507e8d1f71
The webrtc::AudioMixer uses a limiter component. This CL allows changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial. The AGC2 limiter has a float interface. We plan to eventually switch to the AGC2 limiter. Therefore, we will now mix in de-interleaved floats. Float mixing will happen both when using the old limiter and when using the new one. After this CL the mixer will support two limiters. The limiters have different interfaces and need different processing steps. Because of that, we make (rather big) changes to the control flow in FrameCombiner. For a short while, we will mix in deinterleaved floats when using any limiter. Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/ Reverted in https://webrtc-review.googlesource.com/c/src/+/57940 because of both breaking compilation and having a severe error. The error is fixed and a test is added. The compilation issue is fixed. Bug: webrtc:8925 Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0 Reviewed-on: https://webrtc-review.googlesource.com/58085 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22207}
61 lines
2.2 KiB
C++
61 lines
2.2 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/agc2/gain_curve_applier.h"
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#include "common_audio/include/audio_util.h"
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "modules/audio_processing/agc2/agc2_testing_common.h"
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#include "modules/audio_processing/agc2/vector_float_frame.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/gunit.h"
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namespace webrtc {
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TEST(GainCurveApplier, GainCurveApplierShouldConstructAndRun) {
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const int sample_rate_hz = 48000;
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ApmDataDumper apm_data_dumper(0);
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GainCurveApplier gain_curve_applier(sample_rate_hz, &apm_data_dumper);
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VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
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kMaxAbsFloatS16Value);
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gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
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}
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TEST(GainCurveApplier, OutputVolumeAboveThreshold) {
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const int sample_rate_hz = 48000;
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const float input_level =
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(kMaxAbsFloatS16Value + DbfsToFloatS16(test::kLimiterMaxInputLevelDbFs)) /
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2.f;
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ApmDataDumper apm_data_dumper(0);
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GainCurveApplier gain_curve_applier(sample_rate_hz, &apm_data_dumper);
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// Give the level estimator time to adapt.
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for (int i = 0; i < 5; ++i) {
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VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
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input_level);
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gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
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}
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VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
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input_level);
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gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
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rtc::ArrayView<const float> channel =
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vectors_with_float_frame.float_frame_view().channel(0);
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for (const auto& sample : channel) {
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EXPECT_LT(0.9f * kMaxAbsFloatS16Value, sample);
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}
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}
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} // namespace webrtc
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