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}
2017-09-15 06:15:48 +02:00
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
--- a/modules/rtp_rtcp/test/testAPI/test_api.h
+++ b/modules/rtp_rtcp/test/testAPI/test_api.h
@ -0,0 +1,72 @@
+/*
+ *  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.
+ */
+#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
+#define WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
+
+#include "webrtc/api/call/transport.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+// This class sends all its packet straight to the provided RtpRtcp module.
+// with optional packet loss.
+class LoopBackTransport : public Transport {
+ public:
+  LoopBackTransport()
+      : count_(0),
+        packet_loss_(0),
+        rtp_payload_registry_(NULL),
+        rtp_receiver_(NULL),
+        rtp_rtcp_module_(NULL) {}
+  void SetSendModule(RtpRtcp* rtp_rtcp_module,
+                     RTPPayloadRegistry* payload_registry,
+                     RtpReceiver* receiver,
+                     ReceiveStatistics* receive_statistics);
+  void DropEveryNthPacket(int n);
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override;
+  bool SendRtcp(const uint8_t* data, size_t len) override;
+
+ private:
+  int count_;
+  int packet_loss_;
+  ReceiveStatistics* receive_statistics_;
+  RTPPayloadRegistry* rtp_payload_registry_;
+  RtpReceiver* rtp_receiver_;
+  RtpRtcp* rtp_rtcp_module_;
+};
+
+class TestRtpReceiver : public RtpData {
+ public:
+  int32_t OnReceivedPayloadData(
+      const uint8_t* payload_data,
+      size_t payload_size,
+      const webrtc::WebRtcRTPHeader* rtp_header) override;
+
+  const uint8_t* payload_data() const { return payload_data_; }
+  size_t payload_size() const { return payload_size_; }
+  webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; }
+
+ private:
+  uint8_t payload_data_[1500];
+  size_t payload_size_;
+  webrtc::WebRtcRTPHeader rtp_header_;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_