a33f05e8d7
Changes differing from https://webrtc-codereview.appspot.com/37859004: * I put the include_tests==1 stuff of audio_coding.gypi in its own audio_coding_tests.gypi file, including the Android and isolate targets which were incorrectly located in the previous CL * I moved the bwe utilities in remote_bitrate_estimator.gypi into include_tests==1 since they depend on test.gyp after I cleaned up the duplicated inclusion of rtp_file_reader.cc R=stefan@webrtc.org TBR=tina.legrand@webrtc.org TESTED=Passing gyp and compile using: webrtc/build/gyp_webrtc -Dinclude_tests=1 webrtc/build/gyp_webrtc -Dinclude_tests=0 I also setup a Chromium checkout with my checkout mounted in third_party/webrtc and ran build/gyp_chromium successfully. BUG=4185 Review URL: https://webrtc-codereview.appspot.com/33159004 Cr-Commit-Position: refs/heads/master@{#8205} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
124 lines
4.1 KiB
C++
124 lines
4.1 KiB
C++
/*
|
|
* Copyright (c) 2012 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/tools/agc/agc_manager.h"
|
|
|
|
#include "testing/gmock/include/gmock/gmock.h"
|
|
#include "testing/gtest/include/gtest/gtest.h"
|
|
#include "webrtc/modules/audio_processing/agc/mock_agc.h"
|
|
#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
|
|
#include "webrtc/system_wrappers/interface/sleep.h"
|
|
#include "webrtc/test/channel_transport/include/channel_transport.h"
|
|
#include "webrtc/test/testsupport/gtest_disable.h"
|
|
#include "webrtc/voice_engine/include/voe_base.h"
|
|
#include "webrtc/voice_engine/include/voe_external_media.h"
|
|
#include "webrtc/voice_engine/include/voe_network.h"
|
|
#include "webrtc/voice_engine/include/voe_volume_control.h"
|
|
|
|
using ::testing::_;
|
|
using ::testing::AtLeast;
|
|
using ::testing::Mock;
|
|
using ::testing::Return;
|
|
|
|
namespace webrtc {
|
|
|
|
class AgcManagerTest : public ::testing::Test {
|
|
protected:
|
|
AgcManagerTest()
|
|
: voe_(VoiceEngine::Create()),
|
|
base_(VoEBase::GetInterface(voe_)),
|
|
agc_(new MockAgc()),
|
|
manager_(new AgcManager(VoEExternalMedia::GetInterface(voe_),
|
|
VoEVolumeControl::GetInterface(voe_),
|
|
agc_,
|
|
AudioProcessing::Create())),
|
|
channel_(-1) {
|
|
}
|
|
|
|
virtual void SetUp() {
|
|
ASSERT_TRUE(voe_ != NULL);
|
|
ASSERT_TRUE(base_ != NULL);
|
|
ASSERT_EQ(0, base_->Init());
|
|
channel_ = base_->CreateChannel();
|
|
ASSERT_NE(-1, channel_);
|
|
|
|
VoENetwork* network = VoENetwork::GetInterface(voe_);
|
|
ASSERT_TRUE(network != NULL);
|
|
channel_transport_.reset(
|
|
new test::VoiceChannelTransport(network, channel_));
|
|
ASSERT_EQ(0, channel_transport_->SetSendDestination("127.0.0.1", 1234));
|
|
network->Release();
|
|
}
|
|
|
|
virtual void TearDown() {
|
|
channel_transport_.reset(NULL);
|
|
ASSERT_EQ(0, base_->DeleteChannel(channel_));
|
|
ASSERT_EQ(0, base_->Terminate());
|
|
delete manager_;
|
|
// Test that the manager has released all VoE interfaces. The last
|
|
// reference is released in VoiceEngine::Delete.
|
|
EXPECT_EQ(1, base_->Release());
|
|
ASSERT_TRUE(VoiceEngine::Delete(voe_));
|
|
}
|
|
|
|
VoiceEngine* voe_;
|
|
VoEBase* base_;
|
|
MockAgc* agc_;
|
|
scoped_ptr<test::VoiceChannelTransport> channel_transport_;
|
|
// We use a pointer for the manager, so we can tear it down and test
|
|
// base_->Release() in the destructor.
|
|
AgcManager* manager_;
|
|
int channel_;
|
|
};
|
|
|
|
TEST_F(AgcManagerTest, DISABLED_ON_ANDROID(EnableSucceeds)) {
|
|
EXPECT_EQ(0, manager_->Enable(true));
|
|
EXPECT_TRUE(manager_->enabled());
|
|
EXPECT_EQ(0, manager_->Enable(false));
|
|
EXPECT_FALSE(manager_->enabled());
|
|
}
|
|
|
|
TEST_F(AgcManagerTest, DISABLED_ON_ANDROID(ProcessIsNotCalledByDefault)) {
|
|
EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).Times(0);
|
|
EXPECT_CALL(*agc_, Process(_, _, _)).Times(0);
|
|
EXPECT_CALL(*agc_, GetRmsErrorDb(_)).Times(0);
|
|
ASSERT_EQ(0, base_->StartSend(channel_));
|
|
SleepMs(100);
|
|
ASSERT_EQ(0, base_->StopSend(channel_));
|
|
}
|
|
|
|
TEST_F(AgcManagerTest, DISABLED_ProcessIsCalledOnlyWhenEnabled) {
|
|
EXPECT_CALL(*agc_, Reset());
|
|
EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
|
|
.Times(AtLeast(1))
|
|
.WillRepeatedly(Return(0));
|
|
EXPECT_CALL(*agc_, Process(_, _, _))
|
|
.Times(AtLeast(1))
|
|
.WillRepeatedly(Return(0));
|
|
EXPECT_CALL(*agc_, GetRmsErrorDb(_))
|
|
.Times(AtLeast(1))
|
|
.WillRepeatedly(Return(false));
|
|
EXPECT_EQ(0, manager_->Enable(true));
|
|
ASSERT_EQ(0, base_->StartSend(channel_));
|
|
SleepMs(100);
|
|
EXPECT_EQ(0, manager_->Enable(false));
|
|
SleepMs(100);
|
|
Mock::VerifyAndClearExpectations(agc_);
|
|
|
|
EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).Times(0);
|
|
EXPECT_CALL(*agc_, Process(_, _, _)).Times(0);
|
|
EXPECT_CALL(*agc_, GetRmsErrorDb(_)).Times(0);
|
|
SleepMs(100);
|
|
ASSERT_EQ(0, base_->StopSend(channel_));
|
|
}
|
|
|
|
} // namespace webrtc
|