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Diffstat (limited to 'webrtc/modules/audio_coding/test/PacketLossTest.h')
-rw-r--r-- | webrtc/modules/audio_coding/test/PacketLossTest.h | 67 |
1 files changed, 67 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/test/PacketLossTest.h b/webrtc/modules/audio_coding/test/PacketLossTest.h new file mode 100644 index 0000000000..f3570ae1ca --- /dev/null +++ b/webrtc/modules/audio_coding/test/PacketLossTest.h @@ -0,0 +1,67 @@ +/* + * 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_TEST_PACKETLOSSTEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ + +#include <string> +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.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_; + rtc::scoped_ptr<SenderWithFEC> sender_; + rtc::scoped_ptr<ReceiverWithPacketLoss> receiver_; + int expected_loss_rate_; + int actual_loss_rate_; + int burst_length_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ |