blob: a6c56fa78071cc8e2203e7027329821535055a72 [file] [log] [blame]
/*
* 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 "modules/audio_coding/test/PacketLossTest.h"
#include <memory>
#include "test/gtest.h"
#include "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
std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
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