webrtc_m130/webrtc/common_audio/vad/vad_sp_unittest.cc
oprypin 67fdb80837 Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ )
Reason for revert:
Can reland it if backwards compatible API is kept.

Original issue's description:
> Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ )
>
> Reason for revert:
> The API change in audio/utility/audio_frame_operations.h caused breakage. Need to keep backward-compatible API.
>
> Original issue's description:
> > Enable cpplint and fix cpplint errors in webrtc/*audio
> >
> > Change usage accordingly throughout the codebase
> >
> > BUG=webrtc:5268
> >
> > TESTED=Fixed issues reported by:
> > find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py
> >
> > Review-Url: https://codereview.webrtc.org/2683033004
> > Cr-Commit-Position: refs/heads/master@{#17133}
> > Committed: aebe55ca6c
>
> TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5268
>
> Review-Url: https://codereview.webrtc.org/2739143002
> Cr-Commit-Position: refs/heads/master@{#17138}
> Committed: e47c1d3ca1

TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
BUG=webrtc:5268

Review-Url: https://codereview.webrtc.org/2739073003
Cr-Commit-Position: refs/heads/master@{#17144}
2017-03-09 14:25:06 +00:00

77 lines
2.6 KiB
C++

/*
* 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.
*/
#include <stdlib.h>
#include "webrtc/common_audio/vad/vad_unittest.h"
#include "webrtc/test/gtest.h"
#include "webrtc/typedefs.h"
extern "C" {
#include "webrtc/common_audio/vad/vad_core.h"
#include "webrtc/common_audio/vad/vad_sp.h"
}
namespace webrtc {
namespace test {
TEST_F(VadTest, vad_sp) {
VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
const size_t kMaxFrameLenSp = 960; // Maximum frame length in this unittest.
int16_t zeros[kMaxFrameLenSp] = { 0 };
int32_t state[2] = { 0 };
int16_t data_in[kMaxFrameLenSp];
int16_t data_out[kMaxFrameLenSp];
// We expect the first value to be 1600 as long as |frame_counter| is zero,
// which is true for the first iteration.
static const int16_t kReferenceMin[32] = {
1600, 720, 509, 512, 532, 552, 570, 588,
606, 624, 642, 659, 675, 691, 707, 723,
1600, 544, 502, 522, 542, 561, 579, 597,
615, 633, 651, 667, 683, 699, 715, 731
};
// Construct a speech signal that will trigger the VAD in all modes. It is
// known that (i * i) will wrap around, but that doesn't matter in this case.
for (size_t i = 0; i < kMaxFrameLenSp; ++i) {
data_in[i] = static_cast<int16_t>(i * i);
}
// Input values all zeros, expect all zeros out.
WebRtcVad_Downsampling(zeros, data_out, state, kMaxFrameLenSp);
EXPECT_EQ(0, state[0]);
EXPECT_EQ(0, state[1]);
for (size_t i = 0; i < kMaxFrameLenSp / 2; ++i) {
EXPECT_EQ(0, data_out[i]);
}
// Make a simple non-zero data test.
WebRtcVad_Downsampling(data_in, data_out, state, kMaxFrameLenSp);
EXPECT_EQ(207, state[0]);
EXPECT_EQ(2270, state[1]);
ASSERT_EQ(0, WebRtcVad_InitCore(self));
// TODO(bjornv): Replace this part of the test with taking values from an
// array and calculate the reference value here. Make sure the values are not
// ordered.
for (int16_t i = 0; i < 16; ++i) {
int16_t value = 500 * (i + 1);
for (int j = 0; j < kNumChannels; ++j) {
// Use values both above and below initialized value.
EXPECT_EQ(kReferenceMin[i], WebRtcVad_FindMinimum(self, value, j));
EXPECT_EQ(kReferenceMin[i + 16], WebRtcVad_FindMinimum(self, 12000, j));
}
self->frame_counter++;
}
free(self);
}
} // namespace test
} // namespace webrtc