commit b93af8543df5e4eb64a449b7c8f710f422ab3128
author Jakob Ivarsson <jakobi@webrtc.org> Thu Jun 27 08:07:06 2019 +0000
committer Commit Bot <commit-bot@chromium.org> Thu Jun 27 08:07:21 2019 +0000
tree bff2dbc09df22ea96adff91db60ba7a7deac0331
parent 2d821c3cbc664371294fe37c3f7197af2b80fc41

Revert "Reland "Change buffer level filter to store current level in number of samples.""

This reverts commit 0ded32d5a3d7acb9a00c3a1d9941e539aa94eee5.

Reason for revert: breaks downstream projects.

Original change's description:
> Reland "Change buffer level filter to store current level in number of samples."
> 
> This is a reland of 87977dd06e702ed517f26235c12e37bd927527c7
> 
> Original change's description:
> > Change buffer level filter to store current level in number of samples.
> > 
> > The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
> > 
> > Bug: webrtc:10736
> > Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28368}
> 
> Bug: webrtc:10736
> Change-Id: I1ff603e65cdd31c7429f36b035dcc00a17b68f3b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143787
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28393}

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

Change-Id: I570c83ec3a88a24d7a1f883a351748dd71bea015
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144022
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28397}

modules/audio_coding/neteq/neteq_unittest.cc

diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index a89d248..6c67ca8 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -458,16 +458,16 @@
       webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
 
   const std::string output_checksum =
-      PlatformChecksum("998be2e5a707e636af0b6298f54bedfabe72aae1",
-                       "61e238ece4cd3b67d66a0b7047e06b20607dcb79", "not used",
-                       "998be2e5a707e636af0b6298f54bedfabe72aae1",
-                       "4116ac2a6e75baac3194b712d6fabe28b384275e");
+      PlatformChecksum("9652cee1d6771a9cbfda821ae1bbdb41b0dd4dee",
+                       "54a7e32f163663c0af35bf70bf45cefc24ad62ef", "not used",
+                       "9652cee1d6771a9cbfda821ae1bbdb41b0dd4dee",
+                       "79496b0a1ef0a3824f3ee04789748a461bed643f");
 
   const std::string network_stats_checksum =
-      PlatformChecksum("3689c9f0ab9e50cefab3e44c37c3d7aa0de82ca4",
-                       "0a596217fccd8d90eff7d1666b8cc63143eeda12", "not used",
-                       "3689c9f0ab9e50cefab3e44c37c3d7aa0de82ca4",
-                       "3689c9f0ab9e50cefab3e44c37c3d7aa0de82ca4");
+      PlatformChecksum("c59b1f9f282b6d8733cdff975e3c150ca4a47d51",
+                       "bca95e565996a4ffd6e2ac15736e08843bdca93b", "not used",
+                       "c59b1f9f282b6d8733cdff975e3c150ca4a47d51",
+                       "c59b1f9f282b6d8733cdff975e3c150ca4a47d51");
 
   DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
                    FLAG_gen_ref);
@@ -486,17 +486,17 @@
   // Checksum depends on libopus being compiled with or without SSE.
   const std::string maybe_sse =
       "14a63b3c7b925c82296be4bafc71bec85f2915c2|"
-      "eb0b68bddcac00fc85403df64f83126f8ea9bc93";
+      "2c05677daa968d6c68b92adf4affb7cd9bb4d363";
   const std::string output_checksum = PlatformChecksum(
-      maybe_sse, "f95f2a220c9ca5d60b81c4653d46e0de2bee159f",
-      "6f288a03d34958f62496f18fa85655593eef4dbe", maybe_sse, maybe_sse);
+      maybe_sse, "b7b7ed802b0e18ee416973bf3b9ae98599b0181d",
+      "5876e52dda90d5ca433c3726555b907b97c86374", maybe_sse, maybe_sse);
 
   const std::string network_stats_checksum =
-      PlatformChecksum("0b3d34baffaf651812ffaf06ea1b5ce45ea1c47a",
-                       "a71dce66c7bea85ba22d4e29a5298f606f810444",
-                       "7c64e1e915bace7c4bf583484efd64eaf234552f",
-                       "0b3d34baffaf651812ffaf06ea1b5ce45ea1c47a",
-                       "0b3d34baffaf651812ffaf06ea1b5ce45ea1c47a");
+      PlatformChecksum("adb3272498e436d1c019cbfd71610e9510c54497",
+                       "fa935a91abc7291db47428a2d7c5361b98713a92",
+                       "42106aa5267300f709f63737707ef07afd9dac61",
+                       "adb3272498e436d1c019cbfd71610e9510c54497",
+                       "adb3272498e436d1c019cbfd71610e9510c54497");
 
   DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
                    FLAG_gen_ref);
@@ -796,7 +796,7 @@
   const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
   const double kNetworkFreezeTimeMs = 5000.0;
   const bool kGetAudioDuringFreezeRecovery = false;
-  const int kDelayToleranceMs = 60;
+  const int kDelayToleranceMs = 50;
   const int kMaxTimeToSpeechMs = 200;
   LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
                         kGetAudioDuringFreezeRecovery, kDelayToleranceMs,