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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}
63 lines
2.1 KiB
C++
63 lines
2.1 KiB
C++
/*
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* Copyright (c) 2012 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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// We don't test the value of pitch gain and lags as they are created by iSAC
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// routines. However, interpolation of pitch-gain and lags is in a separate
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// class and has its own unit-test.
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#include "modules/audio_processing/vad/vad_audio_proc.h"
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#include <math.h>
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#include <stdio.h>
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#include <string>
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#include "modules/audio_processing/vad/common.h"
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#include "modules/include/module_common_types.h"
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#include "test/gtest.h"
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#include "test/testsupport/fileutils.h"
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namespace webrtc {
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TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) {
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VadAudioProc audioproc;
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std::string peak_file_name =
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test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat");
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FILE* peak_file = fopen(peak_file_name.c_str(), "rb");
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ASSERT_TRUE(peak_file != NULL);
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std::string pcm_file_name =
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test::ResourcePath("audio_processing/agc/agc_audio", "pcm");
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FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb");
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ASSERT_TRUE(pcm_file != NULL);
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// Read 10 ms audio in each iteration.
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const size_t kDataLength = kLength10Ms;
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int16_t data[kDataLength] = {0};
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AudioFeatures features;
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double sp[kMaxNumFrames];
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while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) {
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audioproc.ExtractFeatures(data, kDataLength, &features);
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if (features.num_frames > 0) {
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ASSERT_LT(features.num_frames, kMaxNumFrames);
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// Read reference values.
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const size_t num_frames = features.num_frames;
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ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file));
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for (size_t n = 0; n < features.num_frames; n++)
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EXPECT_NEAR(features.spectral_peak[n], sp[n], 3);
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}
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}
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fclose(peak_file);
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fclose(pcm_file);
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}
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} // namespace webrtc
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