Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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Mirko Bonadei
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575
logging/rtc_event_log/rtc_event_log2text.cc
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575
logging/rtc_event_log/rtc_event_log2text.cc
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@ -0,0 +1,575 @@
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/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <string.h>
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#include <iostream>
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#include <map>
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#include <sstream>
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#include <string>
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#include <utility> // pair
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#include "webrtc/call/video_config.h"
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#include "webrtc/common_types.h"
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#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/flags.h"
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namespace {
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DEFINE_bool(config, true, "Use --noconfig to exclude stream configurations.");
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DEFINE_bool(incoming, true, "Use --noincoming to exclude incoming packets.");
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DEFINE_bool(outgoing, true, "Use --nooutgoing to exclude packets.");
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// TODO(terelius): Note that the media type doesn't work with outgoing packets.
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DEFINE_bool(audio, true, "Use --noaudio to exclude audio packets.");
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// TODO(terelius): Note that the media type doesn't work with outgoing packets.
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DEFINE_bool(video, true, "Use --novideo to exclude video packets.");
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// TODO(terelius): Note that the media type doesn't work with outgoing packets.
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DEFINE_bool(data, true, "Use --nodata to exclude data packets.");
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DEFINE_bool(rtp, true, "Use --nortp to exclude RTP packets.");
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DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets.");
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// TODO(terelius): Allow a list of SSRCs.
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DEFINE_string(ssrc,
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"",
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"Print only packets with this SSRC (decimal or hex, the latter "
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"starting with 0x).");
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DEFINE_bool(help, false, "Prints this message.");
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using MediaType = webrtc::ParsedRtcEventLog::MediaType;
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static uint32_t filtered_ssrc = 0;
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// Parses the input string for a valid SSRC. If a valid SSRC is found, it is
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// written to the static global variable |filtered_ssrc|, and true is returned.
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// Otherwise, false is returned.
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// The empty string must be validated as true, because it is the default value
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// of the command-line flag. In this case, no value is written to the output
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// variable.
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bool ParseSsrc(std::string str) {
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// If the input string starts with 0x or 0X it indicates a hexadecimal number.
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auto read_mode = std::dec;
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if (str.size() > 2 &&
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(str.substr(0, 2) == "0x" || str.substr(0, 2) == "0X")) {
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read_mode = std::hex;
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str = str.substr(2);
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}
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std::stringstream ss(str);
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ss >> read_mode >> filtered_ssrc;
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return str.empty() || (!ss.fail() && ss.eof());
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}
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bool ExcludePacket(webrtc::PacketDirection direction,
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MediaType media_type,
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uint32_t packet_ssrc) {
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if (!FLAG_outgoing && direction == webrtc::kOutgoingPacket)
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return true;
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if (!FLAG_incoming && direction == webrtc::kIncomingPacket)
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return true;
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if (!FLAG_audio && media_type == MediaType::AUDIO)
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return true;
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if (!FLAG_video && media_type == MediaType::VIDEO)
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return true;
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if (!FLAG_data && media_type == MediaType::DATA)
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return true;
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if (strlen(FLAG_ssrc) > 0 && packet_ssrc != filtered_ssrc)
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return true;
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return false;
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}
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const char* StreamInfo(webrtc::PacketDirection direction,
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MediaType media_type) {
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if (direction == webrtc::kOutgoingPacket) {
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if (media_type == MediaType::AUDIO)
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return "(out,audio)";
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else if (media_type == MediaType::VIDEO)
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return "(out,video)";
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else if (media_type == MediaType::DATA)
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return "(out,data)";
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else
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return "(out)";
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}
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if (direction == webrtc::kIncomingPacket) {
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if (media_type == MediaType::AUDIO)
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return "(in,audio)";
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else if (media_type == MediaType::VIDEO)
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return "(in,video)";
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else if (media_type == MediaType::DATA)
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return "(in,data)";
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else
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return "(in)";
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}
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return "(unknown)";
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}
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// Return default values for header extensions, to use on streams without stored
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// mapping data. Currently this only applies to audio streams, since the mapping
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// is not stored in the event log.
