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/transient/dyadic_decimator.h
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@ -18,7 +18,7 @@
namespace webrtc {
// Returns the proper length of the output buffer that you should use for the
// given |in_length| and decimation |odd_sequence|.
// given `in_length` and decimation `odd_sequence`.
// Return -1 on error.
inline size_t GetOutLengthToDyadicDecimate(size_t in_length,
bool odd_sequence) {
@ -34,10 +34,10 @@ inline size_t GetOutLengthToDyadicDecimate(size_t in_length,
// Performs a dyadic decimation: removes every odd/even member of a sequence
// halving its overall length.
// Arguments:
// in: array of |in_length|.
// in: array of `in_length`.
// odd_sequence: If false, the odd members will be removed (1, 3, 5, ...);
// if true, the even members will be removed (0, 2, 4, ...).
// out: array of |out_length|. |out_length| must be large enough to
// out: array of `out_length`. `out_length` must be large enough to
// hold the decimated output. The necessary length can be provided by
// GetOutLengthToDyadicDecimate().
// Must be previously allocated.