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 0659cbfd2d9e9423b2db8b3f2d50028e7ed8ccbd..103d6306fd857ee4d076bbf471c1bd41a8c69db5 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, 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_;
+  TimeInterval* lifetime_;
+};
+
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
@@ -137,8 +161,14 @@ void AudioSendStream::ConfigureStream(
     if (old_config.send_transport) {
       channel_proxy->DeRegisterExternalTransport();
     }
-
-    channel_proxy->RegisterExternalTransport(new_config.send_transport);
+    if (new_config.send_transport) {
+      stream->timed_send_transport_adapter_.reset(new TimedTransport(
+          new_config.send_transport, &stream->active_lifetime_));
+    } else {
+      stream->timed_send_transport_adapter_.reset(nullptr);
+    }
+    channel_proxy->RegisterExternalTransport(
+        stream->timed_send_transport_adapter_.get());
   }
 
   // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
@@ -391,6 +421,10 @@ RtpState AudioSendStream::GetRtpState() const {
   return rtp_rtcp_module_->GetRtpState();
 }
 
+const TimeInterval& AudioSendStream::GetActiveLifetime() const {
+  return active_lifetime_;
+}
+
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());