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1 /* | |
2 * Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ | |
12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ | |
13 | |
14 #include <stdio.h> | |
15 #include <string.h> | |
16 | |
17 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" | |
18 #include "webrtc/modules/audio_coding/main/test/ACMTest.h" | |
19 #include "webrtc/modules/audio_coding/main/test/PCMFile.h" | |
20 #include "webrtc/modules/audio_coding/main/test/RTPFile.h" | |
21 #include "webrtc/typedefs.h" | |
22 | |
23 namespace webrtc { | |
24 | |
25 #define MAX_INCOMING_PAYLOAD 8096 | |
26 | |
27 // TestPacketization callback which writes the encoded payloads to file | |
28 class TestPacketization : public AudioPacketizationCallback { | |
29 public: | |
30 TestPacketization(RTPStream *rtpStream, uint16_t frequency); | |
31 ~TestPacketization(); | |
32 int32_t SendData(const FrameType frameType, | |
33 const uint8_t payloadType, | |
34 const uint32_t timeStamp, | |
35 const uint8_t* payloadData, | |
36 const size_t payloadSize, | |
37 const RTPFragmentationHeader* fragmentation) override; | |
38 | |
39 private: | |
40 static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, | |
41 int16_t seqNo, uint32_t timeStamp, uint32_t ssrc); | |
42 RTPStream* _rtpStream; | |
43 int32_t _frequency; | |
44 int16_t _seqNo; | |
45 }; | |
46 | |
47 class Sender { | |
48 public: | |
49 Sender(); | |
50 void Setup(AudioCodingModule *acm, RTPStream *rtpStream, | |
51 std::string in_file_name, int sample_rate, int channels); | |
52 void Teardown(); | |
53 void Run(); | |
54 bool Add10MsData(); | |
55 | |
56 //for auto_test and logging | |
57 uint8_t testMode; | |
58 uint8_t codeId; | |
59 | |
60 protected: | |
61 AudioCodingModule* _acm; | |
62 | |
63 private: | |
64 PCMFile _pcmFile; | |
65 AudioFrame _audioFrame; | |
66 TestPacketization* _packetization; | |
67 }; | |
68 | |
69 class Receiver { | |
70 public: | |
71 Receiver(); | |
72 virtual ~Receiver() {}; | |
73 void Setup(AudioCodingModule *acm, RTPStream *rtpStream, | |
74 std::string out_file_name, int channels); | |
75 void Teardown(); | |
76 void Run(); | |
77 virtual bool IncomingPacket(); | |
78 bool PlayoutData(); | |
79 | |
80 //for auto_test and logging | |
81 uint8_t codeId; | |
82 uint8_t testMode; | |
83 | |
84 private: | |
85 PCMFile _pcmFile; | |
86 int16_t* _playoutBuffer; | |
87 uint16_t _playoutLengthSmpls; | |
88 int32_t _frequency; | |
89 bool _firstTime; | |
90 | |
91 protected: | |
92 AudioCodingModule* _acm; | |
93 uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD]; | |
94 RTPStream* _rtpStream; | |
95 WebRtcRTPHeader _rtpInfo; | |
96 size_t _realPayloadSizeBytes; | |
97 size_t _payloadSizeBytes; | |
98 uint32_t _nextTime; | |
99 }; | |
100 | |
101 class EncodeDecodeTest : public ACMTest { | |
102 public: | |
103 EncodeDecodeTest(); | |
104 explicit EncodeDecodeTest(int testMode); | |
105 void Perform() override; | |
106 | |
107 uint16_t _playoutFreq; | |
108 uint8_t _testMode; | |
109 | |
110 private: | |
111 std::string EncodeToFile(int fileType, | |
112 int codeId, | |
113 int* codePars, | |
114 int testMode); | |
115 | |
116 protected: | |
117 Sender _sender; | |
118 Receiver _receiver; | |
119 }; | |
120 | |
121 } // namespace webrtc | |
122 | |
123 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ | |
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