Add a histogram metric tracking for how long audio RTP packets are sent

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 1861cc004fb1824e81416e77abc4d7a1c664e966..18036100b95f3165fe46e5216e39df88e0bbbb09 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -51,6 +51,30 @@ void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy,
 }
 }  // namespace
 
+// TODO(saza): Move this declaration further down when we can use
+// std::make_unique.
+class AudioSendStream::TimedTransport : public Transport {
+ public:
+  TimedTransport(Transport* transport, rtc::TimeInterval* time_interval)
+      : transport_(transport), lifetime_(time_interval) {}
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) {
+    if (lifetime_) {
+      lifetime_->Extend();
+    }
+    return transport_->SendRtp(packet, length, options);
+  }
+  bool SendRtcp(const uint8_t* packet, size_t length) {
+    return transport_->SendRtcp(packet, length);
+  }
+  ~TimedTransport() {}
+
+ private:
+  Transport* transport_;
+  rtc::TimeInterval* lifetime_;
+};
+
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
@@ -138,7 +162,11 @@ void AudioSendStream::ConfigureStream(
       channel_proxy->DeRegisterExternalTransport();
     }
 
-    channel_proxy->RegisterExternalTransport(new_config.send_transport);
+    stream->timed_send_transport_adapter_ =

[ossu, 2017/07/14 11:13:15]

I think you can do .reset(new TimedTransport(...)) here.

[saza WebRTC, 2017/07/17 14:27:29]

Done.

+        std::unique_ptr<TimedTransport>(new TimedTransport(
+            new_config.send_transport, &stream->active_lifetime_));
+    channel_proxy->RegisterExternalTransport(
+        stream->timed_send_transport_adapter_.get());
   }
 
   // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
@@ -388,6 +416,10 @@ RtpState AudioSendStream::GetRtpState() const {
   return rtp_rtcp_module_->GetRtpState();
 }
 
+const rtc::TimeInterval* AudioSendStream::GetActiveLifetime() const {
+  return &active_lifetime_;
+}
+
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());