diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc index 840b210d15..7627d3784b 100644 --- a/webrtc/call/rtc_event_log.cc +++ b/webrtc/call/rtc_event_log.cc @@ -38,40 +38,6 @@ namespace webrtc { -// No-op implementation is used if flag is not set, or in tests. -class RtcEventLogNullImpl final : public RtcEventLog { - public: - bool StartLogging(const std::string& file_name, - int64_t max_size_bytes) override { - return false; - } - bool StartLogging(rtc::PlatformFile platform_file, - int64_t max_size_bytes) override { - // The platform_file is open and needs to be closed. - if (!rtc::ClosePlatformFile(platform_file)) { - LOG(LS_ERROR) << "Can't close file."; - } - return false; - } - void StopLogging() override {} - void LogVideoReceiveStreamConfig( - const VideoReceiveStream::Config& config) override {} - void LogVideoSendStreamConfig( - const VideoSendStream::Config& config) override {} - void LogRtpHeader(PacketDirection direction, - MediaType media_type, - const uint8_t* header, - size_t packet_length) override {} - void LogRtcpPacket(PacketDirection direction, - MediaType media_type, - const uint8_t* packet, - size_t length) override {} - void LogAudioPlayout(uint32_t ssrc) override {} - void LogBwePacketLossEvent(int32_t bitrate, - uint8_t fraction_loss, - int32_t total_packets) override {} -}; - #ifdef ENABLE_RTC_EVENT_LOG class RtcEventLogImpl final : public RtcEventLog { diff --git a/webrtc/call/rtc_event_log.h b/webrtc/call/rtc_event_log.h index 7c72dd5ce9..26496e3984 100644 --- a/webrtc/call/rtc_event_log.h +++ b/webrtc/call/rtc_event_log.h @@ -14,6 +14,7 @@ #include #include +#include "webrtc/base/logging.h" #include "webrtc/base/platform_file.h" #include "webrtc/video_receive_stream.h" #include "webrtc/video_send_stream.h" @@ -109,6 +110,40 @@ class RtcEventLog { rtclog::EventStream* result); }; +// No-op implementation is used if flag is not set, or in tests. +class RtcEventLogNullImpl final : public RtcEventLog { + public: + bool StartLogging(const std::string& file_name, + int64_t max_size_bytes) override { + return false; + } + bool StartLogging(rtc::PlatformFile platform_file, + int64_t max_size_bytes) override { + // The platform_file is open and needs to be closed. + if (!rtc::ClosePlatformFile(platform_file)) { + LOG(LS_ERROR) << "Can't close file."; + } + return false; + } + void StopLogging() override {} + void LogVideoReceiveStreamConfig( + const VideoReceiveStream::Config& config) override {} + void LogVideoSendStreamConfig( + const VideoSendStream::Config& config) override {} + void LogRtpHeader(PacketDirection direction, + MediaType media_type, + const uint8_t* header, + size_t packet_length) override {} + void LogRtcpPacket(PacketDirection direction, + MediaType media_type, + const uint8_t* packet, + size_t length) override {} + void LogAudioPlayout(uint32_t ssrc) override {} + void LogBwePacketLossEvent(int32_t bitrate, + uint8_t fraction_loss, + int32_t total_packets) override {} +}; + } // namespace webrtc #endif // WEBRTC_CALL_RTC_EVENT_LOG_H_ diff --git a/webrtc/modules/congestion_controller/congestion_controller.cc b/webrtc/modules/congestion_controller/congestion_controller.cc index b4e5c1486c..3bb55cba58 100644 --- a/webrtc/modules/congestion_controller/congestion_controller.cc +++ b/webrtc/modules/congestion_controller/congestion_controller.cc @@ -304,7 +304,7 @@ void CongestionController::SignalNetworkState(NetworkState state) { void CongestionController::OnSentPacket(const