Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/neteq/tools/rtp_analyze.cc

@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2012 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 <assert.h>
+#include <stdio.h>
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "webrtc/rtc_base/flags.h"
+
+// Define command line flags.
+DEFINE_int(red, 117, "RTP payload type for RED");
+DEFINE_int(audio_level, -1, "Extension ID for audio level (RFC 6464); "
+                            "-1 not to print audio level");
+DEFINE_int(abs_send_time, -1, "Extension ID for absolute sender time; "
+                             "-1 not to print absolute send time");
+DEFINE_bool(help, false, "Print this message");
+
+int main(int argc, char* argv[]) {
+  std::string program_name = argv[0];
+  std::string usage =
+      "Tool for parsing an RTP dump file to text output.\n"
+      "Run " +
+      program_name +
+      " --help for usage.\n"
+      "Example usage:\n" +
+      program_name + " input.rtp output.txt\n\n" +
+      "Output is sent to stdout if no output file is given. " +
+      "Note that this tool can read files with or without payloads.\n";
+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+      FLAG_help || (argc != 2 && argc != 3)) {
+    printf("%s", usage.c_str());
+    if (FLAG_help) {
+      rtc::FlagList::Print(nullptr, false);
+      return 0;
+    }
+    return 1;
+  }
+
+  RTC_CHECK(FLAG_red >= 0 && FLAG_red <= 127);  // Payload type
+  RTC_CHECK(FLAG_audio_level == -1 ||  // Default
+      (FLAG_audio_level > 0 && FLAG_audio_level <= 255));  // Extension ID
+  RTC_CHECK(FLAG_abs_send_time == -1 ||  // Default
+      (FLAG_abs_send_time > 0 && FLAG_abs_send_time <= 255));  // Extension ID
+
+  printf("Input file: %s\n", argv[1]);
+  std::unique_ptr<webrtc::test::RtpFileSource> file_source(
+      webrtc::test::RtpFileSource::Create(argv[1]));
+  assert(file_source.get());
+  // Set RTP extension IDs.
+  bool print_audio_level = false;
+  if (FLAG_audio_level != -1) {
+    print_audio_level = true;
+    file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
+                                            FLAG_audio_level);
+  }
+  bool print_abs_send_time = false;
+  if (FLAG_abs_send_time != -1) {
+    print_abs_send_time = true;
+    file_source->RegisterRtpHeaderExtension(
+        webrtc::kRtpExtensionAbsoluteSendTime, FLAG_abs_send_time);
+  }
+
+  FILE* out_file;
+  if (argc == 3) {
+    out_file = fopen(argv[2], "wt");
+    if (!out_file) {
+      printf("Cannot open output file %s\n", argv[2]);
+      return -1;
+    }
+    printf("Output file: %s\n\n", argv[2]);
+  } else {
+    out_file = stdout;
+  }
+
+  // Print file header.
+  fprintf(out_file, "SeqNo  TimeStamp   SendTime  Size    PT  M       SSRC");
+  if (print_audio_level) {
+    fprintf(out_file, " AuLvl (V)");
+  }
+  if (print_abs_send_time) {
+    fprintf(out_file, " AbsSendTime");
+  }
+  fprintf(out_file, "\n");
+
+  uint32_t max_abs_send_time = 0;
+  int cycles = -1;
+  std::unique_ptr<webrtc::test::Packet> packet;
+  while (true) {
+    packet = file_source->NextPacket();
+    if (!packet.get()) {
+      // End of file reached.
+      break;
+    }
+    // Write packet data to file. Use virtual_packet_length_bytes so that the
+    // correct packet sizes are printed also for RTP header-only dumps.
+    fprintf(out_file,
+            "%5u %10u %10u %5i %5i %2i %#08X",
+            packet->header().sequenceNumber,
+            packet->header().timestamp,
+            static_cast<unsigned int>(packet->time_ms()),
+            static_cast<int>(packet->virtual_packet_length_bytes()),
+            packet->header().payloadType,
+            packet->header().markerBit,
+            packet->header().ssrc);
+    if (print_audio_level && packet->header().extension.hasAudioLevel) {
+      fprintf(out_file,
+              " %5u (%1i)",
+              packet->header().extension.audioLevel,
+              packet->header().extension.voiceActivity);
+    }
+    if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
+      if (cycles == -1) {
+        // Initialize.
+        max_abs_send_time = packet->header().extension.absoluteSendTime;
+        cycles = 0;
+      }
+      // Abs sender time is 24 bit 6.18 fixed point. Shift by 8 to normalize to
+      // 32 bits (unsigned). Calculate the difference between this packet's
+      // send time and the maximum observed. Cast to signed 32-bit to get the
+      // desired wrap-around behavior.
+      if (static_cast<int32_t>(
+              (packet->header().extension.absoluteSendTime << 8) -
+              (max_abs_send_time << 8)) >= 0) {
+        // The difference is non-negative, meaning that this packet is newer
+        // than the previously observed maximum absolute send time.
+        if (packet->header().extension.absoluteSendTime < max_abs_send_time) {
+          // Wrap detected.
+          cycles++;
+        }
+        max_abs_send_time = packet->header().extension.absoluteSendTime;
+      }
+      // Abs sender time is 24 bit 6.18 fixed point. Divide by 2^18 to convert
+      // to floating point representation.
+      double send_time_seconds =
+          static_cast<double>(packet->header().extension.absoluteSendTime) /
+              262144 +
+          64.0 * cycles;
+      fprintf(out_file, " %11f", send_time_seconds);
+    }
+    fprintf(out_file, "\n");
+
+    if (packet->header().payloadType == FLAG_red) {
+      std::list<webrtc::RTPHeader*> red_headers;
+      packet->ExtractRedHeaders(&red_headers);
+      while (!red_headers.empty()) {
+        webrtc::RTPHeader* red = red_headers.front();
+        assert(red);
+        fprintf(out_file,
+                "* %5u %10u %10u %5i\n",
+                red->sequenceNumber,
+                red->timestamp,
+                static_cast<unsigned int>(packet->time_ms()),
+                red->payloadType);
+        red_headers.pop_front();
+        delete red;
+      }
+    }
+  }
+
+  fclose(out_file);
+
+  return 0;
+}