#include <mock_audio_encoder.h>

Inheritance diagram for webrtc::MockAudioEncoder:

Classes
class	CopyEncoding

class	FakeEncoding

Public Member Functions
	~MockAudioEncoder ()

	MOCK_METHOD0 (Die, void())

	MOCK_METHOD1 (Mark, void(std::string desc))

	MOCK_CONST_METHOD0 (SampleRateHz, int())

	MOCK_CONST_METHOD0 (NumChannels, size_t())

	MOCK_CONST_METHOD0 (RtpTimestampRateHz, int())

	MOCK_CONST_METHOD0 (Num10MsFramesInNextPacket, size_t())

	MOCK_CONST_METHOD0 (Max10MsFramesInAPacket, size_t())

	MOCK_CONST_METHOD0 (GetTargetBitrate, int())

	MOCK_METHOD0 (Reset, void())

	MOCK_METHOD1 (SetFec, bool(bool enable))

	MOCK_METHOD1 (SetDtx, bool(bool enable))

	MOCK_METHOD1 (SetApplication, bool(Application application))

	MOCK_METHOD1 (SetMaxPlaybackRate, void(int frequency_hz))

	MOCK_METHOD1 (SetMaxBitrate, void(int max_bps))

	MOCK_METHOD1 (SetMaxPayloadSize, void(int max_payload_size_bytes))

	MOCK_METHOD2 (OnReceivedUplinkBandwidth, void(int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms))

	MOCK_METHOD1 (OnReceivedUplinkPacketLossFraction, void(float uplink_packet_loss_fraction))

	MOCK_METHOD3 (EncodeImpl, EncodedInfo(uint32_t timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded))

	~MockAudioEncoder ()

	MOCK_METHOD0 (Die, void())

	MOCK_METHOD1 (Mark, void(std::string desc))

	MOCK_CONST_METHOD0 (SampleRateHz, int())

	MOCK_CONST_METHOD0 (NumChannels, size_t())

	MOCK_CONST_METHOD0 (RtpTimestampRateHz, int())

	MOCK_CONST_METHOD0 (Num10MsFramesInNextPacket, size_t())

	MOCK_CONST_METHOD0 (Max10MsFramesInAPacket, size_t())

	MOCK_CONST_METHOD0 (GetTargetBitrate, int())

	MOCK_METHOD0 (Reset, void())

	MOCK_METHOD1 (SetFec, bool(bool enable))

	MOCK_METHOD1 (SetDtx, bool(bool enable))

	MOCK_METHOD1 (SetApplication, bool(Application application))

	MOCK_METHOD1 (SetMaxPlaybackRate, void(int frequency_hz))

	MOCK_METHOD1 (SetMaxBitrate, void(int max_bps))

	MOCK_METHOD1 (SetMaxPayloadSize, void(int max_payload_size_bytes))

	MOCK_METHOD2 (OnReceivedUplinkBandwidth, void(int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms))

	MOCK_METHOD1 (OnReceivedUplinkPacketLossFraction, void(float uplink_packet_loss_fraction))

	MOCK_METHOD3 (EncodeImpl, EncodedInfo(uint32_t timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded))

Public Member Functions inherited from webrtc::AudioEncoder
virtual	~AudioEncoder ()=default

virtual int	SampleRateHz () const =0

virtual size_t	NumChannels () const =0

virtual int	RtpTimestampRateHz () const

virtual size_t	Num10MsFramesInNextPacket () const =0

virtual size_t	Max10MsFramesInAPacket () const =0

virtual int	GetTargetBitrate () const =0

EncodedInfo	Encode (uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)

virtual void	Reset ()=0

virtual bool	SetFec (bool enable)

virtual bool	SetDtx (bool enable)

virtual bool	GetDtx () const

virtual bool	SetApplication (Application application)

virtual void	SetMaxPlaybackRate (int frequency_hz)

virtual RTC_DEPRECATED void	SetTargetBitrate (int target_bps)

virtual rtc::ArrayView< std::unique_ptr< AudioEncoder > >	ReclaimContainedEncoders ()

virtual bool	EnableAudioNetworkAdaptor (const std::string &config_string, RtcEventLog event_log, const Clock clock)

virtual void	DisableAudioNetworkAdaptor ()

virtual void	OnReceivedUplinkPacketLossFraction (float uplink_packet_loss_fraction)

virtual void	OnReceivedTargetAudioBitrate (int target_bps)

virtual void	OnReceivedUplinkBandwidth (int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms)

virtual void	OnReceivedRtt (int rtt_ms)

virtual void	OnReceivedOverhead (size_t overhead_bytes_per_packet)

virtual void	SetReceiverFrameLengthRange (int min_frame_length_ms, int max_frame_length_ms)

virtual	~AudioEncoder ()=default

virtual int	SampleRateHz () const =0

virtual size_t	NumChannels () const =0

virtual int	RtpTimestampRateHz () const

virtual size_t	Num10MsFramesInNextPacket () const =0

virtual size_t	Max10MsFramesInAPacket () const =0

virtual int	GetTargetBitrate () const =0

EncodedInfo	Encode (uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)

virtual void	Reset ()=0

virtual bool	SetFec (bool enable)

virtual bool	SetDtx (bool enable)

virtual bool	GetDtx () const

virtual bool	SetApplication (Application application)

virtual void	SetMaxPlaybackRate (int frequency_hz)

virtual RTC_DEPRECATED void	SetTargetBitrate (int target_bps)

virtual rtc::ArrayView< std::unique_ptr< AudioEncoder > >	ReclaimContainedEncoders ()

virtual bool	EnableAudioNetworkAdaptor (const std::string &config_string, RtcEventLog event_log, const Clock clock)

virtual void	DisableAudioNetworkAdaptor ()

virtual void	OnReceivedUplinkPacketLossFraction (float uplink_packet_loss_fraction)

virtual void	OnReceivedTargetAudioBitrate (int target_bps)

virtual void	OnReceivedUplinkBandwidth (int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms)

virtual void	OnReceivedRtt (int rtt_ms)

virtual void	OnReceivedOverhead (size_t overhead_bytes_per_packet)

virtual void	SetReceiverFrameLengthRange (int min_frame_length_ms, int max_frame_length_ms)

Additional Inherited Members
Public Types inherited from webrtc::AudioEncoder
enum	CodecType { CodecType::kOther = 0, CodecType::kOpus = 1, CodecType::kIsac = 2, CodecType::kPcmA = 3, CodecType::kPcmU = 4, CodecType::kG722 = 5, CodecType::kIlbc = 6, CodecType::kMaxLoggedAudioCodecTypes, CodecType::kOther = 0, CodecType::kOpus = 1, CodecType::kIsac = 2, CodecType::kPcmA = 3, CodecType::kPcmU = 4, CodecType::kG722 = 5, CodecType::kIlbc = 6, CodecType::kMaxLoggedAudioCodecTypes }

enum	Application { Application::kSpeech, Application::kAudio, Application::kSpeech, Application::kAudio }

enum	CodecType { CodecType::kOther = 0, CodecType::kOpus = 1, CodecType::kIsac = 2, CodecType::kPcmA = 3, CodecType::kPcmU = 4, CodecType::kG722 = 5, CodecType::kIlbc = 6, CodecType::kMaxLoggedAudioCodecTypes, CodecType::kOther = 0, CodecType::kOpus = 1, CodecType::kIsac = 2, CodecType::kPcmA = 3, CodecType::kPcmU = 4, CodecType::kG722 = 5, CodecType::kIlbc = 6, CodecType::kMaxLoggedAudioCodecTypes }

enum	Application { Application::kSpeech, Application::kAudio, Application::kSpeech, Application::kAudio }

Protected Member Functions inherited from webrtc::AudioEncoder
virtual EncodedInfo	EncodeImpl (uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)=0

virtual EncodedInfo	EncodeImpl (uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)=0

Constructor & Destructor Documentation

◆ ~MockAudioEncoder() [1/2]

webrtc::MockAudioEncoder::~MockAudioEncoder ( )

inline

◆ ~MockAudioEncoder() [2/2]

webrtc::MockAudioEncoder::~MockAudioEncoder ( )

inline

Member Function Documentation

◆ MOCK_CONST_METHOD0() [1/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	SampleRateHz	,
		int()
	)

◆ MOCK_CONST_METHOD0() [2/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	SampleRateHz	,
		int()
	)

◆ MOCK_CONST_METHOD0() [3/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	NumChannels	,
		size_t()
	)

◆ MOCK_CONST_METHOD0() [4/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	NumChannels	,
		size_t()
	)

◆ MOCK_CONST_METHOD0() [5/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	RtpTimestampRateHz	,
		int()
	)

◆ MOCK_CONST_METHOD0() [6/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	RtpTimestampRateHz	,
		int()
	)

◆ MOCK_CONST_METHOD0() [7/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	Num10MsFramesInNextPacket	,
		size_t()
	)

◆ MOCK_CONST_METHOD0() [8/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	Num10MsFramesInNextPacket	,
		size_t()
	)

◆ MOCK_CONST_METHOD0() [9/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	Max10MsFramesInAPacket	,
		size_t()
	)

◆ MOCK_CONST_METHOD0() [10/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	Max10MsFramesInAPacket	,
		size_t()
	)

◆ MOCK_CONST_METHOD0() [11/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	GetTargetBitrate	,
		int()
	)

◆ MOCK_CONST_METHOD0() [12/12]

webrtc::MockAudioEncoder::MOCK_CONST_METHOD0	(	GetTargetBitrate	,
		int()
	)

◆ MOCK_METHOD0() [1/4]

webrtc::MockAudioEncoder::MOCK_METHOD0	(	Die	,
		void()
	)

◆ MOCK_METHOD0() [2/4]

webrtc::MockAudioEncoder::MOCK_METHOD0	(	Die	,
		void()
	)

◆ MOCK_METHOD0() [3/4]

webrtc::MockAudioEncoder::MOCK_METHOD0	(	Reset	,
		void()
	)

◆ MOCK_METHOD0() [4/4]

webrtc::MockAudioEncoder::MOCK_METHOD0	(	Reset	,
		void()
	)

◆ MOCK_METHOD1() [1/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	Mark	,
		void(std::string desc)
	)

◆ MOCK_METHOD1() [2/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	Mark	,
		void(std::string desc)
	)

◆ MOCK_METHOD1() [3/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetFec	,
		bool(bool enable)
	)

◆ MOCK_METHOD1() [4/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetFec	,
		bool(bool enable)
	)

◆ MOCK_METHOD1() [5/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetDtx	,
		bool(bool enable)
	)

◆ MOCK_METHOD1() [6/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetDtx	,
		bool(bool enable)
	)

◆ MOCK_METHOD1() [7/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetApplication	,
		bool(Application application)
	)

◆ MOCK_METHOD1() [8/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetApplication	,
		bool(Application application)
	)

◆ MOCK_METHOD1() [9/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetMaxPlaybackRate	,
		void(int frequency_hz)
	)

◆ MOCK_METHOD1() [10/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetMaxPlaybackRate	,
		void(int frequency_hz)
	)

◆ MOCK_METHOD1() [11/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetMaxBitrate	,
		void(int max_bps)
	)

◆ MOCK_METHOD1() [12/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetMaxBitrate	,
		void(int max_bps)
	)

◆ MOCK_METHOD1() [13/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetMaxPayloadSize	,
		void(int max_payload_size_bytes)
	)

◆ MOCK_METHOD1() [14/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	SetMaxPayloadSize	,
		void(int max_payload_size_bytes)
	)

◆ MOCK_METHOD1() [15/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	OnReceivedUplinkPacketLossFraction	,
		void(float uplink_packet_loss_fraction)
	)

◆ MOCK_METHOD1() [16/16]

webrtc::MockAudioEncoder::MOCK_METHOD1	(	OnReceivedUplinkPacketLossFraction	,
		void(float uplink_packet_loss_fraction)
	)

◆ MOCK_METHOD2() [1/2]

webrtc::MockAudioEncoder::MOCK_METHOD2	(	OnReceivedUplinkBandwidth	,
		void(int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms)
	)

◆ MOCK_METHOD2() [2/2]

webrtc::MockAudioEncoder::MOCK_METHOD2	(	OnReceivedUplinkBandwidth	,
		void(int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms)
	)

◆ MOCK_METHOD3() [1/2]

webrtc::MockAudioEncoder::MOCK_METHOD3	(	EncodeImpl	,
		EncodedInfo(uint32_t timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)
	)

◆ MOCK_METHOD3() [2/2]

webrtc::MockAudioEncoder::MOCK_METHOD3	(	EncodeImpl	,
		EncodedInfo(uint32_t timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)
	)

The documentation for this class was generated from the following file:

DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h

Classes

Public Member Functions

Additional Inherited Members