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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 "audio/test/audio_end_to_end_test.h"
#include <algorithm>
#include <memory>
#include "api/task_queue/task_queue_base.h"
#include "call/fake_network_pipe.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "test/gtest.h"
#include "test/video_test_constants.h"
namespace webrtc {
namespace test {
namespace {
constexpr int kSampleRate = 48000;
} // namespace
AudioEndToEndTest::AudioEndToEndTest()
: EndToEndTest(VideoTestConstants::kDefaultTimeout) {}
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(
AudioDeviceModule* send_audio_device,
AudioDeviceModule* /* recv_audio_device */) {
send_audio_device_ = send_audio_device;
}
void AudioEndToEndTest::ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>* /* receive_configs */) {
// Large bitrate by default.
const SdpAudioFormat kDefaultFormat("opus", 48000, 2, {{"stereo", "1"}});
send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
test::VideoTestConstants::kAudioSendPayloadType, kDefaultFormat);
send_config->min_bitrate_bps = 32000;
send_config->max_bitrate_bps = 32000;
}
void AudioEndToEndTest::OnAudioStreamsCreated(
AudioSendStream* send_stream,
const std::vector<AudioReceiveStreamInterface*>& 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];
}
} // namespace test
} // namespace webrtc
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