rtc::SentPacket& sent_packet) { transport_feedback_adapter_.OnSentPacket(sent_packet.packet_id, - sent_packet.send_time_ms); + sent_packet.send_time_ms); } void CongestionController::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { diff --git a/webrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc b/webrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc index aa0d62e138..6fb5f9ba1f 100644 --- a/webrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc +++ b/webrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc @@ -26,42 +26,13 @@ class NullBitrateObserver : public CongestionController::Observer, uint32_t bitrate) override {} }; -class NullEventLog : public RtcEventLog { - public: - ~NullEventLog() override {} - bool StartLogging(const std::string& file_name, - int64_t max_size_bytes) override { - return true; - } - bool StartLogging(rtc::PlatformFile platform_file, int64_t max_size_bytes) { - return true; - } - void StopLogging() override{}; - void LogVideoReceiveStreamConfig( - const webrtc::VideoReceiveStream::Config& config) override {} - void LogVideoSendStreamConfig( - const webrtc::VideoSendStream::Config& config) override {} - void LogRtpHeader(PacketDirection direction, - MediaType media_type, - const uint8_t* header, - size_t packet_length) override {} - void LogRtcpPacket(PacketDirection direction, - MediaType media_type, - const uint8_t* packet, - size_t length) override {} - void LogAudioPlayout(uint32_t ssrc) override {} - void LogBwePacketLossEvent(int32_t bitrate, - uint8_t fraction_loss, - int32_t total_packets) override {} -}; - void FuzzOneInput(const uint8_t* data, size_t size) { size_t i = 0; if (size < sizeof(int64_t) + sizeof(uint8_t) + sizeof(uint32_t)) return; SimulatedClock clock(data[i++]); NullBitrateObserver observer; - NullEventLog event_log; + RtcEventLogNullImpl event_log; CongestionController cc(&clock, &observer, &observer, &event_log); RemoteBitrateEstimator* rbe = cc.GetRemoteBitrateEstimator(true); RTPHeader header; diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn index abf0ca0482..fb77066c4b 100644 --- a/webrtc/tools/BUILD.gn +++ b/webrtc/tools/BUILD.gn @@ -173,6 +173,7 @@ source_set("agc_test_utils") { if (rtc_include_tests) { if (rtc_enable_protobuf) { executable("event_log_visualizer") { + testonly = true sources = [ "event_log_visualizer/analyzer.cc", "event_log_visualizer/analyzer.h", @@ -194,8 +195,10 @@ if (rtc_include_tests) { defines = [ "ENABLE_RTC_EVENT_LOG" ] deps = [ "../:rtc_event_log_parser", + "../modules/congestion_controller:congestion_controller", "../modules/rtp_rtcp:rtp_rtcp", "../system_wrappers:system_wrappers_default", + "../test:field_trial", "//third_party/gflags", ] } diff --git a/webrtc/tools/DEPS b/webrtc/tools/DEPS index 33df89fe2d..e4b85c8fc8 100644 --- a/webrtc/tools/DEPS +++ b/webrtc/tools/DEPS @@ -3,6 +3,7 @@ include_rules = [ "+webrtc/call", "+webrtc/common_video", "+webrtc/modules/audio_processing", + "+webrtc/modules/congestion_controller", "+webrtc/modules/rtp_rtcp", "+webrtc/system_wrappers", "+webrtc/voice_engine", diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc index e6dd35b6c6..c15de6d7fb 100644 --- a/webrtc/tools/event_log_visualizer/analyzer.cc +++ b/webrtc/tools/event_log_visualizer/analyzer.cc @@ -22,9 +22,12 @@ #include "webrtc/base/checks.h" #include "webrtc/call.h" #include "webrtc/common_types.h" +#include "webrtc/modules/congestion_controller/include/congestion_controller.