AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M131 release

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Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
This commit is contained in:
Arun Raghavan
2024-12-24 19:32:07 -05:00
parent 8bdb53d91c
commit b5c48b97f6
263 changed files with 4628 additions and 20416 deletions
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60
webrtc/modules/audio_processing/include/audio_frame_view.h
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@ -11,7 +11,7 @@
#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
#include "api/array_view.h"
#include "api/audio/audio_view.h"
namespace webrtc {
@ -22,46 +22,44 @@ class AudioFrameView {
// `num_channels` and `channel_size` describe the T**
// `audio_samples`. `audio_samples` is assumed to point to a
// two-dimensional |num_channels * channel_size| array of floats.
//
// Note: The implementation now only requires the first channel pointer.
// The previous implementation retained a pointer to externally owned array
// of channel pointers, but since the channel size and count are provided
// and the array is assumed to be a single two-dimensional array, the other
// channel pointers can be calculated based on that (which is what the class
// now uses `DeinterleavedView<>` internally for).
AudioFrameView(T* const* audio_samples, int num_channels, int channel_size)
: audio_samples_(audio_samples),
num_channels_(num_channels),
channel_size_(channel_size) {
RTC_DCHECK_GE(num_channels_, 0);
RTC_DCHECK_GE(channel_size_, 0);
: view_(num_channels && channel_size ? audio_samples[0] : nullptr,
channel_size,
num_channels) {
RTC_DCHECK_GE(view_.num_channels(), 0);
RTC_DCHECK_GE(view_.samples_per_channel(), 0);
}
// Implicit cast to allow converting Frame<float> to
// Frame<const float>.
// Implicit cast to allow converting AudioFrameView<float> to
// AudioFrameView<const float>.
template <class U>
AudioFrameView(AudioFrameView<U> other)
: audio_samples_(other.data()),
num_channels_(other.num_channels()),
channel_size_(other.samples_per_channel()) {}
AudioFrameView(AudioFrameView<U> other) : view_(other.view()) {}
// Allow constructing AudioFrameView from a DeinterleavedView.
template <class U>
explicit AudioFrameView(DeinterleavedView<U> view) : view_(view) {}
AudioFrameView() = delete;
int num_channels() const { return num_channels_; }
int num_channels() const { return view_.num_channels(); }
int samples_per_channel() const { return view_.samples_per_channel(); }
MonoView<T> channel(int idx) { return view_[idx]; }
MonoView<const T> channel(int idx) const { return view_[idx]; }
MonoView<T> operator[](int idx) { return view_[idx]; }
MonoView<const T> operator[](int idx) const { return view_[idx]; }
int samples_per_channel() const { return channel_size_; }
rtc::ArrayView<T> channel(int idx) {
RTC_DCHECK_LE(0, idx);
RTC_DCHECK_LE(idx, num_channels_);
return rtc::ArrayView<T>(audio_samples_[idx], channel_size_);
}
rtc::ArrayView<const T> channel(int idx) const {
RTC_DCHECK_LE(0, idx);
RTC_DCHECK_LE(idx, num_channels_);
return rtc::ArrayView<const T>(audio_samples_[idx], channel_size_);
}
T* const* data() { return audio_samples_; }
DeinterleavedView<T> view() { return view_; }
DeinterleavedView<const T> view() const { return view_; }
private:
T* const* audio_samples_;
int num_channels_;
int channel_size_;
DeinterleavedView<T> view_;
};
} // namespace webrtc