blob: aed1d1ad20b61fc5d566ef9fb321203e6200f454 [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 <string>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
#include "webrtc/modules/pacing/packet_router.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
#include "webrtc/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;
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 kAbsSendTimeId = 2;
const int kAudioLevelId = 3;
const int kTransportSequenceNumberId = 4;
const int kJitterBufferDelay = -7;
const int kPlayoutBufferDelay = 302;
const unsigned int kSpeechOutputLevel = 99;
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, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
ConfigHelper()
: simulated_clock_(123456),
decoder_factory_(new rtc::RefCountedObject<MockAudioDecoderFactory>),
congestion_controller_(&simulated_clock_,
&bitrate_observer_,
&remote_bitrate_observer_) {
using testing::Invoke;
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_, SetLocalSSRC(kLocalSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
EXPECT_CALL(*channel_proxy_,
SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
.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(congestion_controller_, packet_router())
.WillOnce(Return(&packet_router_));
EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
.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_));
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::kAbsSendTimeUri, kAbsSendTimeId));
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_;
}
MockCongestionController* congestion_controller() {
return &congestion_controller_;
}
MockRemoteBitrateEstimator* remote_bitrate_estimator() {
return &remote_bitrate_estimator_;
}
AudioReceiveStream::Config& config() { return stream_config_; }
rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
MockVoiceEngine& voice_engine() { return voice_engine_; }
MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
void SetupMockForBweFeedback(bool send_side_bwe) {
EXPECT_CALL(congestion_controller_,
GetRemoteBitrateEstimator(send_side_bwe))
.WillOnce(Return(&remote_bitrate_estimator_));
EXPECT_CALL(remote_bitrate_estimator_,
RemoveStream(stream_config_.rtp.remote_ssrc));
}
void SetupMockForGetStats() {
using testing::DoAll;
using testing::SetArgReferee;
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_, GetNetworkStatistics())
.WillOnce(Return(kNetworkStats));
EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
.WillOnce(Return(kAudioDecodeStats));
EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
.WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
}
private:
SimulatedClock simulated_clock_;
PacketRouter packet_router_;
testing::NiceMock<MockCongestionObserver> bitrate_observer_;
testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
MockCongestionController congestion_controller_;
MockRemoteBitrateEstimator remote_bitrate_estimator_;
testing::StrictMock<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
AudioReceiveStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
};
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.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
config.voe_channel_id = kChannelId;
EXPECT_EQ(
"{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, "
"nack: {rtp_history_ms: 0}, extensions: [{uri: "
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
"rtcp_send_transport: nullptr, "
"voe_channel_id: 2}",
config.ToString());
}
TEST(AudioReceiveStreamTest, ConstructDestruct) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
helper.congestion_controller(), helper.config(), helper.audio_state());
}
MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
return arg.extension.hasAbsoluteSendTime ==
expected_extension.hasAbsoluteSendTime &&
arg.extension.absoluteSendTime ==
expected_extension.absoluteSendTime &&
arg.extension.hasTransportSequenceNumber ==
expected_extension.hasTransportSequenceNumber &&
arg.extension.transportSequenceNumber ==
expected_extension.transportSequenceNumber;
}
TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
ConfigHelper helper;
helper.config().rtp.transport_cc = true;
helper.SetupMockForBweFeedback(true);
internal::AudioReceiveStream recv_stream(
helper.congestion_controller(), helper.config(), helper.audio_state());
const int kTransportSequenceNumberValue = 1234;
std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
PacketTime packet_time(5678000, 0);
const size_t kExpectedHeaderLength = 20;
RTPHeaderExtension expected_extension;
expected_extension.hasTransportSequenceNumber = true;
expected_extension.transportSequenceNumber = kTransportSequenceNumberValue;
EXPECT_CALL(*helper.remote_bitrate_estimator(),
IncomingPacket(packet_time.timestamp / 1000,
rtp_packet.size() - kExpectedHeaderLength,
VerifyHeaderExtension(expected_extension)))
.Times(1);
EXPECT_CALL(*helper.channel_proxy(),
ReceivedRTPPacket(&rtp_packet[0],
rtp_packet.size(),
_))
.WillOnce(Return(true));
EXPECT_TRUE(
recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
}
TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
ConfigHelper helper;
helper.config().rtp.transport_cc = true;
helper.SetupMockForBweFeedback(true);
internal::AudioReceiveStream recv_stream(
helper.congestion_controller(), helper.config(), helper.audio_state());
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.congestion_controller(), helper.config(), helper.audio_state());
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(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.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(kCallStats.capture_start_ntp_time_ms_,
stats.capture_start_ntp_time_ms);
}
TEST(AudioReceiveStreamTest, SetGain) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
helper.congestion_controller(), helper.config(), helper.audio_state());
EXPECT_CALL(*helper.channel_proxy(),
SetChannelOutputVolumeScaling(FloatEq(0.765f)));
recv_stream.SetGain(0.765f);
}
} // namespace test
} // namespace webrtc