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Diffstat (limited to 'webrtc/modules/audio_coding/test/PacketLossTest.cc')
-rw-r--r-- | webrtc/modules/audio_coding/test/PacketLossTest.cc | 167 |
1 files changed, 167 insertions, 0 deletions
diff --git a/webrtc/modules/audio_coding/test/PacketLossTest.cc b/webrtc/modules/audio_coding/test/PacketLossTest.cc new file mode 100644 index 0000000000..ad3e83403e --- /dev/null +++ b/webrtc/modules/audio_coding/test/PacketLossTest.cc @@ -0,0 +1,167 @@ +/* + * 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. + */ + +#include "webrtc/modules/audio_coding/test/PacketLossTest.h" + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/common.h" +#include "webrtc/test/testsupport/fileutils.h" + +namespace webrtc { + +ReceiverWithPacketLoss::ReceiverWithPacketLoss() + : loss_rate_(0), + burst_length_(1), + packet_counter_(0), + lost_packet_counter_(0), + burst_lost_counter_(burst_length_) { +} + +void ReceiverWithPacketLoss::Setup(AudioCodingModule *acm, + RTPStream *rtpStream, + std::string out_file_name, + int channels, + int loss_rate, + int burst_length) { + loss_rate_ = loss_rate; + burst_length_ = burst_length; + burst_lost_counter_ = burst_length_; // To prevent first packet gets lost. + std::stringstream ss; + ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_"; + Receiver::Setup(acm, rtpStream, ss.str(), channels); +} + +bool ReceiverWithPacketLoss::IncomingPacket() { + if (!_rtpStream->EndOfFile()) { + if (packet_counter_ == 0) { + _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload, + _payloadSizeBytes, &_nextTime); + if (_realPayloadSizeBytes == 0) { + if (_rtpStream->EndOfFile()) { + packet_counter_ = 0; + return true; + } else { + return false; + } + } + } + + if (!PacketLost()) { + _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo); + } + packet_counter_++; + _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload, + _payloadSizeBytes, &_nextTime); + if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) { + packet_counter_ = 0; + lost_packet_counter_ = 0; + } + } + return true; +} + +bool ReceiverWithPacketLoss::PacketLost() { + if (burst_lost_counter_ < burst_length_) { + lost_packet_counter_++; + burst_lost_counter_++; + return true; + } + + if (lost_packet_counter_ * 100 < loss_rate_ * packet_counter_) { + lost_packet_counter_++; + burst_lost_counter_ = 1; + return true; + } + return false; +} + +SenderWithFEC::SenderWithFEC() + : expected_loss_rate_(0) { +} + +void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream, + std::string in_file_name, int sample_rate, + int channels, int expected_loss_rate) { + Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels); + EXPECT_TRUE(SetFEC(true)); + EXPECT_TRUE(SetPacketLossRate(expected_loss_rate)); +} + +bool SenderWithFEC::SetFEC(bool enable_fec) { + if (_acm->SetCodecFEC(enable_fec) == 0) { + return true; + } + return false; +} + +bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) { + if (_acm->SetPacketLossRate(expected_loss_rate) == 0) { + expected_loss_rate_ = expected_loss_rate; + return true; + } + return false; +} + +PacketLossTest::PacketLossTest(int channels, int expected_loss_rate, + int actual_loss_rate, int burst_length) + : channels_(channels), + in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" : + "audio_coding/teststereo32kHz"), + sample_rate_hz_(32000), + sender_(new SenderWithFEC), + receiver_(new ReceiverWithPacketLoss), + expected_loss_rate_(expected_loss_rate), + actual_loss_rate_(actual_loss_rate), + burst_length_(burst_length) { +} + +void PacketLossTest::Perform() { +#ifndef WEBRTC_CODEC_OPUS + return; +#else + rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0)); + + int codec_id = acm->Codec("opus", 48000, channels_); + + RTPFile rtpFile; + std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(), + "packet_loss_test"); + + // Encode to file + rtpFile.Open(fileName.c_str(), "wb+"); + rtpFile.WriteHeader(); + + sender_->testMode = 0; + sender_->codeId = codec_id; + + sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_, + expected_loss_rate_); + if (acm->SendCodec()) { + sender_->Run(); + } + sender_->Teardown(); + rtpFile.Close(); + + // Decode to file + rtpFile.Open(fileName.c_str(), "rb"); + rtpFile.ReadHeader(); + + receiver_->testMode = 0; + receiver_->codeId = codec_id; + + receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, + actual_loss_rate_, burst_length_); + receiver_->Run(); + receiver_->Teardown(); + rtpFile.Close(); +#endif +} + +} // namespace webrtc |