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1 /* | 1 /* |
2 * Copyright (c) 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 #include <algorithm> | 11 #include <algorithm> |
12 | 12 |
13 #include "gflags/gflags.h" | |
14 #include "webrtc/audio/test/low_bandwidth_audio_test.h" | 13 #include "webrtc/audio/test/low_bandwidth_audio_test.h" |
15 #include "webrtc/common_audio/wav_file.h" | 14 #include "webrtc/common_audio/wav_file.h" |
| 15 #include "webrtc/rtc_base/flags.h" |
| 16 #include "webrtc/system_wrappers/include/sleep.h" |
16 #include "webrtc/test/gtest.h" | 17 #include "webrtc/test/gtest.h" |
17 #include "webrtc/system_wrappers/include/sleep.h" | |
18 #include "webrtc/test/testsupport/fileutils.h" | 18 #include "webrtc/test/testsupport/fileutils.h" |
19 | 19 |
20 | 20 |
21 DEFINE_int32(sample_rate_hz, 16000, | 21 DEFINE_int(sample_rate_hz, 16000, |
22 "Sample rate (Hz) of the produced audio files."); | 22 "Sample rate (Hz) of the produced audio files."); |
23 | 23 |
24 DEFINE_bool(quick, false, | 24 DEFINE_bool(quick, false, |
25 "Don't do the full audio recording. " | 25 "Don't do the full audio recording. " |
26 "Used to quickly check that the test runs without crashing."); | 26 "Used to quickly check that the test runs without crashing."); |
27 | 27 |
28 namespace { | 28 namespace { |
29 | 29 |
30 // Wait half a second between stopping sending and stopping receiving audio. | 30 // Wait half a second between stopping sending and stopping receiving audio. |
31 constexpr int kExtraRecordTimeMs = 500; | 31 constexpr int kExtraRecordTimeMs = 500; |
32 | 32 |
33 std::string FileSampleRateSuffix() { | 33 std::string FileSampleRateSuffix() { |
34 return std::to_string(FLAGS_sample_rate_hz / 1000); | 34 return std::to_string(FLAG_sample_rate_hz / 1000); |
35 } | 35 } |
36 | 36 |
37 } // namespace | 37 } // namespace |
38 | 38 |
39 namespace webrtc { | 39 namespace webrtc { |
40 namespace test { | 40 namespace test { |
41 | 41 |
42 AudioQualityTest::AudioQualityTest() | 42 AudioQualityTest::AudioQualityTest() |
43 : EndToEndTest(CallTest::kDefaultTimeoutMs) {} | 43 : EndToEndTest(CallTest::kDefaultTimeoutMs) {} |
44 | 44 |
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65 } | 65 } |
66 | 66 |
67 std::unique_ptr<test::FakeAudioDevice::Capturer> | 67 std::unique_ptr<test::FakeAudioDevice::Capturer> |
68 AudioQualityTest::CreateCapturer() { | 68 AudioQualityTest::CreateCapturer() { |
69 return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile()); | 69 return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile()); |
70 } | 70 } |
71 | 71 |
72 std::unique_ptr<test::FakeAudioDevice::Renderer> | 72 std::unique_ptr<test::FakeAudioDevice::Renderer> |
73 AudioQualityTest::CreateRenderer() { | 73 AudioQualityTest::CreateRenderer() { |
74 return test::FakeAudioDevice::CreateBoundedWavFileWriter( | 74 return test::FakeAudioDevice::CreateBoundedWavFileWriter( |
75 AudioOutputFile(), FLAGS_sample_rate_hz); | 75 AudioOutputFile(), FLAG_sample_rate_hz); |
76 } | 76 } |
77 | 77 |
78 void AudioQualityTest::OnFakeAudioDevicesCreated( | 78 void AudioQualityTest::OnFakeAudioDevicesCreated( |
79 test::FakeAudioDevice* send_audio_device, | 79 test::FakeAudioDevice* send_audio_device, |
80 test::FakeAudioDevice* recv_audio_device) { | 80 test::FakeAudioDevice* recv_audio_device) { |
81 send_audio_device_ = send_audio_device; | 81 send_audio_device_ = send_audio_device; |
82 } | 82 } |
83 | 83 |
84 FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() { | 84 FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() { |
85 return FakeNetworkPipe::Config(); | 85 return FakeNetworkPipe::Config(); |
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105 std::vector<AudioReceiveStream::Config>* receive_configs) { | 105 std::vector<AudioReceiveStream::Config>* receive_configs) { |
106 // Large bitrate by default. | 106 // Large bitrate by default. |
107 const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2, | 107 const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2, |
108 {{"stereo", "1"}}); | 108 {{"stereo", "1"}}); |
109 send_config->send_codec_spec = | 109 send_config->send_codec_spec = |
110 rtc::Optional<AudioSendStream::Config::SendCodecSpec>( | 110 rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
111 {test::CallTest::kAudioSendPayloadType, kDefaultFormat}); | 111 {test::CallTest::kAudioSendPayloadType, kDefaultFormat}); |
112 } | 112 } |
113 | 113 |
114 void AudioQualityTest::PerformTest() { | 114 void AudioQualityTest::PerformTest() { |
115 if (FLAGS_quick) { | 115 if (FLAG_quick) { |
116 // Let the recording run for a small amount of time to check if it works. | 116 // Let the recording run for a small amount of time to check if it works. |
117 SleepMs(1000); | 117 SleepMs(1000); |
118 } else { | 118 } else { |
119 // Wait until the input audio file is done... | 119 // Wait until the input audio file is done... |
120 send_audio_device_->WaitForRecordingEnd(); | 120 send_audio_device_->WaitForRecordingEnd(); |
121 // and some extra time to account for network delay. | 121 // and some extra time to account for network delay. |
122 SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs); | 122 SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs); |
123 } | 123 } |
124 } | 124 } |
125 | 125 |
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165 } | 165 } |
166 }; | 166 }; |
167 | 167 |
168 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) { | 168 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) { |
169 Mobile2GNetworkTest test; | 169 Mobile2GNetworkTest test; |
170 RunBaseTest(&test); | 170 RunBaseTest(&test); |
171 } | 171 } |
172 | 172 |
173 } // namespace test | 173 } // namespace test |
174 } // namespace webrtc | 174 } // namespace webrtc |
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