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Side by Side Diff: webrtc/audio/audio_receive_stream_unittest.cc

Issue 1414743004: Implement AudioSendStream::GetStats(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: workaround for android build issue Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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54 BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id, 54 BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
55 abs_send_time); 55 abs_send_time);
56 rtp_header_length += kAbsoluteSendTimeLength; 56 rtp_header_length += kAbsoluteSendTimeLength;
57 return rtp_header_length; 57 return rtp_header_length;
58 } 58 }
59 } // namespace 59 } // namespace
60 60
61 namespace webrtc { 61 namespace webrtc {
62 namespace test { 62 namespace test {
63 63
64 TEST(AudioReceiveStreamTest, ConfigToString) {
65 const int kAbsSendTimeId = 3;
66 AudioReceiveStream::Config config;
67 config.rtp.remote_ssrc = 1234;
68 config.rtp.local_ssrc = 5678;
69 config.rtp.extensions.push_back(
70 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
71 config.voe_channel_id = 1;
72 config.combined_audio_video_bwe = true;
73 EXPECT_EQ("{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
74 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, "
75 "receive_transport: nullptr, rtcp_send_transport: nullptr, "
76 "voe_channel_id: 1, combined_audio_video_bwe: true}", config.ToString());
77 }
78
79 TEST(AudioReceiveStreamTest, ConstructDestruct) {
80 MockRemoteBitrateEstimator remote_bitrate_estimator;
81 FakeVoiceEngine voice_engine;
82 AudioReceiveStream::Config config;
83 config.voe_channel_id = 1;
84 internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
85 &voice_engine);
86 }
87
64 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) { 88 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
65 MockRemoteBitrateEstimator remote_bitrate_estimator; 89 MockRemoteBitrateEstimator remote_bitrate_estimator;
66 FakeVoiceEngine voice_engine; 90 FakeVoiceEngine voice_engine;
67 AudioReceiveStream::Config config; 91 AudioReceiveStream::Config config;
68 config.combined_audio_video_bwe = true; 92 config.combined_audio_video_bwe = true;
69 config.voe_channel_id = voice_engine.kReceiveChannelId; 93 config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
70 const int kAbsSendTimeId = 3; 94 const int kAbsSendTimeId = 3;
71 config.rtp.extensions.push_back( 95 config.rtp.extensions.push_back(
72 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); 96 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
73 internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config, 97 internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
74 &voice_engine); 98 &voice_engine);
75 uint8_t rtp_packet[30]; 99 uint8_t rtp_packet[30];
76 const int kAbsSendTimeValue = 1234; 100 const int kAbsSendTimeValue = 1234;
77 CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue); 101 CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
78 PacketTime packet_time(5678000, 0); 102 PacketTime packet_time(5678000, 0);
79 const size_t kExpectedHeaderLength = 20; 103 const size_t kExpectedHeaderLength = 20;
80 EXPECT_CALL(remote_bitrate_estimator, 104 EXPECT_CALL(remote_bitrate_estimator,
81 IncomingPacket(packet_time.timestamp / 1000, 105 IncomingPacket(packet_time.timestamp / 1000,
82 sizeof(rtp_packet) - kExpectedHeaderLength, testing::_, false)) 106 sizeof(rtp_packet) - kExpectedHeaderLength, testing::_, false))
83 .Times(1); 107 .Times(1);
84 EXPECT_TRUE( 108 EXPECT_TRUE(
85 recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time)); 109 recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
86 } 110 }
87 111
88 TEST(AudioReceiveStreamTest, GetStats) { 112 TEST(AudioReceiveStreamTest, GetStats) {
89 const uint32_t kSsrc1 = 667;
90
91 MockRemoteBitrateEstimator remote_bitrate_estimator; 113 MockRemoteBitrateEstimator remote_bitrate_estimator;
92 FakeVoiceEngine voice_engine; 114 FakeVoiceEngine voice_engine;
93 AudioReceiveStream::Config config; 115 AudioReceiveStream::Config config;
94 config.rtp.remote_ssrc = kSsrc1; 116 config.rtp.remote_ssrc = FakeVoiceEngine::kRecvSsrc;
95 config.voe_channel_id = voice_engine.kReceiveChannelId; 117 config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
96 internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config, 118 internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
97 &voice_engine); 119 &voice_engine);
98 120
99 AudioReceiveStream::Stats stats = recv_stream.GetStats(); 121 AudioReceiveStream::Stats stats = recv_stream.GetStats();
