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Unified Diff: webrtc/modules/audio_coding/main/test/PacketLossTest.cc

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years, 1 month ago
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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
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