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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 <string> | 11 #include <string> |
12 #include <utility> | |
12 #include <vector> | 13 #include <vector> |
13 | 14 |
14 #include "webrtc/audio/audio_send_stream.h" | 15 #include "webrtc/audio/audio_send_stream.h" |
15 #include "webrtc/audio/audio_state.h" | 16 #include "webrtc/audio/audio_state.h" |
16 #include "webrtc/audio/conversion.h" | 17 #include "webrtc/audio/conversion.h" |
17 #include "webrtc/base/task_queue.h" | 18 #include "webrtc/base/task_queue.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" |
20 #include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h" | |
19 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" | 21 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
20 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" | 22 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h" |
21 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" | 23 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
22 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_cont roller.h" | 24 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_cont roller.h" |
23 #include "webrtc/modules/pacing/paced_sender.h" | 25 #include "webrtc/modules/pacing/paced_sender.h" |
24 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitra te_estimator.h" | 26 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitra te_estimator.h" |
25 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" | 27 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" |
26 #include "webrtc/test/gtest.h" | 28 #include "webrtc/test/gtest.h" |
27 #include "webrtc/test/mock_voe_channel_proxy.h" | 29 #include "webrtc/test/mock_voe_channel_proxy.h" |
28 #include "webrtc/test/mock_voice_engine.h" | 30 #include "webrtc/test/mock_voice_engine.h" |
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48 const float kResidualEchoLikelihood = -1.0f; | 50 const float kResidualEchoLikelihood = -1.0f; |
49 const unsigned int kSpeechInputLevel = 96; | 51 const unsigned int kSpeechInputLevel = 96; |
50 const CallStatistics kCallStats = { | 52 const CallStatistics kCallStats = { |
51 1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123}; | 53 1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123}; |
52 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354}; | 54 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354}; |
53 const int kTelephoneEventPayloadType = 123; | 55 const int kTelephoneEventPayloadType = 123; |
54 const int kTelephoneEventPayloadFrequency = 65432; | 56 const int kTelephoneEventPayloadFrequency = 65432; |
55 const int kTelephoneEventCode = 45; | 57 const int kTelephoneEventCode = 45; |
56 const int kTelephoneEventDuration = 6789; | 58 const int kTelephoneEventDuration = 6789; |
57 const CodecInst kIsacCodec = {103, "isac", 16000, 320, 1, 32000}; | 59 const CodecInst kIsacCodec = {103, "isac", 16000, 320, 1, 32000}; |
60 const int kIsacPayloadType = 103; | |
61 const SdpAudioFormat kIsacFormat = {"isac", 16000, 1}; | |
kwiberg-webrtc
2017/03/01 12:26:47
The first one can be constexpr.
It makes me sad t
ossu
2017/03/20 18:19:48
Acknowledged.
| |
58 | 62 |
59 class MockLimitObserver : public BitrateAllocator::LimitObserver { | 63 class MockLimitObserver : public BitrateAllocator::LimitObserver { |
60 public: | 64 public: |
61 MOCK_METHOD2(OnAllocationLimitsChanged, | 65 MOCK_METHOD2(OnAllocationLimitsChanged, |
62 void(uint32_t min_send_bitrate_bps, | 66 void(uint32_t min_send_bitrate_bps, |
63 uint32_t max_padding_bitrate_bps)); | 67 uint32_t max_padding_bitrate_bps)); |
64 }; | 68 }; |
65 | 69 |
66 struct ConfigHelper { | 70 struct ConfigHelper { |
67 explicit ConfigHelper(bool audio_bwe_enabled) | 71 explicit ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call) |
kwiberg-webrtc
2017/03/01 12:26:47
Drop "explicit".
ossu
2017/03/02 01:30:28
Sure. Was in a rush!
| |
68 : simulated_clock_(123456), | 72 : simulated_clock_(123456), |
69 stream_config_(nullptr), | 73 stream_config_(nullptr), |
70 congestion_controller_(&simulated_clock_, | 74 congestion_controller_(&simulated_clock_, |
71 &bitrate_observer_, | 75 &bitrate_observer_, |
72 &remote_bitrate_observer_, | 76 &remote_bitrate_observer_, |
73 &event_log_, | 77 &event_log_, |
74 &packet_router_), | 78 &packet_router_), |
75 bitrate_allocator_(&limit_observer_), | 79 bitrate_allocator_(&limit_observer_), |
76 worker_queue_("ConfigHelper_worker_queue") { | 80 worker_queue_("ConfigHelper_worker_queue") { |
77 using testing::Invoke; | 81 using testing::Invoke; |
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89 config.audio_mixer = AudioMixerImpl::Create(); | 93 config.audio_mixer = AudioMixerImpl::Create(); |
90 audio_state_ = AudioState::Create(config); | 94 audio_state_ = AudioState::Create(config); |
91 | 95 |
92 SetupDefaultChannelProxy(audio_bwe_enabled); | 96 SetupDefaultChannelProxy(audio_bwe_enabled); |
93 | 97 |
94 EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) | 98 EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) |
95 .WillOnce(Invoke([this](int channel_id) { | 99 .WillOnce(Invoke([this](int channel_id) { |
96 return channel_proxy_; | 100 return channel_proxy_; |
97 })); | 101 })); |
98 | 102 |
99 SetupMockForSetupSendCodec(); | 103 SetupMockForSetupSendCodec(expect_set_encoder_call); |
100 | 104 |
101 stream_config_.voe_channel_id = kChannelId; | 105 stream_config_.voe_channel_id = kChannelId; |
102 stream_config_.rtp.ssrc = kSsrc; | 106 stream_config_.rtp.ssrc = kSsrc; |
103 stream_config_.rtp.nack.rtp_history_ms = 200; | 107 stream_config_.rtp.nack.rtp_history_ms = 200; |
104 stream_config_.rtp.c_name = kCName; | 108 stream_config_.rtp.c_name = kCName; |
105 stream_config_.rtp.extensions.push_back( | 109 stream_config_.rtp.extensions.push_back( |
106 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); | 110 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); |
107 if (audio_bwe_enabled) { | 111 if (audio_bwe_enabled) { |
108 stream_config_.rtp.extensions.push_back( | 112 stream_config_.rtp.extensions.push_back( |
109 RtpExtension(RtpExtension::kTransportSequenceNumberUri, | 113 RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
110 kTransportSequenceNumberId)); | 114 kTransportSequenceNumberId)); |
111 stream_config_.send_codec_spec.transport_cc_enabled = true; | 115 stream_config_.send_codec_spec.transport_cc_enabled = true; |
112 } | 116 } |
113 // Use ISAC as default codec so as to prevent unnecessary |voice_engine_| | 117 // Use ISAC as default codec so as to prevent unnecessary |voice_engine_| |
114 // calls from the default ctor behavior. | 118 // calls from the default ctor behavior. |
115 stream_config_.send_codec_spec.codec_inst = kIsacCodec; | 119 stream_config_.send_codec_spec.payload_type = kIsacPayloadType; |
120 stream_config_.send_codec_spec.format = kIsacFormat; | |
121 // TODO(ossu): Mock this factory. | |
122 stream_config_.encoder_factory = CreateBuiltinAudioEncoderFactory(); | |
116 stream_config_.min_bitrate_bps = 10000; | 123 stream_config_.min_bitrate_bps = 10000; |
117 stream_config_.max_bitrate_bps = 65000; | 124 stream_config_.max_bitrate_bps = 65000; |
118 } | 125 } |
119 | 126 |
120 AudioSendStream::Config& config() { return stream_config_; } | 127 AudioSendStream::Config& config() { return stream_config_; } |
121 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } | 128 rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } |
122 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } | 129 MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } |
123 PacketRouter* packet_router() { return &packet_router_; } | 130 PacketRouter* packet_router() { return &packet_router_; } |
124 CongestionController* congestion_controller() { | 131 CongestionController* congestion_controller() { |
125 return &congestion_controller_; | 132 return &congestion_controller_; |
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162 EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)).Times(1); | 169 EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)).Times(1); |
163 EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()).Times(1); | 170 EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport()).Times(1); |
164 EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1); | 171 EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1); |
165 EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull())) | 172 EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull())) |
166 .Times(1); // Destructor resets the event log | 173 .Times(1); // Destructor resets the event log |
167 EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(&rtcp_rtt_stats_)).Times(1); | 174 EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(&rtcp_rtt_stats_)).Times(1); |
168 EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(testing::IsNull())) | 175 EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(testing::IsNull())) |
169 .Times(1); // Destructor resets the rtt stats. | 176 .Times(1); // Destructor resets the rtt stats. |
170 } | 177 } |
171 | 178 |
172 void SetupMockForSetupSendCodec() { | 179 void SetupMockForSetupSendCodec(bool expect_set_encoder_call) { |
173 EXPECT_CALL(*channel_proxy_, SetVADStatus(false)) | 180 if (expect_set_encoder_call) { |
174 .WillOnce(Return(true)); | 181 EXPECT_CALL(*channel_proxy_, SetEncoderForMock(_, _)) |
175 EXPECT_CALL(*channel_proxy_, SetCodecFECStatus(false)) | 182 .WillOnce(Return(true)); |
176 .WillOnce(Return(true)); | 183 } |
177 EXPECT_CALL(*channel_proxy_, DisableAudioNetworkAdaptor()); | 184 // These should no longer be called. |
178 // Let |GetSendCodec| return false for the first time to indicate that no | 185 EXPECT_CALL(*channel_proxy_, SetVADStatus(_)).Times(0); |
the sun
2017/03/16 08:48:19
Well, or you could just remove those methods right
ossu
2017/03/20 18:19:48
Cleaned up these, and probably a few other, unused
| |
179 // send codec has been set. | 186 EXPECT_CALL(*channel_proxy_, GetSendCodec(_)).Times(0); |
180 EXPECT_CALL(*channel_proxy_, GetSendCodec(_)).WillOnce(Return(false)); | 187 EXPECT_CALL(*channel_proxy_, SetSendCodec(_)).Times(0); |
181 EXPECT_CALL(*channel_proxy_, SetSendCodec(_)).WillOnce(Return(true)); | 188 EXPECT_CALL(*channel_proxy_, SetCodecFECStatus(false)).Times(0); |
189 EXPECT_CALL(*channel_proxy_, DisableAudioNetworkAdaptor()).Times(0); | |
182 } | 190 } |
191 | |
183 RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; } | 192 RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; } |
184 | 193 |
185 void SetupMockForSendTelephoneEvent() { | 194 void SetupMockForSendTelephoneEvent() { |
186 EXPECT_TRUE(channel_proxy_); | 195 EXPECT_TRUE(channel_proxy_); |
187 EXPECT_CALL(*channel_proxy_, | 196 EXPECT_CALL(*channel_proxy_, |
188 SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType, | 197 SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType, |
189 kTelephoneEventPayloadFrequency)) | 198 kTelephoneEventPayloadFrequency)) |
190 .WillOnce(Return(true)); | 199 .WillOnce(Return(true)); |
191 EXPECT_CALL(*channel_proxy_, | 200 EXPECT_CALL(*channel_proxy_, |
192 SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration)) | 201 SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration)) |
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256 | 265 |
257 TEST(AudioSendStreamTest, ConfigToString) { | 266 TEST(AudioSendStreamTest, ConfigToString) { |
258 AudioSendStream::Config config(nullptr); | 267 AudioSendStream::Config config(nullptr); |
259 config.rtp.ssrc = kSsrc; | 268 config.rtp.ssrc = kSsrc; |
260 config.rtp.c_name = kCName; | 269 config.rtp.c_name = kCName; |
261 config.voe_channel_id = kChannelId; | 270 config.voe_channel_id = kChannelId; |
262 config.min_bitrate_bps = 12000; | 271 config.min_bitrate_bps = 12000; |
263 config.max_bitrate_bps = 34000; | 272 config.max_bitrate_bps = 34000; |
264 config.send_codec_spec.nack_enabled = true; | 273 config.send_codec_spec.nack_enabled = true; |
265 config.send_codec_spec.transport_cc_enabled = false; | 274 config.send_codec_spec.transport_cc_enabled = false; |
266 config.send_codec_spec.enable_codec_fec = true; | |
267 config.send_codec_spec.enable_opus_dtx = false; | |
268 config.send_codec_spec.opus_max_playback_rate = 32000; | |
269 config.send_codec_spec.cng_payload_type = 42; | 275 config.send_codec_spec.cng_payload_type = 42; |
270 config.send_codec_spec.cng_plfreq = 56; | 276 config.send_codec_spec.payload_type = kIsacPayloadType; |
271 config.send_codec_spec.min_ptime_ms = 20; | 277 config.send_codec_spec.format = kIsacFormat; |
272 config.send_codec_spec.max_ptime_ms = 60; | 278 // TODO(ossu): Maybe mock this? |
273 config.send_codec_spec.codec_inst = kIsacCodec; | 279 config.encoder_factory = CreateBuiltinAudioEncoderFactory(); |
274 config.rtp.extensions.push_back( | 280 config.rtp.extensions.push_back( |
275 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); | 281 RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId)); |
276 EXPECT_EQ( | 282 EXPECT_EQ( |
277 "{rtp: {ssrc: 1234, extensions: [{uri: " | 283 "{rtp: {ssrc: 1234, extensions: [{uri: " |
278 "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: " | 284 "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: " |
279 "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, " | 285 "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, " |
280 "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, " | 286 "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, " |
281 "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " | 287 "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, " |
282 "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: " | 288 "cng_payload_type: 42, payload_type: 103, " |
283 "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: " | 289 "format: {name: isac, clockrate_hz: 16000, num_channels: 1, " |
284 "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: " | 290 "parameters: {}}}}", |
285 "320, channels: 1, rate: 32000}}}", | |
286 config.ToString()); | 291 config.ToString()); |
287 } | 292 } |
288 | 293 |
289 TEST(AudioSendStreamTest, ConstructDestruct) { | 294 TEST(AudioSendStreamTest, ConstructDestruct) { |
290 ConfigHelper helper(false); | 295 ConfigHelper helper(false, true); |
291 internal::AudioSendStream send_stream( | 296 internal::AudioSendStream send_stream( |
292 helper.config(), helper.audio_state(), helper.worker_queue(), | 297 helper.config(), helper.audio_state(), helper.worker_queue(), |
293 helper.packet_router(), helper.congestion_controller(), | 298 helper.packet_router(), helper.congestion_controller(), |
294 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 299 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
295 } | 300 } |
296 | 301 |
297 TEST(AudioSendStreamTest, SendTelephoneEvent) { | 302 TEST(AudioSendStreamTest, SendTelephoneEvent) { |
298 ConfigHelper helper(false); | 303 ConfigHelper helper(false, true); |
299 internal::AudioSendStream send_stream( | 304 internal::AudioSendStream send_stream( |
300 helper.config(), helper.audio_state(), helper.worker_queue(), | 305 helper.config(), helper.audio_state(), helper.worker_queue(), |
301 helper.packet_router(), helper.congestion_controller(), | 306 helper.packet_router(), helper.congestion_controller(), |
302 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 307 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
303 helper.SetupMockForSendTelephoneEvent(); | 308 helper.SetupMockForSendTelephoneEvent(); |
304 EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType, | 309 EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType, |
305 kTelephoneEventPayloadFrequency, kTelephoneEventCode, | 310 kTelephoneEventPayloadFrequency, kTelephoneEventCode, |
306 kTelephoneEventDuration)); | 311 kTelephoneEventDuration)); |
307 } | 312 } |
308 | 313 |
309 TEST(AudioSendStreamTest, SetMuted) { | 314 TEST(AudioSendStreamTest, SetMuted) { |
310 ConfigHelper helper(false); | 315 ConfigHelper helper(false, true); |
311 internal::AudioSendStream send_stream( | 316 internal::AudioSendStream send_stream( |
312 helper.config(), helper.audio_state(), helper.worker_queue(), | 317 helper.config(), helper.audio_state(), helper.worker_queue(), |
313 helper.packet_router(), helper.congestion_controller(), | 318 helper.packet_router(), helper.congestion_controller(), |
314 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 319 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
315 EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true)); | 320 EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true)); |
316 send_stream.SetMuted(true); | 321 send_stream.SetMuted(true); |
317 } | 322 } |
318 | 323 |
319 TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) { | 324 TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) { |
320 ConfigHelper helper(true); | 325 ConfigHelper helper(true, true); |
321 internal::AudioSendStream send_stream( | 326 internal::AudioSendStream send_stream( |
322 helper.config(), helper.audio_state(), helper.worker_queue(), | 327 helper.config(), helper.audio_state(), helper.worker_queue(), |
323 helper.packet_router(), helper.congestion_controller(), | 328 helper.packet_router(), helper.congestion_controller(), |
324 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 329 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
325 } | 330 } |
326 | 331 |
327 TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) { | 332 TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) { |
328 ConfigHelper helper(false); | 333 ConfigHelper helper(false, true); |
329 internal::AudioSendStream send_stream( | 334 internal::AudioSendStream send_stream( |
330 helper.config(), helper.audio_state(), helper.worker_queue(), | 335 helper.config(), helper.audio_state(), helper.worker_queue(), |
331 helper.packet_router(), helper.congestion_controller(), | 336 helper.packet_router(), helper.congestion_controller(), |
332 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 337 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
333 } | 338 } |
334 | 339 |
335 TEST(AudioSendStreamTest, GetStats) { | 340 TEST(AudioSendStreamTest, GetStats) { |
336 ConfigHelper helper(false); | 341 ConfigHelper helper(false, true); |
337 internal::AudioSendStream send_stream( | 342 internal::AudioSendStream send_stream( |
338 helper.config(), helper.audio_state(), helper.worker_queue(), | 343 helper.config(), helper.audio_state(), helper.worker_queue(), |
339 helper.packet_router(), helper.congestion_controller(), | 344 helper.packet_router(), helper.congestion_controller(), |
340 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 345 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
341 helper.SetupMockForGetStats(); | 346 helper.SetupMockForGetStats(); |
342 AudioSendStream::Stats stats = send_stream.GetStats(); | 347 AudioSendStream::Stats stats = send_stream.GetStats(); |
343 EXPECT_EQ(kSsrc, stats.local_ssrc); | 348 EXPECT_EQ(kSsrc, stats.local_ssrc); |
344 EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent); | 349 EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent); |
345 EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent); | 350 EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent); |
346 EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost), | 351 EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost), |
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357 EXPECT_EQ(-1, stats.aec_quality_min); | 362 EXPECT_EQ(-1, stats.aec_quality_min); |
358 EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms); | 363 EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms); |
359 EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms); | 364 EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms); |
360 EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss); | 365 EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss); |
361 EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement); | 366 EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement); |
362 EXPECT_EQ(kResidualEchoLikelihood, stats.residual_echo_likelihood); | 367 EXPECT_EQ(kResidualEchoLikelihood, stats.residual_echo_likelihood); |
363 EXPECT_FALSE(stats.typing_noise_detected); | 368 EXPECT_FALSE(stats.typing_noise_detected); |
364 } | 369 } |
365 | 370 |
366 TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) { | 371 TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) { |
367 ConfigHelper helper(false); | 372 ConfigHelper helper(false, true); |
368 internal::AudioSendStream send_stream( | 373 internal::AudioSendStream send_stream( |
369 helper.config(), helper.audio_state(), helper.worker_queue(), | 374 helper.config(), helper.audio_state(), helper.worker_queue(), |
