| /* | 
 |  *  Copyright (c) 2015 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 AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ | 
 | #define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ | 
 |  | 
 | #include <map> | 
 | #include <memory> | 
 | #include <string> | 
 | #include <utility> | 
 | #include <vector> | 
 |  | 
 | #include "api/crypto/frame_decryptor_interface.h" | 
 | #include "api/test/mock_frame_encryptor.h" | 
 | #include "audio/channel_receive.h" | 
 | #include "audio/channel_send.h" | 
 | #include "modules/rtp_rtcp/source/rtp_packet_received.h" | 
 | #include "test/gmock.h" | 
 |  | 
 | namespace webrtc { | 
 | namespace test { | 
 |  | 
 | class MockChannelReceive : public voe::ChannelReceiveInterface { | 
 |  public: | 
 |   MOCK_METHOD(void, SetNACKStatus, (bool enable, int max_packets), (override)); | 
 |   MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override)); | 
 |   MOCK_METHOD(void, | 
 |               RegisterReceiverCongestionControlObjects, | 
 |               (PacketRouter*), | 
 |               (override)); | 
 |   MOCK_METHOD(void, ResetReceiverCongestionControlObjects, (), (override)); | 
 |   MOCK_METHOD(CallReceiveStatistics, GetRTCPStatistics, (), (const, override)); | 
 |   MOCK_METHOD(NetworkStatistics, | 
 |               GetNetworkStatistics, | 
 |               (bool), | 
 |               (const, override)); | 
 |   MOCK_METHOD(AudioDecodingCallStats, | 
 |               GetDecodingCallStatistics, | 
 |               (), | 
 |               (const, override)); | 
 |   MOCK_METHOD(int, GetSpeechOutputLevelFullRange, (), (const, override)); | 
 |   MOCK_METHOD(double, GetTotalOutputEnergy, (), (const, override)); | 
 |   MOCK_METHOD(double, GetTotalOutputDuration, (), (const, override)); | 
 |   MOCK_METHOD(uint32_t, GetDelayEstimate, (), (const, override)); | 
 |   MOCK_METHOD(void, SetSink, (AudioSinkInterface*), (override)); | 
 |   MOCK_METHOD(void, OnRtpPacket, (const RtpPacketReceived& packet), (override)); | 
 |   MOCK_METHOD(void, | 
 |               ReceivedRTCPPacket, | 
 |               (const uint8_t*, size_t length), | 
 |               (override)); | 
 |   MOCK_METHOD(void, SetChannelOutputVolumeScaling, (float scaling), (override)); | 
 |   MOCK_METHOD(AudioMixer::Source::AudioFrameInfo, | 
 |               GetAudioFrameWithInfo, | 
 |               (int sample_rate_hz, AudioFrame*), | 
 |               (override)); | 
 |   MOCK_METHOD(int, PreferredSampleRate, (), (const, override)); | 
 |   MOCK_METHOD(void, SetSourceTracker, (SourceTracker*), (override)); | 
 |   MOCK_METHOD(void, | 
 |               SetAssociatedSendChannel, | 
 |               (const voe::ChannelSendInterface*), | 
 |               (override)); | 
 |   MOCK_METHOD(bool, | 
 |               GetPlayoutRtpTimestamp, | 
 |               (uint32_t*, int64_t*), | 
 |               (const, override)); | 
 |   MOCK_METHOD(void, | 
 |               SetEstimatedPlayoutNtpTimestampMs, | 
 |               (int64_t ntp_timestamp_ms, int64_t time_ms), | 
 |               (override)); | 
 |   MOCK_METHOD(absl::optional<int64_t>, | 
 |               GetCurrentEstimatedPlayoutNtpTimestampMs, | 
 |               (int64_t now_ms), | 
 |               (const, override)); | 
 |   MOCK_METHOD(absl::optional<Syncable::Info>, | 
 |               GetSyncInfo, | 
 |               (), | 
 |               (const, override)); | 
 |   MOCK_METHOD(bool, SetMinimumPlayoutDelay, (int delay_ms), (override)); | 
 |   MOCK_METHOD(bool, SetBaseMinimumPlayoutDelayMs, (int delay_ms), (override)); | 
 |   MOCK_METHOD(int, GetBaseMinimumPlayoutDelayMs, (), (const, override)); | 
 |   MOCK_METHOD((absl::optional<std::pair<int, SdpAudioFormat>>), | 
 |               GetReceiveCodec, | 
 |               (), | 
 |               (const, override)); | 
 |   MOCK_METHOD(void, | 
 |               SetReceiveCodecs, | 
