120 lines
3.6 KiB
C++
120 lines
3.6 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <list>
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#include <memory>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/audio_state.h"
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#include "webrtc/call.h"
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#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
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#include "webrtc/test/mock_voice_engine.h"
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namespace {
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struct CallHelper {
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explicit CallHelper(
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
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: voice_engine_(decoder_factory) {
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webrtc::AudioState::Config audio_state_config;
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audio_state_config.voice_engine = &voice_engine_;
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webrtc::Call::Config config;
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config.audio_state = webrtc::AudioState::Create(audio_state_config);
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call_.reset(webrtc::Call::Create(config));
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}
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webrtc::Call* operator->() { return call_.get(); }
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private:
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testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
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std::unique_ptr<webrtc::Call> call_;
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};
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} // namespace
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namespace webrtc {
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TEST(CallTest, ConstructDestruct) {
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CallHelper call;
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}
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TEST(CallTest, CreateDestroy_AudioSendStream) {
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CallHelper call;
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AudioSendStream::Config config(nullptr);
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config.rtp.ssrc = 42;
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config.voe_channel_id = 123;
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AudioSendStream* stream = call->CreateAudioSendStream(config);
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EXPECT_NE(stream, nullptr);
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call->DestroyAudioSendStream(stream);
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}
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TEST(CallTest, CreateDestroy_AudioReceiveStream) {
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
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CallHelper call(decoder_factory);
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AudioReceiveStream::Config config;
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config.rtp.remote_ssrc = 42;
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config.voe_channel_id = 123;
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config.decoder_factory = decoder_factory;
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AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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call->DestroyAudioReceiveStream(stream);
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}
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TEST(CallTest, CreateDestroy_AudioSendStreams) {
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CallHelper call;
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AudioSendStream::Config config(nullptr);
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config.voe_channel_id = 123;
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std::list<AudioSendStream*> streams;
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for (int i = 0; i < 2; ++i) {
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for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
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config.rtp.ssrc = ssrc;
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AudioSendStream* stream = call->CreateAudioSendStream(config);
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EXPECT_NE(stream, nullptr);
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if (ssrc & 1) {
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streams.push_back(stream);
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} else {
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streams.push_front(stream);
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}
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}
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for (auto s : streams) {
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call->DestroyAudioSendStream(s);
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}
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streams.clear();
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}
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}
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TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory(
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new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>);
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CallHelper call(decoder_factory);
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AudioReceiveStream::Config config;
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config.voe_channel_id = 123;
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config.decoder_factory = decoder_factory;
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std::list<AudioReceiveStream*> streams;
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for (int i = 0; i < 2; ++i) {
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for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
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config.rtp.remote_ssrc = ssrc;
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AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
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EXPECT_NE(stream, nullptr);
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if (ssrc & 1) {
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streams.push_back(stream);
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} else {
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streams.push_front(stream);
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}
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}
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for (auto s : streams) {
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call->DestroyAudioReceiveStream(s);
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}
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streams.clear();
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}
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}
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} // namespace webrtc
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