Add the WebRTC.webkitApiCountPerSession metric.

content/renderer/media/rtc_peer_connection_handler.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/debug/trace_event.h"
 #include "base/logging.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/metrics/histogram.h"
 #include "base/stl_util.h"
 #include "base/strings/utf_string_conversions.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_dependency_factory.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/peer_connection_tracker.h"
 #include "content/renderer/media/remote_media_stream_impl.h"
 #include "content/renderer/media/rtc_data_channel_handler.h"
 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/render_thread_impl.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescriptionRequest.h"
 #include "third_party/WebKit/public/platform/WebRTCVoidRequest.h"
 #include "third_party/WebKit/public/platform/WebURL.h"
@@ skipping 506 matching lines @@
 
 bool RTCPeerConnectionHandler::addStream(
     const blink::WebMediaStream& stream,
     const blink::WebMediaConstraints& options) {
   RTCMediaConstraints constraints(options);
 
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackAddStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
 
+  PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
+
   track_metrics_.AddStream(MediaStreamTrackMetrics::SENT_STREAM,
                            MediaStream::GetAdapter(stream));
 
   // A media stream is connected to a peer connection, enable the
   // peer connection mode for the sources.
   blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
   stream.audioTracks(audio_tracks);
   for (size_t i = 0; i < audio_tracks.size(); ++i) {
     MediaStreamTrack* native_track =
         MediaStreamTrack::GetTrack(audio_tracks[i]);
@@ skipping 18 matching lines @@
 
   return AddStream(stream, &constraints);
 }
 
 void RTCPeerConnectionHandler::removeStream(
     const blink::WebMediaStream& stream) {
   RemoveStream(stream);
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackRemoveStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
+  PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
   track_metrics_.RemoveStream(MediaStreamTrackMetrics::SENT_STREAM,
                               MediaStream::GetAdapter(stream));
 }
 
 void RTCPeerConnectionHandler::getStats(
     const blink::WebRTCStatsRequest& request) {
   scoped_refptr<LocalRTCStatsRequest> inner_request(
       new talk_base::RefCountedObject<LocalRTCStatsRequest>(request));
   getStats(inner_request.get());
 }
@@ skipping 156 matching lines @@
       new RemoteMediaStreamImpl(stream_interface);
   remote_streams_.insert(
       std::pair<webrtc::MediaStreamInterface*, RemoteMediaStreamImpl*> (
           stream_interface, remote_stream));
 
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackAddStream(
         this, remote_stream->webkit_stream(),
         PeerConnectionTracker::SOURCE_REMOTE);
 
+  PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
+
   track_metrics_.AddStream(MediaStreamTrackMetrics::RECEIVED_STREAM,
                            stream_interface);
 
   client_->didAddRemoteStream(remote_stream->webkit_stream());
 }
 
 void RTCPeerConnectionHandler::OnRemoveStream(
     webrtc::MediaStreamInterface* stream_interface) {
   DCHECK(stream_interface);
   RemoteStreamMap::iterator it = remote_streams_.find(stream_interface);
   if (it == remote_streams_.end()) {
     NOTREACHED() << "Stream not found";
     return;
   }
 
   track_metrics_.RemoveStream(MediaStreamTrackMetrics::RECEIVED_STREAM,
                               stream_interface);
+  PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
 
   scoped_ptr<RemoteMediaStreamImpl> remote_stream(it->second);
   const blink::WebMediaStream& webkit_stream = remote_stream->webkit_stream();
   DCHECK(!webkit_stream.isNull());
   remote_streams_.erase(it);
 
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackRemoveStream(
         this, webkit_stream, PeerConnectionTracker::SOURCE_REMOTE);
 
@@ skipping 49 matching lines @@
   webrtc::SessionDescriptionInterface* native_desc =
       dependency_factory_->CreateSessionDescription(type, sdp, error);
 
   LOG_IF(ERROR, !native_desc) << "Failed to create native session description."
                               << " Type: " << type << " SDP: " << sdp;
 
   return native_desc;
 }
 
 }  // namespace content