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/*
* 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_
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