Make rtc_event_log2text handle all events [2/2]
rtc_event_log2text doesn't currently handle all possible RtcEvent-s.
1. Previous CL - to make sure events are not forgotten in the future, change the succession of if-statements to a switch, so that the compiler would complain if events are ever added, but are not handled here.
2. This CL - add handling of currently-unhandled events.
BUG=webrtc:8111
Change-Id: I5c726c077483b5d85cf8060674c8191a90cb84cc
Reviewed-on: https://webrtc-review.googlesource.com/1244
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19869}diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index edba8a8..dc5f8ad 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -167,6 +167,7 @@
# TODO(kwiberg): Remove this dependency.
"../api/audio_codecs:audio_codecs_api",
+ "../modules/audio_coding:audio_network_adaptor_config",
"../modules/rtp_rtcp:rtp_rtcp",
]
if (!build_with_chromium && is_clang) {
diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index 4d96711..7eca099 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/logging/rtc_event_log/rtc_event_log2text.cc
@@ -19,6 +19,7 @@
#include "call/video_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
@@ -40,7 +41,10 @@
namespace {
+DEFINE_bool(unknown, true, "Use --nounknown to exclude unknown events.");
+DEFINE_bool(startstop, true, "Use --nostartstop to exclude start/stop events.");
DEFINE_bool(config, true, "Use --noconfig to exclude stream configurations.");
+DEFINE_bool(bwe, true, "Use --nobwe to exclude BWE events.");
DEFINE_bool(incoming, true, "Use --noincoming to exclude incoming packets.");
DEFINE_bool(outgoing, true, "Use --nooutgoing to exclude packets.");
// TODO(terelius): Note that the media type doesn't work with outgoing packets.
@@ -51,6 +55,10 @@
DEFINE_bool(data, true, "Use --nodata to exclude data packets.");
DEFINE_bool(rtp, true, "Use --nortp to exclude RTP packets.");
DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets.");
+DEFINE_bool(playout, true, "Use --noplayout to exclude audio playout events.");
+DEFINE_bool(ana, true, "Use --noana to exclude ANA events.");
+DEFINE_bool(probe, true, "Use --noprobe to exclude probe events.");
+
// TODO(terelius): Allow a list of SSRCs.
DEFINE_string(ssrc,
"",
@@ -386,20 +394,33 @@
}
for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
+ bool event_recognized = false;
switch (parsed_stream.GetEventType(i)) {
case webrtc::ParsedRtcEventLog::UNKNOWN_EVENT: {
- // TODO(eladalon): Implement in new CL.
- continue;
+ if (FLAG_unknown) {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tUNKNOWN_EVENT"
+ << std::endl;
+ }
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::LOG_START: {
- // TODO(eladalon): Implement in new CL.
- continue;
+ if (FLAG_startstop) {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tLOG_START"
+ << std::endl;
+ }
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::LOG_END: {
- // TODO(eladalon): Implement in new CL.
- continue;
+ if (FLAG_startstop) {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tLOG_END"
+ << std::endl;
+ }
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::RTP_EVENT: {
@@ -422,8 +443,10 @@
MediaType media_type =
parsed_stream.GetMediaType(parsed_header.ssrc, direction);
- if (ExcludePacket(direction, media_type, parsed_header.ssrc))
- continue;
+ if (ExcludePacket(direction, media_type, parsed_header.ssrc)) {
+ event_recognized = true;
+ break;
+ }
std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
<< StreamInfo(direction, media_type)
@@ -456,7 +479,8 @@
}
std::cout << std::endl;
}
- continue;
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::RTCP_EVENT: {
@@ -508,22 +532,47 @@
}
}
}
- continue;
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
- // TODO(eladalon): Implement in new CL.
- continue;
+ if (FLAG_playout) {
+ uint32_t ssrc;
+ parsed_stream.GetAudioPlayout(i, &ssrc);
+ std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_PLAYOUT"
+ << "\tssrc=" << ssrc << std::endl;
+ }
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
- // TODO(eladalon): Implement in new CL.
- continue;
+ if (FLAG_bwe) {
+ int32_t bitrate_bps;
+ uint8_t fraction_loss;
+ int32_t total_packets;
+ parsed_stream.GetLossBasedBweUpdate(i, &bitrate_bps, &fraction_loss,
+ &total_packets);
+ std::cout << parsed_stream.GetTimestamp(i) << "\tBWE(LOSS_BASED)"
+ << "\tbitrate_bps=" << bitrate_bps
+ << "\tfraction_loss=" << fraction_loss
+ << "\ttotal_packets=" << total_packets << std::endl;
+ }
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: {
- // TODO(eladalon): Implement in new CL.
