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Unified Diff: webrtc/logging/rtc_event_log/rtc_event_log2text.cc

Issue 2673403002: New tool for printing basic packet information from an RTC event log to stdout. (Closed)
Patch Set: Created 3 years, 10 months ago
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Index: webrtc/logging/rtc_event_log/rtc_event_log2text.cc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
new file mode 100644
index 0000000000000000000000000000000000000000..b560b180c9c5cb48e9726ed8dd2e5e8cfddf2b31
--- /dev/null
+++ b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
@@ -0,0 +1,186 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <iostream>
+// #include <memory>
ivoc 2017/02/06 16:38:11 Please remove if it's not needed.
terelius 2017/02/06 17:09:43 Done.
+#include <sstream>
+#include <string>
+
+#include "gflags/gflags.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/call/call.h"
+#include "webrtc/common_types.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+
+namespace {
+
+DEFINE_bool(noincoming, false, "Excludes incoming packets.");
+DEFINE_bool(nooutgoing, false, "Excludes incoming packets.");
ivoc 2017/02/06 16:38:11 I assume this should be "Excludes outgoing packets
terelius 2017/02/06 17:09:43 Done.
+// TODO(terelius): Note that the media type don't work with outgoing packets.
ivoc 2017/02/06 16:38:11 doesn't
terelius 2017/02/06 17:09:43 Done.
+DEFINE_bool(noaudio, false, "Excludes audio packets.");
+// TODO(terelius): Note that the media type don't work with outgoing packets.
+DEFINE_bool(novideo, false, "Excludes video packets.");
+// TODO(terelius): Note that the media type don't work with outgoing packets.
+DEFINE_bool(nodata, false, "Excludes data packets.");
+DEFINE_bool(nortp, false, "Excludes RTP packets.");
+DEFINE_bool(nortcp, false, "Excludes RTCP packets.");
+// TODO(terelius): Allow a list of SSRCs.
+DEFINE_string(ssrc,
+ "",
+ "Print only packets with this SSRC (decimal or hex, the latter "
+ "starting with 0x).");
+
+// Parses the input string for a valid SSRC. If a valid SSRC is found, it is
+// written to the output variable |ssrc|, and true is returned. Otherwise,
+// false is returned.
+// The empty string must be validated as true, because it is the default value
+// of the command-line flag. In this case, no value is written to the output
+// variable.
+bool ParseSsrc(std::string str, uint32_t* ssrc) {
+ // If the input string starts with 0x or 0X it indicates a hexadecimal number.
+ auto read_mode = std::dec;
+ if (str.size() > 2 &&
+ (str.substr(0, 2) == "0x" || str.substr(0, 2) == "0X")) {
+ read_mode = std::hex;
+ str = str.substr(2);
+ }
+ std::stringstream ss(str);
+ ss >> read_mode >> *ssrc;
+ return str.empty() || (!ss.fail() && ss.eof());
+}
+
+bool ExcludePacket(webrtc::PacketDirection direction,
+ webrtc::MediaType media_type,
+ uint32_t packet_ssrc,
+ uint32_t filtered_ssrc) {
+ if (FLAGS_nooutgoing && direction == webrtc::kOutgoingPacket)
+ return true;
+ if (FLAGS_noincoming && direction == webrtc::kIncomingPacket)
+ return true;
+ if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
+ return true;
+ if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+ return true;
+ if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
+ return true;
+ if (!FLAGS_ssrc.empty() && packet_ssrc != filtered_ssrc)
+ return true;
+ return false;
+}
+
+std::string StreamInfo(webrtc::PacketDirection direction,
+ webrtc::MediaType media_type) {
+ if (direction == webrtc::kOutgoingPacket) {
+ if (media_type == webrtc::MediaType::AUDIO)
+ return "(out,audio)";
+ if (media_type == webrtc::MediaType::VIDEO)
+ return "(out,video)";
+ if (media_type == webrtc::MediaType::DATA)
+ return "(out,data)";
+ }
+ if (direction == webrtc::kIncomingPacket) {
+ if (media_type == webrtc::MediaType::AUDIO)
+ return "(in,audio)";
+ if (media_type == webrtc::MediaType::VIDEO)
+ return "(in,video)";
+ if (media_type == webrtc::MediaType::DATA)
+ return "(in,data)";
+ }
+ return "()";
ivoc 2017/02/06 16:38:11 I think it would be good to add an RTC_NOTREACHED
terelius 2017/02/06 17:09:43 It isn't unreachable; both direction and media_typ
ivoc 2017/02/07 07:56:52 Acknowledged.
+}
+
+} // namespace
+
+// This utility will convert a stored event log to the rtpdump format.
ivoc 2017/02/06 16:38:11 Please update comment :-)
terelius 2017/02/06 17:09:43 Done.
+int main(int argc, char* argv[]) {
+ std::string program_name = argv[0];
+ std::string usage =
+ "Tool for printing packet information from an RtcEventLog as text.\n"
+ "Run " +
+ program_name +
+ " --helpshort for usage.\n"
+ "Example usage:\n" +
+ program_name + " input.rel\n";
+ google::SetUsageMessage(usage);
+ google::ParseCommandLineFlags(&argc, &argv, true);
+
+ if (argc != 2) {
+ std::cout << google::ProgramUsage();
+ return 0;
+ }
+ std::string input_file = argv[1];
+
+ uint32_t filtered_ssrc = 0;
+ if (!FLAGS_ssrc.empty())
+ RTC_CHECK(ParseSsrc(FLAGS_ssrc, &filtered_ssrc))
+ << "Flag verification has failed.";
+
+ 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++) {
+ // The parsed_stream will assert if the protobuf event is missing
+ // some required fields and we attempt to access them. We could consider
+ // a softer failure option, but it does not seem useful to generate
+ // RTP dumps based on broken event logs.
ivoc 2017/02/06 16:38:11 Please update comment.
terelius 2017/02/06 17:09:43 Done.
+ if (!FLAGS_nortp &&
+ 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::MediaType media_type;
+ parsed_stream.GetRtpHeader(i, &direction, &media_type, header,
+ &header_length, &total_length);
+
+ // 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);
+
+ if (ExcludePacket(direction, media_type, parsed_header.ssrc,
+ filtered_ssrc))
+ continue;
+
+ std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
+ << StreamInfo(direction, media_type)
+ << "\tSSRC=" << parsed_header.ssrc
+ << "\ttimestamp =" << parsed_header.timestamp << std::endl;
+ }
+ if (!FLAGS_nortcp &&
+ parsed_stream.GetEventType(i) ==
+ webrtc::ParsedRtcEventLog::RTCP_EVENT) {
+ size_t length;
+ uint8_t packet[IP_PACKET_SIZE];
+ webrtc::PacketDirection direction;
+ webrtc::MediaType media_type;
+ parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet, &length);
+
+ // Parse header to get SSRC and RTP time.
+ webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet, length);
+ webrtc::RTPHeader parsed_header;
+ rtp_parser.Parse(&parsed_header);
+
+ if (ExcludePacket(direction, media_type, parsed_header.ssrc,
+ filtered_ssrc))
+ continue;
+
+ std::cout << parsed_stream.GetTimestamp(i) << "\tRTCP"
+ << StreamInfo(direction, media_type)
+ << "\tSSRC=" << parsed_header.ssrc
+ << "\ttimestamp =" << parsed_header.timestamp << std::endl;
+ }
+ }
+ return 0;
+}
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