1 files changed, 67 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
new file mode 100644
index 0000000000..e8dc59f740
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -0,0 +1,67 @@
+/*
+ *  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 <memory>
+
+#include "modules/remote_bitrate_estimator/tools/bwe_rtp.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/rtp_file_reader.h"
+
+int main(int argc, char* argv[]) {
+  std::unique_ptr<webrtc::test::RtpFileReader> reader;
+  webrtc::RtpHeaderExtensionMap rtp_header_extensions;
+  if (!ParseArgsAndSetupRtpReader(argc, argv, reader, rtp_header_extensions)) {
+    return -1;
+  }
+
+  bool arrival_time_only = (argc >= 5 && strncmp(argv[4], "-t", 2) == 0);
+
+  fprintf(stdout,
+          "seqnum timestamp ts_offset abs_sendtime recvtime "
+          "markerbit ssrc size original_size\n");
+  int packet_counter = 0;
+  int non_zero_abs_send_time = 0;
+  int non_zero_ts_offsets = 0;
+  webrtc::test::RtpPacket packet;
+  while (reader->NextPacket(&packet)) {
+    webrtc::RtpPacket header(&rtp_header_extensions);
+    header.Parse(packet.data, packet.length);
+    uint32_t abs_send_time = 0;
+    if (header.GetExtension<webrtc::AbsoluteSendTime>(&abs_send_time) &&
+        abs_send_time != 0)
+      ++non_zero_abs_send_time;
+    int32_t toffset = 0;
+    if (header.GetExtension<webrtc::TransmissionOffset>(&toffset) &&
+        toffset != 0)
+      ++non_zero_ts_offsets;
+    if (arrival_time_only) {
+      rtc::StringBuilder ss;
+      ss << static_cast<int64_t>(packet.time_ms) * 1000000;
+      fprintf(stdout, "%s\n", ss.str().c_str());
+    } else {
+      fprintf(stdout, "%u %u %d %u %u %d %u %zu %zu\n", header.SequenceNumber(),
+              header.Timestamp(), toffset, abs_send_time, packet.time_ms,
+              header.Marker(), header.Ssrc(), packet.length,
+              packet.original_length);
+    }
+    ++packet_counter;
+  }
+  fprintf(stderr, "Parsed %d packets\n", packet_counter);
+  fprintf(stderr, "Packets with non-zero absolute send time: %d\n",
+          non_zero_abs_send_time);
+  fprintf(stderr, "Packets with non-zero timestamp offset: %d\n",
+          non_zero_ts_offsets);
+  return 0;
+}