webrtc/modules/audio_processing/agc2/agc2_testing_common_unittest.cc
Alessio Bazzica 980c4601e1 AGC2: retuning and large refactoring
- Bug fix: the desired initial gain quickly dropped to 0 dB hence
  starting a call with a too low level
- New tuning to make AGC2 more robust to VAD mistakes
- Smarter max gain increase speed: to deal with an increased threshold
  of adjacent speech frames, the gain applier temporarily allows a
  faster gain increase to deal with a longer time spent waiting for
  enough speech frames in a row to be observed
- Saturation protector isolated from `AdaptiveModeLevelEstimator` to
  simplify the unit tests for the latter (non bit-exact change)
- AGC2 adaptive digital config: unnecessary params deprecated
- Code readability improvements
- Data dumps clean-up and better naming

Bug: webrtc:7494
Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33736}
2021-04-14 19:01:01 +00:00

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/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/agc2/agc2_testing_common.h"
#include "rtc_base/gunit.h"
namespace webrtc {
TEST(GainController2TestingCommon, LinSpace) {
std::vector<double> points1 = test::LinSpace(-1.0, 2.0, 4);
const std::vector<double> expected_points1{{-1.0, 0.0, 1.0, 2.0}};
EXPECT_EQ(expected_points1, points1);
std::vector<double> points2 = test::LinSpace(0.0, 1.0, 4);
const std::vector<double> expected_points2{{0.0, 1.0 / 3.0, 2.0 / 3.0, 1.0}};
EXPECT_EQ(points2, expected_points2);
}
} // namespace webrtc