Revert of Tool to convert RtcEventLog files to RtpDump format. (patchset #11 id:200001 of https://codereview.webrtc.org/1297653002/ )
Reason for revert:
Breaks Chromium WebRTC FYI bots.
Updating projects from gyp files...
gyp: /b/build/slave/linux/build/src/third_party/gflags/gflags.gyp not found (cwd: /b/build/slave/linux/build)
Error: Command '/usr/bin/python src/build/gyp_chromium' returned non-zero exit status 1 in /b/build/slave/linux/build
Original issue's description:
> Tool to convert RtcEventLog files to RtpDump format.
>
> This is a small utility that reads RtcEventLog files, and converts the RTP headers within it to RtpDump format. All other types of events are ignored. Three command-line flags are supported, --audio-only, --video-only and --data-only. When one of these flags is supplied, only RTP packets that match the requested type are converted.
>
> BUG=webrtc:4741
> R=henrik.lundin@webrtc.org, kjellander@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/35624c2c3686a2ad40daffe073aa78507b0ef88e
> Cr-Commit-Position: refs/heads/master@{#9980}
TBR=henrik.lundin@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,kjellander@webrtc.org,kjellander@google.com,ivoc@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1345983009
Cr-Commit-Position: refs/heads/master@{#9987}
diff --git a/webrtc/test/rtp_file_writer.cc b/webrtc/test/rtp_file_writer.cc
index d9e0586..793e51a 100644
--- a/webrtc/test/rtp_file_writer.cc
+++ b/webrtc/test/rtp_file_writer.cc
@@ -40,6 +40,7 @@
bool WritePacket(const RtpPacket* packet) override {
uint16_t len = static_cast<uint16_t>(packet->length + kPacketHeaderSize);
+ RTC_CHECK_GE(packet->original_length, packet->length);
uint16_t plen = static_cast<uint16_t>(packet->original_length);
uint32_t offset = packet->time_ms;
RTC_CHECK(WriteUint16(len));
diff --git a/webrtc/video/rtc_event_log2rtp_dump.cc b/webrtc/video/rtc_event_log2rtp_dump.cc
deleted file mode 100644
index 4f1d93b..0000000
--- a/webrtc/video/rtc_event_log2rtp_dump.cc
+++ /dev/null
@@ -1,207 +0,0 @@
-/*
- * 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 <sstream>
-#include <string>
-
-#include "gflags/gflags.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/test/rtp_file_writer.h"
-#include "webrtc/video/rtc_event_log.h"
-
-// Files generated at build-time by the protobuf compiler.
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/video/rtc_event_log.pb.h"
-#else
-#include "webrtc/video/rtc_event_log.pb.h"
-#endif
-
-namespace {
-
-DEFINE_bool(noaudio,
- false,
- "Excludes audio packets from the converted RTPdump file.");
-DEFINE_bool(novideo,
- false,
- "Excludes video packets from the converted RTPdump file.");
-DEFINE_bool(nodata,
- false,
- "Excludes data packets from the converted RTPdump file.");
-DEFINE_bool(nortp,
- false,
- "Excludes RTP packets from the converted RTPdump file.");
-DEFINE_bool(nortcp,
- false,
- "Excludes RTCP packets from the converted RTPdump file.");
-DEFINE_string(ssrc,
- "",
- "Store 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());
-}
-
-} // namespace
-
-// This utility will convert a stored event log to the rtpdump format.
-int main(int argc, char* argv[]) {
- std::string program_name = argv[0];
- std::string usage =
- "Tool for converting an RtcEventLog file to an RTP dump file.\n"
- "Run " +
- program_name +
- " --helpshort for usage.\n"
- "Example usage:\n" +
- program_name + " input.rel output.rtp\n";
- google::SetUsageMessage(usage);
- google::ParseCommandLineFlags(&argc, &argv, true);
-
- if (argc != 3) {
- std::cout << google::ProgramUsage();
- return 0;
- }
- std::string input_file = argv[1];
- std::string output_file = argv[2];
-
- uint32_t ssrc_filter = 0;
- if (!FLAGS_ssrc.empty())
- RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter))
- << "Flag verification has failed.";
-
- webrtc::rtclog::EventStream event_stream;
- if (!webrtc::RtcEventLog::ParseRtcEventLog(input_file, &event_stream)) {
- std::cerr << "Error while parsing input file: " << input_file << std::endl;
- return -1;
- }
-
- rtc::scoped_ptr<webrtc::test::RtpFileWriter> rtp_writer(
- webrtc::test::RtpFileWriter::Create(
- webrtc::test::RtpFileWriter::FileFormat::kRtpDump, output_file));
-
- if (!rtp_writer.get()) {
- std::cerr << "Error while opening output file: " << output_file
- << std::endl;
- return -1;
- }
-
- std::cout << "Found " << event_stream.stream_size()
- << " events in the input file." << std::endl;
- int rtp_counter = 0, rtcp_counter = 0;
- bool header_only = false;
- // TODO(ivoc): This can be refactored once the packet interpretation
- // functions are finished.
