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/*
* Copyright (c) 2015 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 <list>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio_state.h"
#include "webrtc/call.h"
#include "webrtc/test/mock_voice_engine.h"
namespace {
struct CallHelper {
CallHelper() {
webrtc::AudioState::Config audio_state_config;
audio_state_config.voice_engine = &voice_engine_;
webrtc::Call::Config config;
config.audio_state = webrtc::AudioState::Create(audio_state_config);
call_.reset(webrtc::Call::Create(config));
}
webrtc::Call* operator->() { return call_.get(); }
private:
testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
rtc::scoped_ptr<webrtc::Call> call_;
};
} // namespace
namespace webrtc {
TEST(CallTest, ConstructDestruct) {
CallHelper call;
}
TEST(CallTest, CreateDestroy_AudioSendStream) {
CallHelper call;
AudioSendStream::Config config(nullptr);
config.rtp.ssrc = 42;
config.voe_channel_id = 123;
AudioSendStream* stream = call->CreateAudioSendStream(config);
EXPECT_NE(stream, nullptr);
call->DestroyAudioSendStream(stream);
}
TEST(CallTest, CreateDestroy_AudioReceiveStream) {
CallHelper call;
AudioReceiveStream::Config config;
config.rtp.remote_ssrc = 42;
config.voe_channel_id = 123;
AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
EXPECT_NE(stream, nullptr);
call->DestroyAudioReceiveStream(stream);
}
TEST(CallTest, CreateDestroy_AudioSendStreams) {
CallHelper call;
AudioSendStream::Config config(nullptr);
config.voe_channel_id = 123;
std::list<AudioSendStream*> streams;
for (int i = 0; i < 2; ++i) {
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
config.rtp.ssrc = ssrc;
AudioSendStream* stream = call->CreateAudioSendStream(config);
EXPECT_NE(stream, nullptr);
if (ssrc & 1) {
streams.push_back(stream);
} else {
streams.push_front(stream);
}
}
for (auto s : streams) {
call->DestroyAudioSendStream(s);
}
streams.clear();
}
}
TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
CallHelper call;
AudioReceiveStream::Config config;
config.voe_channel_id = 123;
std::list<AudioReceiveStream*> streams;
for (int i = 0; i < 2; ++i) {
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
config.rtp.remote_ssrc = ssrc;
AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
EXPECT_NE(stream, nullptr);
if (ssrc & 1) {
streams.push_back(stream);
} else {
streams.push_front(stream);
}
}
for (auto s : streams) {
call->DestroyAudioReceiveStream(s);
}
streams.clear();
}
}
} // namespace webrtc
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