blob: 473b3877804ce31bdb45c7b9a70107270845234c [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 "audio/audio_receive_stream.h"
#include <map>
#include <string>
#include <utility>
#include <vector>
#include "api/test/mock_audio_mixer.h"
#include "api/test/mock_frame_decryptor.h"
#include "audio/conversion.h"
#include "audio/mock_voe_channel_proxy.h"
#include "call/rtp_stream_receiver_controller.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "rtc_base/time_utils.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder_factory.h"
#include "test/mock_transport.h"
namespace webrtc {
namespace test {
namespace {
using ::testing::_;
using ::testing::FloatEq;
using ::testing::Return;
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_neteq_plc = 123;
audio_decode_stats.decoded_codec_plc = 124;
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 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 int64_t kPlayoutNtpTimestampMs = 5678;
const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 78, 890, 123};
const std::pair<int, SdpAudioFormat> kReceiveCodec = {
123,
{"codec_name_recv", 96000, 0}};
const NetworkStatistics kNetworkStats = {
123, 456, false, 789012, 3456, 123, 456, 789, 543, 432,
321, 123, 101, 0, {}, 789, 12, 345, 678, 901,
0, -1, -1, -1, -1, 0, 0, 0, 0};
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
ConfigHelper() : ConfigHelper(new rtc::RefCountedObject<MockAudioMixer>()) {}
explicit ConfigHelper(rtc::scoped_refptr<MockAudioMixer> audio_mixer)
: audio_mixer_(audio_mixer) {
using ::testing::Invoke;
AudioState::Config config;
config.audio_mixer = audio_mixer_;
config.audio_processing = new rtc::RefCountedObject<MockAudioProcessing>();
config.audio_device_module =
new rtc::RefCountedObject<testing::NiceMock<MockAudioDeviceModule>>();
audio_state_ = AudioState::Create(config);
channel_receive_ = new ::testing::StrictMock<MockChannelReceive>();
EXPECT_CALL(*channel_receive_, SetNACKStatus(true, 15)).Times(1);
EXPECT_CALL(*channel_receive_,
RegisterReceiverCongestionControlObjects(&packet_router_))
.Times(1);
EXPECT_CALL(*channel_receive_, ResetReceiverCongestionControlObjects())
.Times(1);
EXPECT_CALL(*channel_receive_, SetAssociatedSendChannel(nullptr)).Times(1);
EXPECT_CALL(*channel_receive_, SetReceiveCodecs(_))
.WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
EXPECT_THAT(codecs, ::testing::IsEmpty());
}));
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_.rtcp_send_transport = &rtcp_send_transport_;
stream_config_.decoder_factory =
new rtc::RefCountedObject<MockAudioDecoderFactory>;
}
std::unique_ptr<internal::AudioReceiveStream> CreateAudioReceiveStream() {
return std::unique_ptr<internal::AudioReceiveStream>(
new internal::AudioReceiveStream(
Clock::GetRealTimeClock(), &rtp_stream_receiver_controller_,
&packet_router_, stream_config_, audio_state_, &event_log_,
std::unique_ptr<voe::ChannelReceiveInterface>(channel_receive_)));
}
AudioReceiveStream::Config& config() { return stream_config_; }
rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; }
MockChannelReceive* channel_receive() { return channel_receive_; }
void SetupMockForGetStats() {
using ::testing::DoAll;
using ::testing::SetArgPointee;
ASSERT_TRUE(channel_receive_);
EXPECT_CALL(*channel_receive_, GetRTCPStatistics())
.WillOnce(Return(kCallStats));
EXPECT_CALL(*channel_receive_, GetDelayEstimate())
.WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
EXPECT_CALL(*channel_receive_, GetSpeechOutputLevelFullRange())
.WillOnce(Return(kSpeechOutputLevel));
EXPECT_CALL(*channel_receive_, GetTotalOutputEnergy())
.WillOnce(Return(kTotalOutputEnergy));
EXPECT_CALL(*channel_receive_, GetTotalOutputDuration())
.WillOnce(Return(kTotalOutputDuration));
EXPECT_CALL(*channel_receive_, GetNetworkStatistics())
.WillOnce(Return(kNetworkStats));
EXPECT_CALL(*channel_receive_, GetDecodingCallStatistics())
.WillOnce(Return(kAudioDecodeStats));
EXPECT_CALL(*channel_receive_, GetReceiveCodec())
.WillOnce(Return(kReceiveCodec));
EXPECT_CALL(*channel_receive_, GetCurrentEstimatedPlayoutNtpTimestampMs(_))
.WillOnce(Return(kPlayoutNtpTimestampMs));
}
private:
PacketRouter packet_router_;
MockRtcEventLog event_log_;
rtc::scoped_refptr<AudioState> audio_state_;
rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
AudioReceiveStream::Config stream_config_;
::testing::StrictMock<MockChannelReceive>* channel_receive_ = nullptr;
RtpStreamReceiverController rtp_stream_receiver_controller_;
MockTransport rtcp_send_transport_;
};
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::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, media_transport_config: {media_transport: "
"null}}",
config.ToString());
}
TEST(AudioReceiveStreamTest, ConstructDestruct) {
ConfigHelper helper;
auto recv_stream = helper.CreateAudioReceiveStream();
}
TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
ConfigHelper helper;
helper.config().rtp.transport_cc = true;
auto recv_stream = helper.CreateAudioReceiveStream();
const int kTransportSequenceNumberValue = 1234;
std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
constexpr int64_t packet_time_us = 5678000;
RtpPacketReceived parsed_packet;
ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
EXPECT_CALL(*helper.channel_receive(),
OnRtpPacket(::testing::Ref(parsed_packet)));
recv_stream->OnRtpPacket(parsed_packet);
}
TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
ConfigHelper helper;
helper.config().rtp.transport_cc = true;
auto recv_stream = helper.CreateAudioReceiveStream();
std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
EXPECT_CALL(*helper.channel_receive(),
ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
.WillOnce(Return());
recv_stream->DeliverRtcp(&rtcp_packet[0], rtcp_packet.size());
}
TEST(AudioReceiveStreamTest, GetStats) {
ConfigHelper helper;
auto recv_stream = helper.CreateAudioReceiveStream();
helper.SetupMockForGetStats();
AudioReceiveStream::Stats stats = recv_stream->GetStats();
EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
EXPECT_EQ(kCallStats.payload_bytes_rcvd, stats.payload_bytes_rcvd);
EXPECT_EQ(kCallStats.header_and_padding_bytes_rcvd,
stats.header_and_padding_bytes_rcvd);
EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
stats.packets_rcvd);
EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
EXPECT_EQ(kReceiveCodec.second.name, stats.codec_name);
EXPECT_EQ(
kCallStats.jitterSamples / (kReceiveCodec.second.clockrate_hz / 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(static_cast<double>(kNetworkStats.jitterBufferDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec),
stats.jitter_buffer_delay_seconds);
EXPECT_EQ(kNetworkStats.jitterBufferEmittedCount,
stats.jitter_buffer_emitted_count);
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_neteq_plc, stats.decoding_plc);
EXPECT_EQ(kAudioDecodeStats.decoded_codec_plc, stats.decoding_codec_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);
EXPECT_EQ(kPlayoutNtpTimestampMs, stats.estimated_playout_ntp_timestamp_ms);
}
TEST(AudioReceiveStreamTest, SetGain) {
ConfigHelper helper;
auto recv_stream = helper.CreateAudioReceiveStream();
EXPECT_CALL(*helper.channel_receive(),
SetChannelOutputVolumeScaling(FloatEq(0.765f)));
recv_stream->SetGain(0.765f);
}
TEST(AudioReceiveStreamTest, StreamsShouldBeAddedToMixerOnceOnStart) {
ConfigHelper helper1;
ConfigHelper helper2(helper1.audio_mixer());
auto recv_stream1 = helper1.CreateAudioReceiveStream();
auto recv_stream2 = helper2.CreateAudioReceiveStream();
EXPECT_CALL(*helper1.channel_receive(), StartPlayout()).Times(1);
EXPECT_CALL(*helper2.channel_receive(), StartPlayout()).Times(1);
EXPECT_CALL(*helper1.channel_receive(), StopPlayout()).Times(1);
EXPECT_CALL(*helper2.channel_receive(), StopPlayout()).Times(1);
EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream1.get()))
.WillOnce(Return(true));
EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream2.get()))
.WillOnce(Return(true));
EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream1.get()))
.Times(1);
EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream2.get()))
.Times(1);
recv_stream1->Start();
recv_stream2->Start();
// One more should not result in any more mixer sources added.
recv_stream1->Start();
// Stop stream before it is being destructed.
recv_stream2->Stop();
}
TEST(AudioReceiveStreamTest, ReconfigureWithSameConfig) {
ConfigHelper helper;
auto recv_stream = helper.CreateAudioReceiveStream();
recv_stream->Reconfigure(helper.config());
}
TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
ConfigHelper helper;
auto recv_stream = helper.CreateAudioReceiveStream();
auto new_config = helper.config();
new_config.rtp.nack.rtp_history_ms = 300 + 20;
new_config.rtp.extensions.clear();
new_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1));
new_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberId + 1));
new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));
MockChannelReceive& channel_receive = *helper.channel_receive();
EXPECT_CALL(channel_receive, SetNACKStatus(true, 15 + 1)).Times(1);
EXPECT_CALL(channel_receive, SetReceiveCodecs(new_config.decoder_map));
recv_stream->Reconfigure(new_config);
}
TEST(AudioReceiveStreamTest, ReconfigureWithFrameDecryptor) {
ConfigHelper helper;
auto recv_stream = helper.CreateAudioReceiveStream();
auto new_config_0 = helper.config();
rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_0(
new rtc::RefCountedObject<MockFrameDecryptor>());
new_config_0.frame_decryptor = mock_frame_decryptor_0;
recv_stream->Reconfigure(new_config_0);
auto new_config_1 = helper.config();
rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_1(
new rtc::RefCountedObject<MockFrameDecryptor>());
new_config_1.frame_decryptor = mock_frame_decryptor_1;
new_config_1.crypto_options.sframe.require_frame_encryption = true;
recv_stream->Reconfigure(new_config_1);
}
} // namespace test
} // namespace webrtc