OLD | NEW |
(Empty) | |
| 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 <algorithm> |
| 12 |
| 13 #include "webrtc/audio/test/audio_end_to_end_test.h" |
| 14 #include "webrtc/system_wrappers/include/sleep.h" |
| 15 #include "webrtc/test/fake_audio_device.h" |
| 16 #include "webrtc/test/gtest.h" |
| 17 |
| 18 namespace webrtc { |
| 19 namespace test { |
| 20 namespace { |
| 21 // Wait half a second between stopping sending and stopping receiving audio. |
| 22 constexpr int kExtraRecordTimeMs = 500; |
| 23 |
| 24 constexpr int kSampleRate = 48000; |
| 25 } // namespace |
| 26 |
| 27 AudioEndToEndTest::AudioEndToEndTest() |
| 28 : EndToEndTest(CallTest::kDefaultTimeoutMs) {} |
| 29 |
| 30 FakeNetworkPipe::Config AudioEndToEndTest::GetNetworkPipeConfig() const { |
| 31 return FakeNetworkPipe::Config(); |
| 32 } |
| 33 |
| 34 size_t AudioEndToEndTest::GetNumVideoStreams() const { |
| 35 return 0; |
| 36 } |
| 37 |
| 38 size_t AudioEndToEndTest::GetNumAudioStreams() const { |
| 39 return 1; |
| 40 } |
| 41 |
| 42 size_t AudioEndToEndTest::GetNumFlexfecStreams() const { |
| 43 return 0; |
| 44 } |
| 45 |
| 46 std::unique_ptr<test::FakeAudioDevice::Capturer> |
| 47 AudioEndToEndTest::CreateCapturer() { |
| 48 return test::FakeAudioDevice::CreatePulsedNoiseCapturer(32000, kSampleRate); |
| 49 } |
| 50 |
| 51 std::unique_ptr<test::FakeAudioDevice::Renderer> |
| 52 AudioEndToEndTest::CreateRenderer() { |
| 53 return test::FakeAudioDevice::CreateDiscardRenderer(kSampleRate); |
| 54 } |
| 55 |
| 56 void AudioEndToEndTest::OnFakeAudioDevicesCreated( |
| 57 test::FakeAudioDevice* send_audio_device, |
| 58 test::FakeAudioDevice* recv_audio_device) { |
| 59 send_audio_device_ = send_audio_device; |
| 60 } |
| 61 |
| 62 test::PacketTransport* AudioEndToEndTest::CreateSendTransport( |
| 63 SingleThreadedTaskQueueForTesting* task_queue, |
| 64 Call* sender_call) { |
| 65 return new test::PacketTransport( |
| 66 task_queue, sender_call, this, test::PacketTransport::kSender, |
| 67 test::CallTest::payload_type_map_, GetNetworkPipeConfig()); |
| 68 } |
| 69 |
| 70 test::PacketTransport* AudioEndToEndTest::CreateReceiveTransport( |
| 71 SingleThreadedTaskQueueForTesting* task_queue) { |
| 72 return new test::PacketTransport( |
| 73 task_queue, nullptr, this, test::PacketTransport::kReceiver, |
| 74 test::CallTest::payload_type_map_, GetNetworkPipeConfig()); |
| 75 } |
| 76 |
| 77 void AudioEndToEndTest::ModifyAudioConfigs( |
| 78 AudioSendStream::Config* send_config, |
| 79 std::vector<AudioReceiveStream::Config>* receive_configs) { |
| 80 // Large bitrate by default. |
| 81 const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2, |
| 82 {{"stereo", "1"}}); |
| 83 send_config->send_codec_spec = |
| 84 rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
| 85 {test::CallTest::kAudioSendPayloadType, kDefaultFormat}); |
| 86 } |
| 87 |
| 88 void AudioEndToEndTest::OnAudioStreamsCreated( |
| 89 AudioSendStream* send_stream, |
| 90 const std::vector<AudioReceiveStream*>& receive_streams) { |
| 91 ASSERT_NE(nullptr, send_stream); |
| 92 ASSERT_EQ(1u, receive_streams.size()); |
| 93 ASSERT_NE(nullptr, receive_streams[0]); |
| 94 send_stream_ = send_stream; |
| 95 receive_stream_ = receive_streams[0]; |
| 96 } |
| 97 |
| 98 void AudioEndToEndTest::PerformTest() { |
| 99 // Wait until the input audio file is done... |
| 100 send_audio_device_->WaitForRecordingEnd(); |
| 101 // and some extra time to account for network delay. |
| 102 SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs); |
| 103 } |
| 104 } // namespace test |
| 105 } // namespace webrtc |
OLD | NEW |