Add separate send-side UMA stats for screenshare and video.

webrtc/video/send_statistics_proxy.cc

 /*
  *  Copyright (c) 2013 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 13 matching lines @@
 namespace {
 // Used by histograms. Values of entries should not be changed.
 enum HistogramCodecType {
   kVideoUnknown = 0,
   kVideoVp8 = 1,
   kVideoVp9 = 2,
   kVideoH264 = 3,
   kVideoMax = 64,
 };
 
+const char* GetUmaPrefix(const VideoEncoderConfig& config) {
+  switch (config.content_type) {
+    case VideoEncoderConfig::ContentType::kRealtimeVideo:
+      return "WebRTC.Video.";
+    case VideoEncoderConfig::ContentType::kScreen:
+      return "WebRTC.Video.Screenshare.";
+    default:
+      RTC_NOTREACHED() << "Unknown content type.";
+      return "";
+  }
+}
+
 HistogramCodecType PayloadNameToHistogramCodecType(
     const std::string& payload_name) {
   if (payload_name == "VP8") {
     return kVideoVp8;
   } else if (payload_name == "VP9") {
     return kVideoVp9;
   } else if (payload_name == "H264") {
     return kVideoH264;
   } else {
     return kVideoUnknown;
   }
 }
 
 void UpdateCodecTypeHistogram(const std::string& payload_name) {
   RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.Encoder.CodecType",
       PayloadNameToHistogramCodecType(payload_name), kVideoMax);
 }
 }  // namespace
 
 
 const int SendStatisticsProxy::kStatsTimeoutMs = 5000;
 
-SendStatisticsProxy::SendStatisticsProxy(Clock* clock,
-                                         const VideoSendStream::Config& config)
+SendStatisticsProxy::SendStatisticsProxy(
+    Clock* clock,
+    const VideoSendStream::Config& config,
+    const VideoEncoderConfig& encoder_config)
     : clock_(clock),
       config_(config),
+      uma_prefix_(GetUmaPrefix(encoder_config)),
       input_frame_rate_tracker_(100u, 10u),
       sent_frame_rate_tracker_(100u, 10u),
       last_sent_frame_timestamp_(0),
       max_sent_width_per_timestamp_(0),
       max_sent_height_per_timestamp_(0) {
   UpdateCodecTypeHistogram(config_.encoder_settings.payload_name);
 }
 
 SendStatisticsProxy::~SendStatisticsProxy() {
   UpdateHistograms();
 }
 
 void SendStatisticsProxy::UpdateHistograms() {
   const int kMinRequiredSamples = 200;
   int in_width = input_width_counter_.Avg(kMinRequiredSamples);
   int in_height = input_height_counter_.Avg(kMinRequiredSamples);
   int in_fps = round(input_frame_rate_tracker_.ComputeTotalRate());
   if (in_width != -1) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.InputWidthInPixels", in_width);
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.InputHeightInPixels", in_height);
-    RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.InputFramesPerSecond", in_fps);
+    RTC_HISTOGRAM_COUNTS_10000(uma_prefix_ + "InputWidthInPixels", in_width);
+    RTC_HISTOGRAM_COUNTS_10000(uma_prefix_ + "InputHeightInPixels", in_height);
+    RTC_HISTOGRAM_COUNTS_100(uma_prefix_ + "InputFramesPerSecond", in_fps);
   }
   int sent_width = sent_width_counter_.Avg(kMinRequiredSamples);
   int sent_height = sent_height_counter_.Avg(kMinRequiredSamples);
   int sent_fps = round(sent_frame_rate_tracker_.ComputeTotalRate());
   if (sent_width != -1) {
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentWidthInPixels", sent_width);
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentHeightInPixels", sent_height);
-    RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.SentFramesPerSecond", sent_fps);
+    RTC_HISTOGRAM_COUNTS_10000(uma_prefix_ + "SentWidthInPixels", sent_width);
+    RTC_HISTOGRAM_COUNTS_10000(uma_prefix_ + "SentHeightInPixels", sent_height);
+    RTC_HISTOGRAM_COUNTS_100(uma_prefix_ + "SentFramesPerSecond", sent_fps);
   }
   int encode_ms = encode_time_counter_.Avg(kMinRequiredSamples);
   if (encode_ms != -1)
-    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.EncodeTimeInMs", encode_ms);
+    RTC_HISTOGRAM_COUNTS_1000(uma_prefix_ + "EncodeTimeInMs", encode_ms);
 
