blob: b79ae19a6c820244ee002e9deb297835f9ae0b72 [file] [log] [blame]
/*
* 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 <map>
#include <string>
#include <vector>
#include "api/test/mock_audio_mixer.h"
#include "audio/audio_receive_stream.h"
#include "audio/conversion.h"
#include "call/rtp_stream_receiver_controller.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder_factory.h"
#include "test/mock_voe_channel_proxy.h"
#include "test/mock_voice_engine.h"
namespace webrtc {
namespace test {
namespace {
using testing::_;
using testing::FloatEq;
using testing::Return;
using testing::ReturnRef;
AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
AudioDecodingCallStats audio_decode_stats;
audio_decode_stats.calls_to_silence_generator = 234;
audio_decode_stats.calls_to_neteq = 567;
audio_decode_stats.decoded_normal = 890;
audio_decode_stats.decoded_plc = 123;
audio_decode_stats.decoded_cng = 456;
audio_decode_stats.decoded_plc_cng = 789;
audio_decode_stats.decoded_muted_output = 987;
return audio_decode_stats;
}
const int kChannelId = 2;
const uint32_t kRemoteSsrc = 1234;
const uint32_t kLocalSsrc = 5678;
const size_t kOneByteExtensionHeaderLength = 4;
const size_t kOneByteExtensionLength = 4;
const int kAudioLevelId = 3;
const int kTransportSequenceNumberId = 4;
const int kJitterBufferDelay = -7;
const int kPlayoutBufferDelay = 302;
const unsigned int kSpeechOutputLevel = 99;
const double kTotalOutputEnergy = 0.25;
const double kTotalOutputDuration = 0.5;
const CallStatistics kCallStats = {
345, 678, 901, 234, -12, 3456, 7890, 567, 890, 123};
const CodecInst kCodecInst = {
123, "codec_name_recv", 96000, -187, 0, -103};
const NetworkStatistics kNetworkStats = {123, 456, false, 789012, 3456, 123, 0,
{}, 789, 12, 345, 678, 901, 0,
-1, -1, -1, -1, -1, 0};
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
ConfigHelper()
: decoder_factory_(new rtc::RefCountedObject<MockAudioDecoderFactory>),
audio_mixer_(new rtc::RefCountedObject<MockAudioMixer>()) {
using testing::Invoke;
EXPECT_CALL(voice_engine_,
RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
EXPECT_CALL(voice_engine_,
DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
EXPECT_CALL(voice_engine_, audio_device_module());
EXPECT_CALL(voice_engine_, audio_transport());
AudioState::Config config;
config.voice_engine = &voice_engine_;
config.audio_mixer = audio_mixer_;
config.audio_processing = new rtc::RefCountedObject<MockAudioProcessing>();
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_, SetLocalSSRC(kLocalSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
EXPECT_CALL(*channel_proxy_,
SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
EnableReceiveTransportSequenceNumber(kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_proxy_,
RegisterReceiverCongestionControlObjects(&packet_router_))
.Times(1);
EXPECT_CALL(*channel_proxy_, ResetReceiverCongestionControlObjects())
.Times(1);
EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
.Times(1);
EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
.Times(1);
EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory())
.WillOnce(ReturnRef(decoder_factory_));
testing::Expectation expect_set =
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_))
.Times(1);
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
.Times(1)
.After(expect_set);
EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_))
.WillRepeatedly(
Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
EXPECT_THAT(codecs, testing::IsEmpty());
}));