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// TODO(ivoc): Remove this once this mapping is stored in the event log for
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// audio streams. Tracking bug: webrtc:6399
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webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
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webrtc::RtpHeaderExtensionMap default_map;
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default_map.Register<webrtc::AudioLevel>(
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webrtc::RtpExtension::kAudioLevelDefaultId);
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default_map.Register<webrtc::TransmissionOffset>(
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webrtc::RtpExtension::kTimestampOffsetDefaultId);
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default_map.Register<webrtc::AbsoluteSendTime>(
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webrtc::RtpExtension::kAbsSendTimeDefaultId);
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default_map.Register<webrtc::VideoOrientation>(
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webrtc::RtpExtension::kVideoRotationDefaultId);
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default_map.Register<webrtc::VideoContentTypeExtension>(
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webrtc::RtpExtension::kVideoContentTypeDefaultId);
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default_map.Register<webrtc::VideoTimingExtension>(
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webrtc::RtpExtension::kVideoTimingDefaultId);
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default_map.Register<webrtc::TransportSequenceNumber>(
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webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
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default_map.Register<webrtc::PlayoutDelayLimits>(
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webrtc::RtpExtension::kPlayoutDelayDefaultId);
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return default_map;
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}
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void PrintSenderReport(const webrtc::ParsedRtcEventLog& parsed_stream,
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const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction) {
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webrtc::rtcp::SenderReport sr;
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if (!sr.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(sr.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, sr.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_SR" << StreamInfo(direction, media_type)
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<< "\tssrc=" << sr.sender_ssrc()
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<< "\ttimestamp=" << sr.rtp_timestamp() << std::endl;
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}
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void PrintReceiverReport(const webrtc::ParsedRtcEventLog& parsed_stream,
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const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction) {
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webrtc::rtcp::ReceiverReport rr;
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if (!rr.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(rr.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, rr.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_RR" << StreamInfo(direction, media_type)
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<< "\tssrc=" << rr.sender_ssrc() << std::endl;
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}
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void PrintXr(const webrtc::ParsedRtcEventLog& parsed_stream,
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const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction) {
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webrtc::rtcp::ExtendedReports xr;
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if (!xr.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(xr.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, xr.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_XR" << StreamInfo(direction, media_type)
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<< "\tssrc=" << xr.sender_ssrc() << std::endl;
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}
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void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction) {
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std::cout << log_timestamp << "\t"
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<< "RTCP_SDES" << StreamInfo(direction, MediaType::ANY)
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<< std::endl;
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RTC_NOTREACHED() << "SDES should have been redacted when writing the log";
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}
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void PrintBye(const webrtc::ParsedRtcEventLog& parsed_stream,
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const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction) {
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webrtc::rtcp::Bye bye;
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if (!bye.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(bye.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, bye.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_BYE" << StreamInfo(direction, media_type)
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<< "\tssrc=" << bye.sender_ssrc() << std::endl;
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}
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void PrintRtpFeedback(const webrtc::ParsedRtcEventLog& parsed_stream,
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const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction) {
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switch (rtcp_block.fmt()) {
