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/rtc_base/swap_queue.h
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@ -17,8 +17,8 @@
#include <utility>
#include <vector>
#include "absl/base/attributes.h"
#include "rtc_base/checks.h"
#include "rtc_base/system/unused.h"
namespace webrtc {
@ -127,7 +127,7 @@ class SwapQueue {
// When specified, the T given in *input must pass the ItemVerifier() test.
// The contents of *input after the call are then also guaranteed to pass the
// ItemVerifier() test.
bool Insert(T* input) RTC_WARN_UNUSED_RESULT {
ABSL_MUST_USE_RESULT bool Insert(T* input) {
RTC_DCHECK(input);
RTC_DCHECK(queue_item_verifier_(*input));
@ -168,7 +168,7 @@ class SwapQueue {
// empty). When specified, The T given in *output must pass the ItemVerifier()
// test and the contents of *output after the call are then also guaranteed to
// pass the ItemVerifier() test.
bool Remove(T* output) RTC_WARN_UNUSED_RESULT {
ABSL_MUST_USE_RESULT bool Remove(T* output) {
RTC_DCHECK(output);
RTC_DCHECK(queue_item_verifier_(*output));