return channel_proxy_;
}));
stream_config_.voe_channel_id = kChannelId;
stream_config_.rtp.local_ssrc = kLocalSsrc;
stream_config_.rtp.remote_ssrc = kRemoteSsrc;
stream_config_.rtp.nack.rtp_history_ms = 300;
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
stream_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
stream_config_.decoder_factory = decoder_factory_;
}
PacketRouter* packet_router() { return &packet_router_; }
MockRtcEventLog* event_log() { return &event_log_; }
AudioReceiveStream::Config& config() { return stream_config_; }
rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; }
MockVoiceEngine& voice_engine() { return voice_engine_; }
MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
RtpStreamReceiverControllerInterface* rtp_stream_receiver_controller() {
return &rtp_stream_receiver_controller_;
}
void SetupMockForGetStats() {
using testing::DoAll;
using testing::SetArgPointee;
ASSERT_TRUE(channel_proxy_);
EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
.WillOnce(Return(kCallStats));
EXPECT_CALL(*channel_proxy_, GetDelayEstimate())
.WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange())
.WillOnce(Return(kSpeechOutputLevel));
EXPECT_CALL(*channel_proxy_, GetTotalOutputEnergy())
.WillOnce(Return(kTotalOutputEnergy));
EXPECT_CALL(*channel_proxy_, GetTotalOutputDuration())
.WillOnce(Return(kTotalOutputDuration));
EXPECT_CALL(*channel_proxy_, GetNetworkStatistics())
.WillOnce(Return(kNetworkStats));
EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
.WillOnce(Return(kAudioDecodeStats));
EXPECT_CALL(*channel_proxy_, GetRecCodec(_))
.WillOnce(DoAll(SetArgPointee<0>(kCodecInst), Return(true)));
}
private:
PacketRouter packet_router_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
MockRtcEventLog event_log_;
testing::StrictMock<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
AudioReceiveStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
RtpStreamReceiverController rtp_stream_receiver_controller_;
};
void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
int id,
uint32_t extension_value,
size_t value_length) {
const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId);
it += 2;
ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4);
it += 2;
const size_t kExtensionDataLength = kOneByteExtensionLength - 1;
uint32_t shifted_value = extension_value
<< (8 * (kExtensionDataLength - value_length));
*it = (id << 4) + (static_cast<uint8_t>(value_length) - 1);
++it;
ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it),
shifted_value);
}
const std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
int extension_id,
uint32_t extension_value,
size_t value_length) {
std::vector<uint8_t> header;
header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength +
kOneByteExtensionLength);
header[0] = 0x80; // Version 2.
header[0] |= 0x10; // Set extension bit.
header[1] = 100; // Payload type.
header[1] |= 0x80; // Marker bit is set.
ByteWriter<uint16_t>::WriteBigEndian(&header[2], 0x1234); // Sequence number.
ByteWriter<uint32_t>::WriteBigEndian(&header[4], 0x5678); // Timestamp.
ByteWriter<uint32_t>::WriteBigEndian(&header[8], 0x4321); // SSRC.
BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id,
extension_value, value_length);
return header;
}
const std::vector<uint8_t> CreateRtcpSenderReport() {
std::vector<uint8_t> packet;
const size_t kRtcpSrLength = 28; // In bytes.
packet.resize(kRtcpSrLength);
packet[0] = 0x80; // Version 2.
packet[1] = 0xc8; // PT = 200, SR.
// Length in number of 32-bit words - 1.
