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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/test/pc/e2e/media | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/pc/e2e/media')
3 files changed, 241 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/pc/e2e/media/media_helper.cc b/third_party/libwebrtc/test/pc/e2e/media/media_helper.cc new file mode 100644 index 0000000000..e945bd4dae --- /dev/null +++ b/third_party/libwebrtc/test/pc/e2e/media/media_helper.cc @@ -0,0 +1,128 @@ +/* + * Copyright (c) 2020 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. + */ +#include "test/pc/e2e/media/media_helper.h" + +#include <string> +#include <utility> + +#include "absl/types/variant.h" +#include "api/media_stream_interface.h" +#include "api/test/create_frame_generator.h" +#include "api/test/pclf/media_configuration.h" +#include "api/test/pclf/peer_configurer.h" +#include "test/frame_generator_capturer.h" +#include "test/platform_video_capturer.h" +#include "test/testsupport/file_utils.h" + +namespace webrtc { +namespace webrtc_pc_e2e { + +void MediaHelper::MaybeAddAudio(TestPeer* peer) { + if (!peer->params().audio_config) { + return; + } + const AudioConfig& audio_config = peer->params().audio_config.value(); + rtc::scoped_refptr<webrtc::AudioSourceInterface> source = + peer->pc_factory()->CreateAudioSource(audio_config.audio_options); + rtc::scoped_refptr<AudioTrackInterface> track = + peer->pc_factory()->CreateAudioTrack(*audio_config.stream_label, + source.get()); + std::string sync_group = audio_config.sync_group + ? audio_config.sync_group.value() + : audio_config.stream_label.value() + "-sync"; + peer->AddTrack(track, {sync_group, *audio_config.stream_label}); +} + +std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>> +MediaHelper::MaybeAddVideo(TestPeer* peer) { + // Params here valid because of pre-run validation. + const Params& params = peer->params(); + const ConfigurableParams& configurable_params = peer->configurable_params(); + std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>> out; + for (size_t i = 0; i < configurable_params.video_configs.size(); ++i) { + const VideoConfig& video_config = configurable_params.video_configs[i]; + // Setup input video source into peer connection. + std::unique_ptr<test::TestVideoCapturer> capturer = CreateVideoCapturer( + video_config, peer->ReleaseVideoSource(i), + video_quality_analyzer_injection_helper_->CreateFramePreprocessor( + params.name.value(), video_config)); + bool is_screencast = + video_config.content_hint == VideoTrackInterface::ContentHint::kText || + video_config.content_hint == + VideoTrackInterface::ContentHint::kDetailed; + rtc::scoped_refptr<TestVideoCapturerVideoTrackSource> source = + rtc::make_ref_counted<TestVideoCapturerVideoTrackSource>( + std::move(capturer), is_screencast); + out.push_back(source); + RTC_LOG(LS_INFO) << "Adding video with video_config.stream_label=" + << video_config.stream_label.value(); + rtc::scoped_refptr<VideoTrackInterface> track = + peer->pc_factory()->CreateVideoTrack(video_config.stream_label.value(), + source.get()); + if (video_config.content_hint.has_value()) { + track->set_content_hint(video_config.content_hint.value()); + } + std::string sync_group = video_config.sync_group + ? video_config.sync_group.value() + : video_config.stream_label.value() + "-sync"; + RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> sender = + peer->AddTrack(track, {sync_group, *video_config.stream_label}); + RTC_CHECK(sender.ok()); + if (video_config.temporal_layers_count || + video_config.degradation_preference) { + RtpParameters rtp_parameters = sender.value()->GetParameters(); + if (video_config.temporal_layers_count) { + for (auto& encoding_parameters : rtp_parameters.encodings) { + encoding_parameters.num_temporal_layers = + video_config.temporal_layers_count; + } + } + if (video_config.degradation_preference) { + rtp_parameters.degradation_preference = + video_config.degradation_preference; + } + RTCError res = sender.value()->SetParameters(rtp_parameters); + RTC_CHECK(res.ok()) << "Failed to set RTP parameters"; + } + } + return out; +} + +std::unique_ptr<test::TestVideoCapturer> MediaHelper::CreateVideoCapturer( + const VideoConfig& video_config, + PeerConfigurer::VideoSource source, + std::unique_ptr<test::TestVideoCapturer::FramePreprocessor> + frame_preprocessor) { + CapturingDeviceIndex* capturing_device_index = + absl::get_if<CapturingDeviceIndex>(&source); + if (capturing_device_index != nullptr) { + std::unique_ptr<test::TestVideoCapturer> capturer = + test::CreateVideoCapturer(video_config.width, video_config.height, + video_config.fps, + static_cast<size_t>(*capturing_device_index)); + RTC_CHECK(capturer) + << "Failed to obtain input stream from capturing device #" + << *capturing_device_index; + capturer->SetFramePreprocessor(std::move(frame_preprocessor)); + return capturer; + } + + auto capturer = std::make_unique<test::FrameGeneratorCapturer>( + clock_, + absl::get<std::unique_ptr<test::FrameGeneratorInterface>>( + std::move(source)), + video_config.fps, *task_queue_factory_); + capturer->SetFramePreprocessor(std::move(frame_preprocessor)); + capturer->Init(); + return capturer; +} + +} // namespace webrtc_pc_e2e +} // namespace webrtc diff --git a/third_party/libwebrtc/test/pc/e2e/media/media_helper.h b/third_party/libwebrtc/test/pc/e2e/media/media_helper.h new file mode 100644 index 0000000000..2d163d009e --- /dev/null +++ b/third_party/libwebrtc/test/pc/e2e/media/media_helper.h @@ -0,0 +1,58 @@ +/* + * Copyright (c) 2020 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_PC_E2E_MEDIA_MEDIA_HELPER_H_ +#define TEST_PC_E2E_MEDIA_MEDIA_HELPER_H_ + +#include <memory> +#include <vector> + +#include "api/test/frame_generator_interface.h" +#include "api/test/pclf/media_configuration.h" +#include "api/test/pclf/peer_configurer.h" +#include "test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h" +#include "test/pc/e2e/media/test_video_capturer_video_track_source.h" +#include "test/pc/e2e/test_peer.h" + +namespace webrtc { +namespace webrtc_pc_e2e { + +class MediaHelper { + public: + MediaHelper(VideoQualityAnalyzerInjectionHelper* + video_quality_analyzer_injection_helper, + TaskQueueFactory* task_queue_factory, + Clock* clock) + : clock_(clock), + task_queue_factory_(task_queue_factory), + video_quality_analyzer_injection_helper_( + video_quality_analyzer_injection_helper) {} + + void MaybeAddAudio(TestPeer* peer); + + std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>> + MaybeAddVideo(TestPeer* peer); + + private: + std::unique_ptr<test::TestVideoCapturer> CreateVideoCapturer( + const VideoConfig& video_config, + PeerConfigurer::VideoSource source, + std::unique_ptr<test::TestVideoCapturer::FramePreprocessor> + frame_preprocessor); + + Clock* const clock_; + TaskQueueFactory* const task_queue_factory_; + VideoQualityAnalyzerInjectionHelper* video_quality_analyzer_injection_helper_; +}; + +} // namespace webrtc_pc_e2e +} // namespace webrtc + +#endif // TEST_PC_E2E_MEDIA_MEDIA_HELPER_H_ diff --git a/third_party/libwebrtc/test/pc/e2e/media/test_video_capturer_video_track_source.h b/third_party/libwebrtc/test/pc/e2e/media/test_video_capturer_video_track_source.h new file mode 100644 index 0000000000..c883a2e8e9 --- /dev/null +++ b/third_party/libwebrtc/test/pc/e2e/media/test_video_capturer_video_track_source.h @@ -0,0 +1,55 @@ +/* + * Copyright (c) 2020 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_PC_E2E_MEDIA_TEST_VIDEO_CAPTURER_VIDEO_TRACK_SOURCE_H_ +#define TEST_PC_E2E_MEDIA_TEST_VIDEO_CAPTURER_VIDEO_TRACK_SOURCE_H_ + +#include <memory> +#include <utility> + +#include "api/video/video_frame.h" +#include "api/video/video_source_interface.h" +#include "pc/video_track_source.h" +#include "test/test_video_capturer.h" + +namespace webrtc { +namespace webrtc_pc_e2e { + +class TestVideoCapturerVideoTrackSource : public VideoTrackSource { + public: + TestVideoCapturerVideoTrackSource( + std::unique_ptr<test::TestVideoCapturer> video_capturer, + bool is_screencast) + : VideoTrackSource(/*remote=*/false), + video_capturer_(std::move(video_capturer)), + is_screencast_(is_screencast) {} + + ~TestVideoCapturerVideoTrackSource() = default; + + void Start() { SetState(kLive); } + + void Stop() { SetState(kMuted); } + + bool is_screencast() const override { return is_screencast_; } + + protected: + rtc::VideoSourceInterface<VideoFrame>* source() override { + return video_capturer_.get(); + } + + private: + std::unique_ptr<test::TestVideoCapturer> video_capturer_; + const bool is_screencast_; +}; + +} // namespace webrtc_pc_e2e +} // namespace webrtc + +#endif // TEST_PC_E2E_MEDIA_TEST_VIDEO_CAPTURER_VIDEO_TRACK_SOURCE_H_ |