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188
TMessagesProj/jni/voip/webrtc/audio/mock_voe_channel_proxy.h
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188
TMessagesProj/jni/voip/webrtc/audio/mock_voe_channel_proxy.h
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/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <utility>
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#include <vector>
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#include "api/crypto/frame_decryptor_interface.h"
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#include "api/test/mock_frame_encryptor.h"
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#include "audio/channel_receive.h"
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#include "audio/channel_send.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockChannelReceive : public voe::ChannelReceiveInterface {
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public:
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MOCK_METHOD(void, SetNACKStatus, (bool enable, int max_packets), (override));
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MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override));
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MOCK_METHOD(void,
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RegisterReceiverCongestionControlObjects,
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(PacketRouter*),
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(override));
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MOCK_METHOD(void, ResetReceiverCongestionControlObjects, (), (override));
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MOCK_METHOD(CallReceiveStatistics, GetRTCPStatistics, (), (const, override));
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MOCK_METHOD(NetworkStatistics,
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GetNetworkStatistics,
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(bool),
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(const, override));
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MOCK_METHOD(AudioDecodingCallStats,
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GetDecodingCallStatistics,
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(),
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(const, override));
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MOCK_METHOD(int, GetSpeechOutputLevelFullRange, (), (const, override));
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MOCK_METHOD(double, GetTotalOutputEnergy, (), (const, override));
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MOCK_METHOD(double, GetTotalOutputDuration, (), (const, override));
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MOCK_METHOD(uint32_t, GetDelayEstimate, (), (const, override));
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MOCK_METHOD(void, SetSink, (AudioSinkInterface*), (override));
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MOCK_METHOD(void, OnRtpPacket, (const RtpPacketReceived& packet), (override));
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MOCK_METHOD(void,
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ReceivedRTCPPacket,
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(const uint8_t*, size_t length),
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(override));
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MOCK_METHOD(void, SetChannelOutputVolumeScaling, (float scaling), (override));
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MOCK_METHOD(AudioMixer::Source::AudioFrameInfo,
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GetAudioFrameWithInfo,
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(int sample_rate_hz, AudioFrame*),
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(override));
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MOCK_METHOD(int, PreferredSampleRate, (), (const, override));
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MOCK_METHOD(void, SetSourceTracker, (SourceTracker*), (override));
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MOCK_METHOD(void,
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SetAssociatedSendChannel,
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(const voe::ChannelSendInterface*),
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(override));
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MOCK_METHOD(bool,
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GetPlayoutRtpTimestamp,
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(uint32_t*, int64_t*),
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(const, override));
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MOCK_METHOD(void,
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SetEstimatedPlayoutNtpTimestampMs,
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(int64_t ntp_timestamp_ms, int64_t time_ms),
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(override));
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MOCK_METHOD(absl::optional<int64_t>,
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GetCurrentEstimatedPlayoutNtpTimestampMs,
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(int64_t now_ms),
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(const, override));
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MOCK_METHOD(absl::optional<Syncable::Info>,
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GetSyncInfo,
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(),
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(const, override));
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MOCK_METHOD(bool, SetMinimumPlayoutDelay, (int delay_ms), (override));
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MOCK_METHOD(bool, SetBaseMinimumPlayoutDelayMs, (int delay_ms), (override));
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MOCK_METHOD(int, GetBaseMinimumPlayoutDelayMs, (), (const, override));
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MOCK_METHOD((absl::optional<std::pair<int, SdpAudioFormat>>),
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GetReceiveCodec,
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(),
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(const, override));
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MOCK_METHOD(void,
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SetReceiveCodecs,
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((const std::map<int, SdpAudioFormat>& codecs)),
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(override));
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MOCK_METHOD(void, StartPlayout, (), (override));
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MOCK_METHOD(void, StopPlayout, (), (override));
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MOCK_METHOD(
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void,
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SetDepacketizerToDecoderFrameTransformer,
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(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
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(override));
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MOCK_METHOD(
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void,
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SetFrameDecryptor,
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(rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
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(override));
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MOCK_METHOD(void, OnLocalSsrcChange, (uint32_t local_ssrc), (override));
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MOCK_METHOD(uint32_t, GetLocalSsrc, (), (const, override));
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};
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class MockChannelSend : public voe::ChannelSendInterface {
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public:
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MOCK_METHOD(void,
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SetEncoder,
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(int payload_type,
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const SdpAudioFormat& encoder_format,
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std::unique_ptr<AudioEncoder> encoder),
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(override));
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MOCK_METHOD(
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void,
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ModifyEncoder,
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(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier),
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(override));
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MOCK_METHOD(void,
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CallEncoder,
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(rtc::FunctionView<void(AudioEncoder*)> modifier),
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(override));
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MOCK_METHOD(void, SetRTCP_CNAME, (absl::string_view c_name), (override));
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MOCK_METHOD(void,
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SetSendAudioLevelIndicationStatus,
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(bool enable, int id),
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(override));
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MOCK_METHOD(void,
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RegisterSenderCongestionControlObjects,
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(RtpTransportControllerSendInterface*),
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(override));
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MOCK_METHOD(void, ResetSenderCongestionControlObjects, (), (override));
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MOCK_METHOD(CallSendStatistics, GetRTCPStatistics, (), (const, override));
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MOCK_METHOD(std::vector<ReportBlockData>,
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GetRemoteRTCPReportBlocks,
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(),
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(const, override));
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MOCK_METHOD(ANAStats, GetANAStatistics, (), (const, override));
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MOCK_METHOD(void,
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RegisterCngPayloadType,
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(int payload_type, int payload_frequency),
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(override));
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MOCK_METHOD(void,
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SetSendTelephoneEventPayloadType,
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(int payload_type, int payload_frequency),
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(override));
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MOCK_METHOD(bool,
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SendTelephoneEventOutband,
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(int event, int duration_ms),
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(override));
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MOCK_METHOD(void,
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OnBitrateAllocation,
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(BitrateAllocationUpdate update),
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(override));
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MOCK_METHOD(void, SetInputMute, (bool muted), (override));
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MOCK_METHOD(void,
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ReceivedRTCPPacket,
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(const uint8_t*, size_t length),
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(override));
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MOCK_METHOD(void,
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ProcessAndEncodeAudio,
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(std::unique_ptr<AudioFrame>),
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(override));
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MOCK_METHOD(RtpRtcpInterface*, GetRtpRtcp, (), (const, override));
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MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
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MOCK_METHOD(int64_t, GetRTT, (), (const, override));
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MOCK_METHOD(void, StartSend, (), (override));
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MOCK_METHOD(void, StopSend, (), (override));
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MOCK_METHOD(void,
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SetFrameEncryptor,
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(rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor),
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(override));
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MOCK_METHOD(
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void,
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SetEncoderToPacketizerFrameTransformer,
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(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
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(override));
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};
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} // namespace test
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} // namespace webrtc
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#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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