| /* |
| * 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_ |