   int key_frames_permille = key_frame_counter_.Permille(kMinRequiredSamples);
   if (key_frames_permille != -1) {
-    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesSentInPermille",
-        key_frames_permille);
+    RTC_HISTOGRAM_COUNTS_1000(uma_prefix_ + "KeyFramesSentInPermille",
+                              key_frames_permille);
   }
   int quality_limited =
       quality_limited_frame_counter_.Percent(kMinRequiredSamples);
   if (quality_limited != -1) {
-    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.QualityLimitedResolutionInPercent",
+    RTC_HISTOGRAM_PERCENTAGE(uma_prefix_ + "QualityLimitedResolutionInPercent",
                              quality_limited);
   }
   int downscales = quality_downscales_counter_.Avg(kMinRequiredSamples);
   if (downscales != -1) {
-    RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.QualityLimitedResolutionDownscales",
-                              downscales, 20);
+    RTC_HISTOGRAM_ENUMERATION(
+        uma_prefix_ + "QualityLimitedResolutionDownscales", downscales, 20);
   }
   int bw_limited = bw_limited_frame_counter_.Percent(kMinRequiredSamples);
   if (bw_limited != -1) {
-    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BandwidthLimitedResolutionInPercent",
-        bw_limited);
+    RTC_HISTOGRAM_PERCENTAGE(
+        uma_prefix_ + "BandwidthLimitedResolutionInPercent", bw_limited);
   }
   int num_disabled = bw_resolutions_disabled_counter_.Avg(kMinRequiredSamples);
   if (num_disabled != -1) {
     RTC_HISTOGRAM_ENUMERATION(
-        "WebRTC.Video.BandwidthLimitedResolutionsDisabled", num_disabled, 10);
+        uma_prefix_ + "BandwidthLimitedResolutionsDisabled", num_disabled, 10);
   }
   int delay_ms = delay_counter_.Avg(kMinRequiredSamples);
   if (delay_ms != -1)
-    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.SendSideDelayInMs", delay_ms);
+    RTC_HISTOGRAM_COUNTS_100000(uma_prefix_ + "SendSideDelayInMs", delay_ms);
 
   int max_delay_ms = max_delay_counter_.Avg(kMinRequiredSamples);
   if (max_delay_ms != -1) {
-    RTC_HISTOGRAM_COUNTS_100000(
-        "WebRTC.Video.SendSideDelayMaxInMs", max_delay_ms);
+    RTC_HISTOGRAM_COUNTS_100000(uma_prefix_ + "SendSideDelayMaxInMs",
+                                max_delay_ms);
   }
 }
 
 void SendStatisticsProxy::OnOutgoingRate(uint32_t framerate, uint32_t bitrate) {
   rtc::CritScope lock(&crit_);
   stats_.encode_frame_rate = framerate;
   stats_.media_bitrate_bps = bitrate;
 }
 
 void SendStatisticsProxy::CpuOveruseMetricsUpdated(
@@ skipping 95 matching lines @@
     if (downscaled) {
       quality_downscales_counter_.Add(
           encoded_image.adapt_reason_.quality_resolution_downscales);
     }
   }
   if (encoded_image.adapt_reason_.bw_resolutions_disabled != -1) {
     bool bw_limited = encoded_image.adapt_reason_.bw_resolutions_disabled > 0;
     bw_limited_frame_counter_.Add(bw_limited);
     if (bw_limited) {
       bw_resolutions_disabled_counter_.Add(
-         encoded_image.adapt_reason_.bw_resolutions_disabled);
+          encoded_image.adapt_reason_.bw_resolutions_disabled);
     }
   }
 
   // TODO(asapersson): This is incorrect if simulcast layers are encoded on
   // different threads and there is no guarantee that one frame of all layers
   // are encoded before the next start.
   if (last_sent_frame_timestamp_ > 0 &&
       encoded_image._timeStamp != last_sent_frame_timestamp_) {
     sent_frame_rate_tracker_.AddSamples(1);
     sent_width_counter_.Add(max_sent_width_per_timestamp_);
@@ skipping 119 matching lines @@
     int min_required_samples) const {
   return Fraction(min_required_samples, 1000.0f);
 }
 
 int SendStatisticsProxy::BoolSampleCounter::Fraction(
     int min_required_samples, float multiplier) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return static_cast<int>((sum * multiplier / num_samples) + 0.5f);
 }
-
 }  // namespace webrtc