| 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_
|
|
|