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" +#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" +#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" #include "webrtc/video_receive_stream.h" #include "webrtc/video_send_stream.h" @@ -92,21 +95,15 @@ bool EventLogAnalyzer::StreamId::operator<(const StreamId& other) const { return true; } if (ssrc_ == other.ssrc_) { - if (media_type_ < other.media_type_) { + if (direction_ < other.direction_) { return true; } - if (media_type_ == other.media_type_) { - if (direction_ < other.direction_) { - return true; - } - } } return false; } bool EventLogAnalyzer::StreamId::operator==(const StreamId& other) const { - return ssrc_ == other.ssrc_ && direction_ == other.direction_ && - media_type_ == other.media_type_; + return ssrc_ == other.ssrc_ && direction_ == other.direction_; } @@ -115,12 +112,11 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) uint64_t first_timestamp = std::numeric_limits::max(); uint64_t last_timestamp = std::numeric_limits::min(); - // Maps a stream identifier consisting of ssrc, direction and MediaType + // Maps a stream identifier consisting of ssrc and direction // to the header extensions used by that stream, std::map extension_maps; PacketDirection direction; - MediaType media_type; uint8_t header[IP_PACKET_SIZE]; size_t header_length; size_t total_length; @@ -140,8 +136,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: { VideoReceiveStream::Config config(nullptr); parsed_log_.GetVideoReceiveConfig(i, &config); - StreamId stream(config.rtp.remote_ssrc, kIncomingPacket, - MediaType::VIDEO); + StreamId stream(config.rtp.remote_ssrc, kIncomingPacket); extension_maps[stream].Erase(); for (size_t j = 0; j < config.rtp.extensions.size(); ++j) { const std::string& extension = config.rtp.extensions[j].uri; @@ -155,7 +150,7 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) VideoSendStream::Config config(nullptr); parsed_log_.GetVideoSendConfig(i, &config); for (auto ssrc : config.rtp.ssrcs) { - StreamId stream(ssrc, kOutgoingPacket, MediaType::VIDEO); + StreamId stream(ssrc, kOutgoingPacket); extension_maps[stream].Erase(); for (size_t j = 0; j < config.rtp.extensions.size(); ++j) { const std::string& extension = config.rtp.extensions[j].uri; @@ -177,13 +172,14 @@ 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); // Parse header to get SSRC. RtpUtility::RtpHeaderParser rtp_parser(header, header_length); RTPHeader parsed_header; rtp_parser.Parse(&parsed_header); - StreamId stream(parsed_header.ssrc, direction, media_type); + StreamId stream(parsed_header.ssrc, direction); // Look up the extension_map and parse it again to get the extensions. if (extension_maps.count(stream) == 1) { RtpHeaderExtensionMap* extension_map = &extension_maps[stream]; @@ -191,10 +187,45 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) } uint64_t timestamp = parsed_log_.GetTimestamp(i); rtp_packets_[stream].push_back( - LoggedRtpPacket(timestamp, parsed_header)); + LoggedRtpPacket(timestamp, parsed_header, total_length)); break; } case ParsedRtcEventLog::RTCP_EVENT: { + uint8_t packet[IP_PACKET_SIZE]; + MediaType media_type; + parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet, + &total_length); + + RtpUtility::RtpHeaderParser rtp_parser(packet, total_length); + RTPHeader parsed_header; + RTC_CHECK(rtp_parser.ParseRtcp(&parsed_header)); + uint32_t ssrc = parsed_header.ssrc; + + RTCPUtility::RTCPParserV2 rtcp_parser(packet, total_length, true); + RTC_CHECK(rtcp_parser.IsValid()); + + RTCPUtility::RTCPPacketTypes