370 helper.packet_router(), helper.congestion_controller(), | 375 helper.packet_router(), helper.congestion_controller(), |
371 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 376 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
372 helper.SetupMockForGetStats(); | 377 helper.SetupMockForGetStats(); |
373 EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); | 378 EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); |
374 | 379 |
375 internal::AudioState* internal_audio_state = | 380 internal::AudioState* internal_audio_state = |
376 static_cast<internal::AudioState*>(helper.audio_state().get()); | 381 static_cast<internal::AudioState*>(helper.audio_state().get()); |
377 VoiceEngineObserver* voe_observer = | 382 VoiceEngineObserver* voe_observer = |
378 static_cast<VoiceEngineObserver*>(internal_audio_state); | 383 static_cast<VoiceEngineObserver*>(internal_audio_state); |
379 voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING); | 384 voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING); |
380 EXPECT_TRUE(send_stream.GetStats().typing_noise_detected); | 385 EXPECT_TRUE(send_stream.GetStats().typing_noise_detected); |
381 voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING); | 386 voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING); |
382 EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); | 387 EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); |
383 } | 388 } |
384 | 389 |
385 TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) { | 390 TEST(AudioSendStreamTest, SendCodecAppliesNetworkAdaptor) { |
386 ConfigHelper helper(false); | 391 ConfigHelper helper(false, true); |
387 auto stream_config = helper.config(); | 392 auto stream_config = helper.config(); |
388 const CodecInst kOpusCodec = {111, "opus", 48000, 960, 2, 64000}; | 393 const webrtc::SdpAudioFormat kOpusFormat = {"opus", 48000, 2}; |
389 stream_config.send_codec_spec.codec_inst = kOpusCodec; | 394 stream_config.send_codec_spec.format = kOpusFormat; |
390 stream_config.send_codec_spec.enable_codec_fec = true; | |
391 stream_config.send_codec_spec.enable_opus_dtx = true; | |
392 stream_config.send_codec_spec.opus_max_playback_rate = 12345; | |
393 stream_config.send_codec_spec.cng_plfreq = 16000; | |
394 stream_config.send_codec_spec.cng_payload_type = 105; | |
395 stream_config.send_codec_spec.min_ptime_ms = 10; | |
396 stream_config.send_codec_spec.max_ptime_ms = 60; | |
397 stream_config.audio_network_adaptor_config = | 395 stream_config.audio_network_adaptor_config = |
398 rtc::Optional<std::string>("abced"); | 396 rtc::Optional<std::string>("abced"); |
399 EXPECT_CALL(*helper.channel_proxy(), SetCodecFECStatus(true)) | |
400 .WillOnce(Return(true)); | |
401 EXPECT_CALL( | |
402 *helper.channel_proxy(), | |
403 SetOpusDtx(stream_config.send_codec_spec.enable_opus_dtx)) | |
404 .WillOnce(Return(true)); | |
405 EXPECT_CALL( | |
406 *helper.channel_proxy(), | |
407 SetOpusMaxPlaybackRate( | |
408 stream_config.send_codec_spec.opus_max_playback_rate)) | |
409 .WillOnce(Return(true)); | |
410 EXPECT_CALL(*helper.channel_proxy(), | |
411 SetSendCNPayloadType( | |
412 stream_config.send_codec_spec.cng_payload_type, | |
413 webrtc::kFreq16000Hz)) | |
414 .WillOnce(Return(true)); | |
415 EXPECT_CALL( | |
416 *helper.channel_proxy(), | |
417 SetReceiverFrameLengthRange(stream_config.send_codec_spec.min_ptime_ms, | |
418 stream_config.send_codec_spec.max_ptime_ms)); | |
419 EXPECT_CALL( | 397 EXPECT_CALL( |
420 *helper.channel_proxy(), | 398 *helper.channel_proxy(), |
421 EnableAudioNetworkAdaptor(*stream_config.audio_network_adaptor_config)); | 399 EnableAudioNetworkAdaptor(*stream_config.audio_network_adaptor_config)); |
422 internal::AudioSendStream send_stream( | 400 internal::AudioSendStream send_stream( |
423 stream_config, helper.audio_state(), helper.worker_queue(), | 401 stream_config, helper.audio_state(), helper.worker_queue(), |
424 helper.packet_router(), helper.congestion_controller(), | 402 helper.packet_router(), helper.congestion_controller(), |
425 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 403 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
426 } | 404 } |
427 | 405 |
428 // VAD is applied when codec is mono and the CNG frequency matches the codec | 406 // VAD is applied when codec is mono and the CNG frequency matches the codec |
