Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
 #include "webrtc/voice_engine/channel.h"
 
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
+#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/output_mixer.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
 namespace voe {
 
 const int kTelephoneEventAttenuationdB = 10;
 
+class RtcEventLogProxy final : public webrtc::RtcEventLog {
+ public:
+  RtcEventLogProxy() : event_log_(nullptr) {}
+
+  void SetBufferDuration(int64_t buffer_duration_us) override {
+    RTC_NOTREACHED();
+  }
+
+  void StartLogging(const std::string& file_name, int duration_ms) override {
+    RTC_NOTREACHED();
+  }
+
+  bool StartLogging(rtc::PlatformFile log_file,
+                    int64_t max_size_bytes) override {
+    RTC_NOTREACHED();
+    return false;
+  }
+
+  void StopLogging() override { RTC_NOTREACHED(); }
+
+  void LogVideoReceiveStreamConfig(
+      const webrtc::VideoReceiveStream::Config& config) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogVideoReceiveStreamConfig(config);
+    }
+  }
+
+  void LogVideoSendStreamConfig(
+      const webrtc::VideoSendStream::Config& config) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogVideoSendStreamConfig(config);
+    }
+  }
+
+  void LogRtpHeader(webrtc::PacketDirection direction,
+                    webrtc::MediaType media_type,
+                    const uint8_t* header,
+                    size_t packet_length) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogRtpHeader(direction, media_type, header, packet_length);
+    }
+  }
+
+  void LogRtcpPacket(webrtc::PacketDirection direction,
+                     webrtc::MediaType media_type,
+                     const uint8_t* packet,
+                     size_t length) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogRtcpPacket(direction, media_type, packet, length);
+    }
+  }
+
+  void LogAudioPlayout(uint32_t ssrc) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogAudioPlayout(ssrc);
+    }
+  }
+
+  void LogBwePacketLossEvent(int32_t bitrate,
+                             uint8_t fraction_loss,
+                             int32_t total_packets) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogBwePacketLossEvent(bitrate, fraction_loss, total_packets);
+    }
+  }
+
+  void SetEventLog(RtcEventLog* event_log) {
+    rtc::CritScope lock(&crit_);
+    event_log_ = event_log;
+  }
+
+ private:
+  rtc::CriticalSection crit_;
+  RtcEventLog* event_log_ GUARDED_BY(crit_);
+  RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogProxy);
+};
+
 class TransportFeedbackProxy : public TransportFeedbackObserver {
  public:
   TransportFeedbackProxy() : feedback_observer_(nullptr) {
     pacer_thread_.DetachFromThread();
     network_thread_.DetachFromThread();
   }
 
   void SetTransportFeedbackObserver(
       TransportFeedbackObserver* feedback_observer) {
     RTC_DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 400 matching lines @@
     return false;
   }
   header.payload_type_frequency =
       rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
   if (header.payload_type_frequency < 0)
     return false;
   return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
 }
 
 int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
-  if (event_log_) {
-    unsigned int ssrc;
-    RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
-    event_log_->LogAudioPlayout(ssrc);
-  }
+  unsigned int ssrc;
+  RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+  event_log_proxy_->LogAudioPlayout(ssrc);
   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
   if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame) ==
       -1) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::GetAudioFrame() PlayoutData10Ms() failed!");
     // In all likelihood, the audio in this frame is garbage. We return an
     // error so that the audio mixer module doesn't add it to the mix. As
     // a result, it won't be played out and the actions skipped here are
     // irrelevant.
     return -1;
@@ skipping 155 matching lines @@
       }
     }
   }
 
   return (highestNeeded);
 }
 
 int32_t Channel::CreateChannel(Channel*& channel,
                                int32_t channelId,
                                uint32_t instanceId,
-                               RtcEventLog* const event_log,
                                const Config& config) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                "Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
                instanceId);
 
-  channel = new Channel(channelId, instanceId, event_log, config);
+  channel = new Channel(channelId, instanceId, config);
   if (channel == NULL) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                  "Channel::CreateChannel() unable to allocate memory for"
                  " channel");
     return -1;
   }
   return 0;
 }
 
 void Channel::PlayNotification(int32_t id, uint32_t durationMs) {
@@ skipping 36 matching lines @@
   assert(id == _outputFileRecorderId);
 
   rtc::CritScope cs(&_fileCritSect);
 
   _outputFileRecording = false;
   WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RecordFileEnded() => output file recorder module is"
                " shutdown");
 }
 
-Channel::Channel(int32_t channelId,
-                 uint32_t instanceId,
-                 RtcEventLog* const event_log,
-                 const Config& config)
+Channel::Channel(int32_t channelId, uint32_t instanceId, const Config& config)
     : _instanceId(instanceId),
       _channelId(channelId),
-      event_log_(event_log),
+      event_log_proxy_(new RtcEventLogProxy()),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(
           new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
       rtp_receive_statistics_(
           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
       rtp_receiver_(
           RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
                                            this,
                                            this,
                                            rtp_payload_registry_.get())),
@@ skipping 80 matching lines @@
   configuration.audio = true;
   configuration.outgoing_transport = this;
   configuration.receive_statistics = rtp_receive_statistics_.get();
   configuration.bandwidth_callback = rtcp_observer_.get();
   if (pacing_enabled_) {
     configuration.paced_sender = rtp_packet_sender_proxy_.get();
     configuration.transport_sequence_number_allocator =
         seq_num_allocator_proxy_.get();
     configuration.transport_feedback_callback = feedback_observer_proxy_.get();
   }
-  configuration.event_log = event_log;
+  configuration.event_log = &(*event_log_proxy_);
 
   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
   _rtpRtcpModule->SetSendingMediaStatus(false);
 
   statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC()));
   rtp_receive_statistics_->RegisterRtcpStatisticsCallback(
       statistics_proxy_.get());
 
   Config audioproc_config;
   audioproc_config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
@@ skipping 2186 matching lines @@
   rtc::CritScope lock(&assoc_send_channel_lock_);
   Channel* channel = associate_send_channel_.channel();
   if (channel && channel->ChannelId() == channel_id) {
     // If this channel is associated with a send channel of the specified
     // Channel ID, disassociate with it.
     ChannelOwner ref(NULL);
     associate_send_channel_ = ref;
   }
 }
 
+void Channel::SetRtcEventLog(RtcEventLog* event_log) {
+  event_log_proxy_->SetEventLog(event_log);
+}
+
 int Channel::RegisterExternalMediaProcessing(ProcessingTypes type,
                                              VoEMediaProcess& processObject) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterExternalMediaProcessing()");
 
   rtc::CritScope cs(&_callbackCritSect);
 
   if (kPlaybackPerChannel == type) {
     if (_outputExternalMediaCallbackPtr) {
       _engineStatisticsPtr->SetLastError(
@@ skipping 493 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc