commit 076724199226d51583c51cca8d875e2d9d7a5dad
author terelius <terelius@webrtc.org> Tue Aug 08 12:40:26 2017
committer Commit Bot <commit-bot@chromium.org> Tue Aug 08 12:40:26 2017
tree d2ada29fe5b2a3375ec5d8bddfb04293a7861a54
parent c87fa15b08ee096f87d8d63c1358b7700bb6697a

Use default header extension map in rtc_event_log2text

Use defaults if the header extension map is missing from the config.

BUG=webrtc:6399

Review-Url: https://codereview.webrtc.org/2983283002
Cr-Original-Commit-Position: refs/heads/master@{#19267}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 007d56229adb8933960bfb08af1090ec20d48072

logging/rtc_event_log/rtc_event_log2text.cc

diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index c17a3b5..c7214e2 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/logging/rtc_event_log/rtc_event_log2text.cc
@@ -16,6 +16,7 @@
 
 #include "gflags/gflags.h"
 #include "webrtc/common_types.h"
+#include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
@@ -31,6 +32,7 @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/rtc_base/checks.h"
 
@@ -119,6 +121,32 @@
   return "(unknown)";
 }
 
+// Return default values for header extensions, to use on streams without stored
+// mapping data. Currently this only applies to audio streams, since the mapping
+// is not stored in the event log.
+// TODO(ivoc): Remove this once this mapping is stored in the event log for
+//             audio streams. Tracking bug: webrtc:6399
+webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
+  webrtc::RtpHeaderExtensionMap default_map;
+  default_map.Register<webrtc::AudioLevel>(
+      webrtc::RtpExtension::kAudioLevelDefaultId);
+  default_map.Register<webrtc::TransmissionOffset>(
+      webrtc::RtpExtension::kTimestampOffsetDefaultId);
+  default_map.Register<webrtc::AbsoluteSendTime>(
+      webrtc::RtpExtension::kAbsSendTimeDefaultId);
+  default_map.Register<webrtc::VideoOrientation>(
+      webrtc::RtpExtension::kVideoRotationDefaultId);
+  default_map.Register<webrtc::VideoContentTypeExtension>(
+      webrtc::RtpExtension::kVideoContentTypeDefaultId);
+  default_map.Register<webrtc::VideoTimingExtension>(
+      webrtc::RtpExtension::kVideoTimingDefaultId);
+  default_map.Register<webrtc::TransportSequenceNumber>(
+      webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
+  default_map.Register<webrtc::PlayoutDelayLimits>(
+      webrtc::RtpExtension::kPlayoutDelayDefaultId);
+  return default_map;
+}
+
 void PrintSenderReport(const webrtc::ParsedRtcEventLog& parsed_stream,
                        const webrtc::rtcp::CommonHeader& rtcp_block,
                        uint64_t log_timestamp,
@@ -344,6 +372,8 @@
   if (!FLAGS_ssrc.empty())
     RTC_CHECK(ParseSsrc(FLAGS_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;
@@ -444,6 +474,9 @@
       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;

rtc_tools/event_log_visualizer/analyzer.cc

diff --git a/rtc_tools/event_log_visualizer/analyzer.cc b/rtc_tools/event_log_visualizer/analyzer.cc
index b42f082..593a3b8 100644
--- a/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/rtc_tools/event_log_visualizer/analyzer.cc
@@ -119,8 +119,20 @@
 webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
   webrtc::RtpHeaderExtensionMap default_map;
   default_map.Register<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
+  default_map.Register<TransmissionOffset>(
+      webrtc::RtpExtension::kTimestampOffsetDefaultId);
   default_map.Register<AbsoluteSendTime>(
       webrtc::RtpExtension::kAbsSendTimeDefaultId);
+  default_map.Register<VideoOrientation>(
+      webrtc::RtpExtension::kVideoRotationDefaultId);
+  default_map.Register<VideoContentTypeExtension>(
+      webrtc::RtpExtension::kVideoContentTypeDefaultId);
+  default_map.Register<VideoTimingExtension>(
+      webrtc::RtpExtension::kVideoTimingDefaultId);
+  default_map.Register<TransportSequenceNumber>(
+      webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
+  default_map.Register<PlayoutDelayLimits>(
+      webrtc::RtpExtension::kPlayoutDelayDefaultId);
   return default_map;
 }