ByteWriter<uint16_t>::WriteBigEndian(&packet[2], 6);
ByteWriter<uint32_t>::WriteBigEndian(&packet[4], kLocalSsrc);
return packet;
}
} // namespace
TEST(AudioReceiveStreamTest, ConfigToString) {
AudioReceiveStream::Config config;
config.rtp.remote_ssrc = kRemoteSsrc;
config.rtp.local_ssrc = kLocalSsrc;
config.voe_channel_id = kChannelId;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
EXPECT_EQ(
"{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
"{rtp_history_ms: 0}, extensions: [{uri: "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
"rtcp_send_transport: null, voe_channel_id: 2}",
config.ToString());
}
TEST(AudioReceiveStreamTest, ConstructDestruct) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
helper.rtp_stream_receiver_controller(),
helper.packet_router(),
helper.config(), helper.audio_state(), helper.event_log());
}
TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
ConfigHelper helper;
helper.config().rtp.transport_cc = true;
internal::AudioReceiveStream recv_stream(
helper.rtp_stream_receiver_controller(),
helper.packet_router(),
helper.config(), helper.audio_state(), helper.event_log());
const int kTransportSequenceNumberValue = 1234;
std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
PacketTime packet_time(5678000, 0);
RtpPacketReceived parsed_packet;
ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
parsed_packet.set_arrival_time_ms((packet_time.timestamp + 500) / 1000);
EXPECT_CALL(*helper.channel_proxy(),
OnRtpPacket(testing::Ref(parsed_packet)));
recv_stream.OnRtpPacket(parsed_packet);
}
TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
ConfigHelper helper;
helper.config().rtp.transport_cc = true;
internal::AudioReceiveStream recv_stream(
helper.rtp_stream_receiver_controller(),
helper.packet_router(),
helper.config(), helper.audio_state(), helper.event_log());
std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
EXPECT_CALL(*helper.channel_proxy(),
ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
.WillOnce(Return(true));
EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size()));
}
TEST(AudioReceiveStreamTest, GetStats) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
helper.rtp_stream_receiver_controller(),
helper.packet_router(),
helper.config(), helper.audio_state(), helper.event_log());
helper.SetupMockForGetStats();
AudioReceiveStream::Stats stats = recv_stream.GetStats();
EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
stats.packets_rcvd);
EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost);
EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum);
EXPECT_EQ(kCallStats.jitterSamples / (kCodecInst.plfreq / 1000),
stats.jitter_ms);
EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms);
EXPECT_EQ(kNetworkStats.preferredBufferSize,
stats.jitter_buffer_preferred_ms);
EXPECT_EQ(static_cast<uint32_t>(kJitterBufferDelay + kPlayoutBufferDelay),
stats.delay_estimate_ms);
EXPECT_EQ(static_cast<int32_t>(kSpeechOutputLevel), stats.audio_level);
EXPECT_EQ(kTotalOutputEnergy, stats.total_output_energy);
EXPECT_EQ(kNetworkStats.totalSamplesReceived, stats.total_samples_received);
EXPECT_EQ(kTotalOutputDuration, stats.total_output_duration);
EXPECT_EQ(kNetworkStats.concealedSamples, stats.concealed_samples);
EXPECT_EQ(kNetworkStats.concealmentEvents, stats.concealment_events);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
stats.speech_expand_rate);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate),
stats.secondary_decoded_rate);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDiscardedRate),
stats.secondary_discarded_rate);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate),
stats.accelerate_rate);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
stats.preemptive_expand_rate);
EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
stats.decoding_calls_to_silence_generator);
EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
EXPECT_EQ(kAudioDecodeStats.decoded_muted_output,
stats.decoding_muted_output);
EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
stats.capture_start_ntp_time_ms);
}
TEST(AudioReceiveStreamTest, SetGain) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
helper.rtp_stream_receiver_controller(),
helper.packet_router(),
helper.config(), helper.audio_state(), helper.event_log());
EXPECT_CALL(*helper.channel_proxy(),
SetChannelOutputVolumeScaling(FloatEq(0.765f)));
recv_stream.SetGain(0.765f);
}
TEST(AudioReceiveStreamTest, StreamShouldNotBeAddedToMixerWhenVoEReturnsError) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
helper.rtp_stream_receiver_controller(),
helper.packet_router(),
helper.config(), helper.audio_state(), helper.event_log());
EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(-1));
EXPECT_CALL(*helper.audio_mixer(), AddSource(_)).Times(0);
recv_stream.Start();
}
TEST(AudioReceiveStreamTest, StreamShouldBeAddedToMixerOnStart) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
helper.rtp_stream_receiver_controller(),
helper.packet_router(),
helper.config(), helper.audio_state(), helper.event_log());
EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0));
EXPECT_CALL(helper.voice_engine(), StopPlayout(_));
EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream))
.WillOnce(Return(true));
recv_stream.Start();
}
} // namespace test
} // namespace webrtc