mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00

There are a few reasons for making this test only: * The code is only used by tests and utilities. * The pure interface has only a single implementation so an interface isn't really needed. (a followup change could remove it altogether) * The implementation always incorporates locking regardless of how the class gets used. See e.g. previous use in the Packet class. * The implementation is a layer on top of RtpUtility::RtpHeaderParser which is sufficient for most production cases. Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36 Bug: none Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29010}
64 lines
2.3 KiB
C++
64 lines
2.3 KiB
C++
/*
|
|
* Copyright (c) 2014 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 <memory>
|
|
|
|
#include "modules/remote_bitrate_estimator/tools/bwe_rtp.h"
|
|
#include "rtc_base/format_macros.h"
|
|
#include "rtc_base/strings/string_builder.h"
|
|
#include "test/rtp_file_reader.h"
|
|
#include "test/rtp_header_parser.h"
|
|
|
|
int main(int argc, char* argv[]) {
|
|
std::unique_ptr<webrtc::test::RtpFileReader> reader;
|
|
std::unique_ptr<webrtc::RtpHeaderParser> parser(ParseArgsAndSetupEstimator(
|
|
argc, argv, nullptr, nullptr, &reader, nullptr, nullptr));
|
|
if (!parser)
|
|
return -1;
|
|
|
|
bool arrival_time_only = (argc >= 5 && strncmp(argv[4], "-t", 2) == 0);
|
|
|
|
fprintf(stdout,
|
|
"seqnum timestamp ts_offset abs_sendtime recvtime "
|
|
"markerbit ssrc size original_size\n");
|
|
int packet_counter = 0;
|
|
int non_zero_abs_send_time = 0;
|
|
int non_zero_ts_offsets = 0;
|
|
webrtc::test::RtpPacket packet;
|
|
while (reader->NextPacket(&packet)) {
|
|
webrtc::RTPHeader header;
|
|
parser->Parse(packet.data, packet.length, &header);
|
|
if (header.extension.absoluteSendTime != 0)
|
|
++non_zero_abs_send_time;
|
|
if (header.extension.transmissionTimeOffset != 0)
|
|
++non_zero_ts_offsets;
|
|
if (arrival_time_only) {
|
|
rtc::StringBuilder ss;
|
|
ss << static_cast<int64_t>(packet.time_ms) * 1000000;
|
|
fprintf(stdout, "%s\n", ss.str().c_str());
|
|
} else {
|
|
fprintf(stdout, "%u %u %d %u %u %d %u %" RTC_PRIuS " %" RTC_PRIuS "\n",
|
|
header.sequenceNumber, header.timestamp,
|
|
header.extension.transmissionTimeOffset,
|
|
header.extension.absoluteSendTime, packet.time_ms,
|
|
header.markerBit, header.ssrc, packet.length,
|
|
packet.original_length);
|
|
}
|
|
++packet_counter;
|
|
}
|
|
fprintf(stderr, "Parsed %d packets\n", packet_counter);
|
|
fprintf(stderr, "Packets with non-zero absolute send time: %d\n",
|
|
non_zero_abs_send_time);
|
|
fprintf(stderr, "Packets with non-zero timestamp offset: %d\n",
|
|
non_zero_ts_offsets);
|
|
return 0;
|
|
}
|