| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "audio/test/audio_bwe_integration_test.h" |
| |
| #include <memory> |
| |
| #include "absl/functional/any_invocable.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "call/fake_network_pipe.h" |
| #include "call/simulated_network.h" |
| #include "common_audio/wav_file.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "system_wrappers/include/sleep.h" |
| #include "test/field_trial.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/file_utils.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| namespace { |
| enum : int { // The first valid value is 1. |
| kTransportSequenceNumberExtensionId = 1, |
| }; |
| |
| // Wait a second between stopping sending and stopping receiving audio. |
| constexpr int kExtraProcessTimeMs = 1000; |
| } // namespace |
| |
| AudioBweTest::AudioBweTest() : EndToEndTest(CallTest::kDefaultTimeout) {} |
| |
| size_t AudioBweTest::GetNumVideoStreams() const { |
| return 0; |
| } |
| size_t AudioBweTest::GetNumAudioStreams() const { |
| return 1; |
| } |
| size_t AudioBweTest::GetNumFlexfecStreams() const { |
| return 0; |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::Capturer> |
| AudioBweTest::CreateCapturer() { |
| return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile()); |
| } |
| |
| void AudioBweTest::OnFakeAudioDevicesCreated( |
| TestAudioDeviceModule* send_audio_device, |
| TestAudioDeviceModule* recv_audio_device) { |
| send_audio_device_ = send_audio_device; |
| } |
| |
| std::unique_ptr<test::PacketTransport> AudioBweTest::CreateSendTransport( |
| TaskQueueBase* task_queue, |
| Call* sender_call) { |
| return std::make_unique<test::PacketTransport>( |
| task_queue, sender_call, this, test::PacketTransport::kSender, |
| test::CallTest::payload_type_map_, |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>(GetNetworkPipeConfig()))); |
| } |
| |
| std::unique_ptr<test::PacketTransport> AudioBweTest::CreateReceiveTransport( |
| TaskQueueBase* task_queue) { |
| return std::make_unique<test::PacketTransport>( |
| task_queue, nullptr, this, test::PacketTransport::kReceiver, |
| test::CallTest::payload_type_map_, |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>(GetNetworkPipeConfig()))); |
| } |
| |
| void AudioBweTest::PerformTest() { |
| send_audio_device_->WaitForRecordingEnd(); |
| SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraProcessTimeMs); |
| } |
| |
| absl::AnyInvocable<void() &&> StatsPollTask(Call* sender_call) { |
| RTC_CHECK(sender_call); |
| return [sender_call] { |
| Call::Stats call_stats = sender_call->GetStats(); |
| EXPECT_GT(call_stats.send_bandwidth_bps, 25000); |
| TaskQueueBase::Current()->PostDelayedTask(StatsPollTask(sender_call), |
| TimeDelta::Millis(100)); |
| }; |
| } |
| |
| class NoBandwidthDropAfterDtx : public AudioBweTest { |
| public: |
| NoBandwidthDropAfterDtx() |
| : sender_call_(nullptr), stats_poller_("stats poller task queue") {} |
| |
| void ModifyAudioConfigs(AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStreamInterface::Config>* |
| receive_configs) override { |
| send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( |
| test::CallTest::kAudioSendPayloadType, |
| {"OPUS", |
| 48000, |
| 2, |
| {{"ptime", "60"}, {"usedtx", "1"}, {"stereo", "1"}}}); |
| |
| send_config->min_bitrate_bps = 6000; |
| send_config->max_bitrate_bps = 100000; |
| send_config->rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
| kTransportSequenceNumberExtensionId)); |
| for (AudioReceiveStreamInterface::Config& recv_config : *receive_configs) { |
| recv_config.rtp.transport_cc = true; |
| recv_config.rtp.extensions = send_config->rtp.extensions; |
| recv_config.rtp.remote_ssrc = send_config->rtp.ssrc; |
| } |
| } |
| |
| std::string AudioInputFile() override { |
| return test::ResourcePath("voice_engine/audio_dtx16", "wav"); |
| } |
| |
| BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() override { |
| BuiltInNetworkBehaviorConfig pipe_config; |
| pipe_config.link_capacity_kbps = 50; |
| pipe_config.queue_length_packets = 1500; |
| pipe_config.queue_delay_ms = 300; |
| return pipe_config; |
| } |
| |
| void OnCallsCreated(Call* sender_call, Call* receiver_call) override { |
| sender_call_ = sender_call; |
| } |
| |
| void PerformTest() override { |
| stats_poller_.PostDelayedTask(StatsPollTask(sender_call_), |
| TimeDelta::Millis(100)); |
| sender_call_->OnAudioTransportOverheadChanged(0); |
| AudioBweTest::PerformTest(); |
| } |
| |
| private: |
| Call* sender_call_; |
| TaskQueueForTest stats_poller_; |
| }; |
| |
| using AudioBweIntegrationTest = CallTest; |
| |
| // TODO(tschumim): This test is flaky when run on android and mac. Re-enable the |
| // test for when the issue is fixed. |
| TEST_F(AudioBweIntegrationTest, DISABLED_NoBandwidthDropAfterDtx) { |
| NoBandwidthDropAfterDtx test; |
| RunBaseTest(&test); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |