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

This CL reduces the allowed jitter in the api calls to a reasonable
level in order to ensure a quicker revery from audio path glitches.

BUG=webrtc:8224, chromium:763775

Review-Url: https://codereview.webrtc.org/3009273002
Cr-Original-Commit-Position: refs/heads/master@{#19772}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: ea154106a81f2f8f08850a19caee09834283a673
diff --git a/modules/audio_processing/aec3/aec3_common.h b/modules/audio_processing/aec3/aec3_common.h
index b8b638d..feb5bdb 100644
--- a/modules/audio_processing/aec3/aec3_common.h
+++ b/modules/audio_processing/aec3/aec3_common.h
@@ -64,7 +64,7 @@
     (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");