError;
+    }
+    if (packet.codecSpecificHeader.codecHeader.VP9.ss_data_available) {
+      if (!last_gof_valid_ ||
+          IsNewerTimestamp(packet.timestamp, last_gof_timestamp_)) {
+        last_gof_.CopyGofInfoVP9(
+            packet.codecSpecificHeader.codecHeader.VP9.gof);
+        last_gof_timestamp_ = packet.timestamp;
+        last_gof_valid_ = true;
+      }
+    }
+    if (last_gof_valid_ &&
+        !packet.codecSpecificHeader.codecHeader.VP9.flexible_mode) {
+      uint8_t gof_idx = packet.codecSpecificHeader.codecHeader.VP9.gof_idx;
+      if (gof_idx != kNoGofIdx) {
+        if (gof_idx >= last_gof_.num_frames_in_gof) {
+          LOG(LS_WARNING) << "Incorrect gof_idx: " << gof_idx;
+          return kGeneralError;
+        }
+        RTPVideoTypeHeader* hdr = const_cast<RTPVideoTypeHeader*>(
+            &packet.codecSpecificHeader.codecHeader);
+        hdr->VP9.temporal_idx = last_gof_.temporal_idx[gof_idx];
+        hdr->VP9.temporal_up_switch = last_gof_.temporal_up_switch[gof_idx];
+      }
+    }
+  }
+
   num_consecutive_old_packets_ = 0;
 
   VCMFrameBuffer* frame;
   FrameList* frame_list;
   const VCMFrameBufferEnum error = GetFrame(packet, &frame, &frame_list);
   if (error != kNoError)
     return error;
 
   int64_t now_ms = clock_->TimeInMilliseconds();
   // We are keeping track of the first and latest seq numbers, and
@@ skipping 616 matching lines @@
   }
   // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
   // that case we don't wait for retransmissions.
   if (high_rtt_nack_threshold_ms_ >= 0 &&
       rtt_ms_ >= high_rtt_nack_threshold_ms_) {
     return false;
   }
   return true;
 }
 }  // namespace webrtc