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case webrtc::rtcp::Nack::kFeedbackMessageType: {
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webrtc::rtcp::Nack nack;
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if (!nack.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(nack.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, nack.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_NACK" << StreamInfo(direction, media_type)
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<< "\tssrc=" << nack.sender_ssrc() << std::endl;
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break;
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}
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case webrtc::rtcp::Tmmbr::kFeedbackMessageType: {
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webrtc::rtcp::Tmmbr tmmbr;
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if (!tmmbr.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(tmmbr.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, tmmbr.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_TMMBR" << StreamInfo(direction, media_type)
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<< "\tssrc=" << tmmbr.sender_ssrc() << std::endl;
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break;
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}
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case webrtc::rtcp::Tmmbn::kFeedbackMessageType: {
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webrtc::rtcp::Tmmbn tmmbn;
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if (!tmmbn.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(tmmbn.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, tmmbn.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_TMMBN" << StreamInfo(direction, media_type)
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<< "\tssrc=" << tmmbn.sender_ssrc() << std::endl;
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break;
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}
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case webrtc::rtcp::RapidResyncRequest::kFeedbackMessageType: {
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webrtc::rtcp::RapidResyncRequest sr_req;
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if (!sr_req.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(sr_req.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, sr_req.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_SRREQ" << StreamInfo(direction, media_type)
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<< "\tssrc=" << sr_req.sender_ssrc() << std::endl;
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break;
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}
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case webrtc::rtcp::TransportFeedback::kFeedbackMessageType: {
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webrtc::rtcp::TransportFeedback transport_feedback;
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if (!transport_feedback.Parse(rtcp_block))
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return;
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MediaType media_type = parsed_stream.GetMediaType(
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transport_feedback.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type,
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transport_feedback.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_NEWFB" << StreamInfo(direction, media_type)
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<< "\tssrc=" << transport_feedback.sender_ssrc() << std::endl;
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break;
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}
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default:
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break;
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}
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}
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void PrintPsFeedback(const webrtc::ParsedRtcEventLog& parsed_stream,
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const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction) {
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switch (rtcp_block.fmt()) {
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case webrtc::rtcp::Pli::kFeedbackMessageType: {
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webrtc::rtcp::Pli pli;
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if (!pli.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(pli.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, pli.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_PLI" << StreamInfo(direction, media_type)
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<< "\tssrc=" << pli.sender_ssrc() << std::endl;
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break;
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}
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case webrtc::rtcp::Fir::kFeedbackMessageType: {
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webrtc::rtcp::Fir fir;
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if (!fir.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(fir.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, fir.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_FIR" << StreamInfo(direction, media_type)
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<< "\tssrc=" << fir.sender_ssrc() << std::endl;
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break;
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}
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case webrtc::rtcp::Remb::kFeedbackMessageType: {
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webrtc::rtcp::Remb remb;
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if (!remb.Parse(rtcp_block))
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return;
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MediaType media_type =
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parsed_stream.GetMediaType(remb.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, remb.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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<< "RTCP_REMB" << StreamInfo(direction, media_type)
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<< "\tssrc=" << remb.sender_ssrc() << std::endl;
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break;
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}
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default:
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break;
|
||||
}
|
||||
}
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} // namespace
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// This utility will print basic information about each packet to stdout.
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// Note that parser will assert if the protobuf event is missing some required
|
||||
// fields and we attempt to access them. We don't handle this at the moment.
|
||||
int main(int argc, char* argv[]) {
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std::string program_name = argv[0];
|
||||
std::string usage =
|
||||
"Tool for printing packet information from an RtcEventLog as text.\n"
|
||||
"Run " +
|
||||
program_name +
|
||||
" --help for usage.\n"
|
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"Example usage:\n" +
|
||||
program_name + " input.rel\n";
|
||||
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
|
||||
FLAG_help || argc != 2) {
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std::cout << usage;
|
||||
if (FLAG_help) {
|
||||
rtc::FlagList::Print(nullptr, false);
|
||||
return 0;
|
||||
}
|
||||
return 1;
|
||||
}
|
||||
std::string input_file = argv[1];
|
||||
|
||||
if (strlen(FLAG_ssrc) > 0)
|
||||
RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed.";
|
||||
|
||||
webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap();
|
||||
|
||||
webrtc::ParsedRtcEventLog parsed_stream;
|
||||
if (!parsed_stream.ParseFile(input_file)) {
|
||||
std::cerr << "Error while parsing input file: " << input_file << std::endl;
|
||||
return -1;
|
||||
}
|
||||
|
||||
for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
|
||||
if (FLAG_config && FLAG_video && FLAG_incoming &&
|
||||
parsed_stream.GetEventType(i) ==
|
||||
webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
|
||||
webrtc::rtclog::StreamConfig config =
|
||||
parsed_stream.GetVideoReceiveConfig(i);
|
||||
std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
|
||||
<< "\tssrc=" << config.remote_ssrc
|
||||
<< "\tfeedback_ssrc=" << config.local_ssrc;
|
||||
std::cout << "\textensions={";
|
||||
for (const auto& extension : config.rtp_extensions) {
|
||||
std::cout << extension.ToString() << ",";
|
||||
}
|
||||
std::cout << "}";
|
||||
std::cout << "\tcodecs={";
|
||||
for (const auto& codec : config.codecs) {
|
||||
std::cout << "{name: " << codec.payload_name
|
||||
<< ", payload_type: " << codec.payload_type
|
||||
<< ", rtx_payload_type: " << codec.rtx_payload_type << "}";
|
||||
}
|
||||
std::cout << "}" << std::endl;
|
||||
}
|
||||
if (FLAG_config && FLAG_video && FLAG_outgoing &&
|
||||
parsed_stream.GetEventType(i) ==
|
||||
webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
|
||||
std::vector<webrtc::rtclog::StreamConfig> configs =
|
||||
parsed_stream.GetVideoSendConfig(i);
|
||||
for (const auto& config : configs) {
|
||||
std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
|
||||
std::cout << "\tssrcs=" << config.local_ssrc;
|
||||
std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
|
||||
std::cout << "\textensions={";
|
||||
for (const auto& extension : config.rtp_extensions) {
|
||||
std::cout << extension.ToString() << ",";
|
||||
}
|
||||
std::cout << "}";
|
||||
std::cout << "\tcodecs={";
|
||||
for (const auto& codec : config.codecs) {
|
||||
std::cout << "{name: " << codec.payload_name
|
||||
<< ", payload_type: " << codec.payload_type
|
||||
<< ", rtx_payload_type: " << codec.rtx_payload_type << "}";
|
||||
}
|
||||
std::cout << "}" << std::endl;
|
||||
}
|
||||
}
|
||||
if (FLAG_config && FLAG_audio && FLAG_incoming &&
|
||||
parsed_stream.GetEventType(i) ==
|
||||
webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
|
||||
webrtc::rtclog::StreamConfig config =
|
||||
parsed_stream.GetAudioReceiveConfig(i);
|
||||
std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
|
||||
<< "\tssrc=" << config.remote_ssrc
|
||||
<< "\tfeedback_ssrc=" << config.local_ssrc;
|
||||
std::cout << "\textensions={";
|
||||
for (const auto& extension : config.rtp_extensions) {
|
||||
std::cout << extension.ToString() << ",";
|
||||
}
|
||||
std::cout << "}";
|
||||
std::cout << "\tcodecs={";
|
||||
for (const auto& codec : config.codecs) {
|
||||
std::cout << "{name: " << codec.payload_name
|
||||
<< ", payload_type: " << codec.payload_type
|
||||
<< ", rtx_payload_type: " << codec.rtx_payload_type << "}";
|
||||
}
|
||||
std::cout << "}" << std::endl;
|
||||
}
|
||||
if (FLAG_config && FLAG_audio && FLAG_outgoing &&
|
||||
parsed_stream.GetEventType(i) ==
|
||||
webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
|
||||
webrtc::rtclog::StreamConfig config = parsed_stream.GetAudioSendConfig(i);
|
||||
std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
|
||||
<< "\tssrc=" << config.local_ssrc;
|
||||
std::cout << "\textensions={";
|
||||
for (const auto& extension : config.rtp_extensions) {
|
||||
std::cout << extension.ToString() << ",";
|
||||
}
|
||||
std::cout << "}";
|
||||
std::cout << "\tcodecs={";
|
||||
for (const auto& codec : config.codecs) {
|
||||
std::cout << "{name: " << codec.payload_name
|
||||
<< ", payload_type: " << codec.payload_type
|
||||
<< ", rtx_payload_type: " << codec.rtx_payload_type << "}";
|
||||
}
|
||||
std::cout << "}" << std::endl;
|
||||
}
|
||||
if (FLAG_rtp &&
|
||||
parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
|
||||
size_t header_length;
|
||||
size_t total_length;
|
||||
uint8_t header[IP_PACKET_SIZE];
|
||||
webrtc::PacketDirection direction;
|
||||
webrtc::RtpHeaderExtensionMap* extension_map = parsed_stream.GetRtpHeader(
|
||||
i, &direction, header, &header_length, &total_length);
|
||||
|
||||
if (extension_map == nullptr)
|
||||
extension_map = &default_map;
|
||||
|
||||
// Parse header to get SSRC and RTP time.
