| /* |
| * 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 MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ |
| #define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ |
| |
| #include <string> |
| |
| #include "absl/strings/string_view.h" |
| #include "modules/audio_coding/test/EncodeDecodeTest.h" |
| |
| namespace webrtc { |
| |
| class ReceiverWithPacketLoss : public Receiver { |
| public: |
| ReceiverWithPacketLoss(); |
| void Setup(AudioCodingModule* acm, |
| RTPStream* rtpStream, |
| absl::string_view out_file_name, |
| int channels, |
| int file_num, |
| 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, |
| absl::string_view in_file_name, |
| int payload_type, |
| SdpAudioFormat format, |
| int expected_loss_rate); |
| bool SetPacketLossRate(int expected_loss_rate); |
| bool SetFEC(bool enable_fec); |
| |
| protected: |
| int expected_loss_rate_; |
| }; |
| |
| class PacketLossTest { |
| 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_; |
| int expected_loss_rate_; |
| int actual_loss_rate_; |
| int burst_length_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ |