1 /* 2 * libjingle 3 * Copyright 2010 Google Inc. 4 * 5 * Redistribution and use in source and binary forms, with or without 6 * modification, are permitted provided that the following conditions are met: 7 * 8 * 1. Redistributions of source code must retain the above copyright notice, 9 * this list of conditions and the following disclaimer. 10 * 2. Redistributions in binary form must reproduce the above copyright notice, 11 * this list of conditions and the following disclaimer in the documentation 12 * and/or other materials provided with the distribution. 13 * 3. The name of the author may not be used to endorse or promote products 14 * derived from this software without specific prior written permission. 15 * 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 26 */ 27 28 #ifndef TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ 29 #define TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ 30 31 #include <list> 32 #include <map> 33 #include <vector> 34 35 #include "talk/media/base/codec.h" 36 #include "talk/media/base/rtputils.h" 37 #include "talk/media/webrtc/fakewebrtccommon.h" 38 #include "talk/media/webrtc/webrtcvoe.h" 39 #include "webrtc/base/basictypes.h" 40 #include "webrtc/base/checks.h" 41 #include "webrtc/base/gunit.h" 42 #include "webrtc/base/stringutils.h" 43 #include "webrtc/config.h" 44 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" 45 #include "webrtc/modules/audio_processing/include/audio_processing.h" 46 47 namespace cricket { 48 49 static const int kOpusBandwidthNb = 4000; 50 static const int kOpusBandwidthMb = 6000; 51 static const int kOpusBandwidthWb = 8000; 52 static const int kOpusBandwidthSwb = 12000; 53 static const int kOpusBandwidthFb = 20000; 54 55 #define WEBRTC_CHECK_CHANNEL(channel) \ 56 if (channels_.find(channel) == channels_.end()) return -1; 57 58 class FakeAudioProcessing : public webrtc::AudioProcessing { 59 public: 60 FakeAudioProcessing() : experimental_ns_enabled_(false) {} 61 62 WEBRTC_STUB(Initialize, ()) 63 WEBRTC_STUB(Initialize, ( 64 int input_sample_rate_hz, 65 int output_sample_rate_hz, 66 int reverse_sample_rate_hz, 67 webrtc::AudioProcessing::ChannelLayout input_layout, 68 webrtc::AudioProcessing::ChannelLayout output_layout, 69 webrtc::AudioProcessing::ChannelLayout reverse_layout)); 70 WEBRTC_STUB(Initialize, ( 71 const webrtc::ProcessingConfig& processing_config)); 72 73 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { 74 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; 75 } 76 77 WEBRTC_STUB_CONST(input_sample_rate_hz, ()); 78 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); 79 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); 80 size_t num_input_channels() const override { return 0; } 81 size_t num_proc_channels() const override { return 0; } 82 size_t num_output_channels() const override { return 0; } 83 size_t num_reverse_channels() const override { return 0; } 84 WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted)); 85 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame)); 86 WEBRTC_STUB(ProcessStream, ( 87 const float* const* src, 88 size_t samples_per_channel, 89 int input_sample_rate_hz, 90 webrtc::AudioProcessing::ChannelLayout input_layout, 91 int output_sample_rate_hz, 92 webrtc::AudioProcessing::ChannelLayout output_layout, 93 float* const* dest)); 94 WEBRTC_STUB(ProcessStream, 95 (const float* const* src, 96 const webrtc::StreamConfig& input_config, 97 const webrtc::StreamConfig& output_config, 98 float* const* dest)); 99 WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame)); 100 WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame)); 101 WEBRTC_STUB(AnalyzeReverseStream, ( 102 const float* const* data, 103 size_t samples_per_channel, 104 int sample_rate_hz, 105 webrtc::AudioProcessing::ChannelLayout layout)); 106 WEBRTC_STUB(ProcessReverseStream, 107 (const float* const* src, 108 const webrtc::StreamConfig& reverse_input_config, 109 const webrtc::StreamConfig& reverse_output_config, 110 float* const* dest)); 111 WEBRTC_STUB(set_stream_delay_ms, (int delay)); 112 WEBRTC_STUB_CONST(stream_delay_ms, ()); 113 WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); 114 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); 115 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); 116 WEBRTC_STUB_CONST(delay_offset_ms, ()); 117 WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize])); 118 WEBRTC_STUB(StartDebugRecording, (FILE* handle)); 119 WEBRTC_STUB(StopDebugRecording, ()); 120 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); 121 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } 122 webrtc::EchoControlMobile* echo_control_mobile() const override { 123 return NULL; 124 } 125 webrtc::GainControl* gain_control() const override { return NULL; } 126 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } 127 webrtc::LevelEstimator* level_estimator() const override { return NULL; } 128 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } 129 webrtc::VoiceDetection* voice_detection() const override { return NULL; } 130 131 bool experimental_ns_enabled() { 132 return experimental_ns_enabled_; 133 } 134 135 private: 136 bool experimental_ns_enabled_; 137 }; 138 139 class FakeWebRtcVoiceEngine 140 : public webrtc::VoEAudioProcessing, 141 public webrtc::VoEBase, public webrtc::VoECodec, 142 public webrtc::VoEHardware, 143 public webrtc::VoENetwork, public webrtc::VoERTP_RTCP, 144 public webrtc::VoEVolumeControl { 145 public: 146 struct Channel { 147 explicit Channel() 148 : external_transport(false), 149 send(false), 150 playout(false), 151 volume_scale(1.0), 152 vad(false), 153 codec_fec(false), 154 max_encoding_bandwidth(0), 155 opus_dtx(false), 156 red(false), 157 nack(false), 158 cn8_type(13), 159 cn16_type(105), 160 red_type(117), 161 nack_max_packets(0), 162 send_ssrc(0), 163 associate_send_channel(-1), 164 recv_codecs(), 165 neteq_capacity(-1), 166 neteq_fast_accelerate(false) { 167 memset(&send_codec, 0, sizeof(send_codec)); 168 } 169 bool external_transport; 170 bool send; 171 bool playout; 172 float volume_scale; 173 bool vad; 174 bool codec_fec; 175 int max_encoding_bandwidth; 176 bool opus_dtx; 177 bool red; 178 bool nack; 179 int cn8_type; 180 int cn16_type; 181 int red_type; 182 int nack_max_packets; 183 uint32_t send_ssrc; 184 int associate_send_channel; 185 std::vector<webrtc::CodecInst> recv_codecs; 186 webrtc::CodecInst send_codec; 187 webrtc::PacketTime last_rtp_packet_time; 188 std::list<std::string> packets; 189 int neteq_capacity; 190 bool neteq_fast_accelerate; 191 }; 192 193 FakeWebRtcVoiceEngine() 194 : inited_(false), 195 last_channel_(-1), 196 fail_create_channel_(false), 197 num_set_send_codecs_(0), 198 ec_enabled_(false), 199 ec_metrics_enabled_(false), 200 cng_enabled_(false), 201 ns_enabled_(false), 202 agc_enabled_(false), 203 highpass_filter_enabled_(false), 204 stereo_swapping_enabled_(false), 205 typing_detection_enabled_(false), 206 ec_mode_(webrtc::kEcDefault), 207 aecm_mode_(webrtc::kAecmSpeakerphone), 208 ns_mode_(webrtc::kNsDefault), 209 agc_mode_(webrtc::kAgcDefault), 210 observer_(NULL), 211 playout_fail_channel_(-1), 212 send_fail_channel_(-1), 213 recording_sample_rate_(-1), 214 playout_sample_rate_(-1) { 215 memset(&agc_config_, 0, sizeof(agc_config_)); 216 } 217 ~FakeWebRtcVoiceEngine() { 218 RTC_CHECK(channels_.empty()); 219 } 220 221 bool ec_metrics_enabled() const { return ec_metrics_enabled_; } 222 223 bool IsInited() const { return inited_; } 224 int GetLastChannel() const { return last_channel_; } 225 int GetNumChannels() const { return static_cast<int>(channels_.size()); } 226 uint32_t GetLocalSSRC(int channel) { 227 return channels_[channel]->send_ssrc; 228 } 229 bool GetPlayout(int channel) { 230 return channels_[channel]->playout; 231 } 232 bool GetSend(int channel) { 233 return channels_[channel]->send; 234 } 235 bool GetVAD(int channel) { 236 return channels_[channel]->vad; 237 } 238 bool GetOpusDtx(int channel) { 239 return channels_[channel]->opus_dtx; 240 } 241 bool GetRED(int channel) { 242 return channels_[channel]->red; 243 } 244 bool GetCodecFEC(int channel) { 245 return channels_[channel]->codec_fec; 246 } 247 int GetMaxEncodingBandwidth(int channel) { 248 return channels_[channel]->max_encoding_bandwidth; 249 } 250 bool GetNACK(int channel) { 251 return channels_[channel]->nack; 252 } 253 int GetNACKMaxPackets(int channel) { 254 return channels_[channel]->nack_max_packets; 255 } 256 const webrtc::PacketTime& GetLastRtpPacketTime(int channel) { 257 RTC_DCHECK(channels_.find(channel) != channels_.end()); 258 return channels_[channel]->last_rtp_packet_time; 259 } 260 int GetSendCNPayloadType(int channel, bool wideband) { 261 return (wideband) ? 262 channels_[channel]->cn16_type : 263 channels_[channel]->cn8_type; 264 } 265 int GetSendREDPayloadType(int channel) { 266 return channels_[channel]->red_type; 267 } 268 bool CheckPacket(int channel, const void* data, size_t len) { 269 bool result = !CheckNoPacket(channel); 270 if (result) { 271 std::string packet = channels_[channel]->packets.front(); 272 result = (packet == std::string(static_cast<const char*>(data), len)); 273 channels_[channel]->packets.pop_front(); 274 } 275 return result; 276 } 277 bool CheckNoPacket(int channel) { 278 return channels_[channel]->packets.empty(); 279 } 280 void TriggerCallbackOnError(int channel_num, int err_code) { 281 RTC_DCHECK(observer_ != NULL); 282 observer_->CallbackOnError(channel_num, err_code); 283 } 284 void set_playout_fail_channel(int channel) { 285 playout_fail_channel_ = channel; 286 } 287 void set_send_fail_channel(int channel) { 288 send_fail_channel_ = channel; 289 } 290 void set_fail_create_channel(bool fail_create_channel) { 291 fail_create_channel_ = fail_create_channel; 292 } 293 int AddChannel(const webrtc::Config& config) { 294 if (fail_create_channel_) { 295 return -1; 296 } 297 Channel* ch = new Channel(); 298 auto db = webrtc::acm2::RentACodec::Database(); 299 ch->recv_codecs.assign(db.begin(), db.end()); 300 if (config.Get<webrtc::NetEqCapacityConfig>().enabled) { 301 ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity; 302 } 303 ch->neteq_fast_accelerate = 304 config.Get<webrtc::NetEqFastAccelerate>().enabled; 305 channels_[++last_channel_] = ch; 306 return last_channel_; 307 } 308 309 int GetNumSetSendCodecs() const { return num_set_send_codecs_; } 310 311 int GetAssociateSendChannel(int channel) { 312 return channels_[channel]->associate_send_channel; 313 } 314 315 WEBRTC_STUB(Release, ()); 316 317 // webrtc::VoEBase 318 