1 files changed, 74 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
new file mode 100644
index 00000000000..40fa6df8ffb
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
@@ -0,0 +1,74 @@
+/*
+ *  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 "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
+
+#include <stdio.h>
+#include <string>
+
+#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.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"
+
+using webrtc::rtpplayer::RtpPacketSourceInterface;
+
+const int kMinBitrateBps = 30000;
+
+bool ParseArgsAndSetupEstimator(int argc,
+                                char** argv,
+                                webrtc::Clock* clock,
+                                webrtc::RemoteBitrateObserver* observer,
+                                RtpPacketSourceInterface** rtp_reader,
+                                webrtc::RtpHeaderParser** parser,
+                                webrtc::RemoteBitrateEstimator** estimator,
+                                std::string* estimator_used) {
+  *rtp_reader = webrtc::rtpplayer::CreateRtpFileReader(argv[3]);
+  if (!*rtp_reader) {
+    fprintf(stderr, "Cannot open input file %s\n", argv[3]);
+    return false;
+  }
+  fprintf(stderr, "Input file: %s\n\n", argv[3]);
+  webrtc::RTPExtensionType extension = webrtc::kRtpExtensionAbsoluteSendTime;
+
+  if (strncmp("tsoffset", argv[1], 8) == 0) {
+    extension = webrtc::kRtpExtensionTransmissionTimeOffset;
+    fprintf(stderr, "Extension: toffset\n");
+  } else {
+    fprintf(stderr, "Extension: abs\n");
+  }
+  int id = atoi(argv[2]);
+
+  // Setup the RTP header parser and the bitrate estimator.
+  *parser = webrtc::RtpHeaderParser::Create();
+  (*parser)->RegisterRtpHeaderExtension(extension, id);
+  if (estimator) {
+    switch (extension) {
+      case webrtc::kRtpExtensionAbsoluteSendTime: {
+          webrtc::AbsoluteSendTimeRemoteBitrateEstimatorFactory factory;
+          *estimator = factory.Create(observer, clock, webrtc::kAimdControl,
+                                      kMinBitrateBps);
+          *estimator_used = "AbsoluteSendTimeRemoteBitrateEstimator";
+          break;
+        }
+      case webrtc::kRtpExtensionTransmissionTimeOffset: {
+          webrtc::RemoteBitrateEstimatorFactory factory;
+          *estimator = factory.Create(observer, clock, webrtc::kAimdControl,
+                                      kMinBitrateBps);
+          *estimator_used = "RemoteBitrateEstimator";
+          break;
+        }
+      default:
+        assert(false);
+    }
+  }
+  return true;
+}