Remove ResetStatistics from RTP feedback.

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.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.
  */
 
@@ skipping 153 matching lines @@
   const RTPHeader& rtp_header,
   const uint8_t* payload,
   size_t payload_length,
   PayloadUnion payload_specific,
   bool in_order) {
   // Trigger our callbacks.
   CheckSSRCChanged(rtp_header);
 
   int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
   bool is_red = false;
-  bool should_reset_statistics = false;
 
-  if (CheckPayloadChanged(rtp_header,
-                          first_payload_byte,
-                          is_red,
-                          &payload_specific,
-                          &should_reset_statistics) == -1) {
+  if (CheckPayloadChanged(rtp_header, first_payload_byte, is_red,
+                          &payload_specific) == -1) {
     if (payload_length == 0) {
       // OK, keep-alive packet.
       return true;
     }
     LOG(LS_WARNING) << "Receiving invalid payload type.";
     return false;
   }
 
-  if (should_reset_statistics) {
-    cb_rtp_feedback_->ResetStatistics(ssrc_);
-  }
-
   WebRtcRTPHeader webrtc_rtp_header;
   memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
   webrtc_rtp_header.header = rtp_header;
   CheckCSRC(webrtc_rtp_header);
 
   size_t payload_data_length = payload_length - rtp_header.paddingLength;
 
   bool is_first_packet_in_frame = false;
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
@@ skipping 66 matching lines @@
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
 
     int8_t last_received_payload_type =
         rtp_payload_registry_->last_received_payload_type();
     if (ssrc_ != rtp_header.ssrc ||
         (last_received_payload_type == -1 && ssrc_ == 0)) {
       // We need the payload_type_ to make the call if the remote SSRC is 0.
       new_ssrc = true;
 
-      cb_rtp_feedback_->ResetStatistics(ssrc_);
-
       last_received_timestamp_ = 0;
       last_received_sequence_number_ = 0;
       last_received_frame_time_ms_ = -1;
 
       // Do we have a SSRC? Then the stream is restarted.
       if (ssrc_ != 0) {
         // Do we have the same codec? Then re-initialize coder.
         if (rtp_header.payloadType == last_received_payload_type) {
           re_initialize_decoder = true;
 
@@ skipping 32 matching lines @@
   }
 }
 
 // Implementation note: must not hold critsect when called.
 // TODO(phoglund): Move as much as possible of this code path into the media
 // specific receivers. Basically this method goes through a lot of trouble to
 // compute something which is only used by the media specific parts later. If
 // this code path moves we can get rid of some of the rtp_receiver ->
 // media_specific interface (such as CheckPayloadChange, possibly get/set
 // last known payload).
-int32_t RtpReceiverImpl::CheckPayloadChanged(
-  const RTPHeader& rtp_header,
-  const int8_t first_payload_byte,
-  bool& is_red,
-  PayloadUnion* specific_payload,
-  bool* should_reset_statistics) {
+int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
+                                             const int8_t first_payload_byte,
+                                             bool& is_red,
+                                             PayloadUnion* specific_payload) {
   bool re_initialize_decoder = false;
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE];
   int8_t payload_type = rtp_header.payloadType;
 
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
 
     int8_t last_received_payload_type =
         rtp_payload_registry_->last_received_payload_type();
@@ skipping 11 matching lines @@
           // longer catch corrupt packets at this level.
           return -1;
         }
 
         // When we receive RED we need to check the real payload type.
         if (payload_type == last_received_payload_type) {
           rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
           return 0;
         }
       }
-      *should_reset_statistics = false;
       bool should_discard_changes = false;
 
       rtp_media_receiver_->CheckPayloadChanged(
-        payload_type, specific_payload, should_reset_statistics,
+        payload_type, specific_payload,
         &should_discard_changes);
 
       if (should_discard_changes) {
         is_red = false;
         return 0;
       }
 
       Payload* payload;
       if (!rtp_payload_registry_->PayloadTypeToPayload(payload_type, payload)) {
         // Not a registered payload type.
@@ skipping 11 matching lines @@
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
 
       if (!payload->audio) {
         bool media_type_unchanged =
             rtp_payload_registry_->ReportMediaPayloadType(payload_type);
         if (media_type_unchanged) {
           // Only reset the decoder if the media codec type has changed.
           re_initialize_decoder = false;
         }
       }
-      if (re_initialize_decoder) {
-        *should_reset_statistics = true;
-      }
     } else {
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
       is_red = false;
     }
   }  // End critsect.
 
   if (re_initialize_decoder) {
     if (-1 == rtp_media_receiver_->InvokeOnInitializeDecoder(
         cb_rtp_feedback_, id_, payload_type, payload_name,
         *specific_payload)) {
@@ skipping 79 matching lines @@
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, true);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(id_, 0, false);
     }
   }
 }
 
 }  // namespace webrtc