WEBRTC_FUNC(RegisterVoiceEngineObserver, ( 319 webrtc::VoiceEngineObserver& observer)) { 320 observer_ = &observer; 321 return 0; 322 } 323 WEBRTC_STUB(DeRegisterVoiceEngineObserver, ()); 324 WEBRTC_FUNC(Init, (webrtc::AudioDeviceModule* adm, 325 webrtc::AudioProcessing* audioproc)) { 326 inited_ = true; 327 return 0; 328 } 329 WEBRTC_FUNC(Terminate, ()) { 330 inited_ = false; 331 return 0; 332 } 333 webrtc::AudioProcessing* audio_processing() override { 334 return &audio_processing_; 335 } 336 WEBRTC_FUNC(CreateChannel, ()) { 337 webrtc::Config empty_config; 338 return AddChannel(empty_config); 339 } 340 WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) { 341 return AddChannel(config); 342 } 343 WEBRTC_FUNC(DeleteChannel, (int channel)) { 344 WEBRTC_CHECK_CHANNEL(channel); 345 for (const auto& ch : channels_) { 346 if (ch.second->associate_send_channel == channel) { 347 ch.second->associate_send_channel = -1; 348 } 349 } 350 delete channels_[channel]; 351 channels_.erase(channel); 352 return 0; 353 } 354 WEBRTC_STUB(StartReceive, (int channel)); 355 WEBRTC_FUNC(StartPlayout, (int channel)) { 356 if (playout_fail_channel_ != channel) { 357 WEBRTC_CHECK_CHANNEL(channel); 358 channels_[channel]->playout = true; 359 return 0; 360 } else { 361 // When playout_fail_channel_ == channel, fail the StartPlayout on this 362 // channel. 363 return -1; 364 } 365 } 366 WEBRTC_FUNC(StartSend, (int channel)) { 367 if (send_fail_channel_ != channel) { 368 WEBRTC_CHECK_CHANNEL(channel); 369 channels_[channel]->send = true; 370 return 0; 371 } else { 372 // When send_fail_channel_ == channel, fail the StartSend on this 373 // channel. 374 return -1; 375 } 376 } 377 WEBRTC_STUB(StopReceive, (int channel)); 378 WEBRTC_FUNC(StopPlayout, (int channel)) { 379 WEBRTC_CHECK_CHANNEL(channel); 380 channels_[channel]->playout = false; 381 return 0; 382 } 383 WEBRTC_FUNC(StopSend, (int channel)) { 384 WEBRTC_CHECK_CHANNEL(channel); 385 channels_[channel]->send = false; 386 return 0; 387 } 388 WEBRTC_STUB(GetVersion, (char version[1024])); 389 WEBRTC_STUB(LastError, ()); 390 WEBRTC_FUNC(AssociateSendChannel, (int channel, 391 int accociate_send_channel)) { 392 WEBRTC_CHECK_CHANNEL(channel); 393 channels_[channel]->associate_send_channel = accociate_send_channel; 394 return 0; 395 } 396 webrtc::RtcEventLog* GetEventLog() { return nullptr; } 397 398 // webrtc::VoECodec 399 WEBRTC_STUB(NumOfCodecs, ()); 400 WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec)); 401 WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) { 402 WEBRTC_CHECK_CHANNEL(channel); 403 // To match the behavior of the real implementation. 404 if (_stricmp(codec.plname, "telephone-event") == 0 || 405 _stricmp(codec.plname, "audio/telephone-event") == 0 || 406 _stricmp(codec.plname, "CN") == 0 || 407 _stricmp(codec.plname, "red") == 0 ) { 408 return -1; 409 } 410 channels_[channel]->send_codec = codec; 411 ++num_set_send_codecs_; 412 return 0; 413 } 414 WEBRTC_FUNC(GetSendCodec, (int channel, webrtc::CodecInst& codec)) { 415 WEBRTC_CHECK_CHANNEL(channel); 416 codec = channels_[channel]->send_codec; 417 return 0; 418 } 419 WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps)); 420 WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec)); 421 WEBRTC_FUNC(SetRecPayloadType, (int channel, 422 const webrtc::CodecInst& codec)) { 423 WEBRTC_CHECK_CHANNEL(channel); 424 Channel* ch = channels_[channel]; 425 if (ch->playout) 426 return -1; // Channel is in use. 427 // Check if something else already