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-rw-r--r--webrtc/audio/audio_receive_stream_unittest.cc364
1 files changed, 265 insertions, 99 deletions
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 4e267f1738..eb008b3045 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -8,154 +8,320 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <string>
+
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h"
#include "webrtc/audio/conversion.h"
+#include "webrtc/call/mock/mock_congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_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/test/fake_voice_engine.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
+#include "webrtc/system_wrappers/include/clock.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 webrtc::ByteWriter;
+using testing::_;
+using testing::Return;
-const size_t kAbsoluteSendTimeLength = 4;
+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;
+}
-void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
- int id,
- uint32_t abs_send_time) {
- const size_t kRtpOneByteHeaderLength = 4;
- const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
- ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId);
+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),
+ call_stats_(&simulated_clock_),
+ congestion_controller_(&process_thread_,
+ &call_stats_,
+ &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_,
+ SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_,
+ SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(
+ nullptr, nullptr, &packet_router_))
+ .Times(1);
+ EXPECT_CALL(congestion_controller_, packet_router())
+ .WillOnce(Return(&packet_router_));
+ EXPECT_CALL(*channel_proxy_,
+ SetCongestionControlObjects(nullptr, nullptr, nullptr))
+ .Times(1);
+ return channel_proxy_;
+ }));
+ stream_config_.voe_channel_id = kChannelId;
+ stream_config_.rtp.local_ssrc = kLocalSsrc;
+ stream_config_.rtp.remote_ssrc = kRemoteSsrc;
+ stream_config_.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+ stream_config_.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+ }
+
+ 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_; }
- const uint32_t kPosLength = 2;
- ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength,
- kAbsoluteSendTimeLength / 4);
+ 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));
+ }
- const uint8_t kLengthOfData = 3;
- buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1);
- ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian(
- buffer + kRtpOneByteHeaderLength + 1, abs_send_time);
+ 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_;
+ CallStats call_stats_;
+ PacketRouter packet_router_;
+ testing::NiceMock<MockBitrateObserver> bitrate_observer_;
+ testing::NiceMock<MockProcessThread> process_thread_;
+ 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) + (value_length - 1);
+ ++it;
+ ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it),
+ shifted_value);
}
-size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
- int extension_id,
- uint32_t abs_send_time) {
+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.
- int32_t rtp_header_length = webrtc::kRtpHeaderSize;
-
- BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
- abs_send_time);
- rtp_header_length += kAbsoluteSendTimeLength;
- return rtp_header_length;
+ 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;
}
} // namespace
-namespace webrtc {
-namespace test {
-
TEST(AudioReceiveStreamTest, ConfigToString) {
- const int kAbsSendTimeId = 3;
AudioReceiveStream::Config config;
- config.rtp.remote_ssrc = 1234;
- config.rtp.local_ssrc = 5678;
+ config.rtp.remote_ssrc = kRemoteSsrc;
+ config.rtp.local_ssrc = kLocalSsrc;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- config.voe_channel_id = 1;
+ config.voe_channel_id = kChannelId;
config.combined_audio_video_bwe = true;
- EXPECT_EQ("{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
- "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, "
+ EXPECT_EQ(
+ "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
+ "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
"receive_transport: nullptr, rtcp_send_transport: nullptr, "
- "voe_channel_id: 1, combined_audio_video_bwe: true}", config.ToString());
+ "voe_channel_id: 2, combined_audio_video_bwe: true}",
+ config.ToString());
}
TEST(AudioReceiveStreamTest, ConstructDestruct) {
- MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
- AudioReceiveStream::Config config;
- config.voe_channel_id = 1;
- internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
- &voice_engine);
+ 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, AudioPacketUpdatesBweWithTimestamp) {
- MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
- AudioReceiveStream::Config config;
- config.combined_audio_video_bwe = true;
