Index: webrtc/modules/audio_coding/main/test/PacketLossTest.cc |
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 f7c96faacd691e994e2768f1eb5c0a6cee501583..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc |
+++ /dev/null |
@@ -1,167 +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_); |
- 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 |