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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6
webrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
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@ -29,7 +29,7 @@ namespace rnn_vad {
// values are written at the end of the buffer.
// The class also provides a view on the most recent M values, where 0 < M <= S
// and by default M = N.
template <typename T, size_t S, size_t N, size_t M = N>
template <typename T, int S, int N, int M = N>
class SequenceBuffer {
static_assert(N <= S,
"The new chunk size cannot be larger than the sequence buffer "
@ -45,8 +45,8 @@ class SequenceBuffer {
SequenceBuffer(const SequenceBuffer&) = delete;
SequenceBuffer& operator=(const SequenceBuffer&) = delete;
~SequenceBuffer() = default;
size_t size() const { return S; }
size_t chunks_size() const { return N; }
int size() const { return S; }
int chunks_size() const { return N; }
// Sets the sequence buffer values to zero.
void Reset() { std::fill(buffer_.begin(), buffer_.end(), 0); }
// Returns a view on the whole buffer.