Index: webrtc/tools/event_log_visualizer/analyzer.cc |
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc |
index a34d855fe2edebc33d5c2223d5047ef6771fdd10..f42855fa92c1a40bf62a021ebae7bc7e28cd1344 100644 |
--- a/webrtc/tools/event_log_visualizer/analyzer.cc |
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc |
@@ -373,9 +373,8 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) |
break; |
} |
case ParsedRtcEventLog::RTP_EVENT: { |
- MediaType media_type; |
- parsed_log_.GetRtpHeader(i, &direction, &media_type, header, |
- &header_length, &total_length); |
+ parsed_log_.GetRtpHeader(i, &direction, header, &header_length, |
+ &total_length); |
// Parse header to get SSRC. |
RtpUtility::RtpHeaderParser rtp_parser(header, header_length); |
RTPHeader parsed_header; |
@@ -399,9 +398,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) |
} |
case ParsedRtcEventLog::RTCP_EVENT: { |
uint8_t packet[IP_PACKET_SIZE]; |
- MediaType media_type; |
- parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet, |
- &total_length); |
+ parsed_log_.GetRtcpPacket(i, &direction, packet, &total_length); |
// Currently incoming RTCP packets are logged twice, both for audio and |
// video. Only act on one of them. Compare against the previous parsed |
// incoming RTCP packet. |
@@ -905,8 +902,7 @@ void EventLogAnalyzer::CreateTotalBitrateGraph( |
for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) { |
ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i); |
if (event_type == ParsedRtcEventLog::RTP_EVENT) { |
- parsed_log_.GetRtpHeader(i, &direction, nullptr, nullptr, nullptr, |
- &total_length); |
+ parsed_log_.GetRtpHeader(i, &direction, nullptr, nullptr, &total_length); |
if (direction == desired_direction) { |
uint64_t timestamp = parsed_log_.GetTimestamp(i); |
packets.push_back(TimestampSize(timestamp, total_length)); |