Upstream version 9.38.198.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / remote_bitrate_estimator / tools / rtp_to_text.cc

diff --git a/src/third_party/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/src/third_party/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
index af4a4d4..a85bca4 100644 (file)
--- a/src/third_party/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ b/src/third_party/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -14,11 +14,8 @@
 #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;
+#include "webrtc/test/rtp_file_reader.h"
 
 int main(int argc, char** argv) {
   if (argc < 4) {
@@ -32,43 +29,44 @@ int main(int argc, char** argv) {
            " output.\n");
     return -1;
   }
-  RtpPacketSourceInterface* reader;
+  webrtc::test::RtpFileReader* reader;
   webrtc::RtpHeaderParser* parser;
   if (!ParseArgsAndSetupEstimator(argc, argv, NULL, NULL, &reader, &parser,
                                   NULL, NULL)) {
     return -1;
   }
   bool arrival_time_only = (argc >= 5 && strncmp(argv[4], "-t", 2) == 0);
-  webrtc::scoped_ptr<RtpPacketSourceInterface> rtp_reader(reader);
+  webrtc::scoped_ptr<webrtc::test::RtpFileReader> rtp_reader(reader);
   webrtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
   fprintf(stdout, "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::test::RtpFileReader::Packet packet;
+  while (rtp_reader->NextPacket(&packet)) {
     webrtc::RTPHeader header;
-    parser->Parse(packet, packet_length, &header);
+    parser->Parse(packet.data, packet.length, &header);
     if (header.extension.absoluteSendTime != 0)
       ++non_zero_abs_send_time;
     if (header.extension.transmissionTimeOffset != 0)
       ++non_zero_ts_offsets;
     if (arrival_time_only) {
       std::stringstream ss;
-      ss << static_cast<int64_t>(time_ms) * 1000000;
+      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 %u\n", header.sequenceNumber,
-              header.timestamp, header.extension.transmissionTimeOffset,
-              header.extension.absoluteSendTime, time_ms, header.markerBit,
-              header.ssrc, packet_length);
+      fprintf(stdout,
+              "%u %u %d %u %u %d %u %d\n",
+              header.sequenceNumber,
+              header.timestamp,
+              header.extension.transmissionTimeOffset,
+              header.extension.absoluteSendTime,
+              packet.time_ms,
+              header.markerBit,
+              header.ssrc,
+              static_cast<int>(packet.length));
     }
-    packet_length = kMaxPacketSize;
     ++packet_counter;
   }
   fprintf(stderr, "Parsed %d packets\n", packet_counter);