AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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17
webrtc/modules/audio_processing/aec3/block_delay_buffer.cc
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@ -41,17 +41,24 @@ void BlockDelayBuffer::DelaySignal(AudioBuffer* frame) {
RTC_DCHECK_EQ(buf_[ch].size(), frame->num_bands());
RTC_DCHECK_EQ(buf_[ch].size(), num_bands);
rtc::ArrayView<float* const> frame_ch(frame->split_bands(ch), num_bands);
const size_t delay = delay_;
for (size_t band = 0; band < num_bands; ++band) {
RTC_DCHECK_EQ(delay_, buf_[ch][band].size());
i = i_start;
for (size_t k = 0; k < frame_length_; ++k) {
const float tmp = buf_[ch][band][i];
buf_[ch][band][i] = frame_ch[band][k];
frame_ch[band][k] = tmp;
// Offloading these pointers and class variables to local variables allows
// the compiler to optimize the below loop when compiling with
// '-fno-strict-aliasing'.
float* buf_ch_band = buf_[ch][band].data();
float* frame_ch_band = frame_ch[band];
i = i < delay_ - 1 ? i + 1 : 0;
for (size_t k = 0, frame_length = frame_length_; k < frame_length; ++k) {
const float tmp = buf_ch_band[i];
buf_ch_band[i] = frame_ch_band[k];
frame_ch_band[k] = tmp;
i = i < delay - 1 ? i + 1 : 0;
}
}
}