
This will fix PRESUBMIT warnings developers will get due to r7014 and r7020. Also some minor style cleanup in: webrtc/modules/audio_coding/main/test/RTPFile.cc webrtc/modules/audio_coding/neteq/test/RTPjitter.cc BUG= R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
121 lines
4.1 KiB
C++
121 lines
4.1 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.
|
|
*/
|
|
|
|
// Unit tests for Normal class.
|
|
|
|
#include "webrtc/modules/audio_coding/neteq/normal.h"
|
|
|
|
#include <vector>
|
|
|
|
#include "testing/gtest/include/gtest/gtest.h"
|
|
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
|
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
|
|
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
|
|
#include "webrtc/modules/audio_coding/neteq/expand.h"
|
|
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
|
|
#include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h"
|
|
#include "webrtc/modules/audio_coding/neteq/random_vector.h"
|
|
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
|
|
|
|
using ::testing::_;
|
|
|
|
namespace webrtc {
|
|
|
|
TEST(Normal, CreateAndDestroy) {
|
|
MockDecoderDatabase db;
|
|
int fs = 8000;
|
|
size_t channels = 1;
|
|
BackgroundNoise bgn(channels);
|
|
SyncBuffer sync_buffer(1, 1000);
|
|
RandomVector random_vector;
|
|
Expand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
|
|
Normal normal(fs, &db, bgn, &expand);
|
|
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
|
|
}
|
|
|
|
TEST(Normal, AvoidDivideByZero) {
|
|
WebRtcSpl_Init();
|
|
MockDecoderDatabase db;
|
|
int fs = 8000;
|
|
size_t channels = 1;
|
|
BackgroundNoise bgn(channels);
|
|
SyncBuffer sync_buffer(1, 1000);
|
|
RandomVector random_vector;
|
|
MockExpand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
|
|
Normal normal(fs, &db, bgn, &expand);
|
|
|
|
int16_t input[1000] = {0};
|
|
scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
|
|
for (size_t i = 0; i < channels; ++i) {
|
|
mute_factor_array[i] = 16384;
|
|
}
|
|
AudioMultiVector output(channels);
|
|
|
|
// Zero input length.
|
|
EXPECT_EQ(
|
|
0,
|
|
normal.Process(input, 0, kModeExpand, mute_factor_array.get(), &output));
|
|
EXPECT_EQ(0u, output.Size());
|
|
|
|
// Try to make energy_length >> scaling = 0;
|
|
EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
|
|
EXPECT_CALL(expand, Process(_));
|
|
EXPECT_CALL(expand, Reset());
|
|
// If input_size_samples < 64, then energy_length in Normal::Process() will
|
|
// be equal to input_size_samples. Since the input is all zeros, decoded_max
|
|
// will be zero, and scaling will be >= 6. Thus, energy_length >> scaling = 0,
|
|
// and using this as a denominator would lead to problems.
|
|
int input_size_samples = 63;
|
|
EXPECT_EQ(input_size_samples,
|
|
normal.Process(input,
|
|
input_size_samples,
|
|
kModeExpand,
|
|
mute_factor_array.get(),
|
|
&output));
|
|
|
|
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
|
|
EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope.
|
|
}
|
|
|
|
TEST(Normal, InputLengthAndChannelsDoNotMatch) {
|
|
WebRtcSpl_Init();
|
|
MockDecoderDatabase db;
|
|
int fs = 8000;
|
|
size_t channels = 2;
|
|
BackgroundNoise bgn(channels);
|
|
SyncBuffer sync_buffer(channels, 1000);
|
|
RandomVector random_vector;
|
|
MockExpand expand(&bgn, &sync_buffer, &random_vector, fs, channels);
|
|
Normal normal(fs, &db, bgn, &expand);
|
|
|
|
int16_t input[1000] = {0};
|
|
scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
|
|
for (size_t i = 0; i < channels; ++i) {
|
|
mute_factor_array[i] = 16384;
|
|
}
|
|
AudioMultiVector output(channels);
|
|
|
|
// Let the number of samples be one sample less than 80 samples per channel.
|
|
size_t input_len = 80 * channels - 1;
|
|
EXPECT_EQ(
|
|
0,
|
|
normal.Process(
|
|
input, input_len, kModeExpand, mute_factor_array.get(), &output));
|
|
EXPECT_EQ(0u, output.Size());
|
|
|
|
EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
|
|
EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope.
|
|
}
|
|
|
|
// TODO(hlundin): Write more tests.
|
|
|
|
} // namespace webrtc
|