commit 74e72c8c9b4278ee507c3651252c92f6c4a5a66b
author Per Åhgren <peah@webrtc.org> Wed Oct 25 21:14:44 2017
committer Commit Bot <commit-bot@chromium.org> Wed Oct 25 21:56:30 2017
tree 20714c1db05b4ad83871348cafcdc0eced57ae08
parent 173fd91b56f5655890e8a6502f909d217e3a9c03

Lowering the threshold for delay change detection in AEC3

This CL lowers the threshold for delay change detection in AEC3.
This makes the delay decisions more stable.

TBR=gustaf@webrtc.org

Bug: chromium:778396,webrtc:8451
Change-Id: I8b015455399d696172b7c0beb033caf508f426e9
Reviewed-on: https://webrtc-review.googlesource.com/15541
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20433}

modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc

diff --git a/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc b/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
index 2c7b9a5..9ff45cd 100644
--- a/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
@@ -62,7 +62,7 @@
     EchoPathDelayEstimator estimator(&data_dumper, EchoCanceller3Config());
 
     rtc::Optional<size_t> estimated_delay_samples;
-    for (size_t k = 0; k < (100 + delay_samples / kBlockSize); ++k) {
+    for (size_t k = 0; k < (150 + delay_samples / kBlockSize); ++k) {
       RandomizeSampleVector(&random_generator, render[0]);
       signal_delay_buffer.Delay(render[0], capture);
       render_delay_buffer->Insert(render);

modules/audio_processing/aec3/matched_filter.cc

diff --git a/modules/audio_processing/aec3/matched_filter.cc b/modules/audio_processing/aec3/matched_filter.cc
index c70e1a9..901281f 100644
--- a/modules/audio_processing/aec3/matched_filter.cc
+++ b/modules/audio_processing/aec3/matched_filter.cc
@@ -368,7 +368,7 @@
             [](float a, float b) -> bool { return a * a < b * b; }));
 
     // Update the lag estimates for the matched filter.
-    const float kMatchingFilterThreshold = 0.1f;
+    const float kMatchingFilterThreshold = 0.05f;
     lag_estimates_[n] = LagEstimate(
         error_sum_anchor - error_sum,
         (lag_estimate > 2 && lag_estimate < (filters_[n].size() - 10) &&

modules/audio_processing/aec3/matched_filter_unittest.cc

diff --git a/modules/audio_processing/aec3/matched_filter_unittest.cc b/modules/audio_processing/aec3/matched_filter_unittest.cc
index d06f010..deaad11 100644
--- a/modules/audio_processing/aec3/matched_filter_unittest.cc
+++ b/modules/audio_processing/aec3/matched_filter_unittest.cc
@@ -142,7 +142,7 @@
         RenderDelayBuffer::Create(3));
 
     // Analyze the correlation between render and capture.
-    for (size_t k = 0; k < (100 + delay_samples / kSubBlockSize); ++k) {
+    for (size_t k = 0; k < (150 + delay_samples / kSubBlockSize); ++k) {
       RandomizeSampleVector(&random_generator, render[0]);
       signal_delay_buffer.Delay(render[0], capture);
       render_delay_buffer->Insert(render);