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

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2237005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4934 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi

@@ -27,5 +27,26 @@
         '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
 }

webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc

@@ -0,0 +1,65 @@
+/*
+ *  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;
+}