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webrtc_m130/test/call_test.h
Elad Alon 389b1672a3 Delete test/constants.h 
It's not possible to use constants.h for all RTP extensions
after the number of extensions exceeds 14, which is the maximum
number of one-byte RTP extensions. This is because some extensions
would have to be assigned a number greater than 14, even if the
test only involves 14 extensions or less.

For uniformity's sake, this CL also edits some files to use an
enum as the files involved in this CL, rather than free-floating
const-ints.

Bug: webrtc:10288
Change-Id: Ib5e58ad72c4d3756f4c4f6521f140ec59617f3f5
Reviewed-on: https://webrtc-review.googlesource.com/c/123048
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26728}
2019-02-17 21:47:41 +00:00

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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 TEST_CALL_TEST_H_
#define TEST_CALL_TEST_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/call.h"
#include "call/rtp_transport_controller_send.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"
#include "test/fake_videorenderer.h"
#include "test/fake_vp8_encoder.h"
#include "test/frame_generator_capturer.h"
#include "test/rtp_rtcp_observer.h"
#include "test/single_threaded_task_queue.h"
namespace webrtc {
namespace test {
class BaseTest;
class CallTest : public ::testing::Test {
public:
CallTest();
virtual ~CallTest();
static constexpr size_t kNumSsrcs = 6;
static const int kNumSimulcastStreams = 3;
static const int kDefaultWidth = 320;
static const int kDefaultHeight = 180;
static const int kDefaultFramerate = 30;
static const int kDefaultTimeoutMs;
static const int kLongTimeoutMs;
enum classPayloadTypes : uint8_t {
kSendRtxPayloadType = 98,
kRtxRedPayloadType = 99,
kVideoSendPayloadType = 100,
kAudioSendPayloadType = 103,
kRedPayloadType = 118,
kUlpfecPayloadType = 119,
kFlexfecPayloadType = 120,
kPayloadTypeH264 = 122,
kPayloadTypeVP8 = 123,
kPayloadTypeVP9 = 124,
kFakeVideoSendPayloadType = 125,
};
static const uint32_t kSendRtxSsrcs[kNumSsrcs];
static const uint32_t kVideoSendSsrcs[kNumSsrcs];
static const uint32_t kAudioSendSsrc;
static const uint32_t kFlexfecSendSsrc;
static const uint32_t kReceiverLocalVideoSsrc;
static const uint32_t kReceiverLocalAudioSsrc;
static const int kNackRtpHistoryMs;
static const uint8_t kDefaultKeepalivePayloadType;
static const std::map<uint8_t, MediaType> payload_type_map_;
protected:
void RegisterRtpExtension(const RtpExtension& extension);
// RunBaseTest overwrites the audio_state of the send and receive Call configs
// to simplify test code.
void RunBaseTest(BaseTest* test);
void CreateCalls();
void CreateCalls(const Call::Config& sender_config,
const Call::Config& receiver_config);
void CreateSenderCall();
void CreateSenderCall(const Call::Config& config);
void CreateReceiverCall(const Call::Config& config);
void DestroyCalls();
void CreateVideoSendConfig(VideoSendStream::Config* video_config,
size_t num_video_streams,
size_t num_used_ssrcs,
Transport* send_transport);
void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
size_t num_flexfec_streams,
Transport* send_transport);
void SetAudioConfig(const AudioSendStream::Config& config);
void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
void SetReceiveUlpFecConfig(VideoReceiveStream::Config* receive_config);
void CreateSendConfig(size_t num_video_streams,
size_t num_audio_streams,
size_t num_flexfec_streams,
Transport* send_transport);
void CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport);
void CreateMatchingVideoReceiveConfigs(
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
bool send_side_bwe,
VideoDecoderFactory* decoder_factory,
absl::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms);
void AddMatchingVideoReceiveConfigs(
std::vector<VideoReceiveStream::Config>* receive_configs,
const VideoSendStream::Config& video_send_config,
Transport* rtcp_send_transport,
bool send_side_bwe,
VideoDecoderFactory* decoder_factory,
absl::optional<size_t> decode_sub_stream,
bool receiver_reference_time_report,
int rtp_history_ms);
void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
static AudioReceiveStream::Config CreateMatchingAudioConfig(
const AudioSendStream::Config& send_config,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
Transport* transport,
std::string sync_group);
void CreateMatchingFecConfig(
Transport* transport,
const VideoSendStream::Config& video_send_config);
void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
float speed,
int framerate,
int width,
int height);
void CreateFrameGeneratorCapturer(int framerate, int width, int height);
void CreateFakeAudioDevices(
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);
void CreateVideoStreams();
void CreateVideoSendStreams();
void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
void CreateAudioStreams();
