| /* |
| * 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 <algorithm> |
| |
| #include "webrtc/audio/test/audio_end_to_end_test.h" |
| #include "webrtc/system_wrappers/include/sleep.h" |
| #include "webrtc/test/fake_audio_device.h" |
| #include "webrtc/test/gtest.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| // Wait half a second between stopping sending and stopping receiving audio. |
| constexpr int kExtraRecordTimeMs = 500; |
| |
| constexpr int kSampleRate = 48000; |
| } // namespace |
| |
| AudioEndToEndTest::AudioEndToEndTest() |
| : EndToEndTest(CallTest::kDefaultTimeoutMs) {} |
| |
| FakeNetworkPipe::Config AudioEndToEndTest::GetNetworkPipeConfig() const { |
| return FakeNetworkPipe::Config(); |
| } |
| |
| size_t AudioEndToEndTest::GetNumVideoStreams() const { |
| return 0; |
| } |
| |
| size_t AudioEndToEndTest::GetNumAudioStreams() const { |
| return 1; |
| } |
| |
| size_t AudioEndToEndTest::GetNumFlexfecStreams() const { |
| return 0; |
| } |
| |
| std::unique_ptr<test::FakeAudioDevice::Capturer> |
| AudioEndToEndTest::CreateCapturer() { |
| return test::FakeAudioDevice::CreatePulsedNoiseCapturer(32000, kSampleRate); |
| } |
| |
| std::unique_ptr<test::FakeAudioDevice::Renderer> |
| AudioEndToEndTest::CreateRenderer() { |
| return test::FakeAudioDevice::CreateDiscardRenderer(kSampleRate); |
| } |
| |
| void AudioEndToEndTest::OnFakeAudioDevicesCreated( |
| test::FakeAudioDevice* send_audio_device, |
| test::FakeAudioDevice* recv_audio_device) { |
| send_audio_device_ = send_audio_device; |
| } |
| |
| test::PacketTransport* AudioEndToEndTest::CreateSendTransport( |
| SingleThreadedTaskQueueForTesting* task_queue, |
| Call* sender_call) { |
| return new test::PacketTransport( |
| task_queue, sender_call, this, test::PacketTransport::kSender, |
| test::CallTest::payload_type_map_, GetNetworkPipeConfig()); |
| } |
| |
| test::PacketTransport* AudioEndToEndTest::CreateReceiveTransport( |
| SingleThreadedTaskQueueForTesting* task_queue) { |
| return new test::PacketTransport( |
| task_queue, nullptr, this, test::PacketTransport::kReceiver, |
| test::CallTest::payload_type_map_, GetNetworkPipeConfig()); |
| } |
| |
| void AudioEndToEndTest::ModifyAudioConfigs( |
| AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStream::Config>* receive_configs) { |
| // Large bitrate by default. |
| const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2, |
| {{"stereo", "1"}}); |
| send_config->send_codec_spec = |
| rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
| {test::CallTest::kAudioSendPayloadType, kDefaultFormat}); |
| } |
| |
| void AudioEndToEndTest::OnAudioStreamsCreated( |
| AudioSendStream* send_stream, |
| const std::vector<AudioReceiveStream*>& receive_streams) { |
| ASSERT_NE(nullptr, send_stream); |
| ASSERT_EQ(1u, receive_streams.size()); |
| ASSERT_NE(nullptr, receive_streams[0]); |
| send_stream_ = send_stream; |
| receive_stream_ = receive_streams[0]; |
| } |
| |
| void AudioEndToEndTest::PerformTest() { |
| // Wait until the input audio file is done... |
| send_audio_device_->WaitForRecordingEnd(); |
| // and some extra time to account for network delay. |
| SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs); |
| } |
| } // namespace test |
| } // namespace webrtc |