diff --git a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
index f7de91fc4b..6bd5cd846e 100644
--- a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
+++ b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
@@ -35,9 +35,6 @@ class StreamStatistician {
 
   virtual uint32_t BitrateReceived() const = 0;
 
-  // Resets all statistics.
-  virtual void ResetStatistics() = 0;
-
   // Returns true if the packet with RTP header |header| is likely to be a
   // retransmitted packet, false otherwise.
   virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 4fe10c8a30..98f7c26b55 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -414,13 +414,6 @@ class RtpRtcp : public Module {
     */
     virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
 
-    /*
-    *   Reset RTP data counters for the sending side
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t ResetSendDataCountersRTP() = 0;
-
     /*
     *   Statistics of the amount of data sent
     *
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index 1587762763..ed7dfe06a0 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -252,8 +252,6 @@ public:
     virtual void OnIncomingCSRCChanged( const int32_t id,
                                         const uint32_t CSRC,
                                         const bool added) = 0;
-
-    virtual void ResetStatistics(uint32_t ssrc) = 0;
 };
 
 class RtpAudioFeedback {
@@ -323,8 +321,6 @@ class NullRtpFeedback : public RtpFeedback {
   void OnIncomingCSRCChanged(const int32_t id,
                              const uint32_t CSRC,
                              const bool added) override {}
-
-  void ResetStatistics(uint32_t ssrc) override {}
 };
 
 // Null object version of RtpData.
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index fce5f1be50..99e5b1c15a 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -161,8 +161,6 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(const uint64_t pictureID));
   MOCK_METHOD1(SendRTCPSliceLossIndication,
       int32_t(const uint8_t pictureID));
-  MOCK_METHOD0(ResetSendDataCountersRTP,
-      int32_t());
   MOCK_CONST_METHOD2(DataCountersRTP,
       int32_t(size_t *bytesSent, uint32_t *packetsSent));
   MOCK_CONST_METHOD2(GetSendStreamDataCounters,
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 3846558975..01116d6e60 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -51,22 +51,6 @@ StreamStatisticianImpl::StreamStatisticianImpl(
       rtcp_callback_(rtcp_callback),
       rtp_callback_(rtp_callback) {}
 
-void StreamStatisticianImpl::ResetStatistics() {
-  CriticalSectionScoped cs(stream_lock_.get());
-  last_report_inorder_packets_ = 0;
-  last_report_old_packets_ = 0;
-  last_report_seq_max_ = 0;
-  last_reported_statistics_ = RtcpStatistics();
-  jitter_q4_ = 0;
-  cumulative_loss_ = 0;
-  jitter_q4_transmission_time_offset_ = 0;
-  received_seq_wraps_ = 0;
-  received_seq_max_ = 0;
-  received_seq_first_ = 0;
-  stored_sum_receive_counters_.Add(receive_counters_);
-  receive_counters_ = StreamDataCounters();
-}
-
 void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
                                             size_t packet_length,
                                             bool retransmitted) {
@@ -320,7 +304,6 @@ void StreamStatisticianImpl::GetReceiveStreamDataCounters(
     StreamDataCounters* data_counters) const {
   CriticalSectionScoped cs(stream_lock_.get());
   *data_counters = receive_counters_;
-  data_counters->Add(stored_sum_receive_counters_);
 }
 
 uint32_t StreamStatisticianImpl::BitrateReceived() const {
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
index cd41744d8d..e9492298b9 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -36,7 +36,6 @@ class StreamStatisticianImpl : public StreamStatistician {
   void GetReceiveStreamDataCounters(
       StreamDataCounters* data_counters) const override;
   uint32_t BitrateReceived() const override;
-  void ResetStatistics() override;
   bool IsRetransmitOfOldPacket(const RTPHeader& header,
                                int64_t min_rtt) const override;
   bool IsPacketInOrder(uint16_t sequence_number) const override;
@@ -85,9 +84,6 @@ class StreamStatisticianImpl : public StreamStatistician {
   size_t received_packet_overhead_;
   StreamDataCounters receive_counters_;
 