void CreateFlexfecStreams();
void ConnectVideoSourcesToStreams();
void AssociateFlexfecStreamsWithVideoStreams();
void DissociateFlexfecStreamsFromVideoStreams();
void Start();
void StartVideoStreams();
void Stop();
void StopVideoStreams();
void DestroyStreams();
void DestroyVideoSendStreams();
void SetFakeVideoCaptureRotation(VideoRotation rotation);
void SetVideoDegradation(DegradationPreference preference);
VideoSendStream::Config* GetVideoSendConfig();
void SetVideoSendConfig(const VideoSendStream::Config& config);
VideoEncoderConfig* GetVideoEncoderConfig();
void SetVideoEncoderConfig(const VideoEncoderConfig& config);
VideoSendStream* GetVideoSendStream();
FlexfecReceiveStream::Config* GetFlexFecConfig();
Clock* const clock_;
std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
std::unique_ptr<Call> sender_call_;
RtpTransportControllerSend* sender_call_transport_controller_;
std::unique_ptr<PacketTransport> send_transport_;
std::vector<VideoSendStream::Config> video_send_configs_;
std::vector<VideoEncoderConfig> video_encoder_configs_;
std::vector<VideoSendStream*> video_send_streams_;
AudioSendStream::Config audio_send_config_;
AudioSendStream* audio_send_stream_;
std::unique_ptr<Call> receiver_call_;
std::unique_ptr<PacketTransport> receive_transport_;
std::vector<VideoReceiveStream::Config> video_receive_configs_;
std::vector<VideoReceiveStream*> video_receive_streams_;
std::vector<AudioReceiveStream::Config> audio_receive_configs_;
std::vector<AudioReceiveStream*> audio_receive_streams_;
std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
test::FrameGeneratorCapturer* frame_generator_capturer_;
std::vector<std::unique_ptr<rtc::VideoSourceInterface<VideoFrame>>>
video_sources_;
DegradationPreference degradation_preference_ =
DegradationPreference::MAINTAIN_FRAMERATE;
std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
test::FunctionVideoEncoderFactory fake_encoder_factory_;
int fake_encoder_max_bitrate_ = -1;
test::FunctionVideoDecoderFactory fake_decoder_factory_;
std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
// Number of simulcast substreams.
size_t num_video_streams_;
size_t num_audio_streams_;
size_t num_flexfec_streams_;
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
test::FakeVideoRenderer fake_renderer_;
SingleThreadedTaskQueueForTesting task_queue_;
private:
absl::optional<RtpExtension> GetRtpExtensionByUri(
const std::string& uri) const;
void AddRtpExtensionByUri(const std::string& uri,
std::vector<RtpExtension>* extensions) const;
std::vector<RtpExtension> rtp_extensions_;
rtc::scoped_refptr<AudioProcessing> apm_send_;
rtc::scoped_refptr<AudioProcessing> apm_recv_;
rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
};
class BaseTest : public RtpRtcpObserver {
public:
BaseTest();
explicit BaseTest(int timeout_ms);
virtual ~BaseTest();
virtual void PerformTest() = 0;
virtual bool ShouldCreateReceivers() const = 0;
virtual size_t GetNumVideoStreams() const;
virtual size_t GetNumAudioStreams() const;
virtual size_t GetNumFlexfecStreams() const;
virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
virtual void OnFakeAudioDevicesCreated(
TestAudioDeviceModule* send_audio_device,
TestAudioDeviceModule* recv_audio_device);
virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);
virtual void OnRtpTransportControllerSendCreated(
RtpTransportControllerSend* controller);
virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
virtual test::PacketTransport* CreateSendTransport(
SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call);
virtual test::PacketTransport* CreateReceiveTransport(
SingleThreadedTaskQueueForTesting* task_queue);
virtual void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config);
virtual void ModifyVideoCaptureStartResolution(int* width,
int* heigt,
int* frame_rate);
virtual void ModifyVideoDegradationPreference(
DegradationPreference* degradation_preference);
virtual void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams);
virtual void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs);
virtual void OnAudioStreamsCreated(
AudioSendStream* send_stream,
const std::vector<AudioReceiveStream*>& receive_streams);
virtual void ModifyFlexfecConfigs(
std::vector<FlexfecReceiveStream::Config>* receive_configs);
virtual void OnFlexfecStreamsCreated(
const std::vector<FlexfecReceiveStream*>& receive_streams);
virtual void OnFrameGeneratorCapturerCreated(
FrameGeneratorCapturer* frame_generator_capturer);
virtual void OnStreamsStopped();
};
class SendTest : public BaseTest {
public:
explicit SendTest(int timeout_ms);
bool ShouldCreateReceivers() const override;
};
class EndToEndTest : public BaseTest {
public:
EndToEndTest();
explicit EndToEndTest(int timeout_ms);
bool ShouldCreateReceivers() const override;
};
} // namespace test
} // namespace webrtc
#endif // TEST_CALL_TEST_H_
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