webrtc::AudioEncoderOpus Class Reference

#include <audio_encoder_opus.h>

Inheritance diagram for webrtc::AudioEncoderOpus:

Classes
struct	Config
class	PacketLossFractionSmoother

Public Types
enum	ApplicationMode { kVoip = 0, kAudio = 1, kVoip = 0, kAudio = 1 }
enum	ApplicationMode { kVoip = 0, kAudio = 1, kVoip = 0, kAudio = 1 }
using	AudioNetworkAdaptorCreator = std::function< std::unique_ptr< AudioNetworkAdaptor >(const std::string &, RtcEventLog *, const Clock *)>
using	AudioNetworkAdaptorCreator = std::function< std::unique_ptr< AudioNetworkAdaptor >(const std::string &, RtcEventLog *, const Clock *)>
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 }

Public Member Functions
	AudioEncoderOpus (const Config &config, AudioNetworkAdaptorCreator &&audio_network_adaptor_creator=nullptr, std::unique_ptr< SmoothingFilter > bitrate_smoother=nullptr)
	AudioEncoderOpus (const CodecInst &codec_inst)
	~AudioEncoderOpus () override
int	SampleRateHz () const override
size_t	NumChannels () const override
size_t	Num10MsFramesInNextPacket () const override
size_t	Max10MsFramesInAPacket () const override
int	GetTargetBitrate () const override
void	Reset () override
bool	SetFec (bool enable) override
bool	SetDtx (bool enable) override
bool	GetDtx () const override
bool	SetApplication (Application application) override
void	SetMaxPlaybackRate (int frequency_hz) override
bool	EnableAudioNetworkAdaptor (const std::string &config_string, RtcEventLog *event_log, const Clock *clock) override
void	DisableAudioNetworkAdaptor () override
void	OnReceivedUplinkPacketLossFraction (float uplink_packet_loss_fraction) override
void	OnReceivedUplinkBandwidth (int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms) override
void	OnReceivedRtt (int rtt_ms) override
void	OnReceivedOverhead (size_t overhead_bytes_per_packet) override
void	SetReceiverFrameLengthRange (int min_frame_length_ms, int max_frame_length_ms) override
rtc::ArrayView< const int >	supported_frame_lengths_ms () const
float	packet_loss_rate () const
ApplicationMode	application () const
bool	fec_enabled () const
size_t	num_channels_to_encode () const
int	next_frame_length_ms () const
	AudioEncoderOpus (const Config &config, AudioNetworkAdaptorCreator &&audio_network_adaptor_creator=nullptr, std::unique_ptr< SmoothingFilter > bitrate_smoother=nullptr)
	AudioEncoderOpus (const CodecInst &codec_inst)
	~AudioEncoderOpus () override
int	SampleRateHz () const override
size_t	NumChannels () const override
size_t	Num10MsFramesInNextPacket () const override
size_t	Max10MsFramesInAPacket () const override
int	GetTargetBitrate () const override
void	Reset () override
bool	SetFec (bool enable) override
bool	SetDtx (bool enable) override
bool	GetDtx () const override
bool	SetApplication (Application application) override
void	SetMaxPlaybackRate (int frequency_hz) override
bool	EnableAudioNetworkAdaptor (const std::string &config_string, RtcEventLog *event_log, const Clock *clock) override
void	DisableAudioNetworkAdaptor () override
void	OnReceivedUplinkPacketLossFraction (float uplink_packet_loss_fraction) override
void	OnReceivedUplinkBandwidth (int target_audio_bitrate_bps, rtc::Optional< int64_t > probing_interval_ms) override
void	OnReceivedRtt (int rtt_ms) override
void	OnReceivedOverhead (size_t overhead_bytes_per_packet) override
void	SetReceiverFrameLengthRange (int min_frame_length_ms, int max_frame_length_ms) override
rtc::ArrayView< const int >	supported_frame_lengths_ms () const
float	packet_loss_rate () const
ApplicationMode	application () const
bool	fec_enabled () const
size_t	num_channels_to_encode () const
int	next_frame_length_ms () const
Public Member Functions inherited from webrtc::AudioEncoder
virtual	~AudioEncoder ()=default
virtual int	RtpTimestampRateHz () const
EncodedInfo	Encode (uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)
virtual rtc::ArrayView< std::unique_ptr< AudioEncoder > >	ReclaimContainedEncoders ()
virtual void	OnReceivedTargetAudioBitrate (int target_bps)
virtual	~AudioEncoder ()=default
virtual int	RtpTimestampRateHz () const
EncodedInfo	Encode (uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded)
virtual rtc::ArrayView< std::unique_ptr< AudioEncoder > >	ReclaimContainedEncoders ()
virtual void	OnReceivedTargetAudioBitrate (int target_bps)

