| /* | |
| * Copyright (c) 2025 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 CALL_TEST_MOCK_AUDIO_RECEIVE_STREAM_H_ | |
| #define CALL_TEST_MOCK_AUDIO_RECEIVE_STREAM_H_ | |
| #include <cstddef> | |
| #include <cstdint> | |
| #include <map> | |
| #include <vector> | |
| #include "api/audio/audio_frame.h" | |
| #include "api/audio/audio_mixer.h" | |
| #include "api/audio_codecs/audio_format.h" | |
| #include "api/crypto/frame_decryptor_interface.h" | |
| #include "api/frame_transformer_interface.h" | |
| #include "api/rtp_headers.h" | |
| #include "api/scoped_refptr.h" | |
| #include "api/transport/rtp/rtp_source.h" | |
| #include "call/audio_receive_stream.h" | |
| #include "test/gmock.h" | |
| namespace webrtc { | |
| namespace test { | |
| class MockAudioReceiveStream : public AudioReceiveStreamInterface, | |
| public AudioMixer::Source { | |
| public: | |
| MOCK_METHOD(uint32_t, remote_ssrc, (), (const override)); | |
| MOCK_METHOD(void, Start, (), (override)); | |
| MOCK_METHOD(void, Stop, (), (override)); | |
| MOCK_METHOD(bool, IsRunning, (), (const override)); | |
| MOCK_METHOD(void, | |
| SetDepacketizerToDecoderFrameTransformer, | |
| (scoped_refptr<FrameTransformerInterface>), | |
| (override)); | |
| MOCK_METHOD(void, | |
| SetDecoderMap, | |
| ((std::map<int, SdpAudioFormat>)), | |
| (override)); | |
| MOCK_METHOD(void, SetNackHistory, (int), (override)); | |
| MOCK_METHOD(void, SetRtcpMode, (RtcpMode), (override)); | |
| MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool), (override)); | |
| MOCK_METHOD(void, SetJitterBufferMaxPackets, (size_t), (override)); | |
| MOCK_METHOD(void, SetJitterBufferFastAccelerate, (bool), (override)); | |
| MOCK_METHOD(void, | |
| SetFrameDecryptor, | |
| (scoped_refptr<FrameDecryptorInterface>), | |
| (override)); | |
| MOCK_METHOD(webrtc::AudioReceiveStreamInterface::Stats, | |
| GetStats, | |
| (bool), | |
| (const override)); | |
| MOCK_METHOD(void, SetSink, (webrtc::AudioSinkInterface*), (override)); | |
| MOCK_METHOD(void, SetGain, (float), (override)); | |
| MOCK_METHOD(bool, SetBaseMinimumPlayoutDelayMs, (int), (override)); | |
| MOCK_METHOD(int, GetBaseMinimumPlayoutDelayMs, (), (const override)); | |
| MOCK_METHOD(std::vector<webrtc::RtpSource>, GetSources, (), (const override)); | |
| // TODO (b/397376626): Create a MockAudioMixerSource, and instead | |
| // have a member variable here. | |
| AudioMixer::Source* source() override { return this; } | |
| MOCK_METHOD(AudioFrameInfo, | |
| GetAudioFrameWithInfo, | |
| (int, AudioFrame*), | |
| (override)); | |
| MOCK_METHOD(int, Ssrc, (), (const override)); | |
| MOCK_METHOD(int, PreferredSampleRate, (), (const override)); | |
| }; | |
| } // namespace test | |
| } // namespace webrtc | |
| #endif // CALL_TEST_MOCK_AUDIO_RECEIVE_STREAM_H_ |