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Diffstat (limited to 'third_party/libwebrtc/pc/test/mock_voice_media_channel.h')
| -rw-r--r-- | third_party/libwebrtc/pc/test/mock_voice_media_channel.h | 163 |
1 files changed, 163 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/test/mock_voice_media_channel.h b/third_party/libwebrtc/pc/test/mock_voice_media_channel.h new file mode 100644 index 0000000000..2e5a8b5801 --- /dev/null +++ b/third_party/libwebrtc/pc/test/mock_voice_media_channel.h @@ -0,0 +1,163 @@ +/* + * Copyright 2021 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 PC_TEST_MOCK_VOICE_MEDIA_CHANNEL_H_ +#define PC_TEST_MOCK_VOICE_MEDIA_CHANNEL_H_ + +#include <memory> +#include <string> +#include <vector> + +#include "api/call/audio_sink.h" +#include "media/base/media_channel.h" +#include "media/base/media_channel_impl.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "rtc_base/gunit.h" +#include "test/gmock.h" +#include "test/gtest.h" + +using ::testing::InvokeWithoutArgs; +using ::testing::Mock; + +namespace cricket { +class MockVoiceMediaChannel : public VoiceMediaChannel { + public: + explicit MockVoiceMediaChannel(webrtc::TaskQueueBase* network_thread) + : VoiceMediaChannel(network_thread) {} + + MOCK_METHOD(void, + SetInterface, + (MediaChannelNetworkInterface * iface), + (override)); + MOCK_METHOD(void, + OnPacketReceived, + (const webrtc::RtpPacketReceived& packet), + (override)); + MOCK_METHOD(void, + OnPacketSent, + (const rtc::SentPacket& sent_packet), + (override)); + MOCK_METHOD(void, OnReadyToSend, (bool ready), (override)); + MOCK_METHOD(void, + OnNetworkRouteChanged, + (absl::string_view transport_name, + const rtc::NetworkRoute& network_route), + (override)); + MOCK_METHOD(bool, AddSendStream, (const StreamParams& sp), (override)); + MOCK_METHOD(bool, RemoveSendStream, (uint32_t ssrc), (override)); + MOCK_METHOD(bool, AddRecvStream, (const StreamParams& sp), (override)); + MOCK_METHOD(bool, RemoveRecvStream, (uint32_t ssrc), (override)); + MOCK_METHOD(void, ResetUnsignaledRecvStream, (), (override)); + MOCK_METHOD(absl::optional<uint32_t>, + GetUnsignaledSsrc, + (), + (const, override)); + MOCK_METHOD(void, OnDemuxerCriteriaUpdatePending, (), (override)); + MOCK_METHOD(void, OnDemuxerCriteriaUpdateComplete, (), (override)); + MOCK_METHOD(int, GetRtpSendTimeExtnId, (), (const, override)); + MOCK_METHOD( + void, + SetFrameEncryptor, + (uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor), + (override)); + MOCK_METHOD( + void, + SetFrameDecryptor, + (uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor), + (override)); + MOCK_METHOD(webrtc::RtpParameters, + GetRtpSendParameters, + (uint32_t ssrc), + (const, override)); + MOCK_METHOD(webrtc::RTCError, + SetRtpSendParameters, + (uint32_t ssrc, + const webrtc::RtpParameters& parameters, + webrtc::SetParametersCallback callback), + (override)); + MOCK_METHOD( + void, + SetEncoderToPacketizerFrameTransformer, + (uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer), + (override)); + MOCK_METHOD( + void, + SetDepacketizerToDecoderFrameTransformer, + (uint32_t ssrc, + rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer), + (override)); + + MOCK_METHOD(bool, + SetSendParameters, + (const AudioSendParameters& params), + (override)); + MOCK_METHOD(bool, + SetRecvParameters, + (const AudioRecvParameters& params), + (override)); + MOCK_METHOD(webrtc::RtpParameters, + GetRtpReceiveParameters, + (uint32_t ssrc), + (const, override)); + MOCK_METHOD(webrtc::RtpParameters, + GetDefaultRtpReceiveParameters, + (), + (const, override)); + MOCK_METHOD(void, SetPlayout, (bool playout), (override)); + MOCK_METHOD(void, SetSend, (bool send), (override)); + MOCK_METHOD(bool, + SetAudioSend, + (uint32_t ssrc, + bool enable, + const AudioOptions* options, + AudioSource* source), + (override)); + MOCK_METHOD(bool, + SetOutputVolume, + (uint32_t ssrc, double volume), + (override)); + MOCK_METHOD(bool, SetDefaultOutputVolume, (double volume), (override)); + MOCK_METHOD(bool, CanInsertDtmf, (), (override)); + MOCK_METHOD(bool, + InsertDtmf, + (uint32_t ssrc, int event, int duration), + (override)); + MOCK_METHOD(bool, GetSendStats, (VoiceMediaSendInfo * info), (override)); + MOCK_METHOD(bool, + GetReceiveStats, + (VoiceMediaReceiveInfo * info, bool get_and_clear_legacy_stats), + (override)); + MOCK_METHOD(void, + SetRawAudioSink, + (uint32_t ssrc, std::unique_ptr<webrtc::AudioSinkInterface> sink), + (override)); + MOCK_METHOD(void, + SetDefaultRawAudioSink, + (std::unique_ptr<webrtc::AudioSinkInterface> sink), + (override)); + MOCK_METHOD(std::vector<webrtc::RtpSource>, + GetSources, + (uint32_t ssrc), + (const, override)); + + MOCK_METHOD(bool, + SetBaseMinimumPlayoutDelayMs, + (uint32_t ssrc, int delay_ms), + (override)); + MOCK_METHOD(absl::optional<int>, + GetBaseMinimumPlayoutDelayMs, + (uint32_t ssrc), + (const, override)); +}; +} // namespace cricket + +#endif // PC_TEST_MOCK_VOICE_MEDIA_CHANNEL_H_ |