mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
70 lines
2.5 KiB
C++
70 lines
2.5 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 <sstream>
|
|
|
|
#include "modules/remote_bitrate_estimator/tools/bwe_rtp.h"
|
|
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
|
|
#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
|
|
#include "rtc_base/format_macros.h"
|
|
#include "test/rtp_file_reader.h"
|
|
|
|
int main(int argc, char** argv) {
|
|
webrtc::test::RtpFileReader* reader;
|
|
webrtc::RtpHeaderParser* parser;
|
|
if (!ParseArgsAndSetupEstimator(argc, argv, NULL, NULL, &reader, &parser,
|
|
NULL, NULL)) {
|
|
return -1;
|
|
}
|
|
bool arrival_time_only = (argc >= 5 && strncmp(argv[4], "-t", 2) == 0);
|
|
std::unique_ptr<webrtc::test::RtpFileReader> rtp_reader(reader);
|
|
std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
|
|
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 (rtp_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) {
|
|
std::stringstream 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 %" PRIuS " %" 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;
|
|
}
|