From f5d7c5891c8d54085edf6108868597af6293e6b0 Mon Sep 17 00:00:00 2001
From: "stefan@webrtc.org"
 <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Mon, 7 Oct 2013 08:42:46 +0000
Subject: [PATCH] Revert r4934 "Add a tool for parsing an RTP file and
 outputting the BWE relevant fields." 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
---
 .../remote_bitrate_estimator.gypi             | 21 ------
 .../tools/rtp_to_text.cc                      | 65 -------------------
 2 files changed, 86 deletions(-)
 delete mode 100644 webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc

diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
index 9f0261949..f8f7e43dd 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
@@ -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
 }
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
deleted file mode 100644
index 41f6549a7..000000000
--- a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ /dev/null
@@ -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;
-}