blob: e6608e21fc7d6dace2a45a8d2e8c031307f07335 [file] [log] [blame]
/*
* 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 <stdint.h>
#include <string.h>
#include <iostream>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
#include "logging/rtc_event_log/rtc_event_processor.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "test/rtp_file_reader.h"
#include "test/rtp_file_writer.h"
namespace {
using MediaType = webrtc::ParsedRtcEventLogNew::MediaType;
WEBRTC_DEFINE_bool(
audio,
true,
"Use --noaudio to exclude audio packets from the converted RTPdump file.");
WEBRTC_DEFINE_bool(
video,
true,
"Use --novideo to exclude video packets from the converted RTPdump file.");
WEBRTC_DEFINE_bool(
data,
true,
"Use --nodata to exclude data packets from the converted RTPdump file.");
WEBRTC_DEFINE_bool(
rtp,
true,
"Use --nortp to exclude RTP packets from the converted RTPdump file.");
WEBRTC_DEFINE_bool(
rtcp,
true,
"Use --nortcp to exclude RTCP packets from the converted RTPdump file.");
WEBRTC_DEFINE_string(
ssrc,
"",
"Store only packets with this SSRC (decimal or hex, the latter "
"starting with 0x).");
WEBRTC_DEFINE_bool(help, false, "Prints this message.");
// 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.
absl::optional<uint32_t> ParseSsrc(std::string str) {
// If the input string starts with 0x or 0X it indicates a hexadecimal number.
uint32_t ssrc;
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;
if (str.empty() || (!ss.fail() && ss.eof()))
return ssrc;
return absl::nullopt;
}
bool ShouldSkipStream(MediaType media_type,
uint32_t ssrc,
absl::optional<uint32_t> ssrc_filter) {
if (!FLAG_audio && media_type == MediaType::AUDIO)
return true;
if (!FLAG_video && media_type == MediaType::VIDEO)
return true;
if (!FLAG_data && media_type == MediaType::DATA)
return true;
if (ssrc_filter.has_value() && ssrc != *ssrc_filter)
return true;
return false;
}
// Convert a LoggedRtpPacketIncoming to a test::RtpPacket. Header extension IDs
// are allocated according to the provided extension map. This might not match
// the extension map used in the actual call.
void ConvertRtpPacket(
const webrtc::LoggedRtpPacketIncoming& incoming,
const webrtc::RtpHeaderExtensionMap& default_extension_map,
webrtc::test::RtpPacket* packet) {
webrtc::RtpPacket reconstructed_packet(&default_extension_map);
reconstructed_packet.SetMarker(incoming.rtp.header.markerBit);
reconstructed_packet.SetPayloadType(incoming.rtp.header.payloadType);
reconstructed_packet.SetSequenceNumber(incoming.rtp.header.sequenceNumber);
reconstructed_packet.SetTimestamp(incoming.rtp.header.timestamp);
reconstructed_packet.SetSsrc(incoming.rtp.header.ssrc);
if (incoming.rtp.header.numCSRCs > 0) {
reconstructed_packet.SetCsrcs(rtc::ArrayView<const uint32_t>(
incoming.rtp.header.arrOfCSRCs, incoming.rtp.header.numCSRCs));
}
// Set extensions.
if (incoming.rtp.header.extension.hasTransmissionTimeOffset)
reconstructed_packet.SetExtension<webrtc::TransmissionOffset>(
incoming.rtp.header.extension.transmissionTimeOffset);
if (incoming.rtp.header.extension.hasAbsoluteSendTime)
reconstructed_packet.SetExtension<webrtc::AbsoluteSendTime>(
incoming.rtp.header.extension.absoluteSendTime);
if (incoming.rtp.header.extension.hasTransportSequenceNumber)
reconstructed_packet.SetExtension<webrtc::TransportSequenceNumber>(
incoming.rtp.header.extension.transportSequenceNumber);
if (incoming.rtp.header.extension.hasAudioLevel)
reconstructed_packet.SetExtension<webrtc::AudioLevel>(
incoming.rtp.header.extension.voiceActivity,
incoming.rtp.header.extension.audioLevel);
if (incoming.rtp.header.extension.hasVideoRotation)
reconstructed_packet.SetExtension<webrtc::VideoOrientation>(
incoming.rtp.header.extension.videoRotation);
if (incoming.rtp.header.extension.hasVideoContentType)
reconstructed_packet.SetExtension<webrtc::VideoContentTypeExtension>(
incoming.rtp.header.extension.videoContentType);
if (incoming.rtp.header.extension.has_video_timing)
reconstructed_packet.SetExtension<webrtc::VideoTimingExtension>(
incoming.rtp.header.extension.video_timing);
RTC_DCHECK_EQ(reconstructed_packet.size(), incoming.rtp.header_length);
RTC_DCHECK_EQ(reconstructed_packet.headers_size(),
incoming.rtp.header_length);
memcpy(packet->data, reconstructed_packet.data(),
reconstructed_packet.headers_size());
packet->length = reconstructed_packet.headers_size();
packet->original_length = incoming.rtp.total_length;
packet->time_ms = incoming.log_time_ms();
// Set padding bit.
