Switch to use new implementation in metrics.h.

webrtc/video/vie_channel.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 183 matching lines @@
 
 void ViEChannel::UpdateHistograms() {
   int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
 
   {
     CriticalSectionScoped cs(crit_.get());
     int64_t elapsed_sec = (now - time_of_first_rtt_ms_) / 1000;
     if (time_of_first_rtt_ms_ != -1 && num_rtts_ > 0 &&
         elapsed_sec > metrics::kMinRunTimeInSeconds) {
       int64_t avg_rtt_ms = (rtt_sum_ms_ + num_rtts_ / 2) / num_rtts_;
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+      RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms);
     }
   }
 
   if (sender_) {
     RtcpPacketTypeCounter rtcp_counter;
     GetSendRtcpPacketTypeCounter(&rtcp_counter);
     int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
     if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
-          "WebRTC.Video.NackPacketsReceivedPerMinute",
-          rtcp_counter.nack_packets * 60 / elapsed_sec);
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
-          "WebRTC.Video.FirPacketsReceivedPerMinute",
-          rtcp_counter.fir_packets * 60 / elapsed_sec);
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
-          "WebRTC.Video.PliPacketsReceivedPerMinute",
-          rtcp_counter.pli_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
+                                 rtcp_counter.nack_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
+                                 rtcp_counter.fir_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
+                                 rtcp_counter.pli_packets * 60 / elapsed_sec);
       if (rtcp_counter.nack_requests > 0) {
-        RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+        RTC_HISTOGRAM_PERCENTAGE(
             "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
             rtcp_counter.UniqueNackRequestsInPercent());
       }
       int fraction_lost = report_block_stats_sender_->FractionLostInPercent();
       if (fraction_lost != -1) {
-        RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.SentPacketsLostInPercent",
-                                        fraction_lost);
+        RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.SentPacketsLostInPercent",
+                                 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_SPARSE_100000(
+      RTC_HISTOGRAM_COUNTS_100000(
           "WebRTC.Video.BitrateSentInKbps",
           static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
                            1000));
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+      RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.MediaBitrateSentInKbps",
           static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+      RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.PaddingBitrateSentInKbps",
           static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
                            1000));
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+      RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.RetransmittedBitrateSentInKbps",
           static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
                            elapsed_sec / 1000));
       if (rtp_rtcp_modules_[0]->RtxSendStatus() != kRtxOff) {
-        RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+        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_modules_[0]->GenericFECStatus(&fec_enabled, &pltype_red,
                                              &pltype_fec);
       if (fec_enabled) {
-        RTC_HISTOGRAM_COUNTS_SPARSE_10000(
-            "WebRTC.Video.FecBitrateSentInKbps",
-            static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec /
-                             1000));
+        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps",
+                                   static_cast<int>(rtp_rtx.fec.TotalBytes() *
+                                                    8 / elapsed_sec / 1000));
       }
     }
   } else if (vie_receiver_.GetRemoteSsrc() > 0) {
     // Get receive stats if we are receiving packets, i.e. there is a remote
     // ssrc.
     RtcpPacketTypeCounter rtcp_counter;
     GetReceiveRtcpPacketTypeCounter(&rtcp_counter);
     int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
     if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
-          "WebRTC.Video.NackPacketsSentPerMinute",
-          rtcp_counter.nack_packets * 60 / elapsed_sec);
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
-          "WebRTC.Video.FirPacketsSentPerMinute",
-          rtcp_counter.fir_packets * 60 / elapsed_sec);
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
-          "WebRTC.Video.PliPacketsSentPerMinute",
-          rtcp_counter.pli_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
+                                 rtcp_counter.nack_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
+                                 rtcp_counter.fir_packets * 60 / elapsed_sec);
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
+                                 rtcp_counter.pli_packets * 60 / elapsed_sec);
       if (rtcp_counter.nack_requests > 0) {
-        RTC_HISTOGRAM_PERCENTAGE_SPARSE(
-            "WebRTC.Video.UniqueNackRequestsSentInPercent",
-            rtcp_counter.UniqueNackRequestsInPercent());
+        RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
+                                 rtcp_counter.UniqueNackRequestsInPercent());
       }
     }
 
     StreamDataCounters rtp;
     StreamDataCounters rtx;
     GetReceiveStreamDataCounters(&rtp, &rtx);
     StreamDataCounters rtp_rtx = rtp;
     rtp_rtx.Add(rtx);
     elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(now) / 1000;
     if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+      RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.BitrateReceivedInKbps",
           static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
                            1000));
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+      RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.MediaBitrateReceivedInKbps",
           static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+      RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.PaddingBitrateReceivedInKbps",
           static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
                            1000));
-      RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+      RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
           static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
                            elapsed_sec / 1000));
       uint32_t ssrc = 0;
       if (vie_receiver_.GetRtxSsrc(&ssrc)) {
-        RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+        RTC_HISTOGRAM_COUNTS_10000(
             "WebRTC.Video.RtxBitrateReceivedInKbps",
             static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
                              1000));
       }
       if (vie_receiver_.IsFecEnabled()) {
-        RTC_HISTOGRAM_COUNTS_SPARSE_10000(
-            "WebRTC.Video.FecBitrateReceivedInKbps",
-            static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec /
-                             1000));
+        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateReceivedInKbps",
+                                   static_cast<int>(rtp_rtx.fec.TotalBytes() *
+                                                    8 / elapsed_sec / 1000));
       }
     }
   }
 }
 
 int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
                                  bool new_stream) {
   RTC_DCHECK(sender_);
   if (video_codec.codecType == kVideoCodecRED ||
       video_codec.codecType == kVideoCodecULPFEC) {
@@ skipping 864 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   CriticalSectionScoped cs(crit_.get());
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc