blob: 1753741c06e8d30b4d9bca5069645a6d8d187612 [file] [log] [blame]
/*
* Copyright 2018 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 VIDEO_END_TO_END_TESTS_MULTI_STREAM_TESTER_H_
#define VIDEO_END_TO_END_TESTS_MULTI_STREAM_TESTER_H_
#include <map>
#include <memory>
#include "api/task_queue/task_queue_base.h"
#include "call/call.h"
#include "test/direct_transport.h"
#include "test/frame_generator_capturer.h"
namespace webrtc {
// Test sets up a Call multiple senders with different resolutions and SSRCs.
// Another is set up to receive all three of these with different renderers.
class MultiStreamTester {
public:
static constexpr size_t kNumStreams = 3;
const uint8_t kVideoPayloadType = 124;
const std::map<uint8_t, MediaType> payload_type_map_ = {
{kVideoPayloadType, MediaType::VIDEO}};
struct CodecSettings {
uint32_t ssrc;
int width;
int height;
} codec_settings[kNumStreams];
MultiStreamTester();
virtual ~MultiStreamTester();
void RunTest();
protected:
virtual void Wait() = 0;
// Note: frame_generator is a point-to-pointer, since the actual instance
// hasn't been created at the time of this call. Only when packets/frames
// start flowing should this be dereferenced.
virtual void UpdateSendConfig(size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator);
virtual void UpdateReceiveConfig(size_t stream_index,
VideoReceiveStream::Config* receive_config);
virtual std::unique_ptr<test::DirectTransport> CreateSendTransport(
TaskQueueBase* task_queue,
Call* sender_call);
virtual std::unique_ptr<test::DirectTransport> CreateReceiveTransport(
TaskQueueBase* task_queue,
Call* receiver_call);
};
} // namespace webrtc
#endif // VIDEO_END_TO_END_TESTS_MULTI_STREAM_TESTER_H_