-  // Stored counter values. Includes sum of reset counter values for the stream.
-  StreamDataCounters stored_sum_receive_counters_;
-
   // Counter values when we sent the last report.
   uint32_t last_report_inorder_packets_;
   uint32_t last_report_old_packets_;
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 8b25bcf615..fe9f71444e 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -143,12 +143,6 @@ TEST_F(ReceiveStatisticsTest, GetReceiveStreamDataCounters) {
   EXPECT_GT(counters.first_packet_time_ms, -1);
   EXPECT_EQ(1u, counters.transmitted.packets);
 
-  statistician->ResetStatistics();
-  // GetReceiveStreamDataCounters includes reset counter values.
-  statistician->GetReceiveStreamDataCounters(&counters);
-  EXPECT_GT(counters.first_packet_time_ms, -1);
-  EXPECT_EQ(1u, counters.transmitted.packets);
-
   receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
   statistician->GetReceiveStreamDataCounters(&counters);
   EXPECT_GT(counters.first_packet_time_ms, -1);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index e19378aa2e..c9a1adf196 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -228,10 +228,8 @@ RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive(
 
 void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
                                            PayloadUnion* specific_payload,
-                                           bool* should_reset_statistics,
                                            bool* should_discard_changes) {
   *should_discard_changes = false;
-  *should_reset_statistics = false;
 
   if (TelephoneEventPayloadType(payload_type)) {
     // Don't do callbacks for DTMF packets.
@@ -244,8 +242,6 @@ void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
                                             &specific_payload->Audio.frequency,
                                             &cng_payload_type_has_changed);
 
-  *should_reset_statistics = cng_payload_type_has_changed;
-
   if (is_cng_payload_type) {
     // Don't do callbacks for DTMF packets.
     *should_discard_changes = true;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index cd146b9332..a7efcbba34 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -93,7 +93,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
   // statistics. In addition we sometimes need to tweak the frequency.
   void CheckPayloadChanged(int8_t payload_type,
                            PayloadUnion* specific_payload,
-                           bool* should_reset_statistics,
                            bool* should_discard_changes) override;
 
   int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 3640933324..6be0c5a827 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -171,13 +171,9 @@ bool RtpReceiverImpl::IncomingRtpPacket(
 
   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;
@@ -186,10 +182,6 @@ bool RtpReceiverImpl::IncomingRtpPacket(
     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;
@@ -276,8 +268,6 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
       // 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;
@@ -330,12 +320,10 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
 // 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];
@@ -367,11 +355,10 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(
           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) {
@@ -403,9 +390,6 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(
           re_initialize_decoder = false;
         }
       }
-      if (re_initialize_decoder) {
-        *should_reset_statistics = true;
-      }
     } else {
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
       is_red = false;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 539c22676d..c904e1fcac 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -73,8 +73,7 @@ class RtpReceiverImpl : public RtpReceiver {
   int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
                               const int8_t first_payload_byte,
                               bool& is_red,
-                              PayloadUnion* payload,
-                              bool* should_reset_statistics);
+                              PayloadUnion* payload);
 
   Clock* clock_;
   RTPPayloadRegistry* rtp_payload_registry_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
index 56dd081fc8..5fae3db028 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
@@ -36,11 +36,9 @@ void RTPReceiverStrategy::SetLastMediaSpecificPayload(
 
 void RTPReceiverStrategy::CheckPayloadChanged(int8_t payload_type,
                                               PayloadUnion* specific_payload,
-                                              bool* should_reset_statistics,
                                               bool* should_discard_changes) {
-  // Default: Keep changes and don't reset statistics.
+  // Default: Keep changes.
   *should_discard_changes = false;
-  *should_reset_statistics = false;
 }
 
 int RTPReceiverStrategy::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index b34ad38cdb..9c09f8e43f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -79,7 +79,6 @@ class RTPReceiverStrategy {
   // reset statistics and/or discard this packet.
   virtual void CheckPayloadChanged(int8_t payload_type,
                                    PayloadUnion* specific_payload,
-                                   bool* should_reset_statistics,
                                    bool* should_discard_changes);
 
   virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 50382ea03c..d2e224dfa4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -540,12 +540,6 @@ int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc,
   return ret;
 }
 
