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author | aluebs <aluebs@webrtc.org> | 2015-06-24 18:45:58 -0700 |
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committer | Commit bot <commit-bot@chromium.org> | 2015-06-25 01:46:03 +0000 |
commit | 518c683f3e413523a458a94b533274bd7f29992d (patch) | |
tree | 63d53981b3181659cc9344491af65722514214b8 /webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc | |
parent | ac4234ccfc964208163d6152977cc1d015448bbb (diff) | |
download | webrtc-518c683f3e413523a458a94b533274bd7f29992d.tar.gz |
Pull the Voice Activity Detector out from the AGC
This change generates bit-exact values when running through audioproc_f than before.
This change was originally uploaded here:
* https://codereview.webrtc.org/1181933002/
* https://codereview.webrtc.org/1177043017/
And reverted because of an ASAN problem in Chrome here:
* https://codereview.webrtc.org/1192863006/
* https://codereview.webrtc.org/1194963003/
TBR=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1211563003
Cr-Commit-Position: refs/heads/master@{#9502}
Diffstat (limited to 'webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc')
-rw-r--r-- | webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc | 63 |
1 files changed, 63 insertions, 0 deletions
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc new file mode 100644 index 0000000000..675af70b45 --- /dev/null +++ b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc @@ -0,0 +1,63 @@ +/* + * Copyright (c) 2012 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. + */ + +// We don't test the value of pitch gain and lags as they are created by iSAC +// routines. However, interpolation of pitch-gain and lags is in a separate +// class and has its own unit-test. + +#include "webrtc/modules/audio_processing/vad/vad_audio_proc.h" + +#include <math.h> +#include <stdio.h> + +#include <string> + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/audio_processing/vad/common.h" +#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/test/testsupport/fileutils.h" + +namespace webrtc { + +TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) { + VadAudioProc audioproc; + + std::string peak_file_name = + test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat"); + FILE* peak_file = fopen(peak_file_name.c_str(), "rb"); + ASSERT_TRUE(peak_file != NULL); + + std::string pcm_file_name = + test::ResourcePath("audio_processing/agc/agc_audio", "pcm"); + FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb"); + ASSERT_TRUE(pcm_file != NULL); + + // Read 10 ms audio in each iteration. + const size_t kDataLength = kLength10Ms; + int16_t data[kDataLength] = {0}; + AudioFeatures features; + double sp[kMaxNumFrames]; + while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) { + audioproc.ExtractFeatures(data, kDataLength, &features); + if (features.num_frames > 0) { + ASSERT_LT(features.num_frames, kMaxNumFrames); + // Read reference values. + const size_t num_frames = features.num_frames; + ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file)); + for (int n = 0; n < features.num_frames; n++) + EXPECT_NEAR(features.spectral_peak[n], sp[n], 3); + } + } + + fclose(peak_file); + fclose(pcm_file); +} + +} // namespace webrtc |