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/call/rtc_event_log.h"
 #include "webrtc/common.h"
 #include "webrtc/config.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"
@@ skipping 455 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);
+  {
+    rtc::CritScope lock(&event_log_lock_);
+    if (event_log_) {
+      unsigned int ssrc;
+      RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+      { event_log_->LogAudioPlayout(ssrc); }

[the sun, 2016/03/03 09:25:13]

Remove the { } on this line.

[ivoc, 2016/03/10 13:15:36]

Hmm, not sure what happened there :-)

+    }
   }
   // 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.
@@ skipping 156 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_(nullptr),
       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,
                                            this,
@@ skipping 88 matching lines @@
   configuration.outgoing_transport = this;
   configuration.audio_messages = 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;
 
   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
 
   statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC()));
   rtp_receive_statistics_->RegisterRtcpStatisticsCallback(
       statistics_proxy_.get());
 
   Config audioproc_config;
   audioproc_config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
   rx_audioproc_.reset(AudioProcessing::Create(audioproc_config));
@@ skipping 2784 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