-// Reset RTP data counters for the sending side.
-int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
-  rtp_sender_.ResetDataCounters();
-  return 0;  // TODO(pwestin): change to void.
-}
-
 // Force a send of an RTCP packet.
 // Normal SR and RR are triggered via the process function.
 int32_t ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 96eee96c0f..9cd7e70263 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -162,8 +162,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   int32_t SendCompoundRTCP(
       const std::set<RTCPPacketType>& rtcpPacketTypes) override;
 
-  int32_t ResetSendDataCountersRTP() override;
-
   // Statistics of the amount of data sent and received.
   int32_t DataCountersRTP(size_t* bytes_sent,
                           uint32_t* packets_sent) const override;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 29b2897b34..0456688a89 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1101,26 +1101,6 @@ uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
   return first_allocated_sequence_number;
 }
 
-void RTPSender::ResetDataCounters() {
-  uint32_t ssrc;
-  uint32_t ssrc_rtx;
-  bool report_rtx;
-  {
-    CriticalSectionScoped ssrc_lock(send_critsect_.get());
-    ssrc = ssrc_;
-    ssrc_rtx = ssrc_rtx_;
-    report_rtx = rtx_ != kRtxOff;
-  }
-  CriticalSectionScoped lock(statistics_crit_.get());
-  rtp_stats_ = StreamDataCounters();
-  rtx_rtp_stats_ = StreamDataCounters();
-  if (rtp_stats_callback_) {
-    rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc);
-    if (report_rtx)
-      rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx);
-  }
-}
-
 void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
                                 StreamDataCounters* rtx_stats) const {
   CriticalSectionScoped lock(statistics_crit_.get());
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 8a624c724b..61a1fb5ec7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -131,8 +131,6 @@ class RTPSender : public RTPSenderInterface {
   void GetDataCounters(StreamDataCounters* rtp_stats,
                        StreamDataCounters* rtx_stats) const;
 
-  void ResetDataCounters();
-
   uint32_t StartTimestamp() const;
   void SetStartTimestamp(uint32_t timestamp, bool force);
 
