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Side by Side Diff: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc

Issue 1696853004: Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include <memory>
12
11 #include "testing/gtest/include/gtest/gtest.h" 13 #include "testing/gtest/include/gtest/gtest.h"
12 #include "webrtc/base/checks.h" 14 #include "webrtc/base/checks.h"
13 #include "webrtc/base/scoped_ptr.h"
14 #include "webrtc/common_types.h" 15 #include "webrtc/common_types.h"
15 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h" 16 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
16 17
17 namespace webrtc { 18 namespace webrtc {
18 19
19 namespace { 20 namespace {
20 const CodecInst kOpusSettings = {105, "opus", 48000, 960, 1, 32000}; 21 const CodecInst kOpusSettings = {105, "opus", 48000, 960, 1, 32000};
21 } // namespace 22 } // namespace
22 23
23 class AudioEncoderOpusTest : public ::testing::Test { 24 class AudioEncoderOpusTest : public ::testing::Test {
24 protected: 25 protected:
25 void CreateCodec(int num_channels) { 26 void CreateCodec(int num_channels) {
26 codec_inst_.channels = num_channels; 27 codec_inst_.channels = num_channels;
27 encoder_.reset(new AudioEncoderOpus(codec_inst_)); 28 encoder_.reset(new AudioEncoderOpus(codec_inst_));
28 auto expected_app = 29 auto expected_app =
29 num_channels == 1 ? AudioEncoderOpus::kVoip : AudioEncoderOpus::kAudio; 30 num_channels == 1 ? AudioEncoderOpus::kVoip : AudioEncoderOpus::kAudio;
30 EXPECT_EQ(expected_app, encoder_->application()); 31 EXPECT_EQ(expected_app, encoder_->application());
31 } 32 }
32 33
33 CodecInst codec_inst_ = kOpusSettings; 34 CodecInst codec_inst_ = kOpusSettings;
34 rtc::scoped_ptr<AudioEncoderOpus> encoder_; 35 std::unique_ptr<AudioEncoderOpus> encoder_;
35 }; 36 };
36 37
37 TEST_F(AudioEncoderOpusTest, DefaultApplicationModeMono) { 38 TEST_F(AudioEncoderOpusTest, DefaultApplicationModeMono) {
38 CreateCodec(1); 39 CreateCodec(1);
39 } 40 }
40 41
41 TEST_F(AudioEncoderOpusTest, DefaultApplicationModeStereo) { 42 TEST_F(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
42 CreateCodec(2); 43 CreateCodec(2);
43 } 44 }
44 45
(...skipping 98 matching lines...) Expand 10 before | Expand all | Expand 10 after
143 144
144 TestSetPacketLossRate(encoder_.get(), I(1.00 , 0.18 + eps), 0.20); 145 TestSetPacketLossRate(encoder_.get(), I(1.00 , 0.18 + eps), 0.20);
145 TestSetPacketLossRate(encoder_.get(), I(0.18 - eps, 0.09 + eps), 0.10); 146 TestSetPacketLossRate(encoder_.get(), I(0.18 - eps, 0.09 + eps), 0.10);
146 TestSetPacketLossRate(encoder_.get(), I(0.09 - eps, 0.04 + eps), 0.05); 147 TestSetPacketLossRate(encoder_.get(), I(0.09 - eps, 0.04 + eps), 0.05);
147 TestSetPacketLossRate(encoder_.get(), I(0.04 - eps, 0.01 + eps), 0.01); 148 TestSetPacketLossRate(encoder_.get(), I(0.04 - eps, 0.01 + eps), 0.01);
148 TestSetPacketLossRate(encoder_.get(), I(0.01 - eps, 0.00 ), 0.00); 149 TestSetPacketLossRate(encoder_.get(), I(0.01 - eps, 0.00 ), 0.00);
149 // clang-format on 150 // clang-format on
150 } 151 }
151 152
152 } // namespace webrtc 153 } // namespace webrtc
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