webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
|
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] |
void webrtc::voe::MixWithSat | ( | int16_t | target[], |
size_t | target_channel, | ||
const int16_t | source[], | ||
size_t | source_channel, | ||
size_t | source_len | ||
) |
void webrtc::voe::RemixAndResample | ( | const AudioFrame & | src_frame, |
PushResampler< int16_t > * | resampler, | ||
AudioFrame * | dst_frame | ||
) |
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 | ||
) |
const int webrtc::voe::kTelephoneEventAttenuationdB = 10 |
const int8_t webrtc::voe::permutation[33] |