This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader. Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead. BUG=webrtc:7538 TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc Review-Url: https://codereview.webrtc.org/2855143002 Cr-Commit-Position: refs/heads/master@{#18324}
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@ -373,9 +373,8 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
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break;
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}
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case ParsedRtcEventLog::RTP_EVENT: {
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MediaType media_type;
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parsed_log_.GetRtpHeader(i, &direction, &media_type, header,
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&header_length, &total_length);
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parsed_log_.GetRtpHeader(i, &direction, header, &header_length,
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&total_length);
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// Parse header to get SSRC.
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RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
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RTPHeader parsed_header;
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@ -399,9 +398,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
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}
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case ParsedRtcEventLog::RTCP_EVENT: {
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uint8_t packet[IP_PACKET_SIZE];
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MediaType media_type;
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parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet,
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&total_length);
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parsed_log_.GetRtcpPacket(i, &direction, packet, &total_length);
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// Currently incoming RTCP packets are logged twice, both for audio and
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// video. Only act on one of them. Compare against the previous parsed
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// incoming RTCP packet.
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@ -905,8 +902,7 @@ void EventLogAnalyzer::CreateTotalBitrateGraph(
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for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
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ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
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if (event_type == ParsedRtcEventLog::RTP_EVENT) {
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parsed_log_.GetRtpHeader(i, &direction, nullptr, nullptr, nullptr,
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&total_length);
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parsed_log_.GetRtpHeader(i, &direction, nullptr, nullptr, &total_length);
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if (direction == desired_direction) {
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uint64_t timestamp = parsed_log_.GetTimestamp(i);
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packets.push_back(TimestampSize(timestamp, total_length));
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