1 /*
2 * Copyright (c) 2017 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 "absl/flags/declare.h"
12 #include "absl/flags/flag.h"
13 #include "api/test/simulated_network.h"
14 #include "audio/test/audio_end_to_end_test.h"
15 #include "system_wrappers/include/sleep.h"
16 #include "test/testsupport/file_utils.h"
17
18 ABSL_DECLARE_FLAG(int, sample_rate_hz);
19 ABSL_DECLARE_FLAG(bool, quick);
20
21 namespace webrtc {
22 namespace test {
23 namespace {
24
FileSampleRateSuffix()25 std::string FileSampleRateSuffix() {
26 return std::to_string(absl::GetFlag(FLAGS_sample_rate_hz) / 1000);
27 }
28
29 class AudioQualityTest : public AudioEndToEndTest {
30 public:
31 AudioQualityTest() = default;
32
33 private:
AudioInputFile() const34 std::string AudioInputFile() const {
35 return test::ResourcePath(
36 "voice_engine/audio_tiny" + FileSampleRateSuffix(), "wav");
37 }
38
AudioOutputFile() const39 std::string AudioOutputFile() const {
40 const ::testing::TestInfo* const test_info =
41 ::testing::UnitTest::GetInstance()->current_test_info();
42 return webrtc::test::OutputPath() + "LowBandwidth_" + test_info->name() +
43 "_" + FileSampleRateSuffix() + ".wav";
44 }
45
CreateCapturer()46 std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override {
47 return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile());
48 }
49
CreateRenderer()50 std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override {
51 return TestAudioDeviceModule::CreateBoundedWavFileWriter(
52 AudioOutputFile(), absl::GetFlag(FLAGS_sample_rate_hz));
53 }
54
PerformTest()55 void PerformTest() override {
56 if (absl::GetFlag(FLAGS_quick)) {
57 // Let the recording run for a small amount of time to check if it works.
58 SleepMs(1000);
59 } else {
60 AudioEndToEndTest::PerformTest();
61 }
62 }
63
OnStreamsStopped()64 void OnStreamsStopped() override {
65 const ::testing::TestInfo* const test_info =
66 ::testing::UnitTest::GetInstance()->current_test_info();
67
68 // Output information about the input and output audio files so that further
69 // processing can be done by an external process.
70 printf("TEST %s %s %s\n", test_info->name(), AudioInputFile().c_str(),
71 AudioOutputFile().c_str());
72 }
73 };
74
75 class Mobile2GNetworkTest : public AudioQualityTest {
ModifyAudioConfigs(AudioSendStream::Config * send_config,std::vector<AudioReceiveStream::Config> * receive_configs)76 void ModifyAudioConfigs(
77 AudioSendStream::Config* send_config,
78 std::vector<AudioReceiveStream::Config>* receive_configs) override {
79 send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
80 test::CallTest::kAudioSendPayloadType,
81 {"OPUS",
82 48000,
83 2,
84 {{"maxaveragebitrate", "6000"}, {"ptime", "60"}, {"stereo", "1"}}});
85 }
86
GetNetworkPipeConfig() const87 BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const override {
88 BuiltInNetworkBehaviorConfig pipe_config;
89 pipe_config.link_capacity_kbps = 12;
90 pipe_config.queue_length_packets = 1500;
91 pipe_config.queue_delay_ms = 400;
92 return pipe_config;
93 }
94 };
95 } // namespace
96
97 using LowBandwidthAudioTest = CallTest;
98
TEST_F(LowBandwidthAudioTest,GoodNetworkHighBitrate)99 TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
100 AudioQualityTest test;
101 RunBaseTest(&test);
102 }
103
TEST_F(LowBandwidthAudioTest,Mobile2GNetwork)104 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
105 Mobile2GNetworkTest test;
106 RunBaseTest(&test);
107 }
108 } // namespace test
109 } // namespace webrtc
110