/*
* 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 , SetRtcpMode, (RtcpMode mode), (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(std::vector<RtpSource>, GetSources, (), (const , 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(std::optional<int64_t>,
GetCurrentEstimatedPlayoutNtpTimestampMs,
(int64_t now_ms),
(const , override));
MOCK_METHOD(std::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((std::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,
const SdpAudioFormat& encoder_format,
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*),
(override));
MOCK_METHOD(void , ResetSenderCongestionControlObjects, (), (override));
MOCK_METHOD(CallSendStatistics, GetRTCPStatistics, (), (const , override));
MOCK_METHOD(std::vector<ReportBlockData>,
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));
MOCK_METHOD(std::optional<DataRate>, GetUsedRate, (), (const , override));
MOCK_METHOD(void ,
RegisterPacketOverhead,
(int packet_byte_overhead),
(override));
};
} // namespace test
} // namespace webrtc
#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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