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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24
webrtc/modules/audio_processing/audio_buffer.h
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@ -32,7 +32,7 @@ enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
class AudioBuffer {
public:
static const int kSplitBandSize = 160;
static const size_t kMaxSampleRate = 384000;
static const int kMaxSampleRate = 384000;
AudioBuffer(size_t input_rate,
size_t input_num_channels,
size_t buffer_rate,
@ -40,12 +40,6 @@ class AudioBuffer {
size_t output_rate,
size_t output_num_channels);
// The constructor below will be deprecated.
AudioBuffer(size_t input_num_frames,
size_t input_num_channels,
size_t buffer_num_frames,
size_t buffer_num_channels,
size_t output_num_frames);
virtual ~AudioBuffer();
AudioBuffer(const AudioBuffer&) = delete;
@ -71,8 +65,8 @@ class AudioBuffer {
// Usage:
// channels()[channel][sample].
// Where:
// 0 <= channel < |buffer_num_channels_|
// 0 <= sample < |buffer_num_frames_|
// 0 <= channel < `buffer_num_channels_`
// 0 <= sample < `buffer_num_frames_`
float* const* channels() { return data_->channels(); }
const float* const* channels_const() const { return data_->channels(); }
@ -80,9 +74,9 @@ class AudioBuffer {
// Usage:
// split_bands(channel)[band][sample].
// Where:
// 0 <= channel < |buffer_num_channels_|
// 0 <= band < |num_bands_|
// 0 <= sample < |num_split_frames_|
// 0 <= channel < `buffer_num_channels_`
// 0 <= band < `num_bands_`
// 0 <= sample < `num_split_frames_`
const float* const* split_bands_const(size_t channel) const {
return split_data_.get() ? split_data_->bands(channel)
: data_->bands(channel);
@ -96,9 +90,9 @@ class AudioBuffer {
// Usage:
// split_channels(band)[channel][sample].
// Where:
// 0 <= band < |num_bands_|
// 0 <= channel < |buffer_num_channels_|
// 0 <= sample < |num_split_frames_|
// 0 <= band < `num_bands_`
// 0 <= channel < `buffer_num_channels_`
// 0 <= sample < `num_split_frames_`
const float* const* split_channels_const(Band band) const {
if (split_data_.get()) {
return split_data_->channels(band);