Searched defs:WEBRTC_SPL_MUL_16_16_RSFT (Results 1 - 3 of 3) sorted by relevance
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| H A D | background_noise.cc | 194 WEBRTC_SPL_MUL_16_16_RSFT(kThresholdIncrement,
126 assert(channel < num_channels_); return channel_parameters_[channel].energy; } void BackgroundNoise::SetMuteFactor(size_t channel, int16_t value) { assert(channel < num_channels_); channel_parameters_[channel].mute_factor = value; } int16_t BackgroundNoise::MuteFactor(size_t channel) const { assert(channel < num_channels_); return channel_parameters_[channel].mute_factor; } const int16_t* BackgroundNoise::Filter(size_t channel) const { assert(channel < num_channels_); return channel_parameters_[channel].filter; } const int16_t* BackgroundNoise::FilterState(size_t channel) const { assert(channel < num_channels_); return channel_parameters_[channel].filter_state; } void BackgroundNoise::SetFilterState(size_t channel, const int16_t* input, size_t length) { assert(channel < num_channels_); length = std::min(length, static_cast<size_t>(kMaxLpcOrder)); memcpy(channel_parameters_[channel].filter_state, input, length * sizeof(int16_t)); } int16_t BackgroundNoise::Scale(size_t channel) const { assert(channel < num_channels_); return channel_parameters_[channel].scale; } int16_t BackgroundNoise::ScaleShift(size_t channel) const { assert(channel < num_channels_); return channel_parameters_[channel].scale_shift; } int32_t BackgroundNoise::CalculateAutoCorrelation( const int16_t* signal, int length, int32_t* auto_correlation) const { int16_t signal_max = WebRtcSpl_MaxAbsValueW16(signal, length); int correlation_scale = kLogVecLen - WebRtcSpl_NormW32(signal_max * signal_max); correlation_scale = std::max(0, correlation_scale); static const int kCorrelationStep = -1; WebRtcSpl_CrossCorrelation(auto_correlation, signal, signal, length, kMaxLpcOrder + 1, correlation_scale, kCorrelationStep); int energy_sample_shift = kLogVecLen - correlation_scale; return auto_correlation[0] >> energy_sample_shift; } void BackgroundNoise::IncrementEnergyThreshold(size_t channel, int32_t sample_energy) { assert(channel < num_channels_); ChannelParameters& parameters = channel_parameters_[channel]; int32_t temp_energy = WEBRTC_SPL_MUL_16_16_RSFT(kThresholdIncrement, parameters.low_energy_update_threshold, 16); temp_energy += kThresholdIncrement * (parameters.energy_update_threshold & 0xFF); temp_energy += (kThresholdIncrement * ((parameters.energy_update_threshold>>8) & 0xFF)) << 8; parameters.low_energy_update_threshold += temp_energy; parameters.energy_update_threshold += kThresholdIncrement * (parameters.energy_update_threshold>>16); parameters.energy_update_threshold += parameters.low_energy_update_threshold >> 16; parameters.low_energy_update_threshold = parameters.low_energy_update_threshold & 0x0FFFF; parameters.max_energy = parameters.max_energy - (parameters.max_energy >> 10); if (sample_energy > parameters.max_energy) argument
|
| /external/chromium_org/third_party/webrtc/common_audio/signal_processing/include/ |
| H A D | signal_processing_library.h | 72 #define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \ macro
101 ((int16_t)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
|
| /external/webrtc/src/common_audio/signal_processing/include/ |
| H A D | signal_processing_library.h | 116 #define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \ macro
162 ((WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
|
Completed in 146 milliseconds