| 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 7469cf7c8529dc8514ec98f1874b204ed1998248..d139c4d780211626dc3aa1d7b957dc98eee8cc7f 100644
|
| --- a/webrtc/logging/rtc_event_log/rtc_event_log.cc
|
| +++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc
|
| @@ -21,7 +21,6 @@
|
| #include "webrtc/base/swap_queue.h"
|
| #include "webrtc/base/thread_checker.h"
|
| #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/remote_bitrate_estimator/include/bwe_defines.h"
|
| @@ -67,16 +66,13 @@ class RtcEventLogImpl final : public RtcEventLog {
|
| void LogAudioReceiveStreamConfig(const rtclog::StreamConfig& config) override;
|
| void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override;
|
| void LogRtpHeader(PacketDirection direction,
|
| - MediaType media_type,
|
| const uint8_t* header,
|
| size_t packet_length) override;
|
| void LogRtpHeader(PacketDirection direction,
|
| - MediaType media_type,
|
| const uint8_t* header,
|
| size_t packet_length,
|
| int probe_cluster_id) override;
|
| void LogRtcpPacket(PacketDirection direction,
|
| - MediaType media_type,
|
| const uint8_t* packet,
|
| size_t length) override;
|
| void LogAudioPlayout(uint32_t ssrc) override;
|
| @@ -132,21 +128,6 @@ rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) {
|
| return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
|
| }
|
|
|
| -rtclog::MediaType ConvertMediaType(MediaType media_type) {
|
| - switch (media_type) {
|
| - case MediaType::ANY:
|
| - return rtclog::MediaType::ANY;
|
| - case MediaType::AUDIO:
|
| - return rtclog::MediaType::AUDIO;
|
| - case MediaType::VIDEO:
|
| - return rtclog::MediaType::VIDEO;
|
| - case MediaType::DATA:
|
| - return rtclog::MediaType::DATA;
|
| - }
|
| - RTC_NOTREACHED();
|
| - return rtclog::ANY;
|
| -}
|
| -
|
| rtclog::DelayBasedBweUpdate::DetectorState ConvertDetectorState(
|
| BandwidthUsage state) {
|
| switch (state) {
|
| @@ -390,15 +371,12 @@ void RtcEventLogImpl::LogAudioSendStreamConfig(
|
| }
|
|
|
| void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
|
| - MediaType media_type,
|
| const uint8_t* header,
|
| size_t packet_length) {
|
| - LogRtpHeader(direction, media_type, header, packet_length,
|
| - PacedPacketInfo::kNotAProbe);
|
| + LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe);
|
| }
|
|
|
| void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
|
| - MediaType media_type,
|
| const uint8_t* header,
|
| size_t packet_length,
|
| int probe_cluster_id) {
|
| @@ -422,7 +400,6 @@ void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
|
| rtp_event->set_timestamp_us(rtc::TimeMicros());
|
| rtp_event->set_type(rtclog::Event::RTP_EVENT);
|
| rtp_event->mutable_rtp_packet()->set_incoming(direction == kIncomingPacket);
|
| - rtp_event->mutable_rtp_packet()->set_type(ConvertMediaType(media_type));
|
| rtp_event->mutable_rtp_packet()->set_packet_length(packet_length);
|
| rtp_event->mutable_rtp_packet()->set_header(header, header_length);
|
| if (probe_cluster_id != PacedPacketInfo::kNotAProbe)
|
| @@ -431,14 +408,12 @@ void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
|
| }
|
|
|
| void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
|
| - MediaType media_type,
|
| const uint8_t* packet,
|
| size_t length) {
|
| std::unique_ptr<rtclog::Event> rtcp_event(new rtclog::Event());
|
| rtcp_event->set_timestamp_us(rtc::TimeMicros());
|
| rtcp_event->set_type(rtclog::Event::RTCP_EVENT);
|
| rtcp_event->mutable_rtcp_packet()->set_incoming(direction == kIncomingPacket);
|
| - rtcp_event->mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
|
|
|
| rtcp::CommonHeader header;
|
| const uint8_t* block_begin = packet;
|
|
|