if (incoming.rtp.header.paddingLength > 0)
packet->data[0] = packet->data[0] | 0x20;
}
} // 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 +
" --help for usage.\n"
"Example usage:\n" +
program_name + " input.rel output.rtp\n";
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
argc != 3) {
std::cout << usage;
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
std::string input_file = argv[1];
std::string output_file = argv[2];
absl::optional<uint32_t> ssrc_filter;
if (strlen(FLAG_ssrc) > 0) {
ssrc_filter = ParseSsrc(FLAG_ssrc);
RTC_CHECK(ssrc_filter.has_value()) << "Failed to read SSRC filter flag.";
}
webrtc::ParsedRtcEventLogNew parsed_stream;
if (!parsed_stream.ParseFile(input_file)) {
std::cerr << "Error while parsing input file: " << input_file << std::endl;
return -1;
}
std::unique_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;
}
int rtp_counter = 0, rtcp_counter = 0;
bool header_only = false;
webrtc::RtpHeaderExtensionMap default_extension_map =
webrtc::ParsedRtcEventLogNew::GetDefaultHeaderExtensionMap();
auto handle_rtp = [&default_extension_map, &rtp_writer, &rtp_counter](
const webrtc::LoggedRtpPacketIncoming& incoming) {
webrtc::test::RtpPacket packet;
ConvertRtpPacket(incoming, default_extension_map, &packet);
rtp_writer->WritePacket(&packet);
rtp_counter++;
};
auto handle_rtcp = [&rtp_writer, &rtcp_counter](
const webrtc::LoggedRtcpPacketIncoming& incoming) {
webrtc::test::RtpPacket packet;
memcpy(packet.data, incoming.rtcp.raw_data.data(),
incoming.rtcp.raw_data.size());
packet.length = incoming.rtcp.raw_data.size();
// For RTCP packets the original_length should be set to 0 in the
// RTPdump format.
packet.original_length = 0;
packet.time_ms = incoming.log_time_ms();
rtp_writer->WritePacket(&packet);
rtcp_counter++;
};
webrtc::RtcEventProcessor event_processor;
for (const auto& stream : parsed_stream.incoming_rtp_packets_by_ssrc()) {
MediaType media_type =
parsed_stream.GetMediaType(stream.ssrc, webrtc::kIncomingPacket);
if (ShouldSkipStream(media_type, stream.ssrc, ssrc_filter))
continue;
auto rtp_view = absl::make_unique<
webrtc::ProcessableEventList<webrtc::LoggedRtpPacketIncoming>>(
stream.incoming_packets.begin(), stream.incoming_packets.end(),
handle_rtp);
event_processor.AddEvents(std::move(rtp_view));
}
// Note that |packet_ssrc| is the sender SSRC. An RTCP message may contain
// report blocks for many streams, thus several SSRCs and they don't
// necessarily have to be of the same media type. We therefore don't
// support filtering of RTCP based on SSRC and media type.
auto rtcp_view = absl::make_unique<
webrtc::ProcessableEventList<webrtc::LoggedRtcpPacketIncoming>>(
parsed_stream.incoming_rtcp_packets().begin(),
parsed_stream.incoming_rtcp_packets().end(), handle_rtcp);
event_processor.AddEvents(std::move(rtcp_view));
event_processor.ProcessEventsInOrder();
std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
<< " RTP packets and " << rtcp_counter << " RTCP packets to the "
<< "output file." << std::endl;
return 0;
}