Protected Member Functions
EncodedInfo	EncodeImpl (uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded) override
EncodedInfo	EncodeImpl (uint32_t rtp_timestamp, rtc::ArrayView< const int16_t > audio, rtc::Buffer *encoded) override

Member Typedef Documentation

AudioNetworkAdaptorCreator [1/2]

using webrtc::AudioEncoderOpus::AudioNetworkAdaptorCreator = std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&, RtcEventLog*, const Clock*)>

AudioNetworkAdaptorCreator [2/2]

using webrtc::AudioEncoderOpus::AudioNetworkAdaptorCreator = std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&, RtcEventLog*, const Clock*)>

Member Enumeration Documentation

ApplicationMode [1/2]

enum webrtc::AudioEncoderOpus::ApplicationMode

Enumerator
kVoip
kAudio
kVoip
kAudio

ApplicationMode [2/2]

enum webrtc::AudioEncoderOpus::ApplicationMode

Enumerator
kVoip
kAudio
kVoip
kAudio

Constructor & Destructor Documentation

AudioEncoderOpus() [1/4]

webrtc::AudioEncoderOpus::AudioEncoderOpus	(	const Config &	config,
		AudioNetworkAdaptorCreator &&	audio_network_adaptor_creator = nullptr,
		std::unique_ptr< SmoothingFilter >	bitrate_smoother = nullptr
	)

AudioEncoderOpus() [2/4]

webrtc::AudioEncoderOpus::AudioEncoderOpus ( const CodecInst &  codec_inst )

explicit

~AudioEncoderOpus() [1/2]

webrtc::AudioEncoderOpus::~AudioEncoderOpus ( )

override

AudioEncoderOpus() [3/4]

webrtc::AudioEncoderOpus::AudioEncoderOpus	(	const Config &	config,
		AudioNetworkAdaptorCreator &&	audio_network_adaptor_creator = nullptr,
		std::unique_ptr< SmoothingFilter >	bitrate_smoother = nullptr
	)

AudioEncoderOpus() [4/4]

webrtc::AudioEncoderOpus::AudioEncoderOpus ( const CodecInst &  codec_inst )

explicit

~AudioEncoderOpus() [2/2]

webrtc::AudioEncoderOpus::~AudioEncoderOpus ( )

override

Member Function Documentation

application() [1/2]

ApplicationMode webrtc::AudioEncoderOpus::application ( ) const

inline

application() [2/2]

ApplicationMode webrtc::AudioEncoderOpus::application ( ) const

inline

DisableAudioNetworkAdaptor() [1/2]

void webrtc::AudioEncoderOpus::DisableAudioNetworkAdaptor ( )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

DisableAudioNetworkAdaptor() [2/2]

void webrtc::AudioEncoderOpus::DisableAudioNetworkAdaptor ( )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

EnableAudioNetworkAdaptor() [1/2]

bool webrtc::AudioEncoderOpus::EnableAudioNetworkAdaptor ( const std::string &  config_string, RtcEventLog *  event_log, const Clock *  clock  )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

EnableAudioNetworkAdaptor() [2/2]

bool webrtc::AudioEncoderOpus::EnableAudioNetworkAdaptor ( const std::string &  config_string, RtcEventLog *  event_log, const Clock *  clock  )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

EncodeImpl() [1/2]

AudioEncoder::EncodedInfo webrtc::AudioEncoderOpus::EncodeImpl ( uint32_t  rtp_timestamp, rtc::ArrayView< const int16_t >  audio, rtc::Buffer *  encoded  )

overrideprotectedvirtual

Implements webrtc::AudioEncoder.

EncodeImpl() [2/2]

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

overrideprotectedvirtual

Implements webrtc::AudioEncoder.

fec_enabled() [1/2]

bool webrtc::AudioEncoderOpus::fec_enabled ( ) const

inline

fec_enabled() [2/2]

bool webrtc::AudioEncoderOpus::fec_enabled ( ) const

inline

GetDtx() [1/2]

bool webrtc::AudioEncoderOpus::GetDtx ( ) const

overridevirtual

Reimplemented from webrtc::AudioEncoder.

GetDtx() [2/2]

bool webrtc::AudioEncoderOpus::GetDtx ( ) const

overridevirtual

Reimplemented from webrtc::AudioEncoder.

