diff options
Diffstat (limited to 'webrtc/modules/audio_coding/main/test/PacketLossTest.cc')
-rw-r--r-- | webrtc/modules/audio_coding/main/test/PacketLossTest.cc | 168 |
1 files changed, 0 insertions, 168 deletions
diff --git a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc b/webrtc/modules/audio_coding/main/test/PacketLossTest.cc deleted file mode 100644 index f19d491d2d..0000000000 --- a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc +++ /dev/null @@ -1,168 +0,0 @@ -/* - * 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/main/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_); - struct CodecInst sendCodecInst; - if (acm->SendCodec(&sendCodecInst) >= 0) { - 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 |