webrtc_m130/video/video_quality_test.h
Per Kjellander 89870ffa95 Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae.

Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104

Original change's description:
> Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
>
> This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425.
>
> Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and  CallPerfTest.Min_Bitrate_VideoAndAudio 
>
>
> Original change's description:
> > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp
> >
> > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
> > Therefore DirectTransport is provided with the extension mapping.
> >
> > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
> >
> >
> > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
> > Bug: webrtc:7135, webrtc:14795
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
> > Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39137}
>
> Bug: webrtc:7135, webrtc:14795, webrtc:14833
> Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Auto-Submit: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#39146}

Bug: webrtc:7135, webrtc:14795, webrtc:14833
Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39157}
2023-01-20 06:32:29 +00:00

146 lines
5.4 KiB
C++

/*
* Copyright (c) 2015 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_VIDEO_QUALITY_TEST_H_
#define VIDEO_VIDEO_QUALITY_TEST_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/fec_controller.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/test/frame_generator_interface.h"
#include "api/test/video_quality_test_fixture.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/fake_network_pipe.h"
#include "media/engine/internal_decoder_factory.h"
#include "media/engine/internal_encoder_factory.h"
#include "test/call_test.h"
#include "test/layer_filtering_transport.h"
#include "video/video_analyzer.h"
#ifdef WEBRTC_WIN
#include "modules/audio_device/win/core_audio_utility_win.h"
#include "rtc_base/win/scoped_com_initializer.h"
#endif
namespace webrtc {
class VideoQualityTest : public test::CallTest,
public VideoQualityTestFixtureInterface {
public:
explicit VideoQualityTest(
std::unique_ptr<InjectionComponents> injection_components);
void RunWithAnalyzer(const Params& params) override;
void RunWithRenderers(const Params& params) override;
const std::map<uint8_t, webrtc::MediaType>& payload_type_map() override {
return payload_type_map_;
}
static void FillScalabilitySettings(
Params* params,
size_t video_idx,
const std::vector<std::string>& stream_descriptors,
int num_streams,
size_t selected_stream,
int num_spatial_layers,
int selected_sl,
InterLayerPredMode inter_layer_pred,
const std::vector<std::string>& sl_descriptors);
// Helper static methods.
static VideoStream DefaultVideoStream(const Params& params, size_t video_idx);
static VideoStream DefaultThumbnailStream();
static std::vector<int> ParseCSV(const std::string& str);
protected:
// No-op implementation to be able to instantiate this class from non-TEST_F
// locations.
void TestBody() override;
// Helper methods accessing only params_.
std::string GenerateGraphTitle() const;
void CheckParamsAndInjectionComponents();
// Helper methods for setting up the call.
void CreateCapturers();
std::unique_ptr<test::FrameGeneratorInterface> CreateFrameGenerator(
size_t video_idx);
void SetupThumbnailCapturers(size_t num_thumbnail_streams);
std::unique_ptr<VideoDecoder> CreateVideoDecoder(
const SdpVideoFormat& format);
std::unique_ptr<VideoEncoder> CreateVideoEncoder(const SdpVideoFormat& format,
VideoAnalyzer* analyzer);
void SetupVideo(Transport* send_transport, Transport* recv_transport);
void SetupThumbnails(Transport* send_transport, Transport* recv_transport);
void StartAudioStreams();
void StartThumbnails();
void StopThumbnails();
void DestroyThumbnailStreams();
// Helper method for creating a real ADM (using hardware) for all platforms.
rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice();
void InitializeAudioDevice(Call::Config* send_call_config,
Call::Config* recv_call_config,
bool use_real_adm);
void SetupAudio(Transport* transport);
void StartEncodedFrameLogs(VideoReceiveStreamInterface* stream);
virtual std::unique_ptr<test::LayerFilteringTransport> CreateSendTransport();
virtual std::unique_ptr<test::DirectTransport> CreateReceiveTransport();
std::vector<std::unique_ptr<rtc::VideoSourceInterface<VideoFrame>>>
thumbnail_capturers_;
Clock* const clock_;
const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
RtcEventLogFactory rtc_event_log_factory_;
test::FunctionVideoDecoderFactory video_decoder_factory_;
std::unique_ptr<VideoDecoderFactory> decoder_factory_;
test::FunctionVideoEncoderFactory video_encoder_factory_;
test::FunctionVideoEncoderFactory video_encoder_factory_with_analyzer_;
std::unique_ptr<VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
std::unique_ptr<VideoEncoderFactory> encoder_factory_;
std::vector<VideoSendStream::Config> thumbnail_send_configs_;
std::vector<VideoEncoderConfig> thumbnail_encoder_configs_;
std::vector<VideoSendStream*> thumbnail_send_streams_;
std::vector<VideoReceiveStreamInterface::Config> thumbnail_receive_configs_;
std::vector<VideoReceiveStreamInterface*> thumbnail_receive_streams_;
int receive_logs_;
int send_logs_;
Params params_;
std::unique_ptr<InjectionComponents> injection_components_;
// Set non-null when running with analyzer.
std::unique_ptr<VideoAnalyzer> analyzer_;
// Note: not same as similarly named member in CallTest. This is the number of
// separate send streams, the one in CallTest is the number of substreams for
// a single send stream.
size_t num_video_streams_;
#ifdef WEBRTC_WIN
// Windows Core Audio based ADM needs to run on a COM initialized thread.
// Only referenced in combination with --audio --use_real_adm flags.
std::unique_ptr<ScopedCOMInitializer> com_initializer_;
#endif
};
} // namespace webrtc
#endif // VIDEO_VIDEO_QUALITY_TEST_H_