webkit  2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
Classes | Functions | Variables
webrtc::voe Namespace Reference

Classes

class  AudioLevel
 
class  Channel
 
class  ChannelManager
 
class  ChannelOwner
 
class  ChannelProxy
 
class  ChannelState
 
class  MonitorModule
 
class  OutputMixer
 
class  RtcEventLogProxy
 
class  RtcpRttStatsProxy
 
class  RtpPacketSenderProxy
 
class  SharedData
 
class  Statistics
 
class  TransmitMixer
 
class  TransportFeedbackProxy
 
class  TransportSequenceNumberProxy
 
class  VoERtcpObserver
 

Functions

void RemixAndResample (const AudioFrame &src_frame, PushResampler< int16_t > *resampler, AudioFrame *dst_frame)
 
void RemixAndResample (const int16_t *src_data, size_t samples_per_channel, size_t num_channels, int sample_rate_hz, PushResampler< int16_t > *resampler, AudioFrame *dst_frame)
 
void MixWithSat (int16_t target[], size_t target_channel, const int16_t source[], size_t source_channel, size_t source_len)
 

Variables

const int kTelephoneEventAttenuationdB = 10
 
const int8_t permutation [33]
 

Function Documentation

◆ MixWithSat()

void webrtc::voe::MixWithSat ( int16_t  target[],
size_t  target_channel,
const int16_t  source[],
size_t  source_channel,
size_t  source_len 
)

◆ RemixAndResample() [1/2]

void webrtc::voe::RemixAndResample ( const AudioFrame src_frame,
PushResampler< int16_t > *  resampler,
AudioFrame dst_frame 
)

◆ RemixAndResample() [2/2]

void webrtc::voe::RemixAndResample ( const int16_t src_data,
size_t  samples_per_channel,
size_t  num_channels,
int  sample_rate_hz,
PushResampler< int16_t > *  resampler,
AudioFrame dst_frame 
)

Variable Documentation

◆ kTelephoneEventAttenuationdB

const int webrtc::voe::kTelephoneEventAttenuationdB = 10

◆ permutation

const int8_t webrtc::voe::permutation[33]
Initial value:
=
{0,1,2,3,4,4,5,5,5,5,6,6,6,6,6,7,7,7,7,8,8,8,9,9,9,9,9,9,9,9,9,9,9}