| /* |
| * 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 "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" |
| #include "webrtc/modules/video_coding/main/test/rtp_file_reader.h" |
| #include "webrtc/modules/video_coding/main/test/rtp_player.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| |
| using webrtc::rtpplayer::RtpPacketSourceInterface; |
| |
| int main(int argc, char** argv) { |
| if (argc < 5) { |
| printf("Usage: rtp_to_text <extension type> <extension id> <input_file.rtp>" |
| " <output_file.rtp>\n"); |
| printf("<extension type> can either be:\n" |
| " abs for absolute send time or\n" |
| " tsoffset for timestamp offset.\n" |
| "<extension id> is the id associated with the extension.\n"); |
| return -1; |
| } |
| RtpPacketSourceInterface* reader; |
| webrtc::RtpHeaderParser* parser; |
| if (!ParseArgsAndSetupEstimator(argc, argv, NULL, NULL, &reader, &parser, |
| NULL, NULL)) { |
| return -1; |
| } |
| webrtc::scoped_ptr<RtpPacketSourceInterface> rtp_reader(reader); |
| webrtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(parser); |
| |
| FILE* out_file = fopen(argv[4], "wt"); |
| if (!out_file) |
| printf("Cannot open output file %s\n", argv[4]); |
| |
| printf("Output file: %s\n\n", argv[4]); |
| fprintf(out_file, "seqnum timestamp ts_offset abs_sendtime recvtime " |
| "markerbit ssrc size\n"); |
| int packet_counter = 0; |
| static const uint32_t kMaxPacketSize = 1500; |
| uint8_t packet_buffer[kMaxPacketSize]; |
| uint8_t* packet = packet_buffer; |
| uint32_t packet_length = kMaxPacketSize; |
| uint32_t time_ms = 0; |
| int non_zero_abs_send_time = 0; |
| int non_zero_ts_offsets = 0; |
| while (rtp_reader->NextPacket(packet, &packet_length, &time_ms) == 0) { |
| webrtc::RTPHeader header = {}; |
| parser->Parse(packet, packet_length, &header); |
| if (header.extension.absoluteSendTime != 0) |
| ++non_zero_abs_send_time; |
| if (header.extension.transmissionTimeOffset != 0) |
| ++non_zero_ts_offsets; |
| fprintf(out_file, "%u %u %d %u %u %d %u %u\n", header.sequenceNumber, |
| header.timestamp, header.extension.transmissionTimeOffset, |
| header.extension.absoluteSendTime, time_ms, header.markerBit, |
| header.ssrc, packet_length); |
| packet_length = kMaxPacketSize; |
| ++packet_counter; |
| } |
| printf("Parsed %d packets\n", packet_counter); |
| printf("Packets with non-zero absolute send time: %d\n", |
| non_zero_abs_send_time); |
| printf("Packets with non-zero timestamp offset: %d\n", |
| non_zero_ts_offsets); |
| return 0; |
| } |