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/*
* 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 "audio/test/audio_end_to_end_test.h"
#include "system_wrappers/include/sleep.h"
#include "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<TestAudioDeviceModule::Capturer>
AudioEndToEndTest::CreateCapturer() {
return TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, kSampleRate);
}
std::unique_ptr<TestAudioDeviceModule::Renderer>
AudioEndToEndTest::CreateRenderer() {
return TestAudioDeviceModule::CreateDiscardRenderer(kSampleRate);
}
void AudioEndToEndTest::OnFakeAudioDevicesCreated(
TestAudioDeviceModule* send_audio_device,
TestAudioDeviceModule* 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 = 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
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