Lowered the allowed jitter in the api calls to a reasonable level

webrtc/modules/audio_processing/aec3/aec3_common.h

Index: webrtc/modules/audio_processing/aec3/aec3_common.h
diff --git a/webrtc/modules/audio_processing/aec3/aec3_common.h b/webrtc/modules/audio_processing/aec3/aec3_common.h
index b8b638d1d8b29fba632a1f375873f4f1ffc14365..feb5bdbf6240f6f9170106c8a7f9f6df41fec355 100644
--- a/webrtc/modules/audio_processing/aec3/aec3_common.h
+++ b/webrtc/modules/audio_processing/aec3/aec3_common.h
@@ -64,7 +64,7 @@ constexpr size_t kRenderDelayBufferSize =
     (3 * kDownsampledRenderBufferSize) / (4 * kSubBlockSize);
 
 constexpr size_t kMinEchoPathDelayBlocks = 5;
-constexpr size_t kMaxApiCallsJitterBlocks = 60;
+constexpr size_t kMaxApiCallsJitterBlocks = 26;
 constexpr size_t kRenderTransferQueueSize = kMaxApiCallsJitterBlocks / 2;
 static_assert(2 * kRenderTransferQueueSize >= kMaxApiCallsJitterBlocks,
               "Requirement to ensure buffer overflow detection");