Support for Signal calls.

Merge in RedPhone

// FREEBIE

jni/webrtc/modules/audio_coding/main/test/PacketLossTest.h

@@ -0,0 +1,66 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+
+#include <string>
+#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class ReceiverWithPacketLoss : public Receiver {
+ public:
+  ReceiverWithPacketLoss();
+  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+             std::string out_file_name, int channels, int loss_rate,
+             int burst_length);
+  bool IncomingPacket() OVERRIDE;
+ protected:
+  bool PacketLost();
+  int loss_rate_;
+  int burst_length_;
+  int packet_counter_;
+  int lost_packet_counter_;
+  int burst_lost_counter_;
+};
+
+class SenderWithFEC : public Sender {
+ public:
+  SenderWithFEC();
+  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+             std::string in_file_name, int sample_rate, int channels,
+             int expected_loss_rate);
+  bool SetPacketLossRate(int expected_loss_rate);
+  bool SetFEC(bool enable_fec);
+ protected:
+  int expected_loss_rate_;
+};
+
+class PacketLossTest : public ACMTest {
+ public:
+  PacketLossTest(int channels, int expected_loss_rate_, int actual_loss_rate,
+                 int burst_length);
+  void Perform();
+ protected:
+  int channels_;
+  std::string in_file_name_;
+  int sample_rate_hz_;
+  scoped_ptr<SenderWithFEC> sender_;
+  scoped_ptr<ReceiverWithPacketLoss> receiver_;
+  int expected_loss_rate_;
+  int actual_loss_rate_;
+  int burst_length_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_