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/aec3_common.h
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@ -28,7 +28,7 @@ enum class Aec3Optimization { kNone, kSse2, kAvx2, kNeon };
constexpr int kNumBlocksPerSecond = 250;
constexpr int kMetricsReportingIntervalBlocks = 10 * kNumBlocksPerSecond;
constexpr int kMetricsComputationBlocks = 7;
constexpr int kMetricsComputationBlocks = 3;
constexpr int kMetricsCollectionBlocks =
kMetricsReportingIntervalBlocks - kMetricsComputationBlocks;
@ -85,10 +85,10 @@ constexpr size_t GetRenderDelayBufferSize(size_t down_sampling_factor,
Aec3Optimization DetectOptimization();
// Computes the log2 of the input in a fast an approximate manner.
float FastApproxLog2f(const float in);
float FastApproxLog2f(float in);
// Returns dB from a power quantity expressed in log2.
float Log2TodB(const float in_log2);
float Log2TodB(float in_log2);
static_assert(1 << kBlockSizeLog2 == kBlockSize,
"Proper number of shifts for blocksize");