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/*
* Copyright (c) 2020 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/voip/audio_ingress.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/call/transport.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "audio/voip/audio_egress.h"
#include "modules/audio_mixer/sine_wave_generator.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_transport.h"
namespace webrtc {
namespace {
using ::testing::Invoke;
using ::testing::NiceMock;
using ::testing::Unused;
constexpr int16_t kAudioLevel = 3004; // Used for sine wave level.
class AudioIngressTest : public ::testing::Test {
public:
const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1};
AudioIngressTest()
: fake_clock_(123456789), wave_generator_(1000.0, kAudioLevel) {
receive_statistics_ = ReceiveStatistics::Create(&fake_clock_);
RtpRtcpInterface::Configuration rtp_config;
rtp_config.clock = &fake_clock_;
rtp_config.audio = true;
rtp_config.receive_statistics = receive_statistics_.get();
rtp_config.rtcp_report_interval_ms = 5000;
rtp_config.outgoing_transport = &transport_;
rtp_config.local_media_ssrc = 0xdeadc0de;
rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
rtp_rtcp_->SetSendingMediaStatus(false);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
task_queue_factory_ = CreateDefaultTaskQueueFactory();
encoder_factory_ = CreateBuiltinAudioEncoderFactory();
decoder_factory_ = CreateBuiltinAudioDecoderFactory();
}
void SetUp() override {
constexpr int kPcmuPayload = 0;
ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), &fake_clock_,
receive_statistics_.get(),
decoder_factory_);
ingress_->SetReceiveCodecs({{kPcmuPayload, kPcmuFormat}});
egress_ = std::make_unique<AudioEgress>(rtp_rtcp_.get(), &fake_clock_,
task_queue_factory_.get());
egress_->SetEncoder(kPcmuPayload, kPcmuFormat,
encoder_factory_->MakeAudioEncoder(
kPcmuPayload, kPcmuFormat, absl::nullopt));
egress_->StartSend();
ingress_->StartPlay();
rtp_rtcp_->SetSendingStatus(true);
}
void TearDown() override {
rtp_rtcp_->SetSendingStatus(false);
ingress_->StopPlay();
egress_->StopSend();
egress_.reset();
ingress_.reset();
}
std::unique_ptr<AudioFrame> GetAudioFrame(int order) {
auto frame = std::make_unique<AudioFrame>();
frame->sample_rate_hz_ = kPcmuFormat.clockrate_hz;
frame->samples_per_channel_ = kPcmuFormat.clockrate_hz / 100; // 10 ms.
frame->num_channels_ = kPcmuFormat.num_channels;
frame->timestamp_ = frame->samples_per_channel_ * order;
wave_generator_.GenerateNextFrame(frame.get());
return frame;
}
SimulatedClock fake_clock_;
SineWaveGenerator wave_generator_;
NiceMock<MockTransport> transport_;
std::unique_ptr<ReceiveStatistics> receive_statistics_;
std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
std::unique_ptr<TaskQueueFactory> task_queue_factory_;
std::unique_ptr<AudioIngress> ingress_;
std::unique_ptr<AudioEgress> egress_;
};
TEST_F(AudioIngressTest, PlayingAfterStartAndStop) {
EXPECT_EQ(ingress_->IsPlaying(), true);
ingress_->StopPlay();
EXPECT_EQ(ingress_->IsPlaying(), false);
}
TEST_F(AudioIngressTest, GetAudioFrameAfterRtpReceived) {
rtc::Event event;
auto handle_rtp = [&](const uint8_t* packet, size_t length, Unused) {
ingress_->ReceivedRTPPacket(rtc::ArrayView<const uint8_t>(packet, length));
event.Set();
return true;
};
EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp));
egress_->SendAudioData(GetAudioFrame(0));
egress_->SendAudioData(GetAudioFrame(1));
event.Wait(/*ms=*/1000);
AudioFrame audio_frame;
EXPECT_EQ(
ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame),
AudioMixer::Source::AudioFrameInfo::kNormal);
EXPECT_FALSE(audio_frame.muted());
EXPECT_EQ(audio_frame.num_channels_, 1u);
EXPECT_EQ(audio_frame.samples_per_channel_,
static_cast<size_t>(kPcmuFormat.clockrate_hz / 100));
EXPECT_EQ(audio_frame.sample_rate_hz_, kPcmuFormat.clockrate_hz);
EXPECT_NE(audio_frame.timestamp_, 0u);
EXPECT_EQ(audio_frame.elapsed_time_ms_, 0);
}
TEST_F(AudioIngressTest, GetSpeechOutputLevelFullRange) {
// Per audio_level's kUpdateFrequency, we need 11 RTP to get audio level.
constexpr int kNumRtp = 11;
int rtp_count = 0;
rtc::Event event;
auto handle_rtp = [&](const uint8_t* packet, size_t length, Unused) {
ingress_->ReceivedRTPPacket(rtc::ArrayView<const uint8_t>(packet, length));
if (++rtp_count == kNumRtp) {
event.Set();
}
return true;
};
EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp));
for (int i = 0; i < kNumRtp * 2; i++) {
egress_->SendAudioData(GetAudioFrame(i));
fake_clock_.AdvanceTimeMilliseconds(10);
}
event.Wait(/*ms=*/1000);
for (int i = 0; i < kNumRtp; ++i) {
AudioFrame audio_frame;
EXPECT_EQ(
ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame),
AudioMixer::Source::AudioFrameInfo::kNormal);
}
EXPECT_EQ(ingress_->GetSpeechOutputLevelFullRange(), kAudioLevel);
}
TEST_F(AudioIngressTest, PreferredSampleRate) {
rtc::Event event;
auto handle_rtp = [&](const uint8_t* packet, size_t length, Unused) {
ingress_->ReceivedRTPPacket(rtc::ArrayView<const uint8_t>(packet, length));
event.Set();
return true;
};
EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp));
egress_->SendAudioData(GetAudioFrame(0));
egress_->SendAudioData(GetAudioFrame(1));
event.Wait(/*ms=*/1000);
AudioFrame audio_frame;
EXPECT_EQ(
ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame),
AudioMixer::Source::AudioFrameInfo::kNormal);
EXPECT_EQ(ingress_->PreferredSampleRate(), kPcmuFormat.clockrate_hz);
}
} // namespace
} // namespace webrtc
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