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 10f5055246..83ce92b420 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -245,7 +245,6 @@ class FakeReceiveStatistics : public NullReceiveStatistics {
     void GetReceiveStreamDataCounters(
         StreamDataCounters* data_counters) const override {}
     uint32_t BitrateReceived() const override { return 0; }
-    void ResetStatistics() override {}
     bool IsRetransmitOfOldPacket(const RTPHeader& header,
                                  int64_t min_rtt) const override {
       return false;
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 2eb7a5c6d0..f4d28eb645 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -229,20 +229,61 @@ void ViEChannel::UpdateHistograms() {
     int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
     if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
       RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
-          rtcp_counter.nack_packets * 60 / elapsed_sec);
+                                 rtcp_counter.nack_packets * 60 / elapsed_sec);
       RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
-          rtcp_counter.fir_packets * 60 / elapsed_sec);
+                                 rtcp_counter.fir_packets * 60 / elapsed_sec);
       RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
-          rtcp_counter.pli_packets * 60 / elapsed_sec);
+                                 rtcp_counter.pli_packets * 60 / elapsed_sec);
       if (rtcp_counter.nack_requests > 0) {
         RTC_HISTOGRAM_PERCENTAGE(
             "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
-                rtcp_counter.UniqueNackRequestsInPercent());
+            rtcp_counter.UniqueNackRequestsInPercent());
       }
       int fraction_lost = report_block_stats_sender_->FractionLostInPercent();
       if (fraction_lost != -1) {
         RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.SentPacketsLostInPercent",
-            fraction_lost);
+                                 fraction_lost);
+      }
+    }
+
+    StreamDataCounters rtp;
+    StreamDataCounters rtx;
+    GetSendStreamDataCounters(&rtp, &rtx);
+    StreamDataCounters rtp_rtx = rtp;
+    rtp_rtx.Add(rtx);
+    elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(
+                      Clock::GetRealTimeClock()->TimeInMilliseconds()) /
+                  1000;
+    if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+      RTC_HISTOGRAM_COUNTS_100000(
+          "WebRTC.Video.BitrateSentInKbps",
+          static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                           1000));
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.MediaBitrateSentInKbps",
+          static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.PaddingBitrateSentInKbps",
+          static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
+                           1000));
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.RetransmittedBitrateSentInKbps",
+          static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
+                           elapsed_sec / 1000));
+      if (rtp_rtcp_->RtxSendStatus() != kRtxOff) {
+        RTC_HISTOGRAM_COUNTS_10000(
+            "WebRTC.Video.RtxBitrateSentInKbps",
+            static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
+                             1000));
+      }
+      bool fec_enabled = false;
+      uint8_t pltype_red;
+      uint8_t pltype_fec;
+      rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
+      if (fec_enabled) {
+        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps",
+                                   static_cast<int>(rtp_rtx.fec.TotalBytes() *
+                                                    8 / elapsed_sec / 1000));
       }
     }
   } else if (vie_receiver_.GetRemoteSsrc() > 0) {
@@ -302,49 +343,6 @@ void ViEChannel::UpdateHistograms() {
   }
 }
 
-void ViEChannel::UpdateHistogramsAtStopSend() {
-  StreamDataCounters rtp;
-  StreamDataCounters rtx;
-  GetSendStreamDataCounters(&rtp, &rtx);
-  StreamDataCounters rtp_rtx = rtp;
-  rtp_rtx.Add(rtx);
-
-  int64_t elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(
-      Clock::GetRealTimeClock()->TimeInMilliseconds()) / 1000;
-  if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
-    return;
-  }
-  RTC_HISTOGRAM_COUNTS_100000(
-      "WebRTC.Video.BitrateSentInKbps",
-      static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
-                       1000));
-  RTC_HISTOGRAM_COUNTS_10000(
-      "WebRTC.Video.MediaBitrateSentInKbps",
-      static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
-  RTC_HISTOGRAM_COUNTS_10000(
-      "WebRTC.Video.PaddingBitrateSentInKbps",
-      static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
-                       1000));
-  RTC_HISTOGRAM_COUNTS_10000(
-      "WebRTC.Video.RetransmittedBitrateSentInKbps",
-      static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec /
-                       1000));
-  if (rtp_rtcp_->RtxSendStatus() != kRtxOff) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateSentInKbps",
-                               static_cast<int>(rtx.transmitted.TotalBytes() *
-                                                8 / elapsed_sec / 1000));
-  }
-  bool fec_enabled = false;
-  uint8_t pltype_red;
-  uint8_t pltype_fec;
-  rtp_rtcp_->GenericFECStatus(fec_enabled, pltype_red, pltype_fec);
-  if (fec_enabled) {
-    RTC_HISTOGRAM_COUNTS_10000(
-        "WebRTC.Video.FecBitrateSentInKbps",
-        static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000));
-  }
-}
-
 int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
                                  bool new_stream) {
   DCHECK(sender_);
@@ -1336,7 +1334,6 @@ int32_t ViEChannel::StartSend() {
 }
 
 int32_t ViEChannel::StopSend() {
-  UpdateHistogramsAtStopSend();
   send_payload_router_->set_active(false);
   CriticalSectionScoped cs(rtp_rtcp_cs_.get());
   rtp_rtcp_->SetSendingMediaStatus(false);
@@ -1350,8 +1347,6 @@ int32_t ViEChannel::StopSend() {
     return -1;
   }
 
