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093824c4d2
This simplifies the following steps: * FrameCombiner infers the sample rate from channel size * Sends the inferred sample rate to FixedDigitalLevelEstimator and Limiter. * Those classes then convert the sample rate to channel size. Along the way perform checks that the derived channel size value is a legal value (which has already been done by FrameCombiner). To: * FrameCombiner sends channel size to FixedDigitalLevelEstimator and Limiter. Bug: chromium:335805780 Change-Id: I6d2953ba5ee99771f3ff5bf4f4a049a8a29b5577 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352581 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42480}
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/limiter.h"
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#include "api/audio/audio_frame.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(Limiter, LimiterShouldConstructAndRun) {
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const size_t samples_per_channel = SampleRateToDefaultChannelSize(48000);
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ApmDataDumper apm_data_dumper(0);
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Limiter limiter(&apm_data_dumper, samples_per_channel, "");
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VectorFloatFrame vectors_with_float_frame(1, samples_per_channel,
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kMaxAbsFloatS16Value);
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limiter.Process(vectors_with_float_frame.float_frame_view());
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}
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TEST(Limiter, OutputVolumeAboveThreshold) {
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const size_t samples_per_channel = SampleRateToDefaultChannelSize(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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Limiter limiter(&apm_data_dumper, samples_per_channel, "");
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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, samples_per_channel,
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input_level);
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limiter.Process(vectors_with_float_frame.float_frame_view());
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}
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VectorFloatFrame vectors_with_float_frame(1, samples_per_channel,
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input_level);
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limiter.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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