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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25
webrtc/modules/audio_processing/include/audio_frame_view.h
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@ -19,15 +19,16 @@ namespace webrtc {
template <class T>
class AudioFrameView {
public:
// |num_channels| and |channel_size| describe the T**
// |audio_samples|. |audio_samples| is assumed to point to a
// `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.
AudioFrameView(T* const* audio_samples,
size_t num_channels,
size_t channel_size)
AudioFrameView(T* const* audio_samples, int num_channels, int channel_size)
: audio_samples_(audio_samples),
num_channels_(num_channels),
channel_size_(channel_size) {}
channel_size_(channel_size) {
RTC_DCHECK_GE(num_channels_, 0);
RTC_DCHECK_GE(channel_size_, 0);
}
// Implicit cast to allow converting Frame<float> to
// Frame<const float>.
@ -39,17 +40,17 @@ class AudioFrameView {
AudioFrameView() = delete;
size_t num_channels() const { return num_channels_; }
int num_channels() const { return num_channels_; }
size_t samples_per_channel() const { return channel_size_; }
int samples_per_channel() const { return channel_size_; }
rtc::ArrayView<T> channel(size_t idx) {
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(size_t idx) const {
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_);
@ -59,8 +60,8 @@ class AudioFrameView {
private:
T* const* audio_samples_;
size_t num_channels_;
size_t channel_size_;
int num_channels_;
int channel_size_;
};
} // namespace webrtc