Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.

webrtc/video/call.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 14 matching lines @@
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/modules/video_render/include/video_render.h"
 #include "webrtc/system_wrappers/interface/cpu_info.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 #include "webrtc/video/audio_receive_stream.h"
+#include "webrtc/video/rtc_event_log.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
+#include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace webrtc {
 
 const int Call::Config::kDefaultStartBitrateBps = 300000;
 
 namespace internal {
 
 class CpuOveruseObserverProxy : public webrtc::CpuOveruseObserver {
  public:
   explicit CpuOveruseObserverProxy(LoadObserver* overuse_callback)
@@ skipping 89 matching lines @@
       GUARDED_BY(receive_crit_);
 
   rtc::scoped_ptr<RWLockWrapper> send_crit_;
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
   std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
 
   rtc::scoped_ptr<CpuOveruseObserverProxy> overuse_observer_proxy_;
 
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
 
+  RtcEventLog* event_log_;
+
   DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
 namespace internal {
 
 Call::Call(const Call::Config& config)
     : num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create()),
       channel_group_(new ChannelGroup(module_process_thread_.get())),
       next_channel_id_(0),
       config_(config),
       network_enabled_(true),
       transport_adapter_(nullptr),
       receive_crit_(RWLockWrapper::CreateRWLock()),
-      send_crit_(RWLockWrapper::CreateRWLock()) {
+      send_crit_(RWLockWrapper::CreateRWLock()),
+      event_log_(nullptr) {
   DCHECK(config.send_transport != nullptr);
 
   DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   DCHECK_GE(config.bitrate_config.start_bitrate_bps,
             config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     DCHECK_GE(config.bitrate_config.max_bitrate_bps,
               config.bitrate_config.start_bitrate_bps);
   }
+  if (config.voice_engine) {
+    VoECodec* voe_codec = VoECodec::GetInterface(config.voice_engine);
+    if (voe_codec) {
+      event_log_ = voe_codec->GetEventLog();
+      voe_codec->Release();
+    }
+  }
 
   Trace::CreateTrace();
   module_process_thread_->Start();
 
   if (config.overuse_callback) {
     overuse_observer_proxy_.reset(
         new CpuOveruseObserverProxy(config.overuse_callback));
   }
 
   SetBitrateControllerConfig(config_.bitrate_config);
@@ skipping 76 matching lines @@
   // This needs to be taken before send_crit_ as both locks need to be held
   // while changing network state.
   rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*send_crit_);
   for (uint32_t ssrc : config.rtp.ssrcs) {
     DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
     video_send_ssrcs_[ssrc] = send_stream;
   }
   video_send_streams_.insert(send_stream);
 
+  if (event_log_)
+    event_log_->LogVideoSendStreamConfig(config);
+
   if (!network_enabled_)
     send_stream->SignalNetworkState(kNetworkDown);
   return send_stream;
 }
 
 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
   DCHECK(send_stream != nullptr);
 
   send_stream->Stop();
@@ skipping 46 matching lines @@
       config.rtp.rtx.begin();
   if (it != config.rtp.rtx.end())
     video_receive_ssrcs_[it->second.ssrc] = receive_stream;
   video_receive_streams_.insert(receive_stream);
 
   ConfigureSync(config.sync_group);
 
   if (!network_enabled_)
     receive_stream->SignalNetworkState(kNetworkDown);
 
+  if (event_log_)
+    event_log_->LogVideoReceiveStreamConfig(config);
+
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
   DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl = nullptr;
   {
     WriteLockScoped write_lock(*receive_crit_);
@@ skipping 141 matching lines @@
                                                  const uint8_t* packet,
                                                  size_t length) {
   // TODO(pbos): Figure out what channel needs it actually.
   //             Do NOT broadcast! Also make sure it's a valid packet.
   //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
   //             there's no receiver of the packet.
   bool rtcp_delivered = false;
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*receive_crit_);
     for (VideoReceiveStream* stream : video_receive_streams_) {
-      if (stream->DeliverRtcp(packet, length))
+      if (stream->DeliverRtcp(packet, length)) {
         rtcp_delivered = true;
+        if (event_log_)
+          event_log_->LogRtcpPacket(true, media_type, packet, length);
+      }
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*send_crit_);
     for (VideoSendStream* stream : video_send_streams_) {
-      if (stream->DeliverRtcp(packet, length))
+      if (stream->DeliverRtcp(packet, length)) {
         rtcp_delivered = true;
+        if (event_log_)
+          event_log_->LogRtcpPacket(false, media_type, packet, length);
+      }
     }
   }
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length) {
   // Minimum RTP header size.
   if (length < 12)
     return DELIVERY_PACKET_ERROR;
 
   uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
 
+  // TODO(ivoc): This should be updated once the splitting of the header is done
+  //             in the RtcEventLog.
+  if (event_log_)
+    event_log_->LogRtpHeader(true, media_type, packet, 20, length);
   ReadLockScoped read_lock(*receive_crit_);
   if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
     auto it = audio_receive_ssrcs_.find(ssrc);
     if (it != audio_receive_ssrcs_.end()) {
       return it->second->DeliverRtp(packet, length) ? DELIVERY_OK
                                                     : DELIVERY_PACKET_ERROR;
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     auto it = video_receive_ssrcs_.find(ssrc);
     if (it != video_receive_ssrcs_.end()) {
       return it->second->DeliverRtp(packet, length) ? DELIVERY_OK
                                                     : DELIVERY_PACKET_ERROR;
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverPacket(MediaType media_type,
                                                    const uint8_t* packet,
                                                    size_t length) {
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length);
 }
 
 }  // namespace internal
 }  // namespace webrtc