blob: 78f0497ffff0cf0afc432650dad4521e07e0aafb [file] [log] [blame]
/*
* 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 <assert.h>
#include <stdio.h>
#include <vector>
#include "google/gflags.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
// Flag validator.
static bool ValidatePayloadType(const char* flagname, int32_t value) {
if (value >= 0 && value <= 127) // Value is ok.
return true;
printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
return false;
}
static bool ValidateExtensionId(const char* flagname, int32_t value) {
if (value > 0 && value <= 255) // Value is ok.
return true;
printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
return false;
}
// Define command line flags.
DEFINE_int32(red, 117, "RTP payload type for RED");
static const bool red_dummy =
google::RegisterFlagValidator(&FLAGS_red, &ValidatePayloadType);
DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)");
static const bool audio_level_dummy =
google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId);
DEFINE_int32(abs_send_time, 3, "Extension ID for absolute sender time");
static const bool abs_send_time_dummy =
google::RegisterFlagValidator(&FLAGS_abs_send_time, &ValidateExtensionId);
int main(int argc, char* argv[]) {
std::string program_name = argv[0];
std::string usage =
"Tool for parsing an RTP dump file to text output.\n"
"Run " +
program_name +
" --helpshort for usage.\n"
"Example usage:\n" +
program_name + " 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 our without payloads.";
google::SetUsageMessage(usage);
google::ParseCommandLineFlags(&argc, &argv, true);
if (argc != 2 && argc != 3) {
// Print usage information.
printf("%s", google::ProgramUsage());
return 0;
}
printf("Input file: %s\n", argv[1]);
rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
webrtc::test::RtpFileSource::Create(argv[1]));
assert(file_source.get());
// Set RTP extension IDs.
bool print_audio_level = false;
if (!google::GetCommandLineFlagInfoOrDie("audio_level").is_default) {
print_audio_level = true;
file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
FLAGS_audio_level);
}
bool print_abs_send_time = false;
if (!google::GetCommandLineFlagInfoOrDie("abs_send_time").is_default) {
print_abs_send_time = true;
file_source->RegisterRtpHeaderExtension(
webrtc::kRtpExtensionAbsoluteSendTime, FLAGS_abs_send_time);
}
FILE* out_file;
if (argc == 3) {
out_file = fopen(argv[2], "wt");
if (!out_file) {
printf("Cannot open output file %s\n", argv[2]);
return -1;
}
printf("Output file: %s\n\n", argv[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;
rtc::scoped_ptr<webrtc::test::Packet> packet;
while (true) {
packet.reset(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.hasAudioLevel) {
fprintf(out_file,
" %5u (%1i)",
packet->header().extension.audioLevel,
packet->header().extension.voiceActivity);
}
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 == FLAGS_red) {
std::list<webrtc::RTPHeader*> red_headers;
packet->ExtractRedHeaders(&red_headers);
while (!red_headers.empty()) {
webrtc::RTPHeader* red = red_headers.front();
assert(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;
}