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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 |
11 #include <map> | 11 #include <map> |
12 #include <string> | 12 #include <string> |
13 #include <vector> | 13 #include <vector> |
14 | 14 |
15 #include "webrtc/api/test/mock_audio_mixer.h" | 15 #include "webrtc/api/test/mock_audio_mixer.h" |
16 #include "webrtc/audio/audio_receive_stream.h" | 16 #include "webrtc/audio/audio_receive_stream.h" |
17 #include "webrtc/audio/conversion.h" | 17 #include "webrtc/audio/conversion.h" |
| 18 #include "webrtc/call/rtp_stream_receiver_controller.h" |
18 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" | 19 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
19 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller
.h" | 20 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller
.h" |
20 #include "webrtc/modules/pacing/packet_router.h" | 21 #include "webrtc/modules/pacing/packet_router.h" |
21 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 22 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
22 #include "webrtc/test/gtest.h" | 23 #include "webrtc/test/gtest.h" |
23 #include "webrtc/test/mock_audio_decoder_factory.h" | 24 #include "webrtc/test/mock_audio_decoder_factory.h" |
24 #include "webrtc/test/mock_voe_channel_proxy.h" | 25 #include "webrtc/test/mock_voe_channel_proxy.h" |
25 #include "webrtc/test/mock_voice_engine.h" | 26 #include "webrtc/test/mock_voice_engine.h" |
26 | 27 |
27 namespace webrtc { | 28 namespace webrtc { |
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130 stream_config_.decoder_factory = decoder_factory_; | 131 stream_config_.decoder_factory = decoder_factory_; |
131 } | 132 } |
132 | 133 |
133 PacketRouter* packet_router() { return &packet_router_; } | 134 PacketRouter* packet_router() { return &packet_router_; } |
134 MockRtcEventLog* event_log() { return &event_log_; } | 135 MockRtcEventLog* event_log() { return &event_log_; } |
135 AudioReceiveStream::Config& config() { return stream_config_; } | 136 AudioReceiveStream::Config& config() { return stream_config_; } |
136 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } | 137 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } |
137 rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; } | 138 rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; } |
138 MockVoiceEngine& voice_engine() { return voice_engine_; } | 139 MockVoiceEngine& voice_engine() { return voice_engine_; } |
139 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } | 140 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } |
| 141 RtpStreamReceiverControllerInterface* rtp_stream_receiver_controller() { |
| 142 return &rtp_stream_receiver_controller_; |
| 143 } |
140 | 144 |
141 void SetupMockForGetStats() { | 145 void SetupMockForGetStats() { |
142 using testing::DoAll; | 146 using testing::DoAll; |
143 using testing::SetArgPointee; | 147 using testing::SetArgPointee; |
144 | 148 |
145 ASSERT_TRUE(channel_proxy_); | 149 ASSERT_TRUE(channel_proxy_); |
146 EXPECT_CALL(*channel_proxy_, GetRTCPStatistics()) | 150 EXPECT_CALL(*channel_proxy_, GetRTCPStatistics()) |
147 .WillOnce(Return(kCallStats)); | 151 .WillOnce(Return(kCallStats)); |
148 EXPECT_CALL(*channel_proxy_, GetDelayEstimate()) | 152 EXPECT_CALL(*channel_proxy_, GetDelayEstimate()) |
149 .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay)); | 153 .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay)); |
150 EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange()) | 154 EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange()) |
151 .WillOnce(Return(kSpeechOutputLevel)); | 155 .WillOnce(Return(kSpeechOutputLevel)); |
152 EXPECT_CALL(*channel_proxy_, GetNetworkStatistics()) | 156 EXPECT_CALL(*channel_proxy_, GetNetworkStatistics()) |
153 .WillOnce(Return(kNetworkStats)); | 157 .WillOnce(Return(kNetworkStats)); |
154 EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics()) | 158 EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics()) |
155 .WillOnce(Return(kAudioDecodeStats)); | 159 .WillOnce(Return(kAudioDecodeStats)); |
156 EXPECT_CALL(*channel_proxy_, GetRecCodec(_)) | 160 EXPECT_CALL(*channel_proxy_, GetRecCodec(_)) |
157 .WillOnce(DoAll(SetArgPointee<0>(kCodecInst), Return(true))); | 161 .WillOnce(DoAll(SetArgPointee<0>(kCodecInst), Return(true))); |
158 } | 162 } |
159 | 163 |
160 private: | 164 private: |
161 PacketRouter packet_router_; | 165 PacketRouter packet_router_; |
162 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 166 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
163 MockRtcEventLog event_log_; | 167 MockRtcEventLog event_log_; |
164 testing::StrictMock<MockVoiceEngine> voice_engine_; | 168 testing::StrictMock<MockVoiceEngine> voice_engine_; |
165 rtc::scoped_refptr<AudioState> audio_state_; | 169 rtc::scoped_refptr<AudioState> audio_state_; |
166 rtc::scoped_refptr<MockAudioMixer> audio_mixer_; | 170 rtc::scoped_refptr<MockAudioMixer> audio_mixer_; |
