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/*
* Copyright (c) 2015 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 <string>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_send_stream.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/call/congestion_controller.h"
#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
#include "webrtc/test/mock_voice_engine.h"
#include "webrtc/video/call_stats.h"
namespace webrtc {
namespace test {
namespace {
using testing::_;
using testing::Return;
const int kChannelId = 1;
const uint32_t kSsrc = 1234;
const char* kCName = "foo_name";
const int kAudioLevelId = 2;
const int kAbsSendTimeId = 3;
const int kTransportSequenceNumberId = 4;
const int kEchoDelayMedian = 254;
const int kEchoDelayStdDev = -3;
const int kEchoReturnLoss = -65;
const int kEchoReturnLossEnhancement = 101;
const unsigned int kSpeechInputLevel = 96;
const CallStatistics kCallStats = {
1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123};
const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
const int kTelephoneEventPayloadType = 123;
const uint8_t kTelephoneEventCode = 45;
const uint32_t kTelephoneEventDuration = 6789;
struct ConfigHelper {
ConfigHelper()
: stream_config_(nullptr),
call_stats_(Clock::GetRealTimeClock()),
process_thread_(ProcessThread::Create("AudioTestThread")),
congestion_controller_(process_thread_.get(),
&call_stats_,
&bitrate_observer_) {
using testing::Invoke;
using testing::StrEq;
EXPECT_CALL(voice_engine_,
RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
EXPECT_CALL(voice_engine_,
DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
AudioState::Config config;
config.voice_engine = &voice_engine_;
audio_state_ = AudioState::Create(config);
EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
.WillOnce(Invoke([this](int channel_id) {
EXPECT_FALSE(channel_proxy_);
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
EXPECT_CALL(*channel_proxy_,
SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
EXPECT_CALL(*channel_proxy_,
SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1);
EXPECT_CALL(*channel_proxy_, EnableSendTransportSequenceNumber(
kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
SetCongestionControlObjects(
congestion_controller_.pacer(),
congestion_controller_.GetTransportFeedbackObserver(),
congestion_controller_.packet_router()))
.Times(1);
EXPECT_CALL(*channel_proxy_,
SetCongestionControlObjects(nullptr, nullptr, nullptr))
.Times(1);
return channel_proxy_;
}));
stream_config_.voe_channel_id = kChannelId;
stream_config_.rtp.ssrc = kSsrc;
stream_config_.rtp.c_name = kCName;
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
stream_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
}
AudioSendStream::Config& config() { return stream_config_; }
rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
CongestionController* congestion_controller() {
return &congestion_controller_;
}
void SetupMockForSendTelephoneEvent() {
EXPECT_TRUE(channel_proxy_);
EXPECT_CALL(*channel_proxy_,
SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType))
.WillOnce(Return(true));
EXPECT_CALL(*channel_proxy_,
SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
.WillOnce(Return(true));
}
void SetupMockForGetStats() {
using testing::DoAll;
using testing::SetArgReferee;
std::vector<ReportBlock> report_blocks;
webrtc::ReportBlock block = kReportBlock;
report_blocks.push_back(block); // Has wrong SSRC.
block.source_SSRC = kSsrc;
report_blocks.push_back(block); // Correct block.
block.fraction_lost = 0;
report_blocks.push_back(block); // Duplicate SSRC, bad fraction_lost.
EXPECT_TRUE(channel_proxy_);
EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
.WillRepeatedly(Return(kCallStats));
EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
.WillRepeatedly(Return(report_blocks));
EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
.WillRepeatedly(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_))
.WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
EXPECT_CALL(voice_engine_, GetEcMetricsStatus(_))
.WillRepeatedly(DoAll(SetArgReferee<0>(true), Return(0)));
EXPECT_CALL(voice_engine_, GetEchoMetrics(_, _, _, _))
.WillRepeatedly(DoAll(SetArgReferee<0>(kEchoReturnLoss),
SetArgReferee<1>(kEchoReturnLossEnhancement),
Return(0)));
EXPECT_CALL(voice_engine_, GetEcDelayMetrics(_, _, _))
.WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian),
SetArgReferee<1>(kEchoDelayStdDev), Return(0)));
}
private:
testing::StrictMock<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
AudioSendStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
CallStats call_stats_;
testing::NiceMock<MockBitrateObserver> bitrate_observer_;
rtc::scoped_ptr<ProcessThread> process_thread_;
CongestionController congestion_controller_;
};
} // namespace
TEST(AudioSendStreamTest, ConfigToString) {
AudioSendStream::Config config(nullptr);
config.rtp.ssrc = kSsrc;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
config.rtp.c_name = kCName;
config.voe_channel_id = kChannelId;
config.cng_payload_type = 42;
config.red_payload_type = 17;
EXPECT_EQ(
"{rtp: {ssrc: 1234, extensions: [{name: "
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
"c_name: foo_name}, voe_channel_id: 1, cng_payload_type: 42, "
"red_payload_type: 17}",
config.ToString());
}
TEST(AudioSendStreamTest, ConstructDestruct) {
ConfigHelper helper;
internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
helper.congestion_controller());
}
TEST(AudioSendStreamTest, SendTelephoneEvent) {
ConfigHelper helper;
internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
helper.congestion_controller());
helper.SetupMockForSendTelephoneEvent();
EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
kTelephoneEventCode, kTelephoneEventDuration));
}
TEST(AudioSendStreamTest, GetStats) {
ConfigHelper helper;
internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
helper.congestion_controller());
helper.SetupMockForGetStats();
AudioSendStream::Stats stats = send_stream.GetStats();
EXPECT_EQ(kSsrc, stats.local_ssrc);
EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost),
stats.packets_lost);
EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
stats.ext_seqnum);
EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
(kCodecInst.plfreq / 1000)),
stats.jitter_ms);
EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
EXPECT_EQ(static_cast<int32_t>(kSpeechInputLevel), stats.audio_level);
EXPECT_EQ(-1, stats.aec_quality_min);
EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms);
EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms);
EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss);
EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement);
EXPECT_FALSE(stats.typing_noise_detected);
}
TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
ConfigHelper helper;
internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
helper.congestion_controller());
helper.SetupMockForGetStats();
EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
internal::AudioState* internal_audio_state =
static_cast<internal::AudioState*>(helper.audio_state().get());
VoiceEngineObserver* voe_observer =
static_cast<VoiceEngineObserver*>(internal_audio_state);
voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
}
} // namespace test
} // namespace webrtc
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