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| 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, | |
| 118 (const char filename[kMaxFilenameSize], int64_t max_size_bytes)); | |
| 119 WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes)); | |
| 120 WEBRTC_STUB(StopDebugRecording, ()); | |
| 121 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); | |
| 122 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } | |
| 123 webrtc::EchoControlMobile* echo_control_mobile() const override { | |
| 124 return NULL; | |
| 125 } | |
| 126 webrtc::GainControl* gain_control() const override { return NULL; } | |
| 127 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } | |
| 128 webrtc::LevelEstimator* level_estimator() const override { return NULL; } | |
| 129 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } | |
| 130 webrtc::VoiceDetection* voice_detection() const override { return NULL; } | |
| 131 | |
| 132 bool experimental_ns_enabled() { | |
| 133 return experimental_ns_enabled_; | |
| 134 } | |
| 135 | |
| 136 private: | |
| 137 bool experimental_ns_enabled_; | |
| 138 }; | |
| 139 | |
| 140 class FakeWebRtcVoiceEngine | |
| 141 : public webrtc::VoEAudioProcessing, | |
| 142 public webrtc::VoEBase, public webrtc::VoECodec, | |
| 143 public webrtc::VoEHardware, | |
| 144 public webrtc::VoENetwork, public webrtc::VoERTP_RTCP, | |
| 145 public webrtc::VoEVolumeControl { | |
| 146 public: | |
| 147 struct Channel { | |
| 148 explicit Channel() | |
| 149 : external_transport(false), | |
| 150 send(false), | |
| 151 playout(false), | |
| 152 volume_scale(1.0), | |
| 153 vad(false), | |
| 154 codec_fec(false), | |
| 155 max_encoding_bandwidth(0), | |
| 156 opus_dtx(false), | |
| 157 red(false), | |
| 158 nack(false), | |
| 159 cn8_type(13), | |
| 160 cn16_type(105), | |
| 161 red_type(117), | |
| 162 nack_max_packets(0), | |
| 163 send_ssrc(0), | |
| 164 associate_send_channel(-1), | |
| 165 recv_codecs(), | |
| 166 neteq_capacity(-1), | |
| 167 neteq_fast_accelerate(false) { | |
| 168 memset(&send_codec, 0, sizeof(send_codec)); | |
| 169 } | |
| 170 bool external_transport; | |
| 171 bool send; | |
| 172 bool playout; | |
| 173 float volume_scale; | |
| 174 bool vad; | |
| 175 bool codec_fec; | |
| 176 int max_encoding_bandwidth; | |
| 177 bool opus_dtx; | |
| 178 bool red; | |
| 179 bool nack; | |
| 180 int cn8_type; | |
| 181 int cn16_type; | |
| 182 int red_type; | |
| 183 int nack_max_packets; | |
| 184 uint32_t send_ssrc; | |
| 185 int associate_send_channel; | |
| 186 std::vector<webrtc::CodecInst> recv_codecs; | |
| 187 webrtc::CodecInst send_codec; | |
| 188 webrtc::PacketTime last_rtp_packet_time; | |
| 189 std::list<std::string> packets; | |
| 190 int neteq_capacity; | |
| 191 bool neteq_fast_accelerate; | |
| 192 }; | |
| 193 | |
| 194 FakeWebRtcVoiceEngine() | |
| 195 : inited_(false), | |
| 196 last_channel_(-1), | |
| 197 fail_create_channel_(false), | |
| 198 num_set_send_codecs_(0), | |
| 199 ec_enabled_(false), | |
| 200 ec_metrics_enabled_(false), | |
| 201 cng_enabled_(false), | |
| 202 ns_enabled_(false), | |
| 203 agc_enabled_(false), | |
| 204 highpass_filter_enabled_(false), | |
| 205 stereo_swapping_enabled_(false), | |
| 206 typing_detection_enabled_(false), | |
| 207 ec_mode_(webrtc::kEcDefault), | |
| 208 aecm_mode_(webrtc::kAecmSpeakerphone), | |
| 209 ns_mode_(webrtc::kNsDefault), | |
| 210 agc_mode_(webrtc::kAgcDefault), | |
| 211 observer_(NULL), | |
| 212 playout_fail_channel_(-1), | |
| 213 send_fail_channel_(-1), | |
| 214 recording_sample_rate_(-1), | |
| 215 playout_sample_rate_(-1) { | |
| 216 memset(&agc_config_, 0, sizeof(agc_config_)); | |
| 217 } | |
| 218 ~FakeWebRtcVoiceEngine() { | |
| 219 RTC_CHECK(channels_.empty()); | |
| 220 } | |
| 221 | |
| 222 bool ec_metrics_enabled() const { return ec_metrics_enabled_; } | |
| 223 | |
| 224 bool IsInited() const { return inited_; } | |
| 225 int GetLastChannel() const { return last_channel_; } | |
| 226 int GetNumChannels() const { return static_cast<int>(channels_.size()); } | |
| 227 uint32_t GetLocalSSRC(int channel) { | |
| 228 return channels_[channel]->send_ssrc; | |
| 229 } | |
| 230 bool GetPlayout(int channel) { | |
| 231 return channels_[channel]->playout; | |
| 232 } | |
| 233 bool GetSend(int channel) { | |
| 234 return channels_[channel]->send; | |
| 235 } | |
| 236 bool GetVAD(int channel) { | |
| 237 return channels_[channel]->vad; | |
| 238 } | |
| 239 bool GetOpusDtx(int channel) { | |
| 240 return channels_[channel]->opus_dtx; | |
| 241 } | |
| 242 bool GetRED(int channel) { | |
| 243 return channels_[channel]->red; | |
| 244 } | |
| 245 bool GetCodecFEC(int channel) { | |
| 246 return channels_[channel]->codec_fec; | |
| 247 } | |
| 248 int GetMaxEncodingBandwidth(int channel) { | |
| 249 return channels_[channel]->max_encoding_bandwidth; | |
| 250 } | |
| 251 bool GetNACK(int channel) { | |
| 252 return channels_[channel]->nack; | |
| 253 } | |
| 254 int GetNACKMaxPackets(int channel) { | |
| 255 return channels_[channel]->nack_max_packets; | |
| 256 } | |
| 257 const webrtc::PacketTime& GetLastRtpPacketTime(int channel) { | |
| 258 RTC_DCHECK(channels_.find(channel) != channels_.end()); | |
| 259 return channels_[channel]->last_rtp_packet_time; | |
| 260 } | |
| 261 int GetSendCNPayloadType(int channel, bool wideband) { | |
| 262 return (wideband) ? | |
| 263 channels_[channel]->cn16_type : | |
| 264 channels_[channel]->cn8_type; | |
| 265 } | |
| 266 int GetSendREDPayloadType(int channel) { | |
| 267 return channels_[channel]->red_type; | |
| 268 } | |
| 269 bool CheckPacket(int channel, const void* data, size_t len) { | |
| 270 bool result = !CheckNoPacket(channel); | |
| 271 if (result) { | |
| 272 std::string packet = channels_[channel]->packets.front(); | |
| 273 result = (packet == std::string(static_cast<const char*>(data), len)); | |
| 274 channels_[channel]->packets.pop_front(); | |
| 275 } | |
| 276 return result; | |
| 277 } | |
| 278 bool CheckNoPacket(int channel) { | |
| 279 return channels_[channel]->packets.empty(); | |
| 280 } | |
| 281 void TriggerCallbackOnError(int channel_num, int err_code) { | |
| 282 RTC_DCHECK(observer_ != NULL); | |
| 283 observer_->CallbackOnError(channel_num, err_code); | |
| 284 } | |
| 285 void set_playout_fail_channel(int channel) { | |
| 286 playout_fail_channel_ = channel; | |
| 287 } | |
| 288 void set_send_fail_channel(int channel) { | |
| 289 send_fail_channel_ = channel; | |
| 290 } | |
| 291 void set_fail_create_channel(bool fail_create_channel) { | |
| 292 fail_create_channel_ = fail_create_channel; | |
| 293 } | |
| 294 int AddChannel(const webrtc::Config& config) { | |
| 295 if (fail_create_channel_) { | |
| 296 return -1; | |
| 297 } | |
| 298 Channel* ch = new Channel(); | |
| 299 auto db = webrtc::acm2::RentACodec::Database(); | |
| 300 ch->recv_codecs.assign(db.begin(), db.end()); | |
| 301 if (config.Get<webrtc::NetEqCapacityConfig>().enabled) { | |
| 302 ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity; | |
| 303 } | |
| 304 ch->neteq_fast_accelerate = | |
| 305 config.Get<webrtc::NetEqFastAccelerate>().enabled; | |
| 306 channels_[++last_channel_] = ch; | |
| 307 return last_channel_; | |
| 308 } | |
| 309 | |
| 310 int GetNumSetSendCodecs() const { return num_set_send_codecs_; } | |
| 311 | |
| 312 int GetAssociateSendChannel(int channel) { | |
| 313 return channels_[channel]->associate_send_channel; | |
| 314 } | |
| 315 | |
| 316 WEBRTC_STUB(Release, ()); | |
| 317 | |
| 318 // webrtc::VoEBase | |
| 319 WEBRTC_FUNC(RegisterVoiceEngineObserver, ( | |
| 320 webrtc::VoiceEngineObserver& observer)) { | |
| 321 observer_ = &observer; | |
| 322 return 0; | |
| 323 } | |
| 324 WEBRTC_STUB(DeRegisterVoiceEngineObserver, ()); | |
| 325 WEBRTC_FUNC(Init, (webrtc::AudioDeviceModule* adm, | |
| 326 webrtc::AudioProcessing* audioproc)) { | |
| 327 inited_ = true; | |
| 328 return 0; | |
| 329 } | |
| 330 WEBRTC_FUNC(Terminate, ()) { | |
| 331 inited_ = false; | |
| 332 return 0; | |
| 333 } | |
| 334 webrtc::AudioProcessing* audio_processing() override { | |
| 335 return &audio_processing_; | |
| 336 } | |
| 337 WEBRTC_FUNC(CreateChannel, ()) { | |
| 338 webrtc::Config empty_config; | |
| 339 return AddChannel(empty_config); | |
| 340 } | |
| 341 WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) { | |
| 342 return AddChannel(config); | |
| 343 } | |
| 344 WEBRTC_FUNC(DeleteChannel, (int channel)) { | |
| 345 WEBRTC_CHECK_CHANNEL(channel); | |
| 346 for (const auto& ch : channels_) { | |
| 347 if (ch.second->associate_send_channel == channel) { | |
| 348 ch.second->associate_send_channel = -1; | |
| 349 } | |
| 350 } | |
| 351 delete channels_[channel]; | |
| 352 channels_.erase(channel); | |
| 353 return 0; | |
| 354 } | |
| 355 WEBRTC_STUB(StartReceive, (int channel)); | |
| 356 WEBRTC_FUNC(StartPlayout, (int channel)) { | |
| 357 if (playout_fail_channel_ != channel) { | |
| 358 WEBRTC_CHECK_CHANNEL(channel); | |
| 359 channels_[channel]->playout = true; | |
| 360 return 0; | |
| 361 } else { | |
| 362 // When playout_fail_channel_ == channel, fail the StartPlayout on this | |
| 363 // channel. | |
| 364 return -1; | |
| 365 } | |
| 366 } | |
| 367 WEBRTC_FUNC(StartSend, (int channel)) { | |
| 368 if (send_fail_channel_ != channel) { | |
| 369 WEBRTC_CHECK_CHANNEL(channel); | |
| 370 channels_[channel]->send = true; | |
| 371 return 0; | |
| 372 } else { | |
| 373 // When send_fail_channel_ == channel, fail the StartSend on this | |
| 374 // channel. | |
| 375 return -1; | |
| 376 } | |
| 377 } | |
| 378 WEBRTC_STUB(StopReceive, (int channel)); | |
| 379 WEBRTC_FUNC(StopPlayout, (int channel)) { | |
| 380 WEBRTC_CHECK_CHANNEL(channel); | |
| 381 channels_[channel]->playout = false; | |
| 382 return 0; | |
| 383 } | |
| 384 WEBRTC_FUNC(StopSend, (int channel)) { | |
| 385 WEBRTC_CHECK_CHANNEL(channel); | |
| 386 channels_[channel]->send = false; | |
| 387 return 0; | |
| 388 } | |
| 389 WEBRTC_STUB(GetVersion, (char version[1024])); | |
| 390 WEBRTC_STUB(LastError, ()); | |
| 391 WEBRTC_FUNC(AssociateSendChannel, (int channel, | |
| 392 int accociate_send_channel)) { | |
| 393 WEBRTC_CHECK_CHANNEL(channel); | |
| 394 channels_[channel]->associate_send_channel = accociate_send_channel; | |
| 395 return 0; | |
| 396 } | |
| 397 webrtc::RtcEventLog* GetEventLog() { return nullptr; } | |
| 398 | |
| 399 // webrtc::VoECodec | |
| 400 WEBRTC_STUB(NumOfCodecs, ()); | |
| 401 WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec)); | |
| 402 WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) { | |
| 403 WEBRTC_CHECK_CHANNEL(channel); | |
| 404 // To match the behavior of the real implementation. | |
| 405 if (_stricmp(codec.plname, "telephone-event") == 0 || | |
| 406 _stricmp(codec.plname, "audio/telephone-event") == 0 || | |
| 407 _stricmp(codec.plname, "CN") == 0 || | |
| 408 _stricmp(codec.plname, "red") == 0 ) { | |
| 409 return -1; | |
| 410 } | |
| 411 channels_[channel]->send_codec = codec; | |
| 412 ++num_set_send_codecs_; | |
| 413 return 0; | |
| 414 } | |
| 415 WEBRTC_FUNC(GetSendCodec, (int channel, webrtc::CodecInst& codec)) { | |
| 416 WEBRTC_CHECK_CHANNEL(channel); | |
| 417 codec = channels_[channel]->send_codec; | |
| 418 return 0; | |
| 419 } | |
| 420 WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps)); | |
| 421 WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec)); | |
| 422 WEBRTC_FUNC(SetRecPayloadType, (int channel, | |
| 423 const webrtc::CodecInst& codec)) { | |
| 424 WEBRTC_CHECK_CHANNEL(channel); | |
| 425 Channel* ch = channels_[channel]; | |
| 426 if (ch->playout) | |
| 427 return -1; // Channel is in use. | |
| 428 // Check if something else already has this slot. | |
| 429 if (codec.pltype != -1) { | |
| 430 for (std::vector<webrtc::CodecInst>::iterator it = | |
| 431 ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) { | |
| 432 if (it->pltype == codec.pltype && | |
| 433 _stricmp(it->plname, codec.plname) != 0) { | |
| 434 return -1; | |
| 435 } | |
| 436 } | |
| 437 } | |
| 438 // Otherwise try to find this codec and update its payload type. | |
| 439 int result = -1; // not found | |
| 440 for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin(); | |
| 441 it != ch->recv_codecs.end(); ++it) { | |
| 442 if (strcmp(it->plname, codec.plname) == 0 && | |
| 443 it->plfreq == codec.plfreq && | |
| 444 it->channels == codec.channels) { | |
| 445 it->pltype = codec.pltype; | |
| 446 result = 0; | |
| 447 } | |
| 448 } | |
| 449 return result; | |
| 450 } | |
| 451 WEBRTC_FUNC(SetSendCNPayloadType, (int channel, int type, | |
| 452 webrtc::PayloadFrequencies frequency)) { | |
| 453 WEBRTC_CHECK_CHANNEL(channel); | |
| 454 if (frequency == webrtc::kFreq8000Hz) { | |
| 455 channels_[channel]->cn8_type = type; | |
| 456 } else if (frequency == webrtc::kFreq16000Hz) { | |
| 457 channels_[channel]->cn16_type = type; | |
| 458 } | |
| 459 return 0; | |
| 460 } | |
| 461 WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) { | |
| 462 WEBRTC_CHECK_CHANNEL(channel); | |
| 463 Channel* ch = channels_[channel]; | |
| 464 for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin(); | |
| 465 it != ch->recv_codecs.end(); ++it) { | |
| 466 if (strcmp(it->plname, codec.plname) == 0 && | |
| 467 it->plfreq == codec.plfreq && | |
| 468 it->channels == codec.channels && | |
| 469 it->pltype != -1) { | |
| 470 codec.pltype = it->pltype; | |
| 471 return 0; | |
| 472 } | |
| 473 } | |
| 474 return -1; // not found | |
| 475 } | |
| 476 WEBRTC_FUNC(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode, | |
| 477 bool disableDTX)) { | |
| 478 WEBRTC_CHECK_CHANNEL(channel); | |
| 479 if (channels_[channel]->send_codec.channels == 2) { | |
| 480 // Replicating VoE behavior; VAD cannot be enabled for stereo. | |
| 481 return -1; | |
| 482 } | |
| 483 channels_[channel]->vad = enable; | |
| 484 return 0; | |
| 485 } | |
| 486 WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled, | |
| 487 webrtc::VadModes& mode, bool& disabledDTX)); | |
| 488 | |
| 489 WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) { | |
| 490 WEBRTC_CHECK_CHANNEL(channel); | |
| 491 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { | |
| 492 // Return -1 if current send codec is not Opus. | |
| 493 // TODO(minyue): Excludes other codecs if they support inband FEC. | |
| 494 return -1; | |
| 495 } | |
| 496 channels_[channel]->codec_fec = enable; | |
| 497 return 0; | |
| 498 } | |
| 499 WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable)) { | |
| 500 WEBRTC_CHECK_CHANNEL(channel); | |
| 501 enable = channels_[channel]->codec_fec; | |
| 502 return 0; | |
| 503 } | |
| 504 | |
| 505 WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) { | |
| 506 WEBRTC_CHECK_CHANNEL(channel); | |
| 507 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { | |
| 508 // Return -1 if current send codec is not Opus. | |
| 509 return -1; | |
| 510 } | |
| 511 if (frequency_hz <= 8000) | |
| 512 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthNb; | |
| 513 else if (frequency_hz <= 12000) | |
| 514 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthMb; | |
| 515 else if (frequency_hz <= 16000) | |
| 516 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthWb; | |
| 517 else if (frequency_hz <= 24000) | |
| 518 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthSwb; | |
| 519 else | |
| 520 channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb; | |
| 521 return 0; | |
| 522 } | |
| 523 | |
| 524 WEBRTC_FUNC(SetOpusDtx, (int channel, bool enable_dtx)) { | |
| 525 WEBRTC_CHECK_CHANNEL(channel); | |
| 526 if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) { | |
| 527 // Return -1 if current send codec is not Opus. | |
| 528 return -1; | |
| 529 } | |
| 530 channels_[channel]->opus_dtx = enable_dtx; | |
| 531 return 0; | |
| 532 } | |
| 533 | |
| 534 // webrtc::VoEHardware | |
| 535 WEBRTC_STUB(GetNumOfRecordingDevices, (int& num)); | |
| 536 WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num)); | |
| 537 WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid)); | |
| 538 WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid)); | |
| 539 WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel)); | |
| 540 WEBRTC_STUB(SetPlayoutDevice, (int)); | |
| 541 WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers)); | |
| 542 WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&)); | |
| 543 WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) { | |
| 544 recording_sample_rate_ = samples_per_sec; | |
| 545 return 0; | |
| 546 } | |
| 547 WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) { | |
| 548 *samples_per_sec = recording_sample_rate_; | |
| 549 return 0; | |
| 550 } | |
| 551 WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) { | |
| 552 playout_sample_rate_ = samples_per_sec; | |
| 553 return 0; | |
| 554 } | |
| 555 WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) { | |
| 556 *samples_per_sec = playout_sample_rate_; | |
| 557 return 0; | |
| 558 } | |
| 559 WEBRTC_STUB(EnableBuiltInAEC, (bool enable)); | |
| 560 virtual bool BuiltInAECIsAvailable() const { return false; } | |
| 561 WEBRTC_STUB(EnableBuiltInAGC, (bool enable)); | |
| 562 virtual bool BuiltInAGCIsAvailable() const { return false; } | |
| 563 WEBRTC_STUB(EnableBuiltInNS, (bool enable)); | |
| 564 virtual bool BuiltInNSIsAvailable() const { return false; } | |
| 565 | |
| 566 // webrtc::VoENetwork | |
| 567 WEBRTC_FUNC(RegisterExternalTransport, (int channel, | |
| 568 webrtc::Transport& transport)) { | |
| 569 WEBRTC_CHECK_CHANNEL(channel); | |
| 570 channels_[channel]->external_transport = true; | |
| 571 return 0; | |
| 572 } | |
| 573 WEBRTC_FUNC(DeRegisterExternalTransport, (int channel)) { | |
| 574 WEBRTC_CHECK_CHANNEL(channel); | |
| 575 channels_[channel]->external_transport = false; | |
| 576 return 0; | |
| 577 } | |
| 578 WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data, | |
| 579 size_t length)) { | |
| 580 WEBRTC_CHECK_CHANNEL(channel); | |
| 581 if (!channels_[channel]->external_transport) return -1; | |
| 582 channels_[channel]->packets.push_back( | |
| 583 std::string(static_cast<const char*>(data), length)); | |
| 584 return 0; | |
| 585 } | |
| 586 WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data, | |
| 587 size_t length, | |
| 588 const webrtc::PacketTime& packet_time)) { | |
| 589 WEBRTC_CHECK_CHANNEL(channel); | |
| 590 if (ReceivedRTPPacket(channel, data, length) == -1) { | |
| 591 return -1; | |
| 592 } | |
| 593 channels_[channel]->last_rtp_packet_time = packet_time; | |
| 594 return 0; | |
| 595 } | |
| 596 | |
| 597 WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data, | |
| 598 size_t length)); | |
| 599 | |
| 600 // webrtc::VoERTP_RTCP | |
| 601 WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) { | |
| 602 WEBRTC_CHECK_CHANNEL(channel); | |
| 603 channels_[channel]->send_ssrc = ssrc; | |
| 604 return 0; | |
| 605 } | |
| 606 WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc)); | |
| 607 WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc)); | |
| 608 WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable, | |
| 609 unsigned char id)); | |
| 610 WEBRTC_STUB(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable, | |
| 611 unsigned char id)); | |
| 612 WEBRTC_STUB(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable, | |
| 613 unsigned char id)); | |
| 614 WEBRTC_STUB(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable, | |
| 615 unsigned char id)); | |
| 616 WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable)); | |
| 617 WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled)); | |
| 618 WEBRTC_STUB(SetRTCP_CNAME, (int channel, const char cname[256])); | |
| 619 WEBRTC_STUB(GetRTCP_CNAME, (int channel, char cname[256])); | |
| 620 WEBRTC_STUB(GetRemoteRTCP_CNAME, (int channel, char* cname)); | |
| 621 WEBRTC_STUB(GetRemoteRTCPData, (int channel, unsigned int& NTPHigh, | |
| 622 unsigned int& NTPLow, | |
| 623 unsigned int& timestamp, | |
| 624 unsigned int& playoutTimestamp, | |
| 625 unsigned int* jitter, | |
| 626 unsigned short* fractionLost)); | |
| 627 WEBRTC_STUB(GetRemoteRTCPReportBlocks, | |
| 628 (int channel, std::vector<webrtc::ReportBlock>* receive_blocks)); | |
| 629 WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs, | |
| 630 unsigned int& maxJitterMs, | |
| 631 unsigned int& discardedPackets)); | |
| 632 WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats)); | |
| 633 WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) { | |
| 634 WEBRTC_CHECK_CHANNEL(channel); | |
| 635 channels_[channel]->red = enable; | |
| 636 channels_[channel]->red_type = redPayloadtype; | |
| 637 return 0; | |
| 638 } | |
| 639 WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) { | |
| 640 WEBRTC_CHECK_CHANNEL(channel); | |
| 641 enable = channels_[channel]->red; | |
| 642 redPayloadtype = channels_[channel]->red_type; | |
| 643 return 0; | |
| 644 } | |
| 645 WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) { | |
| 646 WEBRTC_CHECK_CHANNEL(channel); | |
| 647 channels_[channel]->nack = enable; | |
| 648 channels_[channel]->nack_max_packets = maxNoPackets; | |
| 649 return 0; | |
| 650 } | |
| 651 | |
| 652 // webrtc::VoEVolumeControl | |
| 653 WEBRTC_STUB(SetSpeakerVolume, (unsigned int)); | |
| 654 WEBRTC_STUB(GetSpeakerVolume, (unsigned int&)); | |
| 655 WEBRTC_STUB(SetMicVolume, (unsigned int)); | |
| 656 WEBRTC_STUB(GetMicVolume, (unsigned int&)); | |
| 657 WEBRTC_STUB(SetInputMute, (int, bool)); | |
| 658 WEBRTC_STUB(GetInputMute, (int, bool&)); | |
| 659 WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&)); | |
| 660 WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&)); | |
| 661 WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&)); | |
| 662 WEBRTC_STUB(GetSpeechOutputLevelFullRange, (int, unsigned int&)); | |
| 663 WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) { | |
| 664 WEBRTC_CHECK_CHANNEL(channel); | |
| 665 channels_[channel]->volume_scale= scale; | |
| 666 return 0; | |
| 667 } | |
| 668 WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) { | |
| 669 WEBRTC_CHECK_CHANNEL(channel); | |
| 670 scale = channels_[channel]->volume_scale; | |
| 671 return 0; | |
| 672 } | |
| 673 WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right)); | |
| 674 WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right)); | |
| 675 | |
| 676 // webrtc::VoEAudioProcessing | |
| 677 WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) { | |
| 678 ns_enabled_ = enable; | |
| 679 ns_mode_ = mode; | |
| 680 return 0; | |
| 681 } | |
| 682 WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) { | |
| 683 enabled = ns_enabled_; | |
| 684 mode = ns_mode_; | |
| 685 return 0; | |
| 686 } | |
| 687 | |
| 688 WEBRTC_FUNC(SetAgcStatus, (bool enable, webrtc::AgcModes mode)) { | |
| 689 agc_enabled_ = enable; | |
| 690 agc_mode_ = mode; | |
| 691 return 0; | |
| 692 } | |
| 693 WEBRTC_FUNC(GetAgcStatus, (bool& enabled, webrtc::AgcModes& mode)) { | |
| 694 enabled = agc_enabled_; | |
| 695 mode = agc_mode_; | |
| 696 return 0; | |
| 697 } | |
| 698 | |
| 699 WEBRTC_FUNC(SetAgcConfig, (webrtc::AgcConfig config)) { | |
| 700 agc_config_ = config; | |
| 701 return 0; | |
| 702 } | |
| 703 WEBRTC_FUNC(GetAgcConfig, (webrtc::AgcConfig& config)) { | |
| 704 config = agc_config_; | |
| 705 return 0; | |
| 706 } | |
| 707 WEBRTC_FUNC(SetEcStatus, (bool enable, webrtc::EcModes mode)) { | |
| 708 ec_enabled_ = enable; | |
| 709 ec_mode_ = mode; | |
| 710 return 0; | |
| 711 } | |
| 712 WEBRTC_FUNC(GetEcStatus, (bool& enabled, webrtc::EcModes& mode)) { | |
| 713 enabled = ec_enabled_; | |
| 714 mode = ec_mode_; | |
| 715 return 0; | |
| 716 } | |
| 717 WEBRTC_STUB(EnableDriftCompensation, (bool enable)) | |
| 718 WEBRTC_BOOL_STUB(DriftCompensationEnabled, ()) | |
| 719 WEBRTC_VOID_STUB(SetDelayOffsetMs, (int offset)) | |
| 720 WEBRTC_STUB(DelayOffsetMs, ()); | |
| 721 WEBRTC_FUNC(SetAecmMode, (webrtc::AecmModes mode, bool enableCNG)) { | |
| 722 aecm_mode_ = mode; | |
| 723 cng_enabled_ = enableCNG; | |
| 724 return 0; | |
| 725 } | |
| 726 WEBRTC_FUNC(GetAecmMode, (webrtc::AecmModes& mode, bool& enabledCNG)) { | |
| 727 mode = aecm_mode_; | |
| 728 enabledCNG = cng_enabled_; | |
| 729 return 0; | |
| 730 } | |
| 731 WEBRTC_STUB(SetRxNsStatus, (int channel, bool enable, webrtc::NsModes mode)); | |
| 732 WEBRTC_STUB(GetRxNsStatus, (int channel, bool& enabled, | |
| 733 webrtc::NsModes& mode)); | |
| 734 WEBRTC_STUB(SetRxAgcStatus, (int channel, bool enable, | |
| 735 webrtc::AgcModes mode)); | |
| 736 WEBRTC_STUB(GetRxAgcStatus, (int channel, bool& enabled, | |
| 737 webrtc::AgcModes& mode)); | |
| 738 WEBRTC_STUB(SetRxAgcConfig, (int channel, webrtc::AgcConfig config)); | |
| 739 WEBRTC_STUB(GetRxAgcConfig, (int channel, webrtc::AgcConfig& config)); | |
| 740 | |
| 741 WEBRTC_STUB(RegisterRxVadObserver, (int, webrtc::VoERxVadCallback&)); | |
| 742 WEBRTC_STUB(DeRegisterRxVadObserver, (int channel)); | |
| 743 WEBRTC_STUB(VoiceActivityIndicator, (int channel)); | |
| 744 WEBRTC_FUNC(SetEcMetricsStatus, (bool enable)) { | |
| 745 ec_metrics_enabled_ = enable; | |
| 746 return 0; | |
| 747 } | |
| 748 WEBRTC_STUB(GetEcMetricsStatus, (bool& enabled)); | |
| 749 WEBRTC_STUB(GetEchoMetrics, (int& ERL, int& ERLE, int& RERL, int& A_NLP)); | |
| 750 WEBRTC_STUB(GetEcDelayMetrics, (int& delay_median, int& delay_std, | |
| 751 float& fraction_poor_delays)); | |
| 752 | |
| 753 WEBRTC_STUB(StartDebugRecording, (const char* fileNameUTF8)); | |
| 754 WEBRTC_STUB(StartDebugRecording, (FILE* handle)); | |
| 755 WEBRTC_STUB(StopDebugRecording, ()); | |
| 756 | |
| 757 WEBRTC_FUNC(SetTypingDetectionStatus, (bool enable)) { | |
| 758 typing_detection_enabled_ = enable; | |
| 759 return 0; | |
| 760 } | |
| 761 WEBRTC_FUNC(GetTypingDetectionStatus, (bool& enabled)) { | |
| 762 enabled = typing_detection_enabled_; | |
| 763 return 0; | |
| 764 } | |
| 765 | |
| 766 WEBRTC_STUB(TimeSinceLastTyping, (int& seconds)); | |
| 767 WEBRTC_STUB(SetTypingDetectionParameters, (int timeWindow, | |
| 768 int costPerTyping, | |
| 769 int reportingThreshold, | |
| 770 int penaltyDecay, | |
| 771 int typeEventDelay)); | |
| 772 int EnableHighPassFilter(bool enable) { | |
| 773 highpass_filter_enabled_ = enable; | |
| 774 return 0; | |
| 775 } | |
| 776 bool IsHighPassFilterEnabled() { | |
| 777 return highpass_filter_enabled_; | |
| 778 } | |
| 779 bool IsStereoChannelSwappingEnabled() { | |
| 780 return stereo_swapping_enabled_; | |
| 781 } | |
| 782 void EnableStereoChannelSwapping(bool enable) { | |
| 783 stereo_swapping_enabled_ = enable; | |
| 784 } | |
| 785 int GetNetEqCapacity() const { | |
| 786 auto ch = channels_.find(last_channel_); | |
| 787 ASSERT(ch != channels_.end()); | |
| 788 return ch->second->neteq_capacity; | |
| 789 } | |
| 790 bool GetNetEqFastAccelerate() const { | |
| 791 auto ch = channels_.find(last_channel_); | |
| 792 ASSERT(ch != channels_.end()); | |
| 793 return ch->second->neteq_fast_accelerate; | |
| 794 } | |
| 795 | |
| 796 private: | |
| 797 bool inited_; | |
| 798 int last_channel_; | |
| 799 std::map<int, Channel*> channels_; | |
| 800 bool fail_create_channel_; | |
| 801 int num_set_send_codecs_; // how many times we call SetSendCodec(). | |
| 802 bool ec_enabled_; | |
| 803 bool ec_metrics_enabled_; | |
| 804 bool cng_enabled_; | |
| 805 bool ns_enabled_; | |
| 806 bool agc_enabled_; | |
| 807 bool highpass_filter_enabled_; | |
| 808 bool stereo_swapping_enabled_; | |
| 809 bool typing_detection_enabled_; | |
| 810 webrtc::EcModes ec_mode_; | |
| 811 webrtc::AecmModes aecm_mode_; | |
| 812 webrtc::NsModes ns_mode_; | |
| 813 webrtc::AgcModes agc_mode_; | |
| 814 webrtc::AgcConfig agc_config_; | |
| 815 webrtc::VoiceEngineObserver* observer_; | |
| 816 int playout_fail_channel_; | |
| 817 int send_fail_channel_; | |
| 818 int recording_sample_rate_; | |
| 819 int playout_sample_rate_; | |
| 820 FakeAudioProcessing audio_processing_; | |
| 821 }; | |
| 822 | |
| 823 } // namespace cricket | |
| 824 | |
| 825 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 1587193006:
Move talk/media to webrtc/media (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master