- config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
- const int kAbsSendTimeId = 3;
- config.rtp.extensions.push_back(
- RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
- &voice_engine);
- uint8_t rtp_packet[30];
+ ConfigHelper helper;
+ helper.config().combined_audio_video_bwe = true;
+ helper.SetupMockForBweFeedback(false);
+ internal::AudioReceiveStream recv_stream(
+ helper.congestion_controller(), helper.config(), helper.audio_state());
const int kAbsSendTimeValue = 1234;
- CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
+ std::vector<uint8_t> rtp_packet =
+ CreateRtpHeaderWithOneByteExtension(kAbsSendTimeId, kAbsSendTimeValue, 3);
PacketTime packet_time(5678000, 0);
const size_t kExpectedHeaderLength = 20;
- EXPECT_CALL(remote_bitrate_estimator,
- IncomingPacket(packet_time.timestamp / 1000,
- sizeof(rtp_packet) - kExpectedHeaderLength, testing::_, false))
+ RTPHeaderExtension expected_extension;
+ expected_extension.hasAbsoluteSendTime = true;
+ expected_extension.absoluteSendTime = kAbsSendTimeValue;
+ EXPECT_CALL(*helper.remote_bitrate_estimator(),
+ IncomingPacket(packet_time.timestamp / 1000,
+ rtp_packet.size() - kExpectedHeaderLength,
+ VerifyHeaderExtension(expected_extension), false))
.Times(1);
EXPECT_TRUE(
- recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
+ recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
}
-TEST(AudioReceiveStreamTest, GetStats) {
- MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
- AudioReceiveStream::Config config;
- config.rtp.remote_ssrc = FakeVoiceEngine::kRecvSsrc;
- config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
- internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
- &voice_engine);
+TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
+ ConfigHelper helper;
+ helper.config().combined_audio_video_bwe = true;
+ helper.config().rtp.transport_cc = true;
+ helper.config().rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
+ 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), false))
+ .Times(1);
+ EXPECT_TRUE(
+ recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
+}
+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();
- const CallStatistics& call_stats = FakeVoiceEngine::kRecvCallStats;
- const CodecInst& codec_inst = FakeVoiceEngine::kRecvCodecInst;
- const NetworkStatistics& net_stats = FakeVoiceEngine::kRecvNetworkStats;
- const AudioDecodingCallStats& decode_stats =
- FakeVoiceEngine::kRecvAudioDecodingCallStats;
- EXPECT_EQ(FakeVoiceEngine::kRecvSsrc, stats.remote_ssrc);
- EXPECT_EQ(static_cast<int64_t>(call_stats.bytesReceived), stats.bytes_rcvd);
- EXPECT_EQ(static_cast<uint32_t>(call_stats.packetsReceived),
+ 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(call_stats.cumulativeLost, stats.packets_lost);
- EXPECT_EQ(Q8ToFloat(call_stats.fractionLost), stats.fraction_lost);
- EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
- EXPECT_EQ(call_stats.extendedMax, stats.ext_seqnum);
- EXPECT_EQ(call_stats.jitterSamples / (codec_inst.plfreq / 1000),
+ 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(net_stats.currentBufferSize, stats.jitter_buffer_ms);
- EXPECT_EQ(net_stats.preferredBufferSize, stats.jitter_buffer_preferred_ms);
- EXPECT_EQ(static_cast<uint32_t>(FakeVoiceEngine::kRecvJitterBufferDelay +
- FakeVoiceEngine::kRecvPlayoutBufferDelay), stats.delay_estimate_ms);
- EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kRecvSpeechOutputLevel),
- stats.audio_level);
- EXPECT_EQ(Q14ToFloat(net_stats.currentExpandRate), stats.expand_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentSpeechExpandRate),
+ 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(net_stats.currentSecondaryDecodedRate),
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate),
stats.secondary_decoded_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentAccelerateRate), stats.accelerate_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentPreemptiveRate),
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate),
+ stats.accelerate_rate);
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
stats.preemptive_expand_rate);
- EXPECT_EQ(decode_stats.calls_to_silence_generator,
+ EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
stats.decoding_calls_to_silence_generator);
- EXPECT_EQ(decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq);
- EXPECT_EQ(decode_stats.decoded_normal, stats.decoding_normal);
- EXPECT_EQ(decode_stats.decoded_plc, stats.decoding_plc);
- EXPECT_EQ(decode_stats.decoded_cng, stats.decoding_cng);
- EXPECT_EQ(decode_stats.decoded_plc_cng, stats.decoding_plc_cng);
- EXPECT_EQ(call_stats.capture_start_ntp_time_ms_,
+ 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);
}
} // namespace test
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-14 01:31:39 +0000