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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /media/webrtc/signaling/gtest/MockCall.h | |
parent | Initial commit. (diff) | |
download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | media/webrtc/signaling/gtest/MockCall.h | 375 |
1 files changed, 375 insertions, 0 deletions
diff --git a/media/webrtc/signaling/gtest/MockCall.h b/media/webrtc/signaling/gtest/MockCall.h new file mode 100644 index 0000000000..13785b2024 --- /dev/null +++ b/media/webrtc/signaling/gtest/MockCall.h @@ -0,0 +1,375 @@ +/* This Source Code Form is subject to the terms of the Mozilla Public + * License, v. 2.0. If a copy of the MPL was not distributed with this file, + * You can obtain one at http://mozilla.org/MPL/2.0/. */ + +#ifndef MOCK_CALL_H_ +#define MOCK_CALL_H_ + +#include "gmock/gmock.h" +#include "mozilla/Assertions.h" +#include "mozilla/Maybe.h" +#include "mozilla/media/MediaUtils.h" +#include "WebrtcCallWrapper.h" +#include "PeerConnectionCtx.h" + +// libwebrtc +#include "api/call/audio_sink.h" +#include "call/call.h" + +namespace test { +class MockCallWrapper; + +class MockAudioSendStream : public webrtc::AudioSendStream { + public: + explicit MockAudioSendStream(RefPtr<MockCallWrapper> aCallWrapper) + : mCallWrapper(std::move(aCallWrapper)) {} + + const webrtc::AudioSendStream::Config& GetConfig() const override; + + void Reconfigure(const Config& config) override; + + void Start() override {} + + void Stop() override {} + + void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override { + } + + bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, + int duration_ms) override { + return true; + } + + void SetMuted(bool muted) override {} + + Stats GetStats() const override { return mStats; } + + Stats GetStats(bool has_remote_tracks) const override { return mStats; } + + virtual ~MockAudioSendStream() {} + + const RefPtr<MockCallWrapper> mCallWrapper; + webrtc::AudioSendStream::Stats mStats; +}; + +class MockAudioReceiveStream : public webrtc::AudioReceiveStreamInterface { + public: + explicit MockAudioReceiveStream(RefPtr<MockCallWrapper> aCallWrapper) + : mCallWrapper(std::move(aCallWrapper)) {} + + void Start() override {} + + void Stop() override {} + + bool IsRunning() const override { return true; } + + bool transport_cc() const override { return false; } + + Stats GetStats(bool get_and_clear_legacy_stats) const override { + return mStats; + } + + void SetSink(webrtc::AudioSinkInterface* sink) override {} + + void SetGain(float gain) override {} + + std::vector<webrtc::RtpSource> GetSources() const override { + return mRtpSources; + } + + virtual void SetDepacketizerToDecoderFrameTransformer( + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) + override { + // Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure + // method. + MOZ_ASSERT(false); + } + virtual void SetDecoderMap( + std::map<int, webrtc::SdpAudioFormat> decoder_map) override; + virtual void SetTransportCc(bool use_transport_cc) override { + // Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure + // method. + MOZ_ASSERT(false); + } + virtual void SetNackHistory(int history_ms) override { + // Unimplemented after webrtc.org e2561e17e2 removed the Reconfigure + // method. + MOZ_ASSERT(false); + } + virtual void SetNonSenderRttMeasurement(bool enabled) override {} + void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface> + frame_decryptor) override {} + void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override; + const std::vector<webrtc::RtpExtension>& GetRtpExtensions() const override; + webrtc::RtpHeaderExtensionMap GetRtpExtensionMap() const override; + bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override { return false; } + int GetBaseMinimumPlayoutDelayMs() const override { return 0; } + uint32_t remote_ssrc() const override { return 0; } + + virtual ~MockAudioReceiveStream() {} + + const RefPtr<MockCallWrapper> mCallWrapper; + webrtc::AudioReceiveStreamInterface::Stats mStats; + std::vector<webrtc::RtpSource> mRtpSources; +}; + +class MockVideoSendStream : public webrtc::VideoSendStream { + public: + explicit MockVideoSendStream(RefPtr<MockCallWrapper> aCallWrapper) + : mCallWrapper(std::move(aCallWrapper)) {} + + void Start() override {} + + void Stop() override {} + + bool started() override { return false; } + + void SetSource( + rtc::VideoSourceInterface<webrtc::VideoFrame>* source, + const webrtc::DegradationPreference& degradation_preference) override {} + + void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config) override; + + Stats GetStats() override { return mStats; } + + void UpdateActiveSimulcastLayers( + const std::vector<bool> active_layers) override {} + + void AddAdaptationResource( + rtc::scoped_refptr<webrtc::Resource> resource) override {} + + std::vector<rtc::scoped_refptr<webrtc::Resource>> GetAdaptationResources() + override { + return std::vector<rtc::scoped_refptr<webrtc::Resource>>(); + } + + virtual ~MockVideoSendStream() {} + + const RefPtr<MockCallWrapper> mCallWrapper; + webrtc::VideoSendStream::Stats mStats; +}; + +class MockVideoReceiveStream : public webrtc::VideoReceiveStreamInterface { + public: + explicit MockVideoReceiveStream(RefPtr<MockCallWrapper> aCallWrapper) + : mCallWrapper(std::move(aCallWrapper)) {} + + void Start() override {} + + void Stop() override {} + + bool transport_cc() const override { return false; } + void SetTransportCc(bool use_transport_cc) override {} + + Stats GetStats() const override { return mStats; } + + std::vector<webrtc::RtpSource> GetSources() const override { + return std::vector<webrtc::RtpSource>(); + } + + bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override { return false; } + + int GetBaseMinimumPlayoutDelayMs() const override { return 0; } + + void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface> + frame_decryptor) override {} + + void SetDepacketizerToDecoderFrameTransformer( + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) + override {} + + RecordingState SetAndGetRecordingState(RecordingState state, + bool generate_key_frame) override { + return {}; + } + + void GenerateKeyFrame() override {} + + void SetRtcpMode(webrtc::RtcpMode mode) override {} + + void SetFlexFecProtection(webrtc::RtpPacketSinkInterface* flexfec_sink) override {} + + void SetLossNotificationEnabled(bool enabled) override {} + + void SetNackHistory(webrtc::TimeDelta history) override {} + + void SetProtectionPayloadTypes(int red_payload_type, + int ulpfec_payload_type) override {} + + void SetRtcpXr(Config::Rtp::RtcpXr rtcp_xr) override {} + + virtual void SetAssociatedPayloadTypes( + std::map<int, int> associated_payload_types) override {} + + void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override { + } + webrtc::RtpHeaderExtensionMap GetRtpExtensionMap() const override; + + virtual ~MockVideoReceiveStream() {} + + const RefPtr<MockCallWrapper> mCallWrapper; + webrtc::VideoReceiveStreamInterface::Stats mStats; +}; + +class MockCall : public webrtc::Call { + public: + explicit MockCall(RefPtr<MockCallWrapper> aCallWrapper) + : mCallWrapper(std::move(aCallWrapper)) {} + + webrtc::AudioSendStream* CreateAudioSendStream( + const webrtc::AudioSendStream::Config& config) override { + MOZ_RELEASE_ASSERT(!mAudioSendConfig); + mAudioSendConfig = mozilla::Some(config); + return new MockAudioSendStream(mCallWrapper); + } + + void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override { + mAudioSendConfig = mozilla::Nothing(); + delete static_cast<MockAudioSendStream*>(send_stream); + } + + webrtc::AudioReceiveStreamInterface* CreateAudioReceiveStream( + const webrtc::AudioReceiveStreamInterface::Config& config) override { + MOZ_RELEASE_ASSERT(!mAudioReceiveConfig); + mAudioReceiveConfig = mozilla::Some(config); + return new MockAudioReceiveStream(mCallWrapper); + } + void DestroyAudioReceiveStream( + webrtc::AudioReceiveStreamInterface* receive_stream) override { + mAudioReceiveConfig = mozilla::Nothing(); + delete static_cast<MockAudioReceiveStream*>(receive_stream); + } + + webrtc::VideoSendStream* CreateVideoSendStream( + webrtc::VideoSendStream::Config config, + webrtc::VideoEncoderConfig encoder_config) override { + MOZ_RELEASE_ASSERT(!mVideoSendConfig); + MOZ_RELEASE_ASSERT(!mVideoSendEncoderConfig); + mVideoSendConfig = mozilla::Some(std::move(config)); + mVideoSendEncoderConfig = mozilla::Some(std::move(encoder_config)); + return new MockVideoSendStream(mCallWrapper); + } + + void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override { + mVideoSendConfig = mozilla::Nothing(); + mVideoSendEncoderConfig = mozilla::Nothing(); + delete static_cast<MockVideoSendStream*>(send_stream); + } + + webrtc::VideoReceiveStreamInterface* CreateVideoReceiveStream( + webrtc::VideoReceiveStreamInterface::Config configuration) override { + MOZ_RELEASE_ASSERT(!mVideoReceiveConfig); + mVideoReceiveConfig = mozilla::Some(std::move(configuration)); + return new MockVideoReceiveStream(mCallWrapper); + } + + void DestroyVideoReceiveStream( + webrtc::VideoReceiveStreamInterface* receive_stream) override { + mVideoReceiveConfig = mozilla::Nothing(); + delete static_cast<MockVideoReceiveStream*>(receive_stream); + } + + webrtc::FlexfecReceiveStream* CreateFlexfecReceiveStream( + const webrtc::FlexfecReceiveStream::Config config) override { + return nullptr; + } + + void DestroyFlexfecReceiveStream( + webrtc::FlexfecReceiveStream* receive_stream) override {} + + void AddAdaptationResource( + rtc::scoped_refptr<webrtc::Resource> resource) override {} + + webrtc::PacketReceiver* Receiver() override { return nullptr; } + + webrtc::RtpTransportControllerSendInterface* GetTransportControllerSend() + override { + return nullptr; + } + + Stats GetStats() const override { return mStats; } + + void SignalChannelNetworkState(webrtc::MediaType media, + webrtc::NetworkState state) override {} + + void OnAudioTransportOverheadChanged( + int transport_overhead_per_packet) override {} + + void OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream, + uint32_t local_ssrc) override {} + void OnLocalSsrcUpdated(webrtc::VideoReceiveStreamInterface& stream, + uint32_t local_ssrc) override {} + void OnLocalSsrcUpdated(webrtc::FlexfecReceiveStream& stream, + uint32_t local_ssrc) override {} + + void OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream, + absl::string_view sync_group) override {} + + void OnSentPacket(const rtc::SentPacket& sent_packet) override {} + + void SetClientBitratePreferences( + const webrtc::BitrateSettings& preferences) override {} + + std::vector<webrtc::VideoStream> CreateEncoderStreams(int width, int height) { + return mVideoSendEncoderConfig->video_stream_factory->CreateEncoderStreams( + width, height, *mVideoSendEncoderConfig); + } + + virtual const webrtc::WebRtcKeyValueConfig& trials() const override { + return mUnusedConfig; + } + + virtual webrtc::TaskQueueBase* network_thread() const override { + return nullptr; + } + + virtual webrtc::TaskQueueBase* worker_thread() const override { + return nullptr; + } + + virtual ~MockCall(){}; + + const RefPtr<MockCallWrapper> mCallWrapper; + mozilla::Maybe<webrtc::AudioReceiveStreamInterface::Config> + mAudioReceiveConfig; + mozilla::Maybe<webrtc::AudioSendStream::Config> mAudioSendConfig; + mozilla::Maybe<webrtc::VideoReceiveStreamInterface::Config> + mVideoReceiveConfig; + mozilla::Maybe<webrtc::VideoSendStream::Config> mVideoSendConfig; + mozilla::Maybe<webrtc::VideoEncoderConfig> mVideoSendEncoderConfig; + webrtc::Call::Stats mStats; + webrtc::NoTrialsConfig mUnusedConfig; +}; + +class MockCallWrapper : public mozilla::WebrtcCallWrapper { + public: + MockCallWrapper( + RefPtr<mozilla::SharedWebrtcState> aSharedState, + mozilla::UniquePtr<webrtc::VideoBitrateAllocatorFactory> + aVideoBitrateAllocatorFactory, + mozilla::UniquePtr<webrtc::RtcEventLog> aEventLog, + mozilla::UniquePtr<webrtc::TaskQueueFactory> aTaskQueueFactory, + const mozilla::dom::RTCStatsTimestampMaker& aTimestampMaker, + mozilla::UniquePtr<mozilla::media::ShutdownBlockingTicket> + aShutdownTicket) + : mozilla::WebrtcCallWrapper( + std::move(aSharedState), std::move(aVideoBitrateAllocatorFactory), + std::move(aEventLog), std::move(aTaskQueueFactory), aTimestampMaker, + std::move(aShutdownTicket)) {} + + static RefPtr<MockCallWrapper> Create() { + auto state = mozilla::MakeRefPtr<mozilla::SharedWebrtcState>( + mozilla::AbstractThread::GetCurrent(), webrtc::AudioState::Config(), + nullptr, nullptr); + auto wrapper = mozilla::MakeRefPtr<MockCallWrapper>( + state, nullptr, nullptr, nullptr, + mozilla::dom::RTCStatsTimestampMaker(), nullptr); + wrapper->SetCall(mozilla::WrapUnique(new MockCall(wrapper))); + return wrapper; + } + + MockCall* GetMockCall() { return static_cast<MockCall*>(Call()); } +}; + +} // namespace test +#endif |