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1 /* | 1 /* |
2 * Copyright 2017 The WebRTC Project Authors. All rights reserved. | 2 * Copyright (c) 2017 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 // This is a placeholder for the work oprypin@ is doing on a low-bandwidth | 11 #include <algorithm> |
12 // audio test executable. | |
13 | 12 |
14 int main() { | 13 #include "webrtc/audio/test/low_bandwidth_audio_test.h" |
14 #include "webrtc/common_audio/wav_file.h" | |
15 #include "webrtc/test/gtest.h" | |
16 #include "webrtc/test/run_test.h" | |
17 #include "webrtc/system_wrappers/include/sleep.h" | |
18 #include "webrtc/test/testsupport/fileutils.h" | |
19 | |
20 namespace { | |
21 // Wait half a second between stopping sending and stopping receiving audio. | |
22 constexpr int kExtraRecordTimeMs = 500; | |
23 | |
24 // Large bitrate by default. | |
25 const webrtc::CodecInst kDefaultCodec{120, "OPUS", 48000, 960, 2, 64000}; | |
26 | |
27 // The best that can be done with PESQ. | |
28 constexpr int kAudioFileBitRate = 16000; | |
29 } | |
30 | |
31 namespace webrtc { | |
32 namespace test { | |
33 | |
34 AudioQualityTest::AudioQualityTest() | |
35 : EndToEndTest(CallTest::kDefaultTimeoutMs) {} | |
36 | |
37 size_t AudioQualityTest::GetNumVideoStreams() const { | |
15 return 0; | 38 return 0; |
16 } | 39 } |
40 size_t AudioQualityTest::GetNumAudioStreams() const { | |
41 return 1; | |
42 } | |
43 size_t AudioQualityTest::GetNumFlexfecStreams() const { | |
44 return 0; | |
45 } | |
46 | |
47 std::string AudioQualityTest::AudioInputFile() { | |
48 return test::ResourcePath("voice_engine/audio_tiny16", "wav"); | |
49 } | |
50 | |
51 std::string AudioQualityTest::AudioOutputFile() { | |
52 const ::testing::TestInfo* const test_info = | |
53 ::testing::UnitTest::GetInstance()->current_test_info(); | |
54 return webrtc::test::OutputPath() + | |
55 "LowBandwidth_" + test_info->name() + ".wav"; | |
56 } | |
57 | |
58 std::unique_ptr<test::FakeAudioDevice::Capturer> | |
59 AudioQualityTest::CreateCapturer() { | |
60 return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile()); | |
61 } | |
62 | |
63 std::unique_ptr<test::FakeAudioDevice::Renderer> | |
64 AudioQualityTest::CreateRenderer() { | |
65 return test::FakeAudioDevice::CreateBoundedWavFileWriter( | |
66 AudioOutputFile(), kAudioFileBitRate); | |
67 } | |
68 | |
69 void AudioQualityTest::OnFakeAudioDevicesCreated( | |
70 test::FakeAudioDevice* send_audio_device, | |
71 test::FakeAudioDevice* recv_audio_device) { | |
72 send_audio_device_ = send_audio_device; | |
73 } | |
74 | |
75 FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() { | |
76 return FakeNetworkPipe::Config(); | |
77 } | |
78 | |
79 test::PacketTransport* AudioQualityTest::CreateSendTransport( | |
80 Call* sender_call) { | |
81 return new test::PacketTransport( | |
82 sender_call, this, test::PacketTransport::kSender, | |
83 GetNetworkPipeConfig()); | |
84 } | |
85 | |
86 test::PacketTransport* AudioQualityTest::CreateReceiveTransport() { | |
87 return new test::PacketTransport(nullptr, this, | |
88 test::PacketTransport::kReceiver, GetNetworkPipeConfig()); | |
89 } | |
90 | |
91 void AudioQualityTest::ModifyAudioConfigs( | |
92 AudioSendStream::Config* send_config, | |
93 std::vector<AudioReceiveStream::Config>* receive_configs) { | |
94 send_config->send_codec_spec.codec_inst = kDefaultCodec; | |
95 } | |
96 | |
97 void AudioQualityTest::PerformTest() { | |
98 // Wait until the input audio file is done... | |
99 send_audio_device_->WaitForRecordingEnd(); | |
100 // and some extra time to account for network delay. | |
101 SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs); | |
stefan-webrtc
2017/03/20 14:40:10
Can we perhaps wait until enough audio has been pl
oprypin_webrtc
2017/03/20 15:09:16
I've given this some thought. Relying on the recei
stefan-webrtc
2017/03/20 15:42:32
Ok, convinced!
| |
102 } | |
103 | |
104 void AudioQualityTest::OnTestFinished() { | |
105 const ::testing::TestInfo* const test_info = | |
106 ::testing::UnitTest::GetInstance()->current_test_info(); | |
107 | |
108 // Output information about the input and output audio files so that further | |
109 // processing can be done by an external process. | |
110 printf("TEST %s %s:%s\n", test_info->name(), | |
111 AudioInputFile().c_str(), AudioOutputFile().c_str()); | |
112 } | |
113 | |
114 | |
115 using LowBandwidthAudioTest = CallTest; | |
116 | |
117 TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) { | |
118 AudioQualityTest test; | |
119 RunBaseTest(&test); | |
120 } | |
121 | |
122 | |
123 class Mobile2GNetworkTest : public AudioQualityTest { | |
124 void ModifyAudioConfigs(AudioSendStream::Config* send_config, | |
125 std::vector<AudioReceiveStream::Config>* receive_configs) override { | |
126 send_config->send_codec_spec.codec_inst = CodecInst{ | |
127 120, // pltype | |
128 "OPUS", // plname | |
129 48000, // plfreq | |
130 2880, // pacsize | |
131 1, // channels | |
132 6000 // rate bits/sec | |
133 }; | |
134 } | |
135 | |
136 FakeNetworkPipe::Config GetNetworkPipeConfig() override { | |
137 FakeNetworkPipe::Config pipe_config; | |
138 pipe_config.link_capacity_kbps = 12; | |
139 pipe_config.queue_length_packets = 1500; | |
140 pipe_config.queue_delay_ms = 400; | |
141 return pipe_config; | |
142 } | |
143 }; | |
144 | |
145 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) { | |
146 Mobile2GNetworkTest test; | |
147 RunBaseTest(&test); | |
148 } | |
149 | |
150 } // namespace test | |
151 } // namespace webrtc | |
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