| Index: webrtc/logging/rtc_event_log/rtc_event_log.cc
|
| diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc
|
| index 902ce423436b15633b4146589360a489d2634a7e..b5c907af9b0f1e062f00939ab271fc8cffd92c0e 100644
|
| --- a/webrtc/logging/rtc_event_log/rtc_event_log.cc
|
| +++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc
|
| @@ -21,6 +21,7 @@
|
| #include "webrtc/base/timeutils.h"
|
| #include "webrtc/call/call.h"
|
| #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h"
|
| +#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
|
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
| #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
|
| @@ -28,8 +29,8 @@
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
|
| -#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
|
| +#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
|
| @@ -85,7 +86,7 @@ class RtcEventLogImpl final : public RtcEventLog {
|
| void LogDelayBasedBweUpdate(int32_t bitrate_bps,
|
| BandwidthUsage detector_state) override;
|
| void LogAudioNetworkAdaptation(
|
| - const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override;
|
| + const AudioEncoderRuntimeConfig& config) override;
|
| void LogProbeClusterCreated(int id,
|
| int bitrate_bps,
|
| int min_probes,
|
| @@ -504,7 +505,7 @@ void RtcEventLogImpl::LogDelayBasedBweUpdate(int32_t bitrate_bps,
|
| }
|
|
|
| void RtcEventLogImpl::LogAudioNetworkAdaptation(
|
| - const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
|
| + const AudioEncoderRuntimeConfig& config) {
|
| std::unique_ptr<rtclog::Event> event(new rtclog::Event());
|
| event->set_timestamp_us(rtc::TimeMicros());
|
| event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
|
|
|