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/*
* 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.
*/
#ifndef WEBRTC_TEST_COMMON_CALL_TEST_H_
#define WEBRTC_TEST_COMMON_CALL_TEST_H_
#include <vector>
#include "webrtc/call.h"
#include "webrtc/system_wrappers/include/scoped_vector.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/rtp_rtcp_observer.h"
namespace webrtc {
namespace test {
class BaseTest;
class CallTest : public ::testing::Test {
public:
CallTest();
~CallTest();
static const size_t kNumSsrcs = 3;
static const unsigned int kDefaultTimeoutMs;
static const unsigned int kLongTimeoutMs;
static const uint8_t kSendPayloadType;
static const uint8_t kSendRtxPayloadType;
static const uint8_t kFakeSendPayloadType;
static const uint8_t kRedPayloadType;
static const uint8_t kRtxRedPayloadType;
static const uint8_t kUlpfecPayloadType;
static const uint32_t kSendRtxSsrcs[kNumSsrcs];
static const uint32_t kSendSsrcs[kNumSsrcs];
static const uint32_t kReceiverLocalSsrc;
static const int kNackRtpHistoryMs;
protected:
void RunBaseTest(BaseTest* test, const FakeNetworkPipe::Config& config);
void CreateCalls(const Call::Config& sender_config,
const Call::Config& receiver_config);
void CreateSenderCall(const Call::Config& config);
void CreateReceiverCall(const Call::Config& config);
void DestroyCalls();
void CreateSendConfig(size_t num_streams, Transport* send_transport);
void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
void CreateFrameGeneratorCapturer();
void CreateStreams();
void Start();
void Stop();
void DestroyStreams();
Clock* const clock_;
rtc::scoped_ptr<Call> sender_call_;
rtc::scoped_ptr<PacketTransport> send_transport_;
VideoSendStream::Config send_config_;
VideoEncoderConfig encoder_config_;
VideoSendStream* send_stream_;
rtc::scoped_ptr<Call> receiver_call_;
rtc::scoped_ptr<PacketTransport> receive_transport_;
std::vector<VideoReceiveStream::Config> receive_configs_;
std::vector<VideoReceiveStream*> receive_streams_;
rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
test::FakeEncoder fake_encoder_;
ScopedVector<VideoDecoder> allocated_decoders_;
};
class BaseTest : public RtpRtcpObserver {
public:
explicit BaseTest(unsigned int timeout_ms);
virtual ~BaseTest();
virtual void PerformTest() = 0;
virtual bool ShouldCreateReceivers() const = 0;
virtual size_t GetNumStreams() const;
virtual Call::Config GetSenderCallConfig();
virtual Call::Config GetReceiverCallConfig();
virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
virtual void OnTransportsCreated(PacketTransport* send_transport,
PacketTransport* receive_transport);
virtual void ModifyConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config);
virtual void OnStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams);
virtual void OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer);
};
class SendTest : public BaseTest {
public:
explicit SendTest(unsigned int timeout_ms);
bool ShouldCreateReceivers() const override;
};
class EndToEndTest : public BaseTest {
public:
explicit EndToEndTest(unsigned int timeout_ms);
bool ShouldCreateReceivers() const override;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_TEST_COMMON_CALL_TEST_H_
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