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 "audio/test/audio_bwe_integration_test.h"
12
13 #include <memory>
14
15 #include "api/task_queue/queued_task.h"
16 #include "api/task_queue/task_queue_base.h"
17 #include "call/fake_network_pipe.h"
18 #include "call/simulated_network.h"
19 #include "common_audio/wav_file.h"
20 #include "rtc_base/task_queue_for_test.h"
21 #include "system_wrappers/include/sleep.h"
22 #include "test/field_trial.h"
23 #include "test/gtest.h"
24 #include "test/testsupport/file_utils.h"
25
26 namespace webrtc {
27 namespace test {
28
29 namespace {
30 enum : int { // The first valid value is 1.
31 kTransportSequenceNumberExtensionId = 1,
32 };
33
34 // Wait a second between stopping sending and stopping receiving audio.
35 constexpr int kExtraProcessTimeMs = 1000;
36 } // namespace
37
AudioBweTest()38 AudioBweTest::AudioBweTest() : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
39
GetNumVideoStreams() const40 size_t AudioBweTest::GetNumVideoStreams() const {
41 return 0;
42 }
GetNumAudioStreams() const43 size_t AudioBweTest::GetNumAudioStreams() const {
44 return 1;
45 }
GetNumFlexfecStreams() const46 size_t AudioBweTest::GetNumFlexfecStreams() const {
47 return 0;
48 }
49
50 std::unique_ptr<TestAudioDeviceModule::Capturer>
CreateCapturer()51 AudioBweTest::CreateCapturer() {
52 return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile());
53 }
54
OnFakeAudioDevicesCreated(TestAudioDeviceModule * send_audio_device,TestAudioDeviceModule * recv_audio_device)55 void AudioBweTest::OnFakeAudioDevicesCreated(
56 TestAudioDeviceModule* send_audio_device,
57 TestAudioDeviceModule* recv_audio_device) {
58 send_audio_device_ = send_audio_device;
59 }
60
CreateSendTransport(TaskQueueBase * task_queue,Call * sender_call)61 std::unique_ptr<test::PacketTransport> AudioBweTest::CreateSendTransport(
62 TaskQueueBase* task_queue,
63 Call* sender_call) {
64 return std::make_unique<test::PacketTransport>(
65 task_queue, sender_call, this, test::PacketTransport::kSender,
66 test::CallTest::payload_type_map_,
67 std::make_unique<FakeNetworkPipe>(
68 Clock::GetRealTimeClock(),
69 std::make_unique<SimulatedNetwork>(GetNetworkPipeConfig())));
70 }
71
CreateReceiveTransport(TaskQueueBase * task_queue)72 std::unique_ptr<test::PacketTransport> AudioBweTest::CreateReceiveTransport(
73 TaskQueueBase* task_queue) {
74 return std::make_unique<test::PacketTransport>(
75 task_queue, nullptr, this, test::PacketTransport::kReceiver,
76 test::CallTest::payload_type_map_,
77 std::make_unique<FakeNetworkPipe>(
78 Clock::GetRealTimeClock(),
79 std::make_unique<SimulatedNetwork>(GetNetworkPipeConfig())));
80 }
81
PerformTest()82 void AudioBweTest::PerformTest() {
83 send_audio_device_->WaitForRecordingEnd();
84 SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraProcessTimeMs);
85 }
86
87 class StatsPollTask : public QueuedTask {
88 public:
StatsPollTask(Call * sender_call)89 explicit StatsPollTask(Call* sender_call) : sender_call_(sender_call) {}
90
91 private:
Run()92 bool Run() override {
93 RTC_CHECK(sender_call_);
94 Call::Stats call_stats = sender_call_->GetStats();
95 EXPECT_GT(call_stats.send_bandwidth_bps, 25000);
96 TaskQueueBase::Current()->PostDelayedTask(std::unique_ptr<QueuedTask>(this),
97 100);
98 return false;
99 }
100 Call* sender_call_;
101 };
102
103 class NoBandwidthDropAfterDtx : public AudioBweTest {
104 public:
NoBandwidthDropAfterDtx()105 NoBandwidthDropAfterDtx()
106 : sender_call_(nullptr), stats_poller_("stats poller task queue") {}
107
ModifyAudioConfigs(AudioSendStream::Config * send_config,std::vector<AudioReceiveStream::Config> * receive_configs)108 void ModifyAudioConfigs(
109 AudioSendStream::Config* send_config,
110 std::vector<AudioReceiveStream::Config>* receive_configs) override {
111 send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
112 test::CallTest::kAudioSendPayloadType,
113 {"OPUS",
114 48000,
115 2,
116 {{"ptime", "60"}, {"usedtx", "1"}, {"stereo", "1"}}});
117
118 send_config->min_bitrate_bps = 6000;
119 send_config->max_bitrate_bps = 100000;
120 send_config->rtp.extensions.push_back(
121 RtpExtension(RtpExtension::kTransportSequenceNumberUri,
122 kTransportSequenceNumberExtensionId));
123 for (AudioReceiveStream::Config& recv_config : *receive_configs) {
124 recv_config.rtp.transport_cc = true;
125 recv_config.rtp.extensions = send_config->rtp.extensions;
126 recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
127 }
128 }
129
AudioInputFile()130 std::string AudioInputFile() override {
131 return test::ResourcePath("voice_engine/audio_dtx16", "wav");
132 }
133
GetNetworkPipeConfig()134 BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() override {
135 BuiltInNetworkBehaviorConfig pipe_config;
136 pipe_config.link_capacity_kbps = 50;
137 pipe_config.queue_length_packets = 1500;
138 pipe_config.queue_delay_ms = 300;
139 return pipe_config;
140 }
141
OnCallsCreated(Call * sender_call,Call * receiver_call)142 void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
143 sender_call_ = sender_call;
144 }
145
PerformTest()146 void PerformTest() override {
147 stats_poller_.PostDelayedTask(std::make_unique<StatsPollTask>(sender_call_),
148 100);
149 sender_call_->OnAudioTransportOverheadChanged(0);
150 AudioBweTest::PerformTest();
151 }
152
153 private:
154 Call* sender_call_;
155 TaskQueueForTest stats_poller_;
156 };
157
158 using AudioBweIntegrationTest = CallTest;
159
160 // TODO(tschumim): This test is flaky when run on android and mac. Re-enable the
161 // test for when the issue is fixed.
TEST_F(AudioBweIntegrationTest,DISABLED_NoBandwidthDropAfterDtx)162 TEST_F(AudioBweIntegrationTest, DISABLED_NoBandwidthDropAfterDtx) {
163 webrtc::test::ScopedFieldTrials override_field_trials(
164 "WebRTC-Audio-SendSideBwe/Enabled/"
165 "WebRTC-SendSideBwe-WithOverhead/Enabled/");
166 NoBandwidthDropAfterDtx test;
167 RunBaseTest(&test);
168 }
169
170 } // namespace test
171 } // namespace webrtc
172