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/*
* Copyright (c) 2012 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 <stdio.h>
#include <cstdint>
#include <list>
#include <memory>
#include <string>
#include <vector>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "api/rtp_headers.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/checks.h"
ABSL_FLAG(int, red, 117, "RTP payload type for RED");
ABSL_FLAG(int,
audio_level,
-1,
"Extension ID for audio level (RFC 6464); "
"-1 not to print audio level");
ABSL_FLAG(int,
abs_send_time,
-1,
"Extension ID for absolute sender time; "
"-1 not to print absolute send time");
int main(int argc, char* argv[]) {
std::vector<char*> args = absl::ParseCommandLine(argc, argv);
std::string usage =
"Tool for parsing an RTP dump file to text output.\n"
"Example usage:\n"
"./rtp_analyze input.rtp output.txt\n\n"
"Output is sent to stdout if no output file is given. "
"Note that this tool can read files with or without payloads.\n";
if (args.size() != 2 && args.size() != 3) {
printf("%s", usage.c_str());
return 1;
}
RTC_CHECK(absl::GetFlag(FLAGS_red) >= 0 &&
absl::GetFlag(FLAGS_red) <= 127); // Payload type
RTC_CHECK(absl::GetFlag(FLAGS_audio_level) == -1 || // Default
(absl::GetFlag(FLAGS_audio_level) > 0 &&
absl::GetFlag(FLAGS_audio_level) <= 255)); // Extension ID
RTC_CHECK(absl::GetFlag(FLAGS_abs_send_time) == -1 || // Default
(absl::GetFlag(FLAGS_abs_send_time) > 0 &&
absl::GetFlag(FLAGS_abs_send_time) <= 255)); // Extension ID
printf("Input file: %s\n", args[1]);
std::unique_ptr<webrtc::test::RtpFileSource> file_source(
webrtc::test::RtpFileSource::Create(args[1]));
RTC_DCHECK(file_source.get());
// Set RTP extension IDs.
bool print_audio_level = false;
if (absl::GetFlag(FLAGS_audio_level) != -1) {
print_audio_level = true;
file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
absl::GetFlag(FLAGS_audio_level));
}
bool print_abs_send_time = false;
if (absl::GetFlag(FLAGS_abs_send_time) != -1) {
print_abs_send_time = true;
file_source->RegisterRtpHeaderExtension(
webrtc::kRtpExtensionAbsoluteSendTime,
absl::GetFlag(FLAGS_abs_send_time));
}
FILE* out_file;
if (args.size() == 3) {
out_file = fopen(args[2], "wt");
if (!out_file) {
printf("Cannot open output file %s\n", args[2]);
return -1;
}
printf("Output file: %s\n\n", args[2]);
} else {
out_file = stdout;
}
// Print file header.
fprintf(out_file, "SeqNo TimeStamp SendTime Size PT M SSRC");
if (print_audio_level) {
fprintf(out_file, " AuLvl (V)");
}
if (print_abs_send_time) {
fprintf(out_file, " AbsSendTime");
}
fprintf(out_file, "\n");
uint32_t max_abs_send_time = 0;
int cycles = -1;
std::unique_ptr<webrtc::test::Packet> packet;
while (true) {
packet = file_source->NextPacket();
if (!packet.get()) {
// End of file reached.
break;
}
// Write packet data to file. Use virtual_packet_length_bytes so that the
// correct packet sizes are printed also for RTP header-only dumps.
fprintf(out_file, "%5u %10u %10u %5i %5i %2i %#08X",
packet->header().sequenceNumber, packet->header().timestamp,
static_cast<unsigned int>(packet->time_ms()),
static_cast<int>(packet->virtual_packet_length_bytes()),
packet->header().payloadType, packet->header().markerBit,
packet->header().ssrc);
if (print_audio_level && packet->header().extension.audio_level()) {
fprintf(out_file, " %5d (%1i)",
packet->header().extension.audio_level()->level(),
packet->header().extension.audio_level()->voice_activity());
}
if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
if (cycles == -1) {
// Initialize.
max_abs_send_time = packet->header().extension.absoluteSendTime;
cycles = 0;
}
// Abs sender time is 24 bit 6.18 fixed point. Shift by 8 to normalize to
// 32 bits (unsigned). Calculate the difference between this packet's
// send time and the maximum observed. Cast to signed 32-bit to get the
// desired wrap-around behavior.
if (static_cast<int32_t>(
(packet->header().extension.absoluteSendTime << 8) -
(max_abs_send_time << 8)) >= 0) {
// The difference is non-negative, meaning that this packet is newer
// than the previously observed maximum absolute send time.
if (packet->header().extension.absoluteSendTime < max_abs_send_time) {
// Wrap detected.
cycles++;
}
max_abs_send_time = packet->header().extension.absoluteSendTime;
}
// Abs sender time is 24 bit 6.18 fixed point. Divide by 2^18 to convert
// to floating point representation.
double send_time_seconds =
static_cast<double>(packet->header().extension.absoluteSendTime) /
262144 +
64.0 * cycles;
fprintf(out_file, " %11f", send_time_seconds);
}
fprintf(out_file, "\n");
if (packet->header().payloadType == absl::GetFlag(FLAGS_red)) {
std::list<webrtc::RTPHeader*> red_headers;
packet->ExtractRedHeaders(&red_headers);
while (!red_headers.empty()) {
webrtc::RTPHeader* red = red_headers.front();
RTC_DCHECK(red);
fprintf(out_file, "* %5u %10u %10u %5i\n", red->sequenceNumber,
red->timestamp, static_cast<unsigned int>(packet->time_ms()),
red->payloadType);
red_headers.pop_front();
delete red;
}
}
}
fclose(out_file);
return 0;
}
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