webrtc::acm2::AcmReceiver Class Reference

#include <acm_receiver.h>

Public Member Functions
	AcmReceiver (const AudioCodingModule::Config &config)
	~AcmReceiver ()
int	InsertPacket (const WebRtcRTPHeader &rtp_header, rtc::ArrayView< const uint8_t > incoming_payload)
int	GetAudio (int desired_freq_hz, AudioFrame *audio_frame, bool *muted)
int	AddCodec (int acm_codec_id, uint8_t payload_type, size_t channels, int sample_rate_hz, AudioDecoder *audio_decoder, const std::string &name)
bool	AddCodec (int rtp_payload_type, const SdpAudioFormat &audio_format)
int	SetMinimumDelay (int delay_ms)
int	SetMaximumDelay (int delay_ms)
int	LeastRequiredDelayMs () const
void	ResetInitialDelay ()
rtc::Optional< int >	last_packet_sample_rate_hz () const
int	last_output_sample_rate_hz () const
void	GetNetworkStatistics (NetworkStatistics *statistics)
void	FlushBuffers ()
int	RemoveCodec (uint8_t payload_type)
void	RemoveAllCodecs ()
rtc::Optional< uint32_t >	GetPlayoutTimestamp ()
int	FilteredCurrentDelayMs () const
int	LastAudioCodec (CodecInst *codec) const
rtc::Optional< SdpAudioFormat >	LastAudioFormat () const
int	DecoderByPayloadType (uint8_t payload_type, CodecInst *codec) const
int	EnableNack (size_t max_nack_list_size)
void	DisableNack ()
std::vector< uint16_t >	GetNackList (int64_t round_trip_time_ms) const
void	GetDecodingCallStatistics (AudioDecodingCallStats *stats) const
	AcmReceiver (const AudioCodingModule::Config &config)
	~AcmReceiver ()
int	InsertPacket (const WebRtcRTPHeader &rtp_header, rtc::ArrayView< const uint8_t > incoming_payload)
int	GetAudio (int desired_freq_hz, AudioFrame *audio_frame, bool *muted)
int	AddCodec (int acm_codec_id, uint8_t payload_type, size_t channels, int sample_rate_hz, AudioDecoder *audio_decoder, const std::string &name)
bool	AddCodec (int rtp_payload_type, const SdpAudioFormat &audio_format)
int	SetMinimumDelay (int delay_ms)
int	SetMaximumDelay (int delay_ms)
int	LeastRequiredDelayMs () const
void	ResetInitialDelay ()
rtc::Optional< int >	last_packet_sample_rate_hz () const
int	last_output_sample_rate_hz () const
void	GetNetworkStatistics (NetworkStatistics *statistics)
void	FlushBuffers ()
int	RemoveCodec (uint8_t payload_type)
void	RemoveAllCodecs ()
rtc::Optional< uint32_t >	GetPlayoutTimestamp ()
int	FilteredCurrentDelayMs () const
int	LastAudioCodec (CodecInst *codec) const
rtc::Optional< SdpAudioFormat >	LastAudioFormat () const
int	DecoderByPayloadType (uint8_t payload_type, CodecInst *codec) const
int	EnableNack (size_t max_nack_list_size)
void	DisableNack ()
std::vector< uint16_t >	GetNackList (int64_t round_trip_time_ms) const
void	GetDecodingCallStatistics (AudioDecodingCallStats *stats) const

Constructor & Destructor Documentation

AcmReceiver() [1/2]

webrtc::acm2::AcmReceiver::AcmReceiver ( const AudioCodingModule::Config &  config )

explicit

~AcmReceiver() [1/2]

webrtc::acm2::AcmReceiver::~AcmReceiver

(

)

AcmReceiver() [2/2]

webrtc::acm2::AcmReceiver::AcmReceiver ( const AudioCodingModule::Config &  config )

explicit

~AcmReceiver() [2/2]

webrtc::acm2::AcmReceiver::~AcmReceiver

(

)

Member Function Documentation

AddCodec() [1/4]

int32_t webrtc::acm2::AcmReceiver::AddCodec	(	int	acm_codec_id,
		uint8_t	payload_type,
		size_t	channels,
		int	sample_rate_hz,
		AudioDecoder *	audio_decoder,
		const std::string &	name
	)

AddCodec() [2/4]

int webrtc::acm2::AcmReceiver::AddCodec	(	int	acm_codec_id,
		uint8_t	payload_type,
		size_t	channels,
		int	sample_rate_hz,
		AudioDecoder *	audio_decoder,
		const std::string &	name
	)

AddCodec() [3/4]

bool webrtc::acm2::AcmReceiver::AddCodec	(	int	rtp_payload_type,
		const SdpAudioFormat &	audio_format
	)

AddCodec() [4/4]

bool webrtc::acm2::AcmReceiver::AddCodec	(	int	rtp_payload_type,
		const SdpAudioFormat &	audio_format
	)

DecoderByPayloadType() [1/2]

int webrtc::acm2::AcmReceiver::DecoderByPayloadType	(	uint8_t	payload_type,
		CodecInst *	codec
	)		const

DecoderByPayloadType() [2/2]

int webrtc::acm2::AcmReceiver::DecoderByPayloadType	(	uint8_t	payload_type,
		CodecInst *	codec
	)		const

DisableNack() [1/2]

void webrtc::acm2::AcmReceiver::DisableNack

(

)

DisableNack() [2/2]

void webrtc::acm2::AcmReceiver::DisableNack

(

)

EnableNack() [1/2]

int webrtc::acm2::AcmReceiver::EnableNack

(

size_t

max_nack_list_size

)