has this slot. 428 if (codec.pltype != -1) { 429 for (std::vector<webrtc::CodecInst>::iterator it = 430 ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) { 431 if (it->pltype == codec.pltype && 432 _stricmp(it->plname, codec.plname) != 0) { 433 return -1; 434 } 435 } 436 } 437 // Otherwise try to find this codec and update its payload type. 438 int result = -1; // not found 439 for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin(); 440 it != ch->recv_codecs.end(); ++it) { 441 if (strcmp(it->plname, codec.plname) == 0 && 442 it->plfreq == codec.plfreq && 443 it->channels == codec.channels) { 444 it->pltype = codec.pltype; 445 result = 0; 446 } 447 } 448 return result; 449 } 450 WEBRTC_FUNC(SetSendCNPayloadType, (int channel, int type, 451 webrtc::PayloadFrequencies frequency)) { 452 WEBRTC_CHECK_CHANNEL(channel); 453 if (frequency == webrtc::kFreq8000Hz) { 454 channels_[channel]->cn8_type = type; 455 } else if (frequency == webrtc::kFreq16000Hz) { 456 channels_[channel]->cn16_type = type; 457 } 458 return 0; 459 } 460 WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) { 461 WEBRTC_CHECK_CHANNEL(channel); 462 Channel* ch = channels_[channel]; 463 for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin(); 464 it != ch->recv_codecs.end(); ++it) { 465 if (strcmp(it->plname, codec.plname) == 0 && 466 it->plfreq == codec.plfreq && 467 it->channels == codec.channels && 468 it->pltype != -1) { 469 codec.pltype = it->pltype; 470 return 0; 471 } 472 } 473 return -1; // not found 474 } 475 WEBRTC_FUNC(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode, 476 bool disableDTX)) { 477 WEBRTC_CHECK_CHANNEL(channel); 478 if (channels_[channel]->send_codec.channels == 2) { 479 // Replicating VoE behavior; VAD cannot be enabled for stereo. 480 return -1; 481 } 482 channels_[channel]->vad = enable; 483 return 0; 484 } 485 WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled, 486 webrtc::VadModes& mode, bool& disabledDTX)); 487 488 WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) { 489 WEBRTC_CHECK_CHANNEL(channel); 490 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { 491 // Return -1 if current send codec is not Opus. 492 // TODO(minyue): Excludes other codecs if they support inband FEC. 493 return -1; 494 } 495 channels_[channel]->codec_fec = enable; 496 return 0; 497 } 498 WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable)) { 499 WEBRTC_CHECK_CHANNEL(channel); 500 enable = channels_[channel]->codec_fec; 501 return 0; 502 } 503 504 WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) { 505 WEBRTC_CHECK_CHANNEL(channel); 506 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { 507 // Return -1 if current send codec is not Opus. 508 return -1; 509 } 510 if (frequency_hz <= 8000) 511 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthNb; 512 else if (frequency_hz <= 12000) 513 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthMb; 514 else if (frequency_hz <= 16000) 515 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthWb; 516 else if (frequency_hz <= 24000) 517 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthSwb; 518 else 519 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb; 520 return 0; 521 } 522 523 WEBRTC_FUNC(SetOpusDtx, (int channel, bool enable_dtx)) { 524 WEBRTC_CHECK_CHANNEL(channel); 525 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { 526 // Return -1 if current send codec is not Opus. 527 return -1; 528 } 529 channels_[channel]->opus_dtx = enable_dtx; 530 return 0; 531 } 532 533 // webrtc::VoEHardware 534 WEBRTC_STUB(GetNumOfRecordingDevices, (int& num)); 535 WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num)); 536 WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid)); 537 WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid)); 538 WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel)); 539 WEBRTC_STUB(SetPlayoutDevice, (int)); 540 WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers)); 541 WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&)); 542 WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) { 543 recording_sample_rate_ = samples_per_sec; 544 return 0; 545 } 546 WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) { 547 *samples_per_sec = recording_sample_rate_; 548 return 0; 549 } 550 WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) { 551 playout_sample_rate_ = samples_per_sec; 552 return 0; 553 } 554 WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) { 555 *samples_per_sec = playout_sample_rate_; 556 return 0; 557 } 558 WEBRTC_STUB(EnableBuiltInAEC, (bool enable)); 559 virtual bool BuiltInAECIsAvailable() const { return false; } 560 WEBRTC_STUB(EnableBuiltInAGC, (bool enable)); 561 virtual bool BuiltInAGCIsAvailable() const { return false; } 562 WEBRTC_STUB(EnableBuiltInNS, (bool enable)); 563 virtual bool BuiltInNSIsAvailable() const { return false; } 564 565 // webrtc::VoENetwork 566 WEBRTC_FUNC(RegisterExternalTransport, (int channel, 567 webrtc::Transport& transport)) { 568 WEBRTC_CHECK_CHANNEL(channel); 569 channels_[channel]->external_transport = true; 570 return 0; 571 } 572 WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) { 573 WEBRTC_CHECK_CHANNEL(channel); 574 channels_[channel]->external_transport = false; 575 return 0; 576 } 577 WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data, 578 size_t length)) { 579 WEBRTC_CHECK_CHANNEL(channel); 580 if (!channels_[channel]->external_transport) return -1; 581 channels_[channel]->packets.push_back( 582 std::string(static_cast<const char*>(data), length)); 583 return 0; 584 } 585 WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data, 586 size_t length, 587 const webrtc::PacketTime& packet_time)) { 588 WEBRTC_CHECK_CHANNEL(channel); 589 if (ReceivedRTPPacket(channel, data, length) == -1) { 590 return -1; 591 } 592 channels_[channel]->last_rtp_packet_time = packet_time; 593 return 0; 594 } 595 596 WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data, 597 size_t length)); 598 599 // webrtc::VoERTP_RTCP 600 WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) { 601 WEBRTC_CHECK_CHANNEL(channel); 602 channels_[channel]->send_ssrc = ssrc; 603 return 0; 604 } 605 WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc)); 606 WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc)); 607 WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable, 608 unsigned char id)); 609 WEBRTC_STUB(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable, 610 unsigned char id)); 611 WEBRTC_STUB(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable, 612 unsigned char id)); 613 WEBRTC_STUB(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable, 614 unsigned char id)); 615 WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable)); 616 WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled)); 617 WEBRTC_STUB(SetRTCP_CNAME, (int channel, const char cname[256])); 618 WEBRTC_STUB(GetRTCP_CNAME, (int channel, char cname[256])); 619 WEBRTC_STUB(GetRemoteRTCP_CNAME, (int channel, char* cname)); 620 WEBRTC_STUB(GetRemoteRTCPData, (int channel, unsigned int& NTPHigh, 621 unsigned int& NTPLow, 622 unsigned int& timestamp, 623 unsigned int& playoutTimestamp, 624 unsigned int* jitter, 625 unsigned short* fractionLost)); 626 WEBRTC_STUB(GetRemoteRTCPReportBlocks, 627 (int channel, std::vector<webrtc::ReportBlock>* receive_blocks)); 628 WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs, 629 unsigned int& maxJitterMs, 630 unsigned int& discardedPackets)); 631 WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats)); 632 WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) { 633 WEBRTC_CHECK_CHANNEL(channel); 634 channels_[channel]->red = enable; 635 channels_[channel]->red_type = redPayloadtype; 636 return 0; 637 } 638 WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) { 639 WEBRTC_CHECK_CHANNEL(channel); 640 enable = channels_[channel]->red; 641 redPayloadtype = channels_[channel]->red_type; 642 return 0; 643 } 644 WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) { 645 WEBRTC_CHECK_CHANNEL(channel); 646 channels_[channel]->nack = enable; 647 channels_[channel]->nack_max_packets = maxNoPackets; 648 return 0; 649 } 650 651 // webrtc::VoEVolumeControl 652 WEBRTC_STUB(SetSpeakerVolume, (unsigned int)); 653 WEBRTC_STUB(GetSpeakerVolume, (unsigned int&)); 654 WEBRTC_STUB(SetMicVolume, (unsigned int)); 655 WEBRTC_STUB(GetMicVolume, (unsigned int&)); 656 WEBRTC_STUB(SetInputMute, (int, bool)); 657 WEBRTC_STUB(GetInputMute, (int, bool&)); 658 WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&)); 659 WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&)); 660 WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&)); 661 WEBRTC_STUB(GetSpeechOutputLevelFullRange, (int, unsigned int&)); 662 WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) { 663 WEBRTC_CHECK_CHANNEL(channel); 664 channels_[channel]->volume_scale= scale; 665 return 0; 666 } 667 WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) { 668 WEBRTC_CHECK_CHANNEL(channel); 669 scale = channels_[channel]->volume_scale; 670 return 0; 671 } 672 WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right)); 673 WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right)); 674 675 // webrtc::VoEAudioProcessing 676 WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) { 677 ns_enabled_ = enable; 678 ns_mode_ = mode; 679 return 0; 680 } 681 WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) { 682 enabled = ns_enabled_; 683 mode = ns_mode_; 684 return 0; 685 } 686 687 WEBRTC_FUNC(SetAgcStatus, (bool enable, webrtc::AgcModes mode)) { 688 agc_enabled_ = enable; 689 agc_mode_ = mode; 690 return 0; 691 } 692 WEBRTC_FUNC(GetAgcStatus, (bool& enabled, webrtc::AgcModes& mode)) { 693 enabled = agc_enabled_; 694 mode = agc_mode_; 695 return 0; 696 } 697 698 WEBRTC_FUNC(SetAgcConfig, (webrtc::AgcConfig config)) { 699 agc_config_ = config; 700 return 0; 701 } 702 WEBRTC_FUNC(GetAgcConfig, (webrtc::AgcConfig& config)) { 703 config = agc_config_; 704 return 0; 705 } 706 WEBRTC_FUNC(SetEcStatus, (bool enable, webrtc::EcModes mode)) { 707 ec_enabled_ = enable; 708 ec_mode_ = mode; 709 return 0; 710 } 711 WEBRTC_FUNC(GetEcStatus, (bool& enabled, webrtc::EcModes& mode)) { 712 enabled = ec_enabled_; 713 mode = ec_mode_; 714 return 0; 715 } 716 WEBRTC_STUB(EnableDriftCompensation, (bool enable)) 717 WEBRTC_BOOL_STUB(DriftCompensationEnabled, ()) 718 WEBRTC_VOID_STUB(SetDelayOffsetMs, (int offset)) 719 WEBRTC_STUB(DelayOffsetMs, ()); 720 WEBRTC_FUNC(SetAecmMode, (webrtc::AecmModes