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webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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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] |
1.8.13