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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
105 lines
3.4 KiB
C++
105 lines
3.4 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <algorithm>
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#include "audio/test/audio_end_to_end_test.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/fake_audio_device.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace test {
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namespace {
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// Wait half a second between stopping sending and stopping receiving audio.
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constexpr int kExtraRecordTimeMs = 500;
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constexpr int kSampleRate = 48000;
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} // namespace
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AudioEndToEndTest::AudioEndToEndTest()
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: EndToEndTest(CallTest::kDefaultTimeoutMs) {}
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FakeNetworkPipe::Config AudioEndToEndTest::GetNetworkPipeConfig() const {
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return FakeNetworkPipe::Config();
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}
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size_t AudioEndToEndTest::GetNumVideoStreams() const {
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return 0;
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}
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size_t AudioEndToEndTest::GetNumAudioStreams() const {
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return 1;
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}
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size_t AudioEndToEndTest::GetNumFlexfecStreams() const {
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return 0;
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}
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std::unique_ptr<test::FakeAudioDevice::Capturer>
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AudioEndToEndTest::CreateCapturer() {
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return test::FakeAudioDevice::CreatePulsedNoiseCapturer(32000, kSampleRate);
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}
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std::unique_ptr<test::FakeAudioDevice::Renderer>
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AudioEndToEndTest::CreateRenderer() {
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return test::FakeAudioDevice::CreateDiscardRenderer(kSampleRate);
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}
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void AudioEndToEndTest::OnFakeAudioDevicesCreated(
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test::FakeAudioDevice* send_audio_device,
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test::FakeAudioDevice* recv_audio_device) {
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send_audio_device_ = send_audio_device;
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}
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test::PacketTransport* AudioEndToEndTest::CreateSendTransport(
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SingleThreadedTaskQueueForTesting* task_queue,
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Call* sender_call) {
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return new test::PacketTransport(
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task_queue, sender_call, this, test::PacketTransport::kSender,
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test::CallTest::payload_type_map_, GetNetworkPipeConfig());
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}
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test::PacketTransport* AudioEndToEndTest::CreateReceiveTransport(
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SingleThreadedTaskQueueForTesting* task_queue) {
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return new test::PacketTransport(
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task_queue, nullptr, this, test::PacketTransport::kReceiver,
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test::CallTest::payload_type_map_, GetNetworkPipeConfig());
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}
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void AudioEndToEndTest::ModifyAudioConfigs(
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AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStream::Config>* receive_configs) {
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// Large bitrate by default.
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const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2,
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{{"stereo", "1"}});
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send_config->send_codec_spec =
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rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
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{test::CallTest::kAudioSendPayloadType, kDefaultFormat});
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}
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void AudioEndToEndTest::OnAudioStreamsCreated(
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AudioSendStream* send_stream,
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const std::vector<AudioReceiveStream*>& receive_streams) {
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ASSERT_NE(nullptr, send_stream);
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ASSERT_EQ(1u, receive_streams.size());
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ASSERT_NE(nullptr, receive_streams[0]);
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send_stream_ = send_stream;
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receive_stream_ = receive_streams[0];
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}
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void AudioEndToEndTest::PerformTest() {
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// Wait until the input audio file is done...
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send_audio_device_->WaitForRecordingEnd();
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// and some extra time to account for network delay.
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SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
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}
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} // namespace test
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} // namespace webrtc
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