GetTargetBitrate() [1/2]

int webrtc::AudioEncoderOpus::GetTargetBitrate ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

GetTargetBitrate() [2/2]

int webrtc::AudioEncoderOpus::GetTargetBitrate ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

Max10MsFramesInAPacket() [1/2]

size_t webrtc::AudioEncoderOpus::Max10MsFramesInAPacket ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

Max10MsFramesInAPacket() [2/2]

size_t webrtc::AudioEncoderOpus::Max10MsFramesInAPacket ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

next_frame_length_ms() [1/2]

int webrtc::AudioEncoderOpus::next_frame_length_ms ( ) const

inline

next_frame_length_ms() [2/2]

int webrtc::AudioEncoderOpus::next_frame_length_ms ( ) const

inline

Num10MsFramesInNextPacket() [1/2]

size_t webrtc::AudioEncoderOpus::Num10MsFramesInNextPacket ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

Num10MsFramesInNextPacket() [2/2]

size_t webrtc::AudioEncoderOpus::Num10MsFramesInNextPacket ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

num_channels_to_encode() [1/2]

size_t webrtc::AudioEncoderOpus::num_channels_to_encode ( ) const

inline

num_channels_to_encode() [2/2]

size_t webrtc::AudioEncoderOpus::num_channels_to_encode ( ) const

inline

NumChannels() [1/2]

size_t webrtc::AudioEncoderOpus::NumChannels ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

NumChannels() [2/2]

size_t webrtc::AudioEncoderOpus::NumChannels ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

OnReceivedOverhead() [1/2]

void webrtc::AudioEncoderOpus::OnReceivedOverhead ( size_t  overhead_bytes_per_packet )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

OnReceivedOverhead() [2/2]

void webrtc::AudioEncoderOpus::OnReceivedOverhead ( size_t  overhead_bytes_per_packet )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

OnReceivedRtt() [1/2]

void webrtc::AudioEncoderOpus::OnReceivedRtt ( int  rtt_ms )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

OnReceivedRtt() [2/2]

void webrtc::AudioEncoderOpus::OnReceivedRtt ( int  rtt_ms )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

OnReceivedUplinkBandwidth() [1/2]

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

overridevirtual

Reimplemented from webrtc::AudioEncoder.

OnReceivedUplinkBandwidth() [2/2]

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

overridevirtual

Reimplemented from webrtc::AudioEncoder.

OnReceivedUplinkPacketLossFraction() [1/2]

void webrtc::AudioEncoderOpus::OnReceivedUplinkPacketLossFraction ( float  uplink_packet_loss_fraction )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

OnReceivedUplinkPacketLossFraction() [2/2]

void webrtc::AudioEncoderOpus::OnReceivedUplinkPacketLossFraction ( float  uplink_packet_loss_fraction )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

packet_loss_rate() [1/2]

float webrtc::AudioEncoderOpus::packet_loss_rate ( ) const

inline

packet_loss_rate() [2/2]

float webrtc::AudioEncoderOpus::packet_loss_rate ( ) const

inline

Reset() [1/2]

void webrtc::AudioEncoderOpus::Reset ( )

overridevirtual

Implements webrtc::AudioEncoder.

Reset() [2/2]

void webrtc::AudioEncoderOpus::Reset ( )

overridevirtual

Implements webrtc::AudioEncoder.

SampleRateHz() [1/2]

int webrtc::AudioEncoderOpus::SampleRateHz ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

SampleRateHz() [2/2]

int webrtc::AudioEncoderOpus::SampleRateHz ( ) const

overridevirtual

Implements webrtc::AudioEncoder.

SetApplication() [1/2]

bool webrtc::AudioEncoderOpus::SetApplication ( Application  application )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

SetApplication() [2/2]

bool webrtc::AudioEncoderOpus::SetApplication ( Application  application )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

SetDtx() [1/2]

bool webrtc::AudioEncoderOpus::SetDtx ( bool  enable )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

SetDtx() [2/2]

bool webrtc::AudioEncoderOpus::SetDtx ( bool  enable )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

SetFec() [1/2]

bool webrtc::AudioEncoderOpus::SetFec ( bool  enable )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

SetFec() [2/2]

bool webrtc::AudioEncoderOpus::SetFec ( bool  enable )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

SetMaxPlaybackRate() [1/2]

void webrtc::AudioEncoderOpus::SetMaxPlaybackRate ( int  frequency_hz )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

SetMaxPlaybackRate() [2/2]

void webrtc::AudioEncoderOpus::SetMaxPlaybackRate ( int  frequency_hz )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

SetReceiverFrameLengthRange() [1/2]

void webrtc::AudioEncoderOpus::SetReceiverFrameLengthRange ( int  min_frame_length_ms, int  max_frame_length_ms  )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

SetReceiverFrameLengthRange() [2/2]

void webrtc::AudioEncoderOpus::SetReceiverFrameLengthRange ( int  min_frame_length_ms, int  max_frame_length_ms  )

overridevirtual

Reimplemented from webrtc::AudioEncoder.

supported_frame_lengths_ms() [1/2]

rtc::ArrayView<const int> webrtc::AudioEncoderOpus::supported_frame_lengths_ms ( ) const

inline

supported_frame_lengths_ms() [2/2]

rtc::ArrayView<const int> webrtc::AudioEncoderOpus::supported_frame_lengths_ms ( ) const

inline

---

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

• DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
• Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc