bef77e234f
Measures the duration of each packet loss concealment (a.k.a. expand) event that is not followed by a merge operation. Having decoded and played packet m−1, the next expected packet is m. If packet m arrives after some time of packet loss concealment, we have a delayed packet outage event. However, if instead packet n>m arrives, we have a lost packet outage event. In NetEq, the two outage types results in different operations. Both types start with expand operations to generate audio to play while the buffer is empty. When a lost packet outage happens, the expand operation(s) are followed by one merge operation. For delayed packet outages, merge is not done, and the expand operations are immediately followed by normal operations. This change also includes unit tests for the new statistics. BUG=webrtc:4915, chromium:488124 R=minyue@webrtc.org Review URL: https://codereview.webrtc.org/1290113002 . Cr-Commit-Position: refs/heads/master@{#9725}
127 lines
4.3 KiB
C++
127 lines
4.3 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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// Unit tests for Normal class.
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#include "webrtc/modules/audio_coding/neteq/normal.h"
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#include <vector>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
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#include "webrtc/modules/audio_coding/neteq/background_noise.h"
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#include "webrtc/modules/audio_coding/neteq/expand.h"
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#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
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#include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h"
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#include "webrtc/modules/audio_coding/neteq/random_vector.h"
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#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
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#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
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using ::testing::_;
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namespace webrtc {
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TEST(Normal, CreateAndDestroy) {
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MockDecoderDatabase db;
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int fs = 8000;
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size_t channels = 1;
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BackgroundNoise bgn(channels);
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SyncBuffer sync_buffer(1, 1000);
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RandomVector random_vector;
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StatisticsCalculator statistics;
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Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
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Normal normal(fs, &db, bgn, &expand);
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EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
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}
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TEST(Normal, AvoidDivideByZero) {
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WebRtcSpl_Init();
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MockDecoderDatabase db;
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int fs = 8000;
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size_t channels = 1;
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BackgroundNoise bgn(channels);
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SyncBuffer sync_buffer(1, 1000);
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RandomVector random_vector;
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StatisticsCalculator statistics;
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MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
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channels);
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Normal normal(fs, &db, bgn, &expand);
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int16_t input[1000] = {0};
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rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
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for (size_t i = 0; i < channels; ++i) {
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mute_factor_array[i] = 16384;
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}
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AudioMultiVector output(channels);
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// Zero input length.
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EXPECT_EQ(
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0,
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normal.Process(input, 0, kModeExpand, mute_factor_array.get(), &output));
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EXPECT_EQ(0u, output.Size());
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// Try to make energy_length >> scaling = 0;
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EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
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EXPECT_CALL(expand, Process(_));
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EXPECT_CALL(expand, Reset());
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// If input_size_samples < 64, then energy_length in Normal::Process() will
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// be equal to input_size_samples. Since the input is all zeros, decoded_max
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// will be zero, and scaling will be >= 6. Thus, energy_length >> scaling = 0,
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// and using this as a denominator would lead to problems.
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int input_size_samples = 63;
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EXPECT_EQ(input_size_samples,
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normal.Process(input,
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input_size_samples,
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kModeExpand,
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mute_factor_array.get(),
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&output));
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EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
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EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope.
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}
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TEST(Normal, InputLengthAndChannelsDoNotMatch) {
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WebRtcSpl_Init();
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MockDecoderDatabase db;
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int fs = 8000;
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size_t channels = 2;
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BackgroundNoise bgn(channels);
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SyncBuffer sync_buffer(channels, 1000);
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RandomVector random_vector;
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StatisticsCalculator statistics;
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MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
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channels);
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Normal normal(fs, &db, bgn, &expand);
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int16_t input[1000] = {0};
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rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
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for (size_t i = 0; i < channels; ++i) {
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mute_factor_array[i] = 16384;
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}
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AudioMultiVector output(channels);
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// Let the number of samples be one sample less than 80 samples per channel.
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size_t input_len = 80 * channels - 1;
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EXPECT_EQ(
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0,
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normal.Process(
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input, input_len, kModeExpand, mute_factor_array.get(), &output));
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EXPECT_EQ(0u, output.Size());
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EXPECT_CALL(db, Die()); // Called when |db| goes out of scope.
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EXPECT_CALL(expand, Die()); // Called when |expand| goes out of scope.
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}
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// TODO(hlundin): Write more tests.
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} // namespace webrtc
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