1 /*
2  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
12 #define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
13 
14 #include <string>
15 
16 #include "modules/audio_coding/test/EncodeDecodeTest.h"
17 
18 namespace webrtc {
19 
20 class ReceiverWithPacketLoss : public Receiver {
21  public:
22   ReceiverWithPacketLoss();
23   void Setup(AudioCodingModule* acm,
24              RTPStream* rtpStream,
25              std::string out_file_name,
26              int channels,
27              int file_num,
28              int loss_rate,
29              int burst_length);
30   bool IncomingPacket() override;
31 
32  protected:
33   bool PacketLost();
34   int loss_rate_;
35   int burst_length_;
36   int packet_counter_;
37   int lost_packet_counter_;
38   int burst_lost_counter_;
39 };
40 
41 class SenderWithFEC : public Sender {
42  public:
43   SenderWithFEC();
44   void Setup(AudioCodingModule* acm,
45              RTPStream* rtpStream,
46              std::string in_file_name,
47              int payload_type,
48              SdpAudioFormat format,
49              int expected_loss_rate);
50   bool SetPacketLossRate(int expected_loss_rate);
51   bool SetFEC(bool enable_fec);
52 
53  protected:
54   int expected_loss_rate_;
55 };
56 
57 class PacketLossTest {
58  public:
59   PacketLossTest(int channels,
60                  int expected_loss_rate_,
61                  int actual_loss_rate,
62                  int burst_length);
63   void Perform();
64 
65  protected:
66   int channels_;
67   std::string in_file_name_;
68   int sample_rate_hz_;
69   int expected_loss_rate_;
70   int actual_loss_rate_;
71   int burst_length_;
72 };
73 
74 }  // namespace webrtc
75 
76 #endif  // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
77