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Side by Side Diff: webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc

Issue 2439473002: Delete DataLog abstraction, which was almost unused. (Closed)
Patch Set: Created 4 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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
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25 #include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h" 25 #include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
26 #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isac fix.h" 26 #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isac fix.h"
27 #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isac fix.h" 27 #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isac fix.h"
28 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isa c.h" 28 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isa c.h"
29 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isa c.h" 29 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isa c.h"
30 #include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h" 30 #include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
31 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h" 31 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
32 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h" 32 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
33 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" 33 #include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
34 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h" 34 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
35 #include "webrtc/system_wrappers/include/data_log.h"
36 #include "webrtc/test/gtest.h" 35 #include "webrtc/test/gtest.h"
37 #include "webrtc/test/testsupport/fileutils.h" 36 #include "webrtc/test/testsupport/fileutils.h"
38 37
39 namespace webrtc { 38 namespace webrtc {
40 39
41 namespace { 40 namespace {
42 // The absolute difference between the input and output (the first channel) is 41 // The absolute difference between the input and output (the first channel) is
43 // compared vs |tolerance|. The parameter |delay| is used to correct for codec 42 // compared vs |tolerance|. The parameter |delay| is used to correct for codec
44 // delays. 43 // delays.
45 void CompareInputOutput(const std::vector<int16_t>& input, 44 void CompareInputOutput(const std::vector<int16_t>& input,
46 const std::vector<int16_t>& output, 45 const std::vector<int16_t>& output,
47 size_t num_samples, 46 size_t num_samples,
48 size_t channels, 47 size_t channels,
49 int tolerance, 48 int tolerance,
50 int delay) { 49 int delay) {
51 ASSERT_LE(num_samples, input.size()); 50 ASSERT_LE(num_samples, input.size());
52 ASSERT_LE(num_samples * channels, output.size()); 51 ASSERT_LE(num_samples * channels, output.size());
53 for (unsigned int n = 0; n < num_samples - delay; ++n) { 52 for (unsigned int n = 0; n < num_samples - delay; ++n) {
54 ASSERT_NEAR(input[n], output[channels * n + delay], tolerance) 53 ASSERT_NEAR(input[n], output[channels * n + delay], tolerance)
55 << "Exit test on first diff; n = " << n; 54 << "Exit test on first diff; n = " << n;
56 DataLog::InsertCell("CodecTest", "input", input[n]);
57 DataLog::InsertCell("CodecTest", "output", output[channels * n]);
58 DataLog::NextRow("CodecTest");
59 } 55 }
60 } 56 }
61 57
62 // The absolute difference between the first two channels in |output| is 58 // The absolute difference between the first two channels in |output| is
63 // compared vs |tolerance|. 59 // compared vs |tolerance|.
64 void CompareTwoChannels(const std::vector<int16_t>& output, 60 void CompareTwoChannels(const std::vector<int16_t>& output,
65 size_t samples_per_channel, 61 size_t samples_per_channel,
66 size_t channels, 62 size_t channels,
67 int tolerance) { 63 int tolerance) {
68 ASSERT_GE(channels, 2u); 64 ASSERT_GE(channels, 2u);
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106 payload_type_(17), 102 payload_type_(17),
107 decoder_(NULL) {} 103 decoder_(NULL) {}
108 104
109 virtual ~AudioDecoderTest() {} 105 virtual ~AudioDecoderTest() {}
110 106
111 virtual void SetUp() { 107 virtual void SetUp() {
112 if (audio_encoder_) 108 if (audio_encoder_)
113 codec_input_rate_hz_ = audio_encoder_->SampleRateHz(); 109 codec_input_rate_hz_ = audio_encoder_->SampleRateHz();
114 // Create arrays. 110 // Create arrays.
115 ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0"; 111 ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0";
116 // Logging to view input and output in Matlab.
117 // Use 'gyp -Denable_data_logging=1' to enable logging.
118 DataLog::CreateLog();
119 DataLog::AddTable("CodecTest");
120 DataLog::AddColumn("CodecTest", "input", 1);
121 DataLog::AddColumn("CodecTest", "output", 1);
122 } 112 }
123 113
124 virtual void TearDown() { 114 virtual void TearDown() {
125 delete decoder_; 115 delete decoder_;
126 decoder_ = NULL; 116 decoder_ = NULL;
127 // Close log.
128 DataLog::ReturnLog();
129 } 117 }
130 118
131 virtual void InitEncoder() { } 119 virtual void InitEncoder() { }
132 120
133 // TODO(henrik.lundin) Change return type to size_t once most/all overriding 121 // TODO(henrik.lundin) Change return type to size_t once most/all overriding
134 // implementations are gone. 122 // implementations are gone.
135 virtual int EncodeFrame(const int16_t* input, 123 virtual int EncodeFrame(const int16_t* input,
136 size_t input_len_samples, 124 size_t input_len_samples,
137 rtc::Buffer* output) { 125 rtc::Buffer* output) {
138 AudioEncoder::EncodedInfo encoded_info; 126 AudioEncoder::EncodedInfo encoded_info;
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698 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGnb)); 686 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGnb));
699 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGwb)); 687 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGwb));
700 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGswb32kHz)); 688 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGswb32kHz));
701 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGswb48kHz)); 689 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGswb48kHz));
702 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderArbitrary)); 690 EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderArbitrary));
703 EXPECT_EQ(has_opus, CodecSupported(NetEqDecoder::kDecoderOpus)); 691 EXPECT_EQ(has_opus, CodecSupported(NetEqDecoder::kDecoderOpus));
704 EXPECT_EQ(has_opus, CodecSupported(NetEqDecoder::kDecoderOpus_2ch)); 692 EXPECT_EQ(has_opus, CodecSupported(NetEqDecoder::kDecoderOpus_2ch));
705 } 693 }
706 694
707 } // namespace webrtc 695 } // namespace webrtc
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