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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
93 lines
2.9 KiB
C++
93 lines
2.9 KiB
C++
/*
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* Copyright (c) 2016 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 <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/level_estimator_impl.h"
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#include "modules/audio_processing/test/audio_buffer_tools.h"
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#include "modules/audio_processing/test/bitexactness_tools.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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const int kNumFramesToProcess = 1000;
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// Processes a specified amount of frames, verifies the results and reports
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// any errors.
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void RunBitexactnessTest(int sample_rate_hz,
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size_t num_channels,
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int rms_reference) {
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rtc::CriticalSection crit_capture;
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LevelEstimatorImpl level_estimator(&crit_capture);
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level_estimator.Initialize();
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level_estimator.Enable(true);
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int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
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StreamConfig capture_config(sample_rate_hz, num_channels, false);
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AudioBuffer capture_buffer(
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capture_config.num_frames(), capture_config.num_channels(),
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capture_config.num_frames(), capture_config.num_channels(),
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capture_config.num_frames());
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test::InputAudioFile capture_file(
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test::GetApmCaptureTestVectorFileName(sample_rate_hz));
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std::vector<float> capture_input(samples_per_channel * num_channels);
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for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
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ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
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&capture_file, capture_input);
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test::CopyVectorToAudioBuffer(capture_config, capture_input,
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&capture_buffer);
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level_estimator.ProcessStream(&capture_buffer);
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}
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// Extract test results.
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int rms = level_estimator.RMS();
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// Compare the output to the reference.
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EXPECT_EQ(rms_reference, rms);
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}
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} // namespace
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TEST(LevelEstimatorBitExactnessTest, Mono8kHz) {
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const int kRmsReference = 31;
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RunBitexactnessTest(8000, 1, kRmsReference);
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}
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TEST(LevelEstimatorBitExactnessTest, Mono16kHz) {
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const int kRmsReference = 31;
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RunBitexactnessTest(16000, 1, kRmsReference);
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}
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TEST(LevelEstimatorBitExactnessTest, Mono32kHz) {
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const int kRmsReference = 31;
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RunBitexactnessTest(32000, 1, kRmsReference);
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}
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TEST(LevelEstimatorBitExactnessTest, Mono48kHz) {
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const int kRmsReference = 31;
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RunBitexactnessTest(48000, 1, kRmsReference);
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}
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TEST(LevelEstimatorBitExactnessTest, Stereo16kHz) {
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const int kRmsReference = 30;
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RunBitexactnessTest(16000, 2, kRmsReference);
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}
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} // namespace webrtc
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