 |               ((const std::map<int, SdpAudioFormat>& codecs)), | 
 |               (override)); | 
 |   MOCK_METHOD(void, StartPlayout, (), (override)); | 
 |   MOCK_METHOD(void, StopPlayout, (), (override)); | 
 |   MOCK_METHOD( | 
 |       void, | 
 |       SetDepacketizerToDecoderFrameTransformer, | 
 |       (rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer), | 
 |       (override)); | 
 |   MOCK_METHOD( | 
 |       void, | 
 |       SetFrameDecryptor, | 
 |       (rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor), | 
 |       (override)); | 
 |   MOCK_METHOD(void, OnLocalSsrcChange, (uint32_t local_ssrc), (override)); | 
 |   MOCK_METHOD(uint32_t, GetLocalSsrc, (), (const, override)); | 
 | }; | 
 |  | 
 | class MockChannelSend : public voe::ChannelSendInterface { | 
 |  public: | 
 |   MOCK_METHOD(void, | 
 |               SetEncoder, | 
 |               (int payload_type, std::unique_ptr<AudioEncoder> encoder), | 
 |               (override)); | 
 |   MOCK_METHOD( | 
 |       void, | 
 |       ModifyEncoder, | 
 |       (rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier), | 
 |       (override)); | 
 |   MOCK_METHOD(void, | 
 |               CallEncoder, | 
 |               (rtc::FunctionView<void(AudioEncoder*)> modifier), | 
 |               (override)); | 
 |   MOCK_METHOD(void, SetRTCP_CNAME, (absl::string_view c_name), (override)); | 
 |   MOCK_METHOD(void, | 
 |               SetSendAudioLevelIndicationStatus, | 
 |               (bool enable, int id), | 
 |               (override)); | 
 |   MOCK_METHOD(void, | 
 |               RegisterSenderCongestionControlObjects, | 
 |               (RtpTransportControllerSendInterface*, RtcpBandwidthObserver*), | 
 |               (override)); | 
 |   MOCK_METHOD(void, ResetSenderCongestionControlObjects, (), (override)); | 
 |   MOCK_METHOD(CallSendStatistics, GetRTCPStatistics, (), (const, override)); | 
 |   MOCK_METHOD(std::vector<ReportBlock>, | 
 |               GetRemoteRTCPReportBlocks, | 
 |               (), | 
 |               (const, override)); | 
 |   MOCK_METHOD(ANAStats, GetANAStatistics, (), (const, override)); | 
 |   MOCK_METHOD(void, | 
 |               RegisterCngPayloadType, | 
 |               (int payload_type, int payload_frequency), | 
 |               (override)); | 
 |   MOCK_METHOD(void, | 
 |               SetSendTelephoneEventPayloadType, | 
 |               (int payload_type, int payload_frequency), | 
 |               (override)); | 
 |   MOCK_METHOD(bool, | 
 |               SendTelephoneEventOutband, | 
 |               (int event, int duration_ms), | 
 |               (override)); | 
 |   MOCK_METHOD(void, | 
 |               OnBitrateAllocation, | 
 |               (BitrateAllocationUpdate update), | 
 |               (override)); | 
 |   MOCK_METHOD(void, SetInputMute, (bool muted), (override)); | 
 |   MOCK_METHOD(void, | 
 |               ReceivedRTCPPacket, | 
 |               (const uint8_t*, size_t length), | 
 |               (override)); | 
 |   MOCK_METHOD(void, | 
 |               ProcessAndEncodeAudio, | 
 |               (std::unique_ptr<AudioFrame>), | 
 |               (override)); | 
 |   MOCK_METHOD(RtpRtcpInterface*, GetRtpRtcp, (), (const, override)); | 
 |   MOCK_METHOD(int, GetTargetBitrate, (), (const, override)); | 
 |   MOCK_METHOD(int64_t, GetRTT, (), (const, override)); | 
 |   MOCK_METHOD(void, StartSend, (), (override)); | 
 |   MOCK_METHOD(void, StopSend, (), (override)); | 
 |   MOCK_METHOD(void, | 
 |               SetFrameEncryptor, | 
 |               (rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor), | 
 |               (override)); | 
 |   MOCK_METHOD( | 
 |       void, | 
 |       SetEncoderToPacketizerFrameTransformer, | 
 |       (rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer), | 
 |       (override)); | 
 | }; | 
 | }  // namespace test | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ |