Index: webrtc/call/rtc_event_log.cc |
diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc |
index 7627d3784bb8630da8d55a48fd6a1f7e159a0b19..840b210d15378dbbdf1ee079886674fe6064d27b 100644 |
--- a/webrtc/call/rtc_event_log.cc |
+++ b/webrtc/call/rtc_event_log.cc |
@@ -37,6 +37,40 @@ |
#endif |
namespace webrtc { |
+ |
+// No-op implementation is used if flag is not set, or in tests. |
+class RtcEventLogNullImpl final : public RtcEventLog { |
+ public: |
+ bool StartLogging(const std::string& file_name, |
+ int64_t max_size_bytes) override { |
+ return false; |
+ } |
+ bool StartLogging(rtc::PlatformFile platform_file, |
+ int64_t max_size_bytes) override { |
+ // The platform_file is open and needs to be closed. |
+ if (!rtc::ClosePlatformFile(platform_file)) { |
+ LOG(LS_ERROR) << "Can't close file."; |
+ } |
+ return false; |
+ } |
+ void StopLogging() override {} |
+ void LogVideoReceiveStreamConfig( |
+ const VideoReceiveStream::Config& config) override {} |
+ void LogVideoSendStreamConfig( |
+ const VideoSendStream::Config& config) override {} |
+ void LogRtpHeader(PacketDirection direction, |
+ MediaType media_type, |
+ const uint8_t* header, |
+ size_t packet_length) override {} |
+ void LogRtcpPacket(PacketDirection direction, |
+ MediaType media_type, |
+ const uint8_t* packet, |
+ size_t length) override {} |
+ void LogAudioPlayout(uint32_t ssrc) override {} |
+ void LogBwePacketLossEvent(int32_t bitrate, |
+ uint8_t fraction_loss, |
+ int32_t total_packets) override {} |
+}; |
#ifdef ENABLE_RTC_EVENT_LOG |