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https://github.com/mollyim/webrtc.git
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665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
66 lines
2.5 KiB
C++
66 lines
2.5 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stdio.h>
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#include <memory>
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#include <sstream>
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#include "modules/remote_bitrate_estimator/tools/bwe_rtp.h"
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#include "modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
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#include "rtc_base/format_macros.h"
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#include "test/rtp_file_reader.h"
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int main(int argc, char* argv[]) {
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webrtc::test::RtpFileReader* reader;
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webrtc::RtpHeaderParser* parser;
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if (!ParseArgsAndSetupEstimator(argc, argv, NULL, NULL, &reader, &parser,
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NULL, NULL)) {
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return -1;
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}
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bool arrival_time_only = (argc >= 5 && strncmp(argv[4], "-t", 2) == 0);
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std::unique_ptr<webrtc::test::RtpFileReader> rtp_reader(reader);
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std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
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fprintf(stdout,
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"seqnum timestamp ts_offset abs_sendtime recvtime "
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"markerbit ssrc size original_size\n");
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int packet_counter = 0;
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int non_zero_abs_send_time = 0;
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int non_zero_ts_offsets = 0;
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webrtc::test::RtpPacket packet;
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while (rtp_reader->NextPacket(&packet)) {
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webrtc::RTPHeader header;
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parser->Parse(packet.data, packet.length, &header);
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if (header.extension.absoluteSendTime != 0)
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++non_zero_abs_send_time;
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if (header.extension.transmissionTimeOffset != 0)
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++non_zero_ts_offsets;
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if (arrival_time_only) {
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std::stringstream ss;
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ss << static_cast<int64_t>(packet.time_ms) * 1000000;
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fprintf(stdout, "%s\n", ss.str().c_str());
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} else {
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fprintf(stdout, "%u %u %d %u %u %d %u %" PRIuS " %" PRIuS "\n",
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header.sequenceNumber, header.timestamp,
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header.extension.transmissionTimeOffset,
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header.extension.absoluteSendTime, packet.time_ms,
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header.markerBit, header.ssrc, packet.length,
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packet.original_length);
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}
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++packet_counter;
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}
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fprintf(stderr, "Parsed %d packets\n", packet_counter);
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fprintf(stderr, "Packets with non-zero absolute send time: %d\n",
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non_zero_abs_send_time);
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fprintf(stderr, "Packets with non-zero timestamp offset: %d\n",
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non_zero_ts_offsets);
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return 0;
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}
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