xref: /aosp_15_r20/external/webrtc/modules/audio_coding/test/TwoWayCommunication.cc (revision d9f758449e529ab9291ac668be2861e7a55c2422)
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 #include "TwoWayCommunication.h"
12 
13 #include <stdio.h>
14 #include <string.h>
15 
16 #include <memory>
17 
18 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
19 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
20 #include "modules/audio_coding/test/PCMFile.h"
21 #include "test/gtest.h"
22 #include "test/testsupport/file_utils.h"
23 
24 namespace webrtc {
25 
26 #define MAX_FILE_NAME_LENGTH_BYTE 500
27 
TwoWayCommunication()28 TwoWayCommunication::TwoWayCommunication()
29     : _acmA(AudioCodingModule::Create(
30           AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
31       _acmRefA(AudioCodingModule::Create(
32           AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {
33   AudioCodingModule::Config config;
34   // The clicks will be more obvious if time-stretching is not allowed.
35   // TODO(henrik.lundin) Really?
36   config.neteq_config.for_test_no_time_stretching = true;
37   config.decoder_factory = CreateBuiltinAudioDecoderFactory();
38   _acmB.reset(AudioCodingModule::Create(config));
39   _acmRefB.reset(AudioCodingModule::Create(config));
40 }
41 
~TwoWayCommunication()42 TwoWayCommunication::~TwoWayCommunication() {
43   delete _channel_A2B;
44   delete _channel_B2A;
45   delete _channelRef_A2B;
46   delete _channelRef_B2A;
47   _inFileA.Close();
48   _inFileB.Close();
49   _outFileA.Close();
50   _outFileB.Close();
51   _outFileRefA.Close();
52   _outFileRefB.Close();
53 }
54 
SetUpAutotest(AudioEncoderFactory * const encoder_factory,const SdpAudioFormat & format1,const int payload_type1,const SdpAudioFormat & format2,const int payload_type2)55 void TwoWayCommunication::SetUpAutotest(
56     AudioEncoderFactory* const encoder_factory,
57     const SdpAudioFormat& format1,
58     const int payload_type1,
59     const SdpAudioFormat& format2,
60     const int payload_type2) {
61   //--- Set A codecs
62   _acmA->SetEncoder(
63       encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
64   _acmA->SetReceiveCodecs({{payload_type2, format2}});
65 
66   //--- Set ref-A codecs
67   _acmRefA->SetEncoder(
68       encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
69   _acmRefA->SetReceiveCodecs({{payload_type2, format2}});
70 
71   //--- Set B codecs
72   _acmB->SetEncoder(
73       encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
74   _acmB->SetReceiveCodecs({{payload_type1, format1}});
75 
76   //--- Set ref-B codecs
77   _acmRefB->SetEncoder(
78       encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
79   _acmRefB->SetReceiveCodecs({{payload_type1, format1}});
80 
81   uint16_t frequencyHz;
82 
83   //--- Input A and B
84   std::string in_file_name =
85       webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
86   frequencyHz = 16000;
87   _inFileA.Open(in_file_name, frequencyHz, "rb");
88   _inFileB.Open(in_file_name, frequencyHz, "rb");
89 
90   //--- Output A
91   std::string output_file_a = webrtc::test::OutputPath() + "outAutotestA.pcm";
92   frequencyHz = 16000;
93   _outFileA.Open(output_file_a, frequencyHz, "wb");
94   std::string output_ref_file_a =
95       webrtc::test::OutputPath() + "ref_outAutotestA.pcm";
96   _outFileRefA.Open(output_ref_file_a, frequencyHz, "wb");
97 
98   //--- Output B
99   std::string output_file_b = webrtc::test::OutputPath() + "outAutotestB.pcm";
100   frequencyHz = 16000;
101   _outFileB.Open(output_file_b, frequencyHz, "wb");
102   std::string output_ref_file_b =
103       webrtc::test::OutputPath() + "ref_outAutotestB.pcm";
104   _outFileRefB.Open(output_ref_file_b, frequencyHz, "wb");
105 
106   //--- Set A-to-B channel
107   _channel_A2B = new Channel;
108   _acmA->RegisterTransportCallback(_channel_A2B);
109   _channel_A2B->RegisterReceiverACM(_acmB.get());
110   //--- Do the same for the reference
111   _channelRef_A2B = new Channel;
112   _acmRefA->RegisterTransportCallback(_channelRef_A2B);
113   _channelRef_A2B->RegisterReceiverACM(_acmRefB.get());
114 
115   //--- Set B-to-A channel
116   _channel_B2A = new Channel;
117   _acmB->RegisterTransportCallback(_channel_B2A);
118   _channel_B2A->RegisterReceiverACM(_acmA.get());
119   //--- Do the same for reference
120   _channelRef_B2A = new Channel;
121   _acmRefB->RegisterTransportCallback(_channelRef_B2A);
122   _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
123 }
124 
Perform()125 void TwoWayCommunication::Perform() {
126   const SdpAudioFormat format1("ISAC", 16000, 1);
127   const SdpAudioFormat format2("L16", 8000, 1);
128   constexpr int payload_type1 = 17, payload_type2 = 18;
129 
130   auto encoder_factory = CreateBuiltinAudioEncoderFactory();
131 
132   SetUpAutotest(encoder_factory.get(), format1, payload_type1, format2,
133                 payload_type2);
134 
135   unsigned int msecPassed = 0;
136   unsigned int secPassed = 0;
137 
138   int32_t outFreqHzA = _outFileA.SamplingFrequency();
139   int32_t outFreqHzB = _outFileB.SamplingFrequency();
140 
141   AudioFrame audioFrame;
142 
143   // In the following loop we tests that the code can handle misuse of the APIs.
144   // In the middle of a session with data flowing between two sides, called A
145   // and B, APIs will be called, and the code should continue to run, and be
146   // able to recover.
147   while (!_inFileA.EndOfFile() && !_inFileB.EndOfFile()) {
148     msecPassed += 10;
149     EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
150     EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
151     EXPECT_GE(_acmRefA->Add10MsData(audioFrame), 0);
152 
153     EXPECT_GT(_inFileB.Read10MsData(audioFrame), 0);
154 
155     EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
156     EXPECT_GE(_acmRefB->Add10MsData(audioFrame), 0);
157     bool muted;
158     EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
159     ASSERT_FALSE(muted);
160     _outFileA.Write10MsData(audioFrame);
161     EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
162     ASSERT_FALSE(muted);
163     _outFileRefA.Write10MsData(audioFrame);
164     EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
165     ASSERT_FALSE(muted);
166     _outFileB.Write10MsData(audioFrame);
167     EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
168     ASSERT_FALSE(muted);
169     _outFileRefB.Write10MsData(audioFrame);
170 
171     // Update time counters each time a second of data has passed.
172     if (msecPassed >= 1000) {
173       msecPassed = 0;
174       secPassed++;
175     }
176     // Re-register send codec on side B.
177     if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
178       _acmB->SetEncoder(encoder_factory->MakeAudioEncoder(
179           payload_type2, format2, absl::nullopt));
180     }
181     // Initialize receiver on side A.
182     if (((secPassed % 7) == 6) && (msecPassed == 0))
183       EXPECT_EQ(0, _acmA->InitializeReceiver());
184     // Re-register codec on side A.
185     if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
186       _acmA->SetReceiveCodecs({{payload_type2, format2}});
187     }
188   }
189 }
190 
191 }  // namespace webrtc
192