Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(15)

Unified Diff: webrtc/logging/rtc_event_log/rtc_event_log.h

Issue 2693213005: Rename some variables and methods in RTC event log. (Closed)
Patch Set: Unset-upstream Created 3 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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..f1bbcbba785e79808863accd1eac61c5641fc909 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"
@@ -111,10 +112,14 @@ class RtcEventLog {
virtual void LogAudioPlayout(uint32_t ssrc) = 0;
// Logs a bitrate update from the bandwidth estimator based on packet loss.
- virtual void LogBwePacketLossEvent(int32_t bitrate,
+ virtual void LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets) = 0;
+ // Logs a bitrate update from the bandwidth estimator based on delay changes.
+ virtual void LogDelayBasedBweUpdate(int32_t bitrate_bps,
+ BandwidthUsage detector_state) = 0;
+
// Logs audio encoder re-configuration driven by audio network adaptor.
virtual void LogAudioNetworkAdaptation(
const AudioNetworkAdaptor::EncoderRuntimeConfig& config) = 0;
@@ -157,11 +162,13 @@ class RtcEventLogNullImpl final : public RtcEventLog {
const uint8_t* packet,
size_t length) override {}
void LogAudioPlayout(uint32_t ssrc) override {}
- void LogBwePacketLossEvent(int32_t bitrate,
+ void LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets) override {}
+ void LogDelayBasedBweUpdate(int32_t bitrate_bps,
+ BandwidthUsage detector_state) override {}
void LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override{};
+ const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override {}
};
} // namespace webrtc
« no previous file with comments | « webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h ('k') | webrtc/logging/rtc_event_log/rtc_event_log.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698