blob: aac7c0f1c1a7c9f87d611024111af8f0435cca3c [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 "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h"
#include "webrtc/audio/conversion.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/mock_voe_channel_proxy.h"
#include "webrtc/test/mock_voice_engine.h"
namespace webrtc {
namespace test {
namespace {
using testing::_;
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_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 kAbsoluteSendTimeLength = 4;
const int kAbsSendTimeId = 2;
const int kAudioLevelId = 3;
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, -198, -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() {
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);
return channel_proxy_;
}));
EXPECT_CALL(voice_engine_,
SetReceiveAbsoluteSenderTimeStatus(kChannelId, true, kAbsSendTimeId))
.WillOnce(Return(0));
EXPECT_CALL(voice_engine_,
SetReceiveAudioLevelIndicationStatus(kChannelId, true, kAudioLevelId))
.WillOnce(Return(0));
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));
}
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_; }
void SetupMockForGetStats() {
using testing::DoAll;
using testing::SetArgPointee;
using testing::SetArgReferee;
EXPECT_CALL(voice_engine_, GetRTCPStatistics(kChannelId, _))
.WillOnce(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
.WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
EXPECT_CALL(voice_engine_, GetDelayEstimate(kChannelId, _, _))
.WillOnce(DoAll(SetArgPointee<1>(kJitterBufferDelay),
SetArgPointee<2>(kPlayoutBufferDelay), Return(0)));
EXPECT_CALL(voice_engine_,
GetSpeechOutputLevelFullRange(kChannelId, _)).WillOnce(
DoAll(SetArgReferee<1>(kSpeechOutputLevel), Return(0)));
EXPECT_CALL(voice_engine_, GetNetworkStatistics(kChannelId, _))
.WillOnce(DoAll(SetArgReferee<1>(kNetworkStats), Return(0)));
EXPECT_CALL(voice_engine_, GetDecodingCallStatistics(kChannelId, _))
.WillOnce(DoAll(SetArgPointee<1>(kAudioDecodeStats), Return(0)));
}
private:
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 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 uint32_t kPosLength = 2;
ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength,
kAbsoluteSendTimeLength / 4);
const uint8_t kLengthOfData = 3;
buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1);
ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian(
buffer + kRtpOneByteHeaderLength + 1, abs_send_time);
}
size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
int extension_id,
uint32_t abs_send_time) {
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;
}
} // namespace
TEST(AudioReceiveStreamTest, ConfigToString) {
AudioReceiveStream::Config config;
config.rtp.remote_ssrc = kRemoteSsrc;
config.rtp.local_ssrc = kLocalSsrc;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
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: 2}]}, "
"receive_transport: nullptr, rtcp_send_transport: nullptr, "
"voe_channel_id: 2, combined_audio_video_bwe: true}",
config.ToString());
}
TEST(AudioReceiveStreamTest, ConstructDestruct) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
}
TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
ConfigHelper helper;
helper.config().combined_audio_video_bwe = true;
internal::AudioReceiveStream recv_stream(
helper.remote_bitrate_estimator(), helper.config(), helper.audio_state());
uint8_t rtp_packet[30];
const int kAbsSendTimeValue = 1234;
CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
PacketTime packet_time(5678000, 0);
const size_t kExpectedHeaderLength = 20;
EXPECT_CALL(*helper.remote_bitrate_estimator(),
IncomingPacket(packet_time.timestamp / 1000,
sizeof(rtp_packet) - kExpectedHeaderLength,
testing::_, false))
.Times(1);
EXPECT_TRUE(
recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
}
TEST(AudioReceiveStreamTest, GetStats) {
ConfigHelper helper;
internal::AudioReceiveStream recv_stream(
helper.remote_bitrate_estimator(), 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);
}
} // namespace test
} // namespace webrtc