429 // sample rate. | 407 // clock rate. |
430 TEST(AudioSendStreamTest, SendCodecCanApplyVad) { | 408 TEST(AudioSendStreamTest, SendCodecCanApplyVad) { |
431 ConfigHelper helper(false); | 409 ConfigHelper helper(false, false); |
432 auto stream_config = helper.config(); | 410 auto stream_config = helper.config(); |
433 const CodecInst kG722Codec = {9, "g722", 8000, 160, 1, 16000}; | 411 stream_config.send_codec_spec.payload_type = 9; |
434 stream_config.send_codec_spec.codec_inst = kG722Codec; | 412 stream_config.send_codec_spec.format = {"g722", 8000, 1}; |
435 stream_config.send_codec_spec.cng_plfreq = 8000; | |
436 stream_config.send_codec_spec.cng_payload_type = 105; | 413 stream_config.send_codec_spec.cng_payload_type = 105; |
437 EXPECT_CALL(*helper.channel_proxy(), SetVADStatus(true)) | 414 |
438 .WillOnce(Return(true)); | 415 using ::testing::Invoke; |
416 std::unique_ptr<AudioEncoder> stolen_encoder; | |
417 EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _)) | |
418 .WillOnce(Invoke([&stolen_encoder]( | |
419 int payload_type, std::unique_ptr<AudioEncoder>& encoder) { | |
420 stolen_encoder = std::move(encoder); | |
421 return true; | |
422 })); | |
423 | |
439 internal::AudioSendStream send_stream( | 424 internal::AudioSendStream send_stream( |
440 stream_config, helper.audio_state(), helper.worker_queue(), | 425 stream_config, helper.audio_state(), helper.worker_queue(), |
441 helper.packet_router(), helper.congestion_controller(), | 426 helper.packet_router(), helper.congestion_controller(), |
442 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 427 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
428 | |
429 // We cannot truly determine if the encoder created is an AudioEncoderCng. It | |
430 // is the only reasonable implementation that will return something from | |
431 // ReclaimContainedEncoders, though. | |
432 ASSERT_TRUE(stolen_encoder); | |
433 EXPECT_FALSE(stolen_encoder->ReclaimContainedEncoders().empty()); | |
443 } | 434 } |
444 | 435 |
445 TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) { | 436 TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) { |
446 ConfigHelper helper(false); | 437 ConfigHelper helper(false, true); |
447 internal::AudioSendStream send_stream( | 438 internal::AudioSendStream send_stream( |
448 helper.config(), helper.audio_state(), helper.worker_queue(), | 439 helper.config(), helper.audio_state(), helper.worker_queue(), |
449 helper.packet_router(), helper.congestion_controller(), | 440 helper.packet_router(), helper.congestion_controller(), |
450 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 441 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
451 EXPECT_CALL(*helper.channel_proxy(), | 442 EXPECT_CALL(*helper.channel_proxy(), |
452 SetBitrate(helper.config().max_bitrate_bps, _)); | 443 SetBitrate(helper.config().max_bitrate_bps, _)); |
453 send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50, | 444 send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50, |
454 6000); | 445 6000); |
455 } | 446 } |
456 | 447 |
457 TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) { | 448 TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) { |
458 ConfigHelper helper(false); | 449 ConfigHelper helper(false, true); |
459 internal::AudioSendStream send_stream( | 450 internal::AudioSendStream send_stream( |
460 helper.config(), helper.audio_state(), helper.worker_queue(), | 451 helper.config(), helper.audio_state(), helper.worker_queue(), |
461 helper.packet_router(), helper.congestion_controller(), | 452 helper.packet_router(), helper.congestion_controller(), |
462 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); | 453 helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats()); |
463 EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000)); | 454 EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000)); |
464 send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000); | 455 send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000); |
465 } | 456 } |
466 | 457 |
467 } // namespace test | 458 } // namespace test |
468 } // namespace webrtc | 459 } // namespace webrtc |
OLD | NEW |