Revert "Make relative arrival delay mode default in NetEq delay manager."

This reverts commit 77c71d1488b1c821b2b3481f23a3264f1b1d37a5.

Reason for revert: breaking downstream projects

Original change's description:
> Make relative arrival delay mode default in NetEq delay manager.
> 
> Bug: webrtc:10333
> Change-Id: I9b1e0bec0b1813cf31259492f83eb2ca86a44d3f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150782
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29075}

TBR=henrik.lundin@webrtc.org,srte@webrtc.org,minyue@webrtc.org,jakobi@webrtc.org

Change-Id: I67c5b9c7a6e854d3aac379aa4d98bfeb5425d312
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151642
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29078}
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 769ec56..0db6fc7 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -456,16 +456,16 @@
       webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
 
   const std::string output_checksum =
-      PlatformChecksum("6ae9f643dc3e5f3452d28a772eef7e00e74158bc",
-                       "f4374430e870d66268c1b8e22fb700eb072d567e", "not used",
-                       "6ae9f643dc3e5f3452d28a772eef7e00e74158bc",
-                       "8d73c98645917cdeaaa01c20cf095ccc5a10b2b5");
+      PlatformChecksum("998be2e5a707e636af0b6298f54bedfabe72aae1",
+                       "61e238ece4cd3b67d66a0b7047e06b20607dcb79", "not used",
+                       "998be2e5a707e636af0b6298f54bedfabe72aae1",
+                       "4116ac2a6e75baac3194b712d6fabe28b384275e");
 
   const std::string network_stats_checksum =
-      PlatformChecksum("3d186ea7e243abfdbd3d39b8ebf8f02a318117e4",
-                       "0b725774133da5dd823f2046663c12a76e0dbd79", "not used",
-                       "3d186ea7e243abfdbd3d39b8ebf8f02a318117e4",
-                       "3d186ea7e243abfdbd3d39b8ebf8f02a318117e4");
+      PlatformChecksum("5e5230b2d5042eccd197dac29edade1cc233586c",
+                       "2183564f11b53259ba7f86f48f4df3d7d653c678", "not used",
+                       "5e5230b2d5042eccd197dac29edade1cc233586c",
+                       "5e5230b2d5042eccd197dac29edade1cc233586c");
 
   DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
                    absl::GetFlag(FLAGS_gen_ref));
@@ -733,7 +733,7 @@
   const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
   const double kNetworkFreezeTimeMs = 0.0;
   const bool kGetAudioDuringFreezeRecovery = false;
-  const int kDelayToleranceMs = 40;
+  const int kDelayToleranceMs = 20;
   const int kMaxTimeToSpeechMs = 100;
   LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
                         kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
@@ -757,7 +757,7 @@
   const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
   const double kNetworkFreezeTimeMs = 5000.0;
   const bool kGetAudioDuringFreezeRecovery = false;
-  const int kDelayToleranceMs = 40;
+  const int kDelayToleranceMs = 20;
   const int kMaxTimeToSpeechMs = 100;
   LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
                         kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
@@ -769,7 +769,7 @@
   const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
   const double kNetworkFreezeTimeMs = 5000.0;
   const bool kGetAudioDuringFreezeRecovery = true;
-  const int kDelayToleranceMs = 40;
+  const int kDelayToleranceMs = 20;
   const int kMaxTimeToSpeechMs = 100;
   LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
                         kGetAudioDuringFreezeRecovery, kDelayToleranceMs,