100 const CallStatistics& call_stats = voice_engine.GetRecvCallStats(); 122 const CallStatistics& call_stats = FakeVoiceEngine::kRecvCallStats;
101 const CodecInst& codec_inst = voice_engine.GetRecvRecCodecInst(); 123 const CodecInst& codec_inst = FakeVoiceEngine::kRecvCodecInst;
102 const NetworkStatistics& net_stats = voice_engine.GetRecvNetworkStats(); 124 const NetworkStatistics& net_stats = FakeVoiceEngine::kRecvNetworkStats;
103 const AudioDecodingCallStats& decode_stats = 125 const AudioDecodingCallStats& decode_stats =
104 voice_engine.GetRecvAudioDecodingCallStats(); 126 FakeVoiceEngine::kRecvAudioDecodingCallStats;
105 EXPECT_EQ(kSsrc1, stats.remote_ssrc); 127 EXPECT_EQ(FakeVoiceEngine::kRecvSsrc, stats.remote_ssrc);
106 EXPECT_EQ(static_cast<int64_t>(call_stats.bytesReceived), stats.bytes_rcvd); 128 EXPECT_EQ(static_cast<int64_t>(call_stats.bytesReceived), stats.bytes_rcvd);
107 EXPECT_EQ(static_cast<uint32_t>(call_stats.packetsReceived), 129 EXPECT_EQ(static_cast<uint32_t>(call_stats.packetsReceived),
108 stats.packets_rcvd); 130 stats.packets_rcvd);
109 EXPECT_EQ(call_stats.cumulativeLost, stats.packets_lost); 131 EXPECT_EQ(call_stats.cumulativeLost, stats.packets_lost);
110 EXPECT_EQ(static_cast<float>(call_stats.fractionLost) / 256, 132 EXPECT_EQ(Q8ToFloat(call_stats.fractionLost), stats.fraction_lost);
111 stats.fraction_lost);
112 EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name); 133 EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
113 EXPECT_EQ(call_stats.extendedMax, stats.ext_seqnum); 134 EXPECT_EQ(call_stats.extendedMax, stats.ext_seqnum);
114 EXPECT_EQ(call_stats.jitterSamples / (codec_inst.plfreq / 1000), 135 EXPECT_EQ(call_stats.jitterSamples / (codec_inst.plfreq / 1000),
115 stats.jitter_ms); 136 stats.jitter_ms);
116 EXPECT_EQ(net_stats.currentBufferSize, stats.jitter_buffer_ms); 137 EXPECT_EQ(net_stats.currentBufferSize, stats.jitter_buffer_ms);
117 EXPECT_EQ(net_stats.preferredBufferSize, stats.jitter_buffer_preferred_ms); 138 EXPECT_EQ(net_stats.preferredBufferSize, stats.jitter_buffer_preferred_ms);
118 EXPECT_EQ(static_cast<uint32_t>(voice_engine.kRecvJitterBufferDelay + 139 EXPECT_EQ(static_cast<uint32_t>(FakeVoiceEngine::kRecvJitterBufferDelay +
119 voice_engine.kRecvPlayoutBufferDelay), stats.delay_estimate_ms); 140 FakeVoiceEngine::kRecvPlayoutBufferDelay), stats.delay_estimate_ms);
120 EXPECT_EQ(static_cast<int32_t>(voice_engine.kRecvSpeechOutputLevel), 141 EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kRecvSpeechOutputLevel),
121 stats.audio_level); 142 stats.audio_level);
122 EXPECT_EQ(Q14ToFloat(net_stats.currentExpandRate), stats.expand_rate); 143 EXPECT_EQ(Q14ToFloat(net_stats.currentExpandRate), stats.expand_rate);
123 EXPECT_EQ(Q14ToFloat(net_stats.currentSpeechExpandRate), 144 EXPECT_EQ(Q14ToFloat(net_stats.currentSpeechExpandRate),
124 stats.speech_expand_rate); 145 stats.speech_expand_rate);
125 EXPECT_EQ(Q14ToFloat(net_stats.currentSecondaryDecodedRate), 146 EXPECT_EQ(Q14ToFloat(net_stats.currentSecondaryDecodedRate),
126 stats.secondary_decoded_rate); 147 stats.secondary_decoded_rate);
127 EXPECT_EQ(Q14ToFloat(net_stats.currentAccelerateRate), stats.accelerate_rate); 148 EXPECT_EQ(Q14ToFloat(net_stats.currentAccelerateRate), stats.accelerate_rate);
128 EXPECT_EQ(Q14ToFloat(net_stats.currentPreemptiveRate), 149 EXPECT_EQ(Q14ToFloat(net_stats.currentPreemptiveRate),
129 stats.preemptive_expand_rate); 150 stats.preemptive_expand_rate);
130 EXPECT_EQ(decode_stats.calls_to_silence_generator, 151 EXPECT_EQ(decode_stats.calls_to_silence_generator,
131 stats.decoding_calls_to_silence_generator); 152 stats.decoding_calls_to_silence_generator);
132 EXPECT_EQ(decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq); 153 EXPECT_EQ(decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq);
133 EXPECT_EQ(decode_stats.decoded_normal, stats.decoding_normal); 154 EXPECT_EQ(decode_stats.decoded_normal, stats.decoding_normal);
134 EXPECT_EQ(decode_stats.decoded_plc, stats.decoding_plc); 155 EXPECT_EQ(decode_stats.decoded_plc, stats.decoding_plc);
135 EXPECT_EQ(decode_stats.decoded_cng, stats.decoding_cng); 156 EXPECT_EQ(decode_stats.decoded_cng, stats.decoding_cng);
136 EXPECT_EQ(decode_stats.decoded_plc_cng, stats.decoding_plc_cng); 157 EXPECT_EQ(decode_stats.decoded_plc_cng, stats.decoding_plc_cng);
137 EXPECT_EQ(call_stats.capture_start_ntp_time_ms_, 158 EXPECT_EQ(call_stats.capture_start_ntp_time_ms_,
138 stats.capture_start_ntp_time_ms); 159 stats.capture_start_ntp_time_ms);
139 } 160 }
140 } // namespace test 161 } // namespace test
141 } // namespace webrtc 162 } // namespace webrtc
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