-  // Reset.
-  rtp_rtcp_->ResetSendDataCountersRTP();
   if (rtp_rtcp_->SetSendingStatus(false) != 0) {
     return -1;
   }
@@ -1359,7 +1354,6 @@ int32_t ViEChannel::StopSend() {
        it != simulcast_rtp_rtcp_.end();
        it++) {
     RtpRtcp* rtp_rtcp = *it;
-    rtp_rtcp->ResetSendDataCountersRTP();
     rtp_rtcp->SetSendingStatus(false);
   }
   return 0;
@@ -1698,13 +1692,6 @@ void ViEChannel::OnIncomingCSRCChanged(const int32_t id,
   CriticalSectionScoped cs(callback_cs_.get());
 }
 
-void ViEChannel::ResetStatistics(uint32_t ssrc) {
-  StreamStatistician* statistician =
-      vie_receiver_.GetReceiveStatistics()->GetStatistician(ssrc);
-  if (statistician)
-    statistician->ResetStatistics();
-}
-
 void ViEChannel::RegisterSendFrameCountObserver(
     FrameCountObserver* observer) {
   send_frame_count_observer_.Set(observer);
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 12f23fb35e..426cc3a700 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -258,7 +258,6 @@ class ViEChannel : public VCMFrameTypeCallback,
   virtual void OnIncomingCSRCChanged(const int32_t id,
                                      const uint32_t CSRC,
                                      const bool added);
-  virtual void ResetStatistics(uint32_t);
 
   int32_t SetRemoteSSRCType(const StreamType usage, const uint32_t SSRC);
 
@@ -375,7 +374,6 @@ class ViEChannel : public VCMFrameTypeCallback,
   void SetRtxSendStatus(bool enable);
 
   void UpdateHistograms();
-  void UpdateHistogramsAtStopSend();
 
   // ViEChannel exposes methods that allow to modify observers and callbacks
   // to be modified. Such an API-style is cumbersome to implement and maintain
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 6dd64c7634..8992425b53 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -74,11 +74,6 @@ class StatisticsProxy : public RtcpStatisticsCallback {
 
   void CNameChanged(const char* cname, uint32_t ssrc) override {}
 
-  void ResetStatistics() {
-    CriticalSectionScoped cs(stats_lock_.get());
-    stats_ = ChannelStatistics();
-  }
-
   ChannelStatistics GetStats() {
     CriticalSectionScoped cs(stats_lock_.get());
     return stats_;
@@ -340,15 +335,6 @@ void Channel::OnIncomingCSRCChanged(int32_t id,
                  id, CSRC, added);
 }
 
-void Channel::ResetStatistics(uint32_t ssrc) {
-  StreamStatistician* statistician =
-      rtp_receive_statistics_->GetStatistician(ssrc);
-  if (statistician) {
-    statistician->ResetStatistics();
-  }
-  statistics_proxy_->ResetStatistics();
-}
-
 int32_t
 Channel::OnInitializeDecoder(
     int32_t id,
@@ -1206,8 +1192,7 @@ Channel::StopSend()
 
     // Reset sending SSRC and sequence number and triggers direct transmission
     // of RTCP BYE
-    if (_rtpRtcpModule->SetSendingStatus(false) == -1 ||
-        _rtpRtcpModule->ResetSendDataCountersRTP() == -1)
+    if (_rtpRtcpModule->SetSendingStatus(false) == -1)
     {
         _engineStatisticsPtr->SetLastError(
             VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index bb9ded9f05..398087eefb 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -368,7 +368,6 @@ public:
                                 uint32_t rate) override;
     void OnIncomingSSRCChanged(int32_t id, uint32_t ssrc) override;
     void OnIncomingCSRCChanged(int32_t id, uint32_t CSRC, bool added) override;
-    void ResetStatistics(uint32_t ssrc) override;
 
     // From RtpAudioFeedback in the RTP/RTCP module
     void OnPlayTelephoneEvent(int32_t id,