- for (int i = 0; i < event_stream.stream_size(); i++) {
- const webrtc::rtclog::Event& event = event_stream.stream(i);
- if (!FLAGS_nortp && event.has_type() && event.type() == event.RTP_EVENT) {
- if (event.has_timestamp_us() && event.has_rtp_packet() &&
- event.rtp_packet().has_header() &&
- event.rtp_packet().header().size() >= 12 &&
- event.rtp_packet().has_packet_length() &&
- event.rtp_packet().has_type()) {
- const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet();
- if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO)
- continue;
- if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO)
- continue;
- if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA)
- continue;
- if (!FLAGS_ssrc.empty()) {
- const uint32_t packet_ssrc =
- webrtc::ByteReader<uint32_t>::ReadBigEndian(
- reinterpret_cast<const uint8_t*>(rtp_packet.header().data() +
- 8));
- if (packet_ssrc != ssrc_filter)
- continue;
- }
-
- webrtc::test::RtpPacket packet;
- packet.length = rtp_packet.header().size();
- if (packet.length > packet.kMaxPacketBufferSize) {
- std::cout << "Skipping packet with size " << packet.length
- << ", the maximum supported size is "
- << packet.kMaxPacketBufferSize << std::endl;
- continue;
- }
- packet.original_length = rtp_packet.packet_length();
- if (packet.original_length > packet.length)
- header_only = true;
- packet.time_ms = event.timestamp_us() / 1000;
- memcpy(packet.data, rtp_packet.header().data(), packet.length);
- rtp_writer->WritePacket(&packet);
- rtp_counter++;
- } else {
- std::cout << "Skipping malformed event." << std::endl;
- }
- }
- if (!FLAGS_nortcp && event.has_type() && event.type() == event.RTCP_EVENT) {
- if (event.has_timestamp_us() && event.has_rtcp_packet() &&
- event.rtcp_packet().has_type() &&
- event.rtcp_packet().has_packet_data() &&
- event.rtcp_packet().packet_data().size() > 0) {
- const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
- if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO)
- continue;
- if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO)
- continue;
- if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA)
- continue;
- if (!FLAGS_ssrc.empty()) {
- const uint32_t packet_ssrc =
- webrtc::ByteReader<uint32_t>::ReadBigEndian(
- reinterpret_cast<const uint8_t*>(
- rtcp_packet.packet_data().data() + 4));
- if (packet_ssrc != ssrc_filter)
- continue;
- }
-
- webrtc::test::RtpPacket packet;
- packet.length = rtcp_packet.packet_data().size();
- if (packet.length > packet.kMaxPacketBufferSize) {
- std::cout << "Skipping packet with size " << packet.length
- << ", the maximum supported size is "
- << packet.kMaxPacketBufferSize << std::endl;
- continue;
- }
- // For RTCP packets the original_length should be set to 0 in the
- // RTPdump format.
- packet.original_length = 0;
- packet.time_ms = event.timestamp_us() / 1000;
- memcpy(packet.data, rtcp_packet.packet_data().data(), packet.length);
- rtp_writer->WritePacket(&packet);
- rtcp_counter++;
- } else {
- std::cout << "Skipping malformed event." << std::endl;
- }
- }
- }
- std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
- << " RTP packets and " << rtcp_counter << " RTCP packets to the "
- << "output file." << std::endl;
- return 0;
-}
diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp
index 4f4f100..12b14ee 100644
--- a/webrtc/webrtc.gyp
+++ b/webrtc/webrtc.gyp
@@ -29,17 +29,6 @@
},
'includes': ['build/protoc.gypi'],
},
- {
- 'target_name': 'rtc_event_log2rtp_dump',
- 'type': 'executable',
- 'sources': ['video/rtc_event_log2rtp_dump.cc',],
- 'dependencies': [
- '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
- 'rtc_event_log',
- 'rtc_event_log_proto',
- 'test/test.gyp:rtp_test_utils'
- ],
- }
],
}],
],
