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/aec3/render_signal_analyzer.h
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@ -20,7 +20,6 @@
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/aec3/render_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -30,6 +29,9 @@ class RenderSignalAnalyzer {
explicit RenderSignalAnalyzer(const EchoCanceller3Config& config);
~RenderSignalAnalyzer();
RenderSignalAnalyzer(const RenderSignalAnalyzer&) = delete;
RenderSignalAnalyzer& operator=(const RenderSignalAnalyzer&) = delete;
// Updates the render signal analysis with the most recent render signal.
void Update(const RenderBuffer& render_buffer,
const absl::optional<size_t>& delay_partitions);
@ -53,8 +55,6 @@ class RenderSignalAnalyzer {
std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_;
absl::optional<int> narrow_peak_band_;
size_t narrow_peak_counter_;
RTC_DISALLOW_COPY_AND_ASSIGN(RenderSignalAnalyzer);
};
} // namespace webrtc