| Index: webrtc/call/rtc_event_log.cc
|
| diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc
|
| index 7627d3784bb8630da8d55a48fd6a1f7e159a0b19..840b210d15378dbbdf1ee079886674fe6064d27b 100644
|
| --- a/webrtc/call/rtc_event_log.cc
|
| +++ b/webrtc/call/rtc_event_log.cc
|
| @@ -37,6 +37,40 @@
|
| #endif
|
|
|
| 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
|
|
|
|
|