Index: webrtc/call/rtc_event_log.h |
diff --git a/webrtc/call/rtc_event_log.h b/webrtc/call/rtc_event_log.h |
index 26496e3984424829510492c94ee5e34906caf5a9..7c72dd5ce995a4443a25af429154c4ce7f651bab 100644 |
--- a/webrtc/call/rtc_event_log.h |
+++ b/webrtc/call/rtc_event_log.h |
@@ -14,7 +14,6 @@ |
#include <memory> |
#include <string> |
-#include "webrtc/base/logging.h" |
#include "webrtc/base/platform_file.h" |
#include "webrtc/video_receive_stream.h" |
#include "webrtc/video_send_stream.h" |
@@ -110,40 +109,6 @@ |
rtclog::EventStream* result); |
}; |
-// 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 {} |
-}; |
- |
} // namespace webrtc |
#endif // WEBRTC_CALL_RTC_EVENT_LOG_H_ |