Index: webrtc/call/rtc_event_log.cc |
diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc |
index 840b210d15378dbbdf1ee079886674fe6064d27b..7627d3784bb8630da8d55a48fd6a1f7e159a0b19 100644 |
--- a/webrtc/call/rtc_event_log.cc |
+++ b/webrtc/call/rtc_event_log.cc |
@@ -38,40 +38,6 @@ |
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 |
class RtcEventLogImpl final : public RtcEventLog { |