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

webrtc/audio/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 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 33 matching lines @@
 namespace {
 void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy,
                  rtc::FunctionView<void(AudioEncoder*)> lambda) {
   channel_proxy->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
     RTC_DCHECK(encoder_ptr);
     lambda(encoder_ptr->get());
   });
 }
 }  // 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,
     rtc::TaskQueue* worker_queue,
     RtpTransportControllerSendInterface* transport,
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats,
     const rtc::Optional<RtpState>& suspended_rtp_state)
     : worker_queue_(worker_queue),
@@ skipping 67 matching lines @@
     channel_proxy->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
                                  new_config.rtp.nack.rtp_history_ms / 20);
   }
 
   if (first_time ||
       new_config.send_transport != old_config.send_transport) {
     if (old_config.send_transport) {
       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
   // reserved for padding and MUST NOT be used as a local identifier.
   // So it should be safe to use 0 here to indicate "not configured".
   struct ExtensionIds {
     int audio_level = 0;
     int transport_sequence_number = 0;
   };
 
@@ skipping 229 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   transport_->send_side_cc()->SetTransportOverhead(
       transport_overhead_per_packet);
   channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
 }
 
 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());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 
 // Apply current codec settings to a single voe::Channel used for sending.
 bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
@@ skipping 206 matching lines @@
     if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
       LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
                        "RTP/RTCP module";
     }
   }
 }
 
 
 }  // namespace internal
 }  // namespace webrtc