Index: webrtc/logging/rtc_event_log/rtc_event_log.h |
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.h b/webrtc/logging/rtc_event_log/rtc_event_log.h |
index 5d221d4ef8e09285c00bfeb2b932b8928b3cd0dc..766fd89bc20bb975a5b998c30c26f8264a383891 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log.h |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.h |
@@ -18,6 +18,7 @@ |
#include "webrtc/call/audio_receive_stream.h" |
#include "webrtc/call/audio_send_stream.h" |
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" |
+#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" |
#include "webrtc/video_receive_stream.h" |
#include "webrtc/video_send_stream.h" |
@@ -115,6 +116,10 @@ class RtcEventLog { |
uint8_t fraction_loss, |
int32_t total_packets) = 0; |
+ // Logs a bitrate update from the bandwidth estimator based on delay changes. |
+ virtual void LogBwePacketDelayEvent(int32_t bitrate, |
+ BandwidthUsage detector_state) = 0; |
+ |
// Logs audio encoder re-configuration driven by audio network adaptor. |
virtual void LogAudioNetworkAdaptation( |
const AudioNetworkAdaptor::EncoderRuntimeConfig& config) = 0; |
@@ -160,8 +165,10 @@ class RtcEventLogNullImpl final : public RtcEventLog { |
void LogBwePacketLossEvent(int32_t bitrate, |
uint8_t fraction_loss, |
int32_t total_packets) override {} |
+ void LogBwePacketDelayEvent(int32_t bitrate, |
+ BandwidthUsage detector_state) override {} |
void LogAudioNetworkAdaptation( |
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override{}; |
+ const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override {} |
}; |
} // namespace webrtc |