| 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));
|
|
|