Add support for parsing header only RTP dumps with bwe_rtp_play.

Also adds support for printing the original_length in rtp_to_text.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7812 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc

@@ -62,47 +62,60 @@ int main(int argc, char** argv) {
 
   // Process the file.
   int packet_counter = 0;
-  int64_t next_process_time_ms = 0;
   int64_t next_rtp_time_ms = 0;
   int64_t first_rtp_time_ms = -1;
-  int non_zero_abs_send_time = 0;
+  int abs_send_time_count = 0;
-  int non_zero_ts_offsets = 0;
+  int ts_offset_count = 0;
+  webrtc::test::RtpPacket packet;
+  if (!rtp_reader->NextPacket(&packet)) {
+    printf("No RTP packet found\n");
+    return 0;
+  }
+  first_rtp_time_ms = packet.time_ms;
+  packet.time_ms = packet.time_ms - first_rtp_time_ms;
   while (true) {
     if (next_rtp_time_ms <= clock.TimeInMilliseconds()) {
-      webrtc::test::RtpPacket packet;
+      webrtc::RTPHeader header;
+      parser->Parse(packet.data, packet.length, &header);
+      if (header.extension.hasAbsoluteSendTime)
+        ++abs_send_time_count;
+      if (header.extension.hasTransmissionTimeOffset)
+        ++ts_offset_count;
+      size_t packet_length = packet.length;
+      // Some RTP dumps only include the header, in which case packet.length
+      // is equal to the header length. In those cases packet.original_length
+      // usually contains the original packet length.
+      if (packet.original_length > 0) {
+        packet_length = packet.original_length;
+      }
+      rbe->IncomingPacket(clock.TimeInMilliseconds(),
+                          packet_length - header.headerLength,
+                          header);
+      ++packet_counter;
       if (!rtp_reader->NextPacket(&packet)) {
         break;
       }
-      if (first_rtp_time_ms == -1)
-        first_rtp_time_ms = packet.time_ms;
       packet.time_ms = packet.time_ms - first_rtp_time_ms;
-      webrtc::RTPHeader header;
+      next_rtp_time_ms = packet.time_ms;
-      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;
-      rbe->IncomingPacket(clock.TimeInMilliseconds(),
-                          packet.length - header.headerLength,
-                          header);
-      ++packet_counter;
     }
-    next_process_time_ms = rbe->TimeUntilNextProcess() +
+    int time_until_process_ms = rbe->TimeUntilNextProcess();
-        clock.TimeInMilliseconds();
+    if (time_until_process_ms <= 0) {
-    if (next_process_time_ms <= clock.TimeInMilliseconds()) {
       rbe->Process();
     }
     int time_until_next_event =
-        std::min(next_process_time_ms, next_rtp_time_ms) -
+        std::min(rbe->TimeUntilNextProcess(),
-        clock.TimeInMilliseconds();
+                 static_cast<int>(next_rtp_time_ms -
+                                  clock.TimeInMilliseconds()));
     clock.AdvanceTimeMilliseconds(std::max(time_until_next_event, 0));
   }
   printf("Parsed %d packets\nTime passed: %u ms\n", packet_counter,
          static_cast<uint32_t>(clock.TimeInMilliseconds()));
   printf("Estimator used: %s\n", estimator_used.c_str());
-  printf("Packets with non-zero absolute send time: %d\n",
+  printf("Packets with absolute send time: %d\n",
-         non_zero_abs_send_time);
+         abs_send_time_count);
-  printf("Packets with non-zero timestamp offset: %d\n",
+  printf("Packets with timestamp offset: %d\n",
-         non_zero_ts_offsets);
+         ts_offset_count);
+  printf("Packets with no extension: %d\n",
+         packet_counter - ts_offset_count - abs_send_time_count);
   return 0;
 }

webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc

@@ -11,6 +11,7 @@
 #include <stdio.h>
 #include <sstream>
 
+#include "webrtc/base/format_macros.h"
 #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"
@@ -39,7 +40,7 @@ int main(int argc, char** argv) {
   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");
+          "markerbit ssrc size original_size\n");
   int packet_counter = 0;
   int non_zero_abs_send_time = 0;
   int non_zero_ts_offsets = 0;
@@ -57,7 +58,7 @@ int main(int argc, char** argv) {
       fprintf(stdout, "%s\n", ss.str().c_str());
     } else {
       fprintf(stdout,
-              "%u %u %d %u %u %d %u %d\n",
+              "%u %u %d %u %u %d %u %" PRIuS " %" PRIuS "\n",
               header.sequenceNumber,
               header.timestamp,
               header.extension.transmissionTimeOffset,
@@ -65,7 +66,8 @@ int main(int argc, char** argv) {
               packet.time_ms,
               header.markerBit,
               header.ssrc,
-              static_cast<int>(packet.length));
+              packet.length,
+              packet.original_length);
     }
     ++packet_counter;
   }