- continue;
+ if (FLAG_bwe) {
+ auto bwe_update = parsed_stream.GetDelayBasedBweUpdate(i);
+ std::cout << parsed_stream.GetTimestamp(i) << "\tBWE(DELAY_BASED)"
+ << "\tbitrate_bps=" << bwe_update.bitrate_bps
+ << "\tdetector_state="
+ << static_cast<int>(bwe_update.detector_state) << std::endl;
+ }
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
@@ -547,7 +596,8 @@
}
std::cout << "}" << std::endl;
}
- continue;
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
@@ -573,7 +623,8 @@
std::cout << "}" << std::endl;
}
}
- continue;
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
@@ -597,7 +648,8 @@
}
std::cout << "}" << std::endl;
}
- continue;
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
@@ -620,26 +672,80 @@
}
std::cout << "}" << std::endl;
}
- continue;
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
- // TODO(eladalon): Implement in new CL.
- continue;
+ if (FLAG_ana) {
+ webrtc::AudioEncoderRuntimeConfig ana_config;
+ parsed_stream.GetAudioNetworkAdaptation(i, &ana_config);
+ std::stringstream ss;
+ ss << parsed_stream.GetTimestamp(i) << "\tANA_UPDATE";
+ if (ana_config.bitrate_bps) {
+ ss << "\tbitrate_bps=" << *ana_config.bitrate_bps;
+ }
+ if (ana_config.frame_length_ms) {
+ ss << "\tframe_length_ms=" << *ana_config.frame_length_ms;
+ }
+ if (ana_config.uplink_packet_loss_fraction) {
+ ss << "\tuplink_packet_loss_fraction="
+ << *ana_config.uplink_packet_loss_fraction;
+ }
+ if (ana_config.enable_fec) {
+ ss << "\tenable_fec=" << *ana_config.enable_fec;
+ }
+ if (ana_config.enable_dtx) {
+ ss << "\tenable_dtx=" << *ana_config.enable_dtx;
+ }
+ if (ana_config.num_channels) {
+ ss << "\tnum_channels=" << *ana_config.num_channels;
+ }
+ std::cout << ss.str() << std::endl;
+ }
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::BWE_PROBE_CLUSTER_CREATED_EVENT: {
- // TODO(eladalon): Implement in new CL.
- continue;
+ if (FLAG_probe) {
+ webrtc::ParsedRtcEventLog::BweProbeClusterCreatedEvent probe_event =
+ parsed_stream.GetBweProbeClusterCreated(i);
+ std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_CREATED("
+ << probe_event.id << ")"
+ << "\tbitrate_bps=" << probe_event.bitrate_bps
+ << "\tmin_packets=" << probe_event.min_packets
+ << "\tmin_bytes=" << probe_event.min_bytes << std::endl;
+ }
+ event_recognized = true;
+ break;
}
case webrtc::ParsedRtcEventLog::BWE_PROBE_RESULT_EVENT: {
- // TODO(eladalon): Implement in new CL.
- continue;
+ if (FLAG_probe) {
+ webrtc::ParsedRtcEventLog::BweProbeResultEvent probe_result =
+ parsed_stream.GetBweProbeResult(i);
+ if (probe_result.failure_reason) {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_SUCCESS("
+ << probe_result.id << ")"
+ << "\tfailure_reason=" << *probe_result.failure_reason
+ << std::endl;
+ } else {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_SUCCESS("
+ << probe_result.id << ")"
+ << "\tbitrate_bps=" << *probe_result.bitrate_bps
+ << std::endl;
+ }
+ }
+ event_recognized = true;
+ break;
}
}
- RTC_NOTREACHED();
+ if (!event_recognized) {
+ std::cout << "Unrecognized event (" << parsed_stream.GetEventType(i)
+ << ")" << std::endl;
+ }
}
return 0;
}
diff --git a/logging/rtc_event_log/rtc_event_log_parser.cc b/logging/rtc_event_log/rtc_event_log_parser.cc
index 87e35eb..cb4fd80 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -75,7 +75,6 @@
case rtclog::Event::BWE_PROBE_RESULT_EVENT:
return ParsedRtcEventLog::EventType::BWE_PROBE_RESULT_EVENT;
}
- RTC_NOTREACHED();
return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
}
