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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 <cstddef>
#include <cstdint>
#include <map>
#include <memory>
#include <optional>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/audio/audio_frame.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/call/audio_sink.h"
#include "api/call/bitrate_allocation.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/function_view.h"
#include "api/rtp_headers.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/units/data_rate.h"
#include "audio/channel_receive.h"
#include "audio/channel_send.h"
#include "call/syncable.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.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(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,
(webrtc::scoped_refptr<webrtc::FrameTransformerInterface>
frame_transformer),
(override));
MOCK_METHOD(
void,
SetFrameDecryptor,
(webrtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
(override));
MOCK_METHOD(void, OnLocalSsrcChange, (uint32_t local_ssrc), (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,
(webrtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier),
(override));
MOCK_METHOD(void,
CallEncoder,
(webrtc::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(void, StartSend, (), (override));
MOCK_METHOD(void, StopSend, (), (override));
MOCK_METHOD(void,
SetFrameEncryptor,
(webrtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor),
(override));
MOCK_METHOD(void,
SetEncoderToPacketizerFrameTransformer,
(webrtc::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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