| /* |
| * 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 |