xref: /aosp_15_r20/external/webrtc/modules/audio_coding/test/PacketLossTest.h (revision d9f758449e529ab9291ac668be2861e7a55c2422)
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 "absl/strings/string_view.h"
17 #include "modules/audio_coding/test/EncodeDecodeTest.h"
18 
19 namespace webrtc {
20 
21 class ReceiverWithPacketLoss : public Receiver {
22  public:
23   ReceiverWithPacketLoss();
24   void Setup(AudioCodingModule* acm,
25              RTPStream* rtpStream,
26              absl::string_view out_file_name,
27              int channels,
28              int file_num,
29              int loss_rate,
30              int burst_length);
31   bool IncomingPacket() override;
32 
33  protected:
34   bool PacketLost();
35   int loss_rate_;
36   int burst_length_;
37   int packet_counter_;
38   int lost_packet_counter_;
39   int burst_lost_counter_;
40 };
41 
42 class SenderWithFEC : public Sender {
43  public:
44   SenderWithFEC();
45   void Setup(AudioCodingModule* acm,
46              RTPStream* rtpStream,
47              absl::string_view in_file_name,
48              int payload_type,
49              SdpAudioFormat format,
50              int expected_loss_rate);
51   bool SetPacketLossRate(int expected_loss_rate);
52   bool SetFEC(bool enable_fec);
53 
54  protected:
55   int expected_loss_rate_;
56 };
57 
58 class PacketLossTest {
59  public:
60   PacketLossTest(int channels,
61                  int expected_loss_rate_,
62                  int actual_loss_rate,
63                  int burst_length);
64   void Perform();
65 
66  protected:
67   int channels_;
68   std::string in_file_name_;
69   int sample_rate_hz_;
70   int expected_loss_rate_;
71   int actual_loss_rate_;
72   int burst_length_;
73 };
74 
75 }  // namespace webrtc
76 
77 #endif  // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
78