packet_type = rtcp_parser.Begin(); + while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) { + switch (packet_type) { + case RTCPUtility::RTCPPacketTypes::kTransportFeedback: { + // Currently feedback is logged twice, both for audio and video. + // Only act on one of them. + if (media_type == MediaType::VIDEO) { + std::unique_ptr rtcp_packet( + rtcp_parser.ReleaseRtcpPacket()); + StreamId stream(ssrc, direction); + uint64_t timestamp = parsed_log_.GetTimestamp(i); + rtcp_packets_[stream].push_back(LoggedRtcpPacket( + timestamp, kRtcpTransportFeedback, std::move(rtcp_packet))); + } + break; + } + default: + break; + } + rtcp_parser.Iterate(); + packet_type = rtcp_parser.PacketType(); + } break; } case ParsedRtcEventLog::LOG_START: { @@ -232,6 +263,33 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log) end_time_ = last_timestamp; } +class BitrateObserver : public CongestionController::Observer, + public RemoteBitrateObserver { + public: + BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {} + + void OnNetworkChanged(uint32_t bitrate_bps, + uint8_t fraction_loss, + int64_t rtt_ms) override { + last_bitrate_bps_ = bitrate_bps; + bitrate_updated_ = true; + } + + void OnReceiveBitrateChanged(const std::vector& ssrcs, + uint32_t bitrate) override {} + + uint32_t last_bitrate_bps() const { return last_bitrate_bps_; } + bool GetAndResetBitrateUpdated() { + bool bitrate_updated = bitrate_updated_; + bitrate_updated_ = false; + return bitrate_updated; + } + + private: + uint32_t last_bitrate_bps_; + bool bitrate_updated_; +}; + void EventLogAnalyzer::CreatePacketGraph(PacketDirection desired_direction, Plot* plot) { std::map time_series; @@ -675,5 +733,113 @@ void EventLogAnalyzer::CreateStreamBitrateGraph( } } +void EventLogAnalyzer::CreateBweGraph(Plot* plot) { + std::map outgoing_rtp; + std::map incoming_rtcp; + + for (const auto& kv : rtp_packets_) { + if (kv.first.GetDirection() == PacketDirection::kOutgoingPacket) { + for (const LoggedRtpPacket& rtp_packet : kv.second) + outgoing_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet)); + } + } + + for (const auto& kv : rtcp_packets_) { + if (kv.first.GetDirection() == PacketDirection::kIncomingPacket) { + for (const LoggedRtcpPacket& rtcp_packet : kv.second) + incoming_rtcp.insert( + std::make_pair(rtcp_packet.timestamp, &rtcp_packet)); + } + } + + SimulatedClock clock(0); + BitrateObserver observer; + RtcEventLogNullImpl null_event_log; + CongestionController cc(&clock, &observer, &observer, &null_event_log); + // TODO(holmer): Log the call config and use that here instead. + static const uint32_t kDefaultStartBitrateBps = 300000; + cc.SetBweBitrates(0, kDefaultStartBitrateBps, -1); + + TimeSeries time_series; + time_series.label = "BWE"; + time_series.style = LINE_DOT_GRAPH; + uint32_t max_y = 10; + uint32_t min_y = 0; + + auto rtp_iterator = outgoing_rtp.begin(); + auto rtcp_iterator = incoming_rtcp.begin(); + + auto NextRtpTime = [&]() { + if (rtp_iterator != outgoing_rtp.end()) + return static_cast(rtp_iterator->first); + return std::numeric_limits::max(); + }; + + auto NextRtcpTime = [&]() { + if (rtcp_iterator != incoming_rtcp.end()) + return static_cast(rtcp_iterator->first); + return std::numeric_limits::max(); + }; + + auto NextProcessTime = [&]() { + if (rtcp_iterator != incoming_rtcp.end() || + rtp_iterator != outgoing_rtp.end()) { + return clock.TimeInMicroseconds() + + std::max(cc.TimeUntilNextProcess() * 