EnableNack() [2/2]

int webrtc::acm2::AcmReceiver::EnableNack

(

size_t

max_nack_list_size

)

FilteredCurrentDelayMs() [1/2]

int webrtc::acm2::AcmReceiver::FilteredCurrentDelayMs

(

)

const

FilteredCurrentDelayMs() [2/2]

int webrtc::acm2::AcmReceiver::FilteredCurrentDelayMs

(

)

const

FlushBuffers() [1/2]

void webrtc::acm2::AcmReceiver::FlushBuffers

(

)

FlushBuffers() [2/2]

void webrtc::acm2::AcmReceiver::FlushBuffers

(

)

GetAudio() [1/2]

int webrtc::acm2::AcmReceiver::GetAudio	(	int	desired_freq_hz,
		AudioFrame *	audio_frame,
		bool *	muted
	)

GetAudio() [2/2]

int webrtc::acm2::AcmReceiver::GetAudio	(	int	desired_freq_hz,
		AudioFrame *	audio_frame,
		bool *	muted
	)

GetDecodingCallStatistics() [1/2]

void webrtc::acm2::AcmReceiver::GetDecodingCallStatistics

(

AudioDecodingCallStats *

stats

)

const

GetDecodingCallStatistics() [2/2]

void webrtc::acm2::AcmReceiver::GetDecodingCallStatistics

(

AudioDecodingCallStats *

stats

)

const

GetNackList() [1/2]

std::vector<uint16_t> webrtc::acm2::AcmReceiver::GetNackList

(

int64_t

round_trip_time_ms

)

const

GetNackList() [2/2]

std::vector< uint16_t > webrtc::acm2::AcmReceiver::GetNackList

(

int64_t

round_trip_time_ms

)

const

GetNetworkStatistics() [1/2]

void webrtc::acm2::AcmReceiver::GetNetworkStatistics

(

NetworkStatistics *

statistics

)

GetNetworkStatistics() [2/2]

void webrtc::acm2::AcmReceiver::GetNetworkStatistics

(

NetworkStatistics *

statistics

)

GetPlayoutTimestamp() [1/2]

rtc::Optional<uint32_t> webrtc::acm2::AcmReceiver::GetPlayoutTimestamp

(

)

GetPlayoutTimestamp() [2/2]

rtc::Optional< uint32_t > webrtc::acm2::AcmReceiver::GetPlayoutTimestamp

(

)

InsertPacket() [1/2]

int webrtc::acm2::AcmReceiver::InsertPacket	(	const WebRtcRTPHeader &	rtp_header,
		rtc::ArrayView< const uint8_t >	incoming_payload
	)

InsertPacket() [2/2]

int webrtc::acm2::AcmReceiver::InsertPacket	(	const WebRtcRTPHeader &	rtp_header,
		rtc::ArrayView< const uint8_t >	incoming_payload
	)

last_output_sample_rate_hz() [1/2]

int webrtc::acm2::AcmReceiver::last_output_sample_rate_hz

(

)

const

last_output_sample_rate_hz() [2/2]

int webrtc::acm2::AcmReceiver::last_output_sample_rate_hz

(

)

const

last_packet_sample_rate_hz() [1/2]

rtc::Optional< int > webrtc::acm2::AcmReceiver::last_packet_sample_rate_hz

(

)

const

last_packet_sample_rate_hz() [2/2]

rtc::Optional<int> webrtc::acm2::AcmReceiver::last_packet_sample_rate_hz

(

)

const

LastAudioCodec() [1/2]

int webrtc::acm2::AcmReceiver::LastAudioCodec

(

CodecInst *

codec

)

const

LastAudioCodec() [2/2]

int webrtc::acm2::AcmReceiver::LastAudioCodec

(

CodecInst *

codec

)

const

LastAudioFormat() [1/2]

rtc::Optional< SdpAudioFormat > webrtc::acm2::AcmReceiver::LastAudioFormat

(

)

const

LastAudioFormat() [2/2]

rtc::Optional<SdpAudioFormat> webrtc::acm2::AcmReceiver::LastAudioFormat

(

)

const

LeastRequiredDelayMs() [1/2]

int webrtc::acm2::AcmReceiver::LeastRequiredDelayMs

(

)

const

LeastRequiredDelayMs() [2/2]

int webrtc::acm2::AcmReceiver::LeastRequiredDelayMs

(

)

const

RemoveAllCodecs() [1/2]

void webrtc::acm2::AcmReceiver::RemoveAllCodecs

(

)

RemoveAllCodecs() [2/2]

void webrtc::acm2::AcmReceiver::RemoveAllCodecs

(

)

RemoveCodec() [1/2]

int webrtc::acm2::AcmReceiver::RemoveCodec

(

uint8_t

payload_type

)

RemoveCodec() [2/2]

int webrtc::acm2::AcmReceiver::RemoveCodec

(

uint8_t

payload_type

)

ResetInitialDelay() [1/2]

void webrtc::acm2::AcmReceiver::ResetInitialDelay

(

)

ResetInitialDelay() [2/2]

void webrtc::acm2::AcmReceiver::ResetInitialDelay

(

)

SetMaximumDelay() [1/2]

int webrtc::acm2::AcmReceiver::SetMaximumDelay

(

int

delay_ms

)

SetMaximumDelay() [2/2]

int webrtc::acm2::AcmReceiver::SetMaximumDelay

(

int

delay_ms

)

SetMinimumDelay() [1/2]

int webrtc::acm2::AcmReceiver::SetMinimumDelay

(

int

delay_ms

)

SetMinimumDelay() [2/2]

int webrtc::acm2::AcmReceiver::SetMinimumDelay

(

int

delay_ms

)

---

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

• DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/acm2/acm_receiver.h
• Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.cc