| Index: webrtc/call/rtc_event_log.h
|
| diff --git a/webrtc/call/rtc_event_log.h b/webrtc/call/rtc_event_log.h
|
| index 7c72dd5ce995a4443a25af429154c4ce7f651bab..26496e3984424829510492c94ee5e34906caf5a9 100644
|
| --- a/webrtc/call/rtc_event_log.h
|
| +++ b/webrtc/call/rtc_event_log.h
|
| @@ -14,6 +14,7 @@
|
| #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"
|
| @@ -109,6 +110,40 @@ class RtcEventLog {
|
| 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_
|
|
|