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1 /*
2  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
3  *
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
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
12 #define MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
13 
14 #include <memory>
15 #include <set>
16 #include <string>
17 
18 #include "api/audio/audio_frame.h"
19 #include "api/neteq/neteq.h"
20 #include "api/rtp_headers.h"
21 #include "modules/audio_coding/neteq/tools/packet.h"
22 #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
23 #include "system_wrappers/include/clock.h"
24 #include "test/gtest.h"
25 
26 namespace webrtc {
27 
28 class NetEqDecodingTest : public ::testing::Test {
29  protected:
30   // NetEQ must be polled for data once every 10 ms.
31   // Thus, none of the constants below can be changed.
32   static constexpr int kTimeStepMs = 10;
33   static constexpr size_t kBlockSize8kHz = kTimeStepMs * 8;
34   static constexpr size_t kBlockSize16kHz = kTimeStepMs * 16;
35   static constexpr size_t kBlockSize32kHz = kTimeStepMs * 32;
36   static constexpr size_t kBlockSize48kHz = kTimeStepMs * 48;
37   static constexpr int kInitSampleRateHz = 8000;
38 
39   NetEqDecodingTest();
40   virtual void SetUp();
41   virtual void TearDown();
42   void OpenInputFile(const std::string& rtp_file);
43   void Process();
44 
45   void DecodeAndCompare(const std::string& rtp_file,
46                         const std::string& output_checksum,
47                         const std::string& network_stats_checksum,
48                         bool gen_ref);
49 
50   static void PopulateRtpInfo(int frame_index,
51                               int timestamp,
52                               RTPHeader* rtp_info);
53   static void PopulateCng(int frame_index,
54                           int timestamp,
55                           RTPHeader* rtp_info,
56                           uint8_t* payload,
57                           size_t* payload_len);
58 
59   void WrapTest(uint16_t start_seq_no,
60                 uint32_t start_timestamp,
61                 const std::set<uint16_t>& drop_seq_numbers,
62                 bool expect_seq_no_wrap,
63                 bool expect_timestamp_wrap);
64 
65   void LongCngWithClockDrift(double drift_factor,
66                              double network_freeze_ms,
67                              bool pull_audio_during_freeze,
68                              int delay_tolerance_ms,
69                              int max_time_to_speech_ms);
70 
71   SimulatedClock clock_;
72   std::unique_ptr<NetEq> neteq_;
73   NetEq::Config config_;
74   std::unique_ptr<test::RtpFileSource> rtp_source_;
75   std::unique_ptr<test::Packet> packet_;
76   AudioFrame out_frame_;
77   int output_sample_rate_;
78   int algorithmic_delay_ms_;
79 };
80 
81 class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
82  public:
NetEqDecodingTestTwoInstances()83   NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
84 
85   void SetUp() override;
86 
87   void CreateSecondInstance();
88 
89  protected:
90   std::unique_ptr<NetEq> neteq2_;
91   NetEq::Config config2_;
92 };
93 
94 }  // namespace webrtc
95 #endif  // MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
96