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PacketLossTest.h (2259B)

      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 "api/audio_codecs/audio_format.h"
     18 #include "api/environment/environment.h"
     19 #include "api/neteq/neteq.h"
     20 #include "modules/audio_coding/include/audio_coding_module.h"
     21 #include "modules/audio_coding/test/EncodeDecodeTest.h"
     22 #include "modules/audio_coding/test/RTPFile.h"
     23 
     24 namespace webrtc {
     25 
     26 class ReceiverWithPacketLoss : public Receiver {
     27 public:
     28  ReceiverWithPacketLoss();
     29  void Setup(NetEq* neteq,
     30             RTPStream* rtpStream,
     31             absl::string_view out_file_name,
     32             int channels,
     33             int file_num,
     34             int loss_rate,
     35             int burst_length);
     36  bool IncomingPacket() override;
     37 
     38 protected:
     39  bool PacketLost();
     40  int loss_rate_;
     41  int burst_length_;
     42  int packet_counter_;
     43  int lost_packet_counter_;
     44  int burst_lost_counter_;
     45 };
     46 
     47 class SenderWithFEC : public Sender {
     48 public:
     49  SenderWithFEC();
     50  void Setup(const Environment& env,
     51             AudioCodingModule* acm,
     52             RTPStream* rtpStream,
     53             absl::string_view in_file_name,
     54             int payload_type,
     55             SdpAudioFormat format,
     56             int expected_loss_rate);
     57  bool SetPacketLossRate(int expected_loss_rate);
     58  bool SetFEC(bool enable_fec);
     59 
     60 protected:
     61  int expected_loss_rate_;
     62 };
     63 
     64 class PacketLossTest {
     65 public:
     66  PacketLossTest(int channels,
     67                 int expected_loss_rate_,
     68                 int actual_loss_rate,
     69                 int burst_length);
     70  void Perform();
     71 
     72 protected:
     73  int channels_;
     74  std::string in_file_name_;
     75  int sample_rate_hz_;
     76  int expected_loss_rate_;
     77  int actual_loss_rate_;
     78  int burst_length_;
     79 };
     80 
     81 }  // namespace webrtc
     82 
     83 #endif  // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_