167 AudioReceiveStream::Config stream_config_; | 171 AudioReceiveStream::Config stream_config_; |
168 testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; | 172 testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; |
| 173 RtpStreamReceiverController rtp_stream_receiver_controller_; |
169 }; | 174 }; |
170 | 175 |
171 void BuildOneByteExtension(std::vector<uint8_t>::iterator it, | 176 void BuildOneByteExtension(std::vector<uint8_t>::iterator it, |
172 int id, | 177 int id, |
173 uint32_t extension_value, | 178 uint32_t extension_value, |
174 size_t value_length) { | 179 size_t value_length) { |
175 const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE; | 180 const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE; |
176 ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId); | 181 ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId); |
177 it += 2; | 182 it += 2; |
178 | 183 |
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231 "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: " | 236 "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: " |
232 "{rtp_history_ms: 0}, extensions: [{uri: " | 237 "{rtp_history_ms: 0}, extensions: [{uri: " |
233 "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, " | 238 "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, " |
234 "rtcp_send_transport: null, voe_channel_id: 2}", | 239 "rtcp_send_transport: null, voe_channel_id: 2}", |
235 config.ToString()); | 240 config.ToString()); |
236 } | 241 } |
237 | 242 |
238 TEST(AudioReceiveStreamTest, ConstructDestruct) { | 243 TEST(AudioReceiveStreamTest, ConstructDestruct) { |
239 ConfigHelper helper; | 244 ConfigHelper helper; |
240 internal::AudioReceiveStream recv_stream( | 245 internal::AudioReceiveStream recv_stream( |
| 246 helper.rtp_stream_receiver_controller(), |
241 helper.packet_router(), | 247 helper.packet_router(), |
242 helper.config(), helper.audio_state(), helper.event_log()); | 248 helper.config(), helper.audio_state(), helper.event_log()); |
243 } | 249 } |
244 | 250 |
245 TEST(AudioReceiveStreamTest, ReceiveRtpPacket) { | 251 TEST(AudioReceiveStreamTest, ReceiveRtpPacket) { |
246 ConfigHelper helper; | 252 ConfigHelper helper; |
247 helper.config().rtp.transport_cc = true; | 253 helper.config().rtp.transport_cc = true; |
248 internal::AudioReceiveStream recv_stream( | 254 internal::AudioReceiveStream recv_stream( |
| 255 helper.rtp_stream_receiver_controller(), |
249 helper.packet_router(), | 256 helper.packet_router(), |
250 helper.config(), helper.audio_state(), helper.event_log()); | 257 helper.config(), helper.audio_state(), helper.event_log()); |
251 const int kTransportSequenceNumberValue = 1234; | 258 const int kTransportSequenceNumberValue = 1234; |
252 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( | 259 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( |
253 kTransportSequenceNumberId, kTransportSequenceNumberValue, 2); | 260 kTransportSequenceNumberId, kTransportSequenceNumberValue, 2); |
254 PacketTime packet_time(5678000, 0); | 261 PacketTime packet_time(5678000, 0); |
255 | 262 |
256 RtpPacketReceived parsed_packet; | 263 RtpPacketReceived parsed_packet; |
257 ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size())); | 264 ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size())); |
258 parsed_packet.set_arrival_time_ms((packet_time.timestamp + 500) / 1000); | 265 parsed_packet.set_arrival_time_ms((packet_time.timestamp + 500) / 1000); |
259 | 266 |
260 EXPECT_CALL(*helper.channel_proxy(), | 267 EXPECT_CALL(*helper.channel_proxy(), |
261 OnRtpPacket(testing::Ref(parsed_packet))); | 268 OnRtpPacket(testing::Ref(parsed_packet))); |
262 | 269 |
263 recv_stream.OnRtpPacket(parsed_packet); | 270 recv_stream.OnRtpPacket(parsed_packet); |
264 } | 271 } |
265 | 272 |
266 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) { | 273 TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) { |
267 ConfigHelper helper; | 274 ConfigHelper helper; |
268 helper.config().rtp.transport_cc = true; | 275 helper.config().rtp.transport_cc = true; |
269 internal::AudioReceiveStream recv_stream( | 276 internal::AudioReceiveStream recv_stream( |
| 277 helper.rtp_stream_receiver_controller(), |
270 helper.packet_router(), | 278 helper.packet_router(), |
271 helper.config(), helper.audio_state(), helper.event_log()); | 279 helper.config(), helper.audio_state(), helper.event_log()); |
272 | 280 |
273 std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport(); | 281 std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport(); |
274 EXPECT_CALL(*helper.channel_proxy(), | 282 EXPECT_CALL(*helper.channel_proxy(), |
275 ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size())) | 283 ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size())) |
276 .WillOnce(Return(true)); | 284 .WillOnce(Return(true)); |
277 EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size())); | 285 EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size())); |
278 } | 286 } |
279 | 287 |
280 TEST(AudioReceiveStreamTest, GetStats) { | 288 TEST(AudioReceiveStreamTest, GetStats) { |
281 ConfigHelper helper; | 289 ConfigHelper helper; |
282 internal::AudioReceiveStream recv_stream( | 290 internal::AudioReceiveStream recv_stream( |
| 291 helper.rtp_stream_receiver_controller(), |
283 helper.packet_router(), | 292 helper.packet_router(), |
284 helper.config(), helper.audio_state(), helper.event_log()); | 293 helper.config(), helper.audio_state(), helper.event_log()); |
285 helper.SetupMockForGetStats(); | 294 helper.SetupMockForGetStats(); |
286 AudioReceiveStream::Stats stats = recv_stream.GetStats(); | 295 AudioReceiveStream::Stats stats = recv_stream.GetStats(); |
287 EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc); | 296 EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc); |
288 EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd); | 297 EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd); |
289 EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived), | 298 EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived), |
290 stats.packets_rcvd); | 299 stats.packets_rcvd); |
291 EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost); | 300 EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost); |
292 EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost); | 301 EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost); |
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318 EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng); | 327 EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng); |
319 EXPECT_EQ(kAudioDecodeStats.decoded_muted_output, | 328 EXPECT_EQ(kAudioDecodeStats.decoded_muted_output, |
320 stats.decoding_muted_output); | 329 stats.decoding_muted_output); |
321 EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_, | 330 EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_, |
322 stats.capture_start_ntp_time_ms); | 331 stats.capture_start_ntp_time_ms); |
323 } | 332 } |
324 | 333 |
325 TEST(AudioReceiveStreamTest, SetGain) { | 334 TEST(AudioReceiveStreamTest, SetGain) { |
326 ConfigHelper helper; | 335 ConfigHelper helper; |
327 internal::AudioReceiveStream recv_stream( | 336 internal::AudioReceiveStream recv_stream( |
| 337 helper.rtp_stream_receiver_controller(), |
328 helper.packet_router(), | 338 helper.packet_router(), |
329 helper.config(), helper.audio_state(), helper.event_log()); | 339 helper.config(), helper.audio_state(), helper.event_log()); |
330 EXPECT_CALL(*helper.channel_proxy(), | 340 EXPECT_CALL(*helper.channel_proxy(), |
331 SetChannelOutputVolumeScaling(FloatEq(0.765f))); | 341 SetChannelOutputVolumeScaling(FloatEq(0.765f))); |
332 recv_stream.SetGain(0.765f); | 342 recv_stream.SetGain(0.765f); |
333 } | 343 } |
334 | 344 |
335 TEST(AudioReceiveStreamTest, StreamShouldNotBeAddedToMixerWhenVoEReturnsError) { | 345 TEST(AudioReceiveStreamTest, StreamShouldNotBeAddedToMixerWhenVoEReturnsError) { |
336 ConfigHelper helper; | 346 ConfigHelper helper; |
337 internal::AudioReceiveStream recv_stream( | 347 internal::AudioReceiveStream recv_stream( |
| 348 helper.rtp_stream_receiver_controller(), |
338 helper.packet_router(), | 349 helper.packet_router(), |
339 helper.config(), helper.audio_state(), helper.event_log()); | 350 helper.config(), helper.audio_state(), helper.event_log()); |
340 | 351 |
341 EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(-1)); | 352 EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(-1)); |
342 EXPECT_CALL(*helper.audio_mixer(), AddSource(_)).Times(0); | 353 EXPECT_CALL(*helper.audio_mixer(), AddSource(_)).Times(0); |
343 | 354 |
344 recv_stream.Start(); | 355 recv_stream.Start(); |
345 } | 356 } |
346 | 357 |
347 TEST(AudioReceiveStreamTest, StreamShouldBeAddedToMixerOnStart) { | 358 TEST(AudioReceiveStreamTest, StreamShouldBeAddedToMixerOnStart) { |
348 ConfigHelper helper; | 359 ConfigHelper helper; |
349 internal::AudioReceiveStream recv_stream( | 360 internal::AudioReceiveStream recv_stream( |
| 361 helper.rtp_stream_receiver_controller(), |
350 helper.packet_router(), | 362 helper.packet_router(), |
351 helper.config(), helper.audio_state(), helper.event_log()); | 363 helper.config(), helper.audio_state(), helper.event_log()); |
352 | 364 |
353 EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0)); | 365 EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0)); |
354 EXPECT_CALL(helper.voice_engine(), StopPlayout(_)); | 366 EXPECT_CALL(helper.voice_engine(), StopPlayout(_)); |
355 EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream)) | 367 EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream)) |
356 .WillOnce(Return(true)); | 368 .WillOnce(Return(true)); |
357 | 369 |
358 recv_stream.Start(); | 370 recv_stream.Start(); |
359 } | 371 } |
360 } // namespace test | 372 } // namespace test |
361 } // namespace webrtc | 373 } // namespace webrtc |
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