|
||||
webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
|
||||
webrtc::RTPHeader parsed_header;
|
||||
rtp_parser.Parse(&parsed_header, extension_map);
|
||||
MediaType media_type =
|
||||
parsed_stream.GetMediaType(parsed_header.ssrc, direction);
|
||||
|
||||
if (ExcludePacket(direction, media_type, parsed_header.ssrc))
|
||||
continue;
|
||||
|
||||
std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
|
||||
<< StreamInfo(direction, media_type)
|
||||
<< "\tssrc=" << parsed_header.ssrc
|
||||
<< "\ttimestamp=" << parsed_header.timestamp;
|
||||
if (parsed_header.extension.hasAbsoluteSendTime) {
|
||||
std::cout << "\tAbsSendTime="
|
||||
<< parsed_header.extension.absoluteSendTime;
|
||||
}
|
||||
if (parsed_header.extension.hasVideoContentType) {
|
||||
std::cout << "\tContentType="
|
||||
<< static_cast<int>(parsed_header.extension.videoContentType);
|
||||
}
|
||||
if (parsed_header.extension.hasVideoRotation) {
|
||||
std::cout << "\tRotation="
|
||||
<< static_cast<int>(parsed_header.extension.videoRotation);
|
||||
}
|
||||
if (parsed_header.extension.hasTransportSequenceNumber) {
|
||||
std::cout << "\tTransportSeq="
|
||||
<< parsed_header.extension.transportSequenceNumber;
|
||||
}
|
||||
if (parsed_header.extension.hasTransmissionTimeOffset) {
|
||||
std::cout << "\tTransmTimeOffset="
|
||||
<< parsed_header.extension.transmissionTimeOffset;
|
||||
}
|
||||
if (parsed_header.extension.hasAudioLevel) {
|
||||
std::cout << "\tAudioLevel=" << parsed_header.extension.audioLevel;
|
||||
}
|
||||
std::cout << std::endl;
|
||||
}
|
||||
if (FLAG_rtcp && parsed_stream.GetEventType(i) ==
|
||||
webrtc::ParsedRtcEventLog::RTCP_EVENT) {
|
||||
size_t length;
|
||||
uint8_t packet[IP_PACKET_SIZE];
|
||||
webrtc::PacketDirection direction;
|
||||
parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
|
||||
|
||||
webrtc::rtcp::CommonHeader rtcp_block;
|
||||
const uint8_t* packet_end = packet + length;
|
||||
for (const uint8_t* next_block = packet; next_block != packet_end;
|
||||
next_block = rtcp_block.NextPacket()) {
|
||||
ptrdiff_t remaining_blocks_size = packet_end - next_block;
|
||||
RTC_DCHECK_GT(remaining_blocks_size, 0);
|
||||
if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
|
||||
break;
|
||||
}
|
||||
|
||||
uint64_t log_timestamp = parsed_stream.GetTimestamp(i);
|
||||
switch (rtcp_block.type()) {
|
||||
case webrtc::rtcp::SenderReport::kPacketType:
|
||||
PrintSenderReport(parsed_stream, rtcp_block, log_timestamp,
|
||||
direction);
|
||||
break;
|
||||
case webrtc::rtcp::ReceiverReport::kPacketType:
|
||||
PrintReceiverReport(parsed_stream, rtcp_block, log_timestamp,
|
||||
direction);
|
||||
break;
|
||||
case webrtc::rtcp::Sdes::kPacketType:
|
||||
PrintSdes(rtcp_block, log_timestamp, direction);
|
||||
break;
|
||||
case webrtc::rtcp::ExtendedReports::kPacketType:
|
||||
PrintXr(parsed_stream, rtcp_block, log_timestamp, direction);
|
||||
break;
|
||||
case webrtc::rtcp::Bye::kPacketType:
|
||||
PrintBye(parsed_stream, rtcp_block, log_timestamp, direction);
|
||||
break;
|
||||
case webrtc::rtcp::Rtpfb::kPacketType:
|
||||
PrintRtpFeedback(parsed_stream, rtcp_block, log_timestamp,
|
||||
direction);
|
||||
break;
|
||||
case webrtc::rtcp::Psfb::kPacketType:
|
||||
PrintPsFeedback(parsed_stream, rtcp_block, log_timestamp,
|
||||
direction);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
Reference in New Issue
Block a user