mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-16 23:30:48 +01:00

This is a reland of fb82fcc7f9
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org
Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
240 lines
8.7 KiB
C++
240 lines
8.7 KiB
C++
/*
|
|
* 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 <memory>
|
|
#include <vector>
|
|
|
|
#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/frame_generator_capturer.h"
|
|
#include "test/function_video_encoder_factory.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:
|
|
// RunBaseTest overwrites the audio_state of the send and receive Call configs
|
|
// to simplify test code.
|
|
void RunBaseTest(BaseTest* test);
|
|
|
|
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 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 CreateSendConfig(size_t num_video_streams,
|
|
size_t num_audio_streams,
|
|
size_t num_flexfec_streams,
|
|
Transport* send_transport);
|
|
|
|
std::vector<VideoReceiveStream::Config> CreateMatchingVideoReceiveConfigs(
|
|
const VideoSendStream::Config& video_send_config,
|
|
Transport* rtcp_send_transport);
|
|
void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
|
|
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 CreateAudioStreams();
|
|
void CreateFlexfecStreams();
|
|
|
|
void AssociateFlexfecStreamsWithVideoStreams();
|
|
void DissociateFlexfecStreamsFromVideoStreams();
|
|
|
|
void Start();
|
|
void Stop();
|
|
void DestroyStreams();
|
|
void SetFakeVideoCaptureRotation(VideoRotation rotation);
|
|
|
|
Clock* const clock_;
|
|
|
|
std::unique_ptr<webrtc::RtcEventLog> event_log_;
|
|
std::unique_ptr<Call> sender_call_;
|
|
RtpTransportControllerSend* sender_call_transport_controller_;
|
|
std::unique_ptr<PacketTransport> send_transport_;
|
|
VideoSendStream::Config video_send_config_;
|
|
VideoEncoderConfig video_encoder_config_;
|
|
VideoSendStream* video_send_stream_;
|
|
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_;
|
|
|
|
std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
|
|
test::FunctionVideoEncoderFactory fake_encoder_factory_;
|
|
int fake_encoder_max_bitrate_ = -1;
|
|
std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_;
|
|
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:
|
|
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(unsigned 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 Call::Config GetSenderCallConfig();
|
|
virtual Call::Config GetReceiverCallConfig();
|
|
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 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();
|
|
|
|
std::unique_ptr<webrtc::RtcEventLog> event_log_;
|
|
};
|
|
|
|
class SendTest : public BaseTest {
|
|
public:
|
|
explicit SendTest(unsigned int timeout_ms);
|
|
|
|
bool ShouldCreateReceivers() const override;
|
|
};
|
|
|
|
class EndToEndTest : public BaseTest {
|
|
public:
|
|
EndToEndTest();
|
|
explicit EndToEndTest(unsigned int timeout_ms);
|
|
|
|
bool ShouldCreateReceivers() const override;
|
|
};
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|
|
|
|
#endif // TEST_CALL_TEST_H_
|