mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
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++
/*
|
|
* Copyright (c) 2016 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 <vector>
|
|
|
|
#include "api/array_view.h"
|
|
#include "modules/audio_processing/audio_buffer.h"
|
|
#include "modules/audio_processing/level_estimator_impl.h"
|
|
#include "modules/audio_processing/test/audio_buffer_tools.h"
|
|
#include "modules/audio_processing/test/bitexactness_tools.h"
|
|
#include "test/gtest.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
const int kNumFramesToProcess = 1000;
|
|
|
|
// Processes a specified amount of frames, verifies the results and reports
|
|
// any errors.
|
|
void RunBitexactnessTest(int sample_rate_hz,
|
|
size_t num_channels,
|
|
int rms_reference) {
|
|
rtc::CriticalSection crit_capture;
|
|
LevelEstimatorImpl level_estimator(&crit_capture);
|
|
level_estimator.Initialize();
|
|
level_estimator.Enable(true);
|
|
|
|
int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
|
|
StreamConfig capture_config(sample_rate_hz, num_channels, false);
|
|
AudioBuffer capture_buffer(
|
|
capture_config.num_frames(), capture_config.num_channels(),
|
|
capture_config.num_frames(), capture_config.num_channels(),
|
|
capture_config.num_frames());
|
|
|
|
test::InputAudioFile capture_file(
|
|
test::GetApmCaptureTestVectorFileName(sample_rate_hz));
|
|
std::vector<float> capture_input(samples_per_channel * num_channels);
|
|
for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
|
|
ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
|
|
&capture_file, capture_input);
|
|
|
|
test::CopyVectorToAudioBuffer(capture_config, capture_input,
|
|
&capture_buffer);
|
|
|
|
level_estimator.ProcessStream(&capture_buffer);
|
|
}
|
|
|
|
// Extract test results.
|
|
int rms = level_estimator.RMS();
|
|
|
|
// Compare the output to the reference.
|
|
EXPECT_EQ(rms_reference, rms);
|
|
}
|
|
|
|
} // namespace
|
|
|
|
TEST(LevelEstimatorBitExactnessTest, Mono8kHz) {
|
|
const int kRmsReference = 31;
|
|
|
|
RunBitexactnessTest(8000, 1, kRmsReference);
|
|
}
|
|
|
|
TEST(LevelEstimatorBitExactnessTest, Mono16kHz) {
|
|
const int kRmsReference = 31;
|
|
|
|
RunBitexactnessTest(16000, 1, kRmsReference);
|
|
}
|
|
|
|
TEST(LevelEstimatorBitExactnessTest, Mono32kHz) {
|
|
const int kRmsReference = 31;
|
|
|
|
RunBitexactnessTest(32000, 1, kRmsReference);
|
|
}
|
|
|
|
TEST(LevelEstimatorBitExactnessTest, Mono48kHz) {
|
|
const int kRmsReference = 31;
|
|
|
|
RunBitexactnessTest(48000, 1, kRmsReference);
|
|
}
|
|
|
|
TEST(LevelEstimatorBitExactnessTest, Stereo16kHz) {
|
|
const int kRmsReference = 30;
|
|
|
|
RunBitexactnessTest(16000, 2, kRmsReference);
|
|
}
|
|
|
|
} // namespace webrtc
|