generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields."

Browse Source 
Revert r4935 "Fix build error in r4934."

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2364004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4936 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2013-10-07 08:42:46 +00:00
parent 611e5141cb
commit f5d7c5891c
2 changed files with 0 additions and 86 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

21
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
View File

@ -27,26 +27,5 @@
'rtp_to_ntp.cc',
], # source
},
{
'target_name': 'bwe_rtp_to_text',
'type': 'executable',
'includes': [
'../rtp_rtcp/source/rtp_rtcp.gypi',
],
'dependencies': [
'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'rtp_rtcp',
],
'direct_dependent_settings': {
'include_dirs': [
'include',
],
},
'sources': [
'tools/rtp_to_text.cc',
'<(webrtc_root)/modules/video_coding/main/test/rtp_file_reader.cc',
'<(webrtc_root)/modules/video_coding/main/test/rtp_file_reader.h',
], # source
},
], # targets
}

65
webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
View File

@ -1,65 +0,0 @@
/*
* Copyright (c) 2013 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/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 namespace webrtc::rtpplayer;
const uint32_t kMaxPacketSize = 1500;
const int kDefaultTransmissionTimeOffsetExtensionId = 2;
int main(int argc, char** argv) {
if (argc < 2) {
printf("Usage: rtp_to_text <input_file.rtp> <output_file.rtp>\n");
return -1;
}
webrtc::scoped_ptr<RtpPacketSourceInterface> rtp_reader(
CreateRtpFileReader(argv[1]));
if (!rtp_reader.get()) {
printf("Cannot open input file %s\n", argv[1]);
return -1;
}
uint8_t packet_buffer[kMaxPacketSize];
uint8_t* packet = packet_buffer;
uint32_t packet_length = kMaxPacketSize;
uint32_t time_ms = 0;
FILE* out_file = fopen(argv[2], "wt");
if (!out_file) {
printf("Cannot open output file %s\n", argv[2]);
return -1;
}
printf("Input file: %s, Output file: %s\n\n", argv[1], argv[2]);
fprintf(out_file, "seqnum timestamp ts_offset abs_sendtime recvtime "
"markerbit ssrc size\n");
webrtc::scoped_ptr<webrtc::RtpHeaderParser> parser(
webrtc::RtpHeaderParser::Create());
parser->RegisterRtpHeaderExtension(
webrtc::kRtpExtensionTransmissionTimeOffset,
kDefaultTransmissionTimeOffsetExtensionId);
int packet_counter = 0;
while (rtp_reader->NextPacket(packet, &packet_length, &time_ms) == 0) {
webrtc::RTPHeader header;
parser->Parse(packet, packet_length, &header);
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);
return 0;
}