1000, 0); + } + return std::numeric_limits::max(); + }; + + int64_t time_us = std::min(NextRtpTime(), NextRtcpTime()); + while (time_us != std::numeric_limits::max()) { + clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds()); + if (clock.TimeInMicroseconds() >= NextRtcpTime()) { + clock.AdvanceTimeMilliseconds(rtcp_iterator->first / 1000 - + clock.TimeInMilliseconds()); + const LoggedRtcpPacket& rtcp = *rtcp_iterator->second; + if (rtcp.type == kRtcpTransportFeedback) { + cc.GetTransportFeedbackObserver()->OnTransportFeedback( + *static_cast(rtcp.packet.get())); + } + ++rtcp_iterator; + } + if (clock.TimeInMicroseconds() >= NextRtpTime()) { + clock.AdvanceTimeMilliseconds(rtp_iterator->first / 1000 - + clock.TimeInMilliseconds()); + const LoggedRtpPacket& rtp = *rtp_iterator->second; + if (rtp.header.extension.hasTransportSequenceNumber) { + RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber); + cc.GetTransportFeedbackObserver()->AddPacket( + rtp.header.extension.transportSequenceNumber, rtp.total_length, 0); + rtc::SentPacket sent_packet( + rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000); + cc.OnSentPacket(sent_packet); + } + ++rtp_iterator; + } + if (clock.TimeInMicroseconds() >= NextProcessTime()) + cc.Process(); + if (observer.GetAndResetBitrateUpdated()) { + uint32_t y = observer.last_bitrate_bps() / 1000; + max_y = std::max(max_y, y); + min_y = std::min(min_y, y); + float x = static_cast(clock.TimeInMicroseconds() - begin_time_) / + 1000000; + time_series.points.emplace_back(x, y); + } + time_us = std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()}); + } + // Add the data set to the plot. + plot->series.push_back(std::move(time_series)); + + plot->xaxis_min = kDefaultXMin; + plot->xaxis_max = (end_time_ - begin_time_) / 1000000 * kXMargin; + plot->xaxis_label = "Time (s)"; + plot->yaxis_min = min_y - (kYMargin - 1) / 2 * (max_y - min_y); + plot->yaxis_max = max_y + (kYMargin - 1) / 2 * (max_y - min_y); + plot->yaxis_label = "Bitrate (kbps)"; + plot->title = "BWE"; +} + } // namespace plotting } // namespace webrtc diff --git a/webrtc/tools/event_log_visualizer/analyzer.h b/webrtc/tools/event_log_visualizer/analyzer.h index 0b92c10e10..da3b206e10 100644 --- a/webrtc/tools/event_log_visualizer/analyzer.h +++ b/webrtc/tools/event_log_visualizer/analyzer.h @@ -13,8 +13,12 @@ #include #include +#include +#include #include "webrtc/call/rtc_event_log_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" #include "webrtc/tools/event_log_visualizer/plot_base.h" namespace webrtc { @@ -41,30 +45,41 @@ class EventLogAnalyzer { void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot); + void CreateBweGraph(Plot* plot); + private: class StreamId { public: - StreamId(uint32_t ssrc, - webrtc::PacketDirection direction, - webrtc::MediaType media_type) - : ssrc_(ssrc), direction_(direction), media_type_(media_type) {} + StreamId(uint32_t ssrc, webrtc::PacketDirection direction) + : ssrc_(ssrc), direction_(direction) {} bool operator<(const StreamId& other) const; bool operator==(const StreamId& other) const; uint32_t GetSsrc() const { return ssrc_; } webrtc::PacketDirection GetDirection() const { return direction_; } - webrtc::MediaType GetMediaType() const { return media_type_; } private: uint32_t ssrc_; webrtc::PacketDirection direction_; - webrtc::MediaType media_type_; }; struct LoggedRtpPacket { - LoggedRtpPacket(uint64_t timestamp, RTPHeader