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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); |
| 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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