| Index: webrtc/call/rtc_event_log.cc
|
| diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc
|
| index 840b210d15378dbbdf1ee079886674fe6064d27b..7627d3784bb8630da8d55a48fd6a1f7e159a0b19 100644
|
| --- a/webrtc/call/rtc_event_log.cc
|
| +++ b/webrtc/call/rtc_event_log.cc
|
| @@ -38,40 +38,6 @@
|
|
|
| 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
|
|
|
| class RtcEventLogImpl final : public RtcEventLog {
|
|
|