Rename neteq4 folder to neteq

Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.

This CL effectively reverts r6257 "Rename neteq4 folder to neteq".

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc

@@ -0,0 +1,147 @@
+/*
+ *  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 <vector>
+
+#include "google/gflags.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+// Flag validator.
+static bool ValidatePayloadType(const char* flagname, int32_t value) {
+  if (value >= 0 && value <= 127)  // Value is ok.
+    return true;
+  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
+  return false;
+}
+static bool ValidateExtensionId(const char* flagname, int32_t value) {
+  if (value > 0 && value <= 255)  // Value is ok.
+    return true;
+  printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
+  return false;
+}
+
+// Define command line flags.
+DEFINE_int32(red, 117, "RTP payload type for RED");
+static const bool red_dummy =
+    google::RegisterFlagValidator(&FLAGS_red, &ValidatePayloadType);
+DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)");
+static const bool audio_level_dummy =
+    google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId);
+
+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 +
+      " --helpshort 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 our without payloads.";
+  google::SetUsageMessage(usage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+
+  if (argc != 2 && argc != 3) {
+    // Print usage information.
+    printf("%s", google::ProgramUsage());
+    return 0;
+  }
+
+  FILE* in_file = fopen(argv[1], "rb");
+  if (!in_file) {
+    printf("Cannot open input file %s\n", argv[1]);
+    return -1;
+  }
+  printf("Input file: %s\n", argv[1]);
+  webrtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
+      webrtc::test::RtpFileSource::Create(argv[1]));
+  assert(file_source.get());
+  // Set RTP extension ID.
+  bool print_audio_level = false;
+  if (!google::GetCommandLineFlagInfoOrDie("audio_level").is_default) {
+    print_audio_level = true;
+    file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
+                                            FLAGS_audio_level);
+  }
+
+  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)");
+  }
+  fprintf(out_file, "\n");
+
+  webrtc::scoped_ptr<webrtc::test::Packet> packet;
+  while (!file_source->EndOfFile()) {
+    packet.reset(file_source->NextPacket());
+    if (!packet.get()) {
+      // This is probably an RTCP packet. Move on to the next one.
+      continue;
+    }
+    assert(packet.get());
+    // Write packet data to file.
+    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->packet_length_bytes()),
+            packet->header().payloadType,
+            packet->header().markerBit,
+            packet->header().ssrc);
+    if (print_audio_level && packet->header().extension.hasAudioLevel) {
+      // |audioLevel| consists of one bit for "V" and then 7 bits level.
+      fprintf(out_file,
+              " %5u (%1i)",
+              packet->header().extension.audioLevel & 0x7F,
+              (packet->header().extension.audioLevel & 0x80) == 0 ? 0 : 1);
+    }
+    fprintf(out_file, "\n");
+
+    if (packet->header().payloadType == FLAGS_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(in_file);
+  fclose(out_file);
+
+  return 0;
+}