| 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..a3359692eb8488dd2796929aaa278476156ee563 100644
|
| --- a/webrtc/call/rtc_event_log.h
|
| +++ b/webrtc/call/rtc_event_log.h
|
| @@ -109,6 +109,34 @@ 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;
|
| + 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_
|
|
|