header) - : timestamp(timestamp), header(header) {} + LoggedRtpPacket(uint64_t timestamp, RTPHeader header, size_t total_length) + : timestamp(timestamp), header(header), total_length(total_length) {} uint64_t timestamp; RTPHeader header; + size_t total_length; + }; + + struct LoggedRtcpPacket { + LoggedRtcpPacket(uint64_t timestamp, + RTCPPacketType rtcp_type, + std::unique_ptr rtcp_packet) + : timestamp(timestamp), + type(rtcp_type), + packet(std::move(rtcp_packet)) {} + uint64_t timestamp; + RTCPPacketType type; + std::unique_ptr packet; }; struct BwePacketLossEvent { @@ -85,6 +100,8 @@ class EventLogAnalyzer { // if the stream has been configured. std::map> rtp_packets_; + std::map> rtcp_packets_; + // A list of all updates from the send-side loss-based bandwidth estimator. std::vector bwe_loss_updates_; diff --git a/webrtc/tools/event_log_visualizer/generate_timeseries.cc b/webrtc/tools/event_log_visualizer/generate_timeseries.cc index d2139475fa..02e9ad3727 100644 --- a/webrtc/tools/event_log_visualizer/generate_timeseries.cc +++ b/webrtc/tools/event_log_visualizer/generate_timeseries.cc @@ -43,6 +43,10 @@ DEFINE_bool(plot_total_bitrate, DEFINE_bool(plot_stream_bitrate, false, "Plot the bitrate used by each stream."); +DEFINE_bool(plot_bwe, + false, + "Run the bandwidth estimator with the logged rtp and rtcp and plot " + "the output."); int main(int argc, char* argv[]) { std::string program_name = argv[0]; @@ -132,6 +136,10 @@ int main(int argc, char* argv[]) { } } + if (FLAGS_plot_all || FLAGS_plot_bwe) { + analyzer.CreateBweGraph(collection->append_new_plot()); + } + collection->draw(); return 0; diff --git a/webrtc/tools/event_log_visualizer/plot_base.h b/webrtc/tools/event_log_visualizer/plot_base.h index 925dcecf3d..0bfdf8d323 100644 --- a/webrtc/tools/event_log_visualizer/plot_base.h +++ b/webrtc/tools/event_log_visualizer/plot_base.h @@ -18,7 +18,7 @@ namespace webrtc { namespace plotting { -enum PlotStyle { LINE_GRAPH, BAR_GRAPH }; +enum PlotStyle { LINE_GRAPH, LINE_DOT_GRAPH, BAR_GRAPH }; struct TimeSeriesPoint { TimeSeriesPoint(float x, float y) : x(x), y(y) {} diff --git a/webrtc/tools/event_log_visualizer/plot_python.cc b/webrtc/tools/event_log_visualizer/plot_python.cc index 916bc2e6a6..211d33679c 100644 --- a/webrtc/tools/event_log_visualizer/plot_python.cc +++ b/webrtc/tools/event_log_visualizer/plot_python.cc @@ -11,6 +11,7 @@ #include "webrtc/tools/event_log_visualizer/plot_python.h" #include +#include namespace webrtc { namespace plotting { @@ -58,6 +59,11 @@ void PythonPlot::draw() { } else if (series[i].style == LINE_GRAPH) { printf("plt.plot(x%zu, y%zu, color=rgb_colors[%zu], label=\'%s\')\n", i, i, i, series[i].label.c_str()); + } else if (series[i].style == LINE_DOT_GRAPH) { + printf( + "plt.plot(x%zu, y%zu, color=rgb_colors[%zu], label=\'%s\', " + "marker='.')\n", + i, i, i, series[i].label.c_str()); } else { printf("raise Exception(\"Unknown graph type\")\n"); } diff --git a/webrtc/tools/tools.gyp b/webrtc/tools/tools.gyp index 5c11370a20..615b93deed 100644 --- a/webrtc/tools/tools.gyp +++ b/webrtc/tools/tools.gyp @@ -110,8 +110,10 @@ 'type': 'executable', 'dependencies': [ '<(webrtc_root)/webrtc.gyp:rtc_event_log_parser', + '<(webrtc_root)/modules/modules.gyp:congestion_controller', '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', '<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default', + '<(webrtc_root)/test/test.gyp:field_trial', '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', ], 'sources': [