mode, bool enableCNG)) { 721 aecm_mode_ = mode; 722 cng_enabled_ = enableCNG; 723 return 0; 724 } 725 WEBRTC_FUNC(GetAecmMode, (webrtc::AecmModes& mode, bool& enabledCNG)) { 726 mode = aecm_mode_; 727 enabledCNG = cng_enabled_; 728 return 0; 729 } 730 WEBRTC_STUB(SetRxNsStatus, (int channel, bool enable, webrtc::NsModes mode)); 731 WEBRTC_STUB(GetRxNsStatus, (int channel, bool& enabled, 732 webrtc::NsModes& mode)); 733 WEBRTC_STUB(SetRxAgcStatus, (int channel, bool enable, 734 webrtc::AgcModes mode)); 735 WEBRTC_STUB(GetRxAgcStatus, (int channel, bool& enabled, 736 webrtc::AgcModes& mode)); 737 WEBRTC_STUB(SetRxAgcConfig, (int channel, webrtc::AgcConfig config)); 738 WEBRTC_STUB(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config)); 739 740 WEBRTC_STUB(RegisterRxVadObserver, (int, webrtc::VoERxVadCallback&)); 741 WEBRTC_STUB(DeRegisterRxVadObserver, (int channel)); 742 WEBRTC_STUB(VoiceActivityIndicator, (int channel)); 743 WEBRTC_FUNC(SetEcMetricsStatus, (bool enable)) { 744 ec_metrics_enabled_ = enable; 745 return 0; 746 } 747 WEBRTC_STUB(GetEcMetricsStatus, (bool& enabled)); 748 WEBRTC_STUB(GetEchoMetrics, (int& ERL, int& ERLE, int& RERL, int& A_NLP)); 749 WEBRTC_STUB(GetEcDelayMetrics, (int& delay_median, int& delay_std, 750 float& fraction_poor_delays)); 751 752 WEBRTC_STUB(StartDebugRecording, (const char* fileNameUTF8)); 753 WEBRTC_STUB(StartDebugRecording, (FILE* handle)); 754 WEBRTC_STUB(StopDebugRecording, ()); 755 756 WEBRTC_FUNC(SetTypingDetectionStatus, (bool enable)) { 757 typing_detection_enabled_ = enable; 758 return 0; 759 } 760 WEBRTC_FUNC(GetTypingDetectionStatus, (bool& enabled)) { 761 enabled = typing_detection_enabled_; 762 return 0; 763 } 764 765 WEBRTC_STUB(TimeSinceLastTyping, (int& seconds)); 766 WEBRTC_STUB(SetTypingDetectionParameters, (int timeWindow, 767 int costPerTyping, 768 int reportingThreshold, 769 int penaltyDecay, 770 int typeEventDelay)); 771 int EnableHighPassFilter(bool enable) { 772 highpass_filter_enabled_ = enable; 773 return 0; 774 } 775 bool IsHighPassFilterEnabled() { 776 return highpass_filter_enabled_; 777 } 778 bool IsStereoChannelSwappingEnabled() { 779 return stereo_swapping_enabled_; 780 } 781 void EnableStereoChannelSwapping(bool enable) { 782 stereo_swapping_enabled_ = enable; 783 } 784 int GetNetEqCapacity() const { 785 auto ch = channels_.find(last_channel_); 786 ASSERT(ch != channels_.end()); 787 return ch->second->neteq_capacity; 788 } 789 bool GetNetEqFastAccelerate() const { 790 auto ch = channels_.find(last_channel_); 791 ASSERT(ch != channels_.end()); 792 return ch->second->neteq_fast_accelerate; 793 } 794 795 private: 796 bool inited_; 797 int last_channel_; 798 std::map<int, Channel*> channels_; 799 bool fail_create_channel_; 800 int num_set_send_codecs_; // how many times we call SetSendCodec(). 801 bool ec_enabled_; 802 bool ec_metrics_enabled_; 803 bool cng_enabled_; 804 bool ns_enabled_; 805 bool agc_enabled_; 806 bool highpass_filter_enabled_; 807 bool stereo_swapping_enabled_; 808 bool typing_detection_enabled_; 809 webrtc::EcModes ec_mode_; 810 webrtc::AecmModes aecm_mode_; 811 webrtc::NsModes ns_mode_; 812 webrtc::AgcModes agc_mode_; 813 webrtc::AgcConfig agc_config_; 814 webrtc::VoiceEngineObserver* observer_; 815 int playout_fail_channel_; 816 int send_fail_channel_; 817 int recording_sample_rate_; 818 int playout_sample_rate_; 819 FakeAudioProcessing audio_processing_; 820 }; 821 822 } // namespace cricket 823 824 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ 825
