andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 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 |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
turaj@webrtc.org | ead8a5b | 2013-02-12 21:42:18 +0000 | [diff] [blame] | 11 | #include "webrtc/voice_engine/channel.h" |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 12 | |
turaj@webrtc.org | ead8a5b | 2013-02-12 21:42:18 +0000 | [diff] [blame] | 13 | #include "webrtc/modules/audio_device/include/audio_device.h" |
| 14 | #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| 15 | #include "webrtc/modules/utility/interface/audio_frame_operations.h" |
| 16 | #include "webrtc/modules/utility/interface/process_thread.h" |
| 17 | #include "webrtc/modules/utility/interface/rtp_dump.h" |
| 18 | #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| 19 | #include "webrtc/system_wrappers/interface/logging.h" |
| 20 | #include "webrtc/system_wrappers/interface/trace.h" |
| 21 | #include "webrtc/voice_engine/include/voe_base.h" |
| 22 | #include "webrtc/voice_engine/include/voe_external_media.h" |
| 23 | #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| 24 | #include "webrtc/voice_engine/output_mixer.h" |
| 25 | #include "webrtc/voice_engine/statistics.h" |
| 26 | #include "webrtc/voice_engine/transmit_mixer.h" |
| 27 | #include "webrtc/voice_engine/utility.h" |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 28 | |
| 29 | #if defined(_WIN32) |
| 30 | #include <Qos.h> |
| 31 | #endif |
| 32 | |
andrew@webrtc.org | d898c01 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 33 | namespace webrtc { |
| 34 | namespace voe { |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 35 | |
| 36 | WebRtc_Word32 |
| 37 | Channel::SendData(FrameType frameType, |
| 38 | WebRtc_UWord8 payloadType, |
| 39 | WebRtc_UWord32 timeStamp, |
| 40 | const WebRtc_UWord8* payloadData, |
| 41 | WebRtc_UWord16 payloadSize, |
| 42 | const RTPFragmentationHeader* fragmentation) |
| 43 | { |
| 44 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 45 | "Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u," |
| 46 | " payloadSize=%u, fragmentation=0x%x)", |
| 47 | frameType, payloadType, timeStamp, payloadSize, fragmentation); |
| 48 | |
| 49 | if (_includeAudioLevelIndication) |
| 50 | { |
| 51 | assert(_rtpAudioProc.get() != NULL); |
| 52 | // Store current audio level in the RTP/RTCP module. |
| 53 | // The level will be used in combination with voice-activity state |
| 54 | // (frameType) to add an RTP header extension |
| 55 | _rtpRtcpModule->SetAudioLevel(_rtpAudioProc->level_estimator()->RMS()); |
| 56 | } |
| 57 | |
| 58 | // Push data from ACM to RTP/RTCP-module to deliver audio frame for |
| 59 | // packetization. |
| 60 | // This call will trigger Transport::SendPacket() from the RTP/RTCP module. |
| 61 | if (_rtpRtcpModule->SendOutgoingData((FrameType&)frameType, |
| 62 | payloadType, |
| 63 | timeStamp, |
| 64 | // Leaving the time when this frame was |
| 65 | // received from the capture device as |
| 66 | // undefined for voice for now. |
| 67 | -1, |
| 68 | payloadData, |
| 69 | payloadSize, |
| 70 | fragmentation) == -1) |
| 71 | { |
| 72 | _engineStatisticsPtr->SetLastError( |
| 73 | VE_RTP_RTCP_MODULE_ERROR, kTraceWarning, |
| 74 | "Channel::SendData() failed to send data to RTP/RTCP module"); |
| 75 | return -1; |
| 76 | } |
| 77 | |
| 78 | _lastLocalTimeStamp = timeStamp; |
| 79 | _lastPayloadType = payloadType; |
| 80 | |
| 81 | return 0; |
| 82 | } |
| 83 | |
| 84 | WebRtc_Word32 |
| 85 | Channel::InFrameType(WebRtc_Word16 frameType) |
| 86 | { |
| 87 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 88 | "Channel::InFrameType(frameType=%d)", frameType); |
| 89 | |
| 90 | CriticalSectionScoped cs(&_callbackCritSect); |
| 91 | // 1 indicates speech |
| 92 | _sendFrameType = (frameType == 1) ? 1 : 0; |
| 93 | return 0; |
| 94 | } |
| 95 | |
| 96 | #ifdef WEBRTC_DTMF_DETECTION |
| 97 | int |
| 98 | Channel::IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool end) |
| 99 | { |
| 100 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 101 | "Channel::IncomingDtmf(digitDtmf=%u, end=%d)", |
| 102 | digitDtmf, end); |
| 103 | |
| 104 | if (digitDtmf != 999) |
| 105 | { |
| 106 | CriticalSectionScoped cs(&_callbackCritSect); |
| 107 | if (_telephoneEventDetectionPtr) |
| 108 | { |
| 109 | _telephoneEventDetectionPtr->OnReceivedTelephoneEventInband( |
| 110 | _channelId, digitDtmf, end); |
| 111 | } |
| 112 | } |
| 113 | |
| 114 | return 0; |
| 115 | } |
| 116 | #endif |
| 117 | |
| 118 | WebRtc_Word32 |
| 119 | Channel::OnRxVadDetected(const int vadDecision) |
| 120 | { |
| 121 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 122 | "Channel::OnRxVadDetected(vadDecision=%d)", vadDecision); |
| 123 | |
| 124 | CriticalSectionScoped cs(&_callbackCritSect); |
| 125 | if (_rxVadObserverPtr) |
| 126 | { |
| 127 | _rxVadObserverPtr->OnRxVad(_channelId, vadDecision); |
| 128 | } |
| 129 | |
| 130 | return 0; |
| 131 | } |
| 132 | |
| 133 | int |
| 134 | Channel::SendPacket(int channel, const void *data, int len) |
| 135 | { |
| 136 | channel = VoEChannelId(channel); |
| 137 | assert(channel == _channelId); |
| 138 | |
| 139 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 140 | "Channel::SendPacket(channel=%d, len=%d)", channel, len); |
| 141 | |
| 142 | if (_transportPtr == NULL) |
| 143 | { |
| 144 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 145 | "Channel::SendPacket() failed to send RTP packet due to" |
| 146 | " invalid transport object"); |
| 147 | return -1; |
| 148 | } |
| 149 | |
| 150 | // Insert extra RTP packet using if user has called the InsertExtraRTPPacket |
| 151 | // API |
| 152 | if (_insertExtraRTPPacket) |
| 153 | { |
| 154 | WebRtc_UWord8* rtpHdr = (WebRtc_UWord8*)data; |
| 155 | WebRtc_UWord8 M_PT(0); |
| 156 | if (_extraMarkerBit) |
| 157 | { |
| 158 | M_PT = 0x80; // set the M-bit |
| 159 | } |
| 160 | M_PT += _extraPayloadType; // set the payload type |
| 161 | *(++rtpHdr) = M_PT; // modify the M|PT-byte within the RTP header |
| 162 | _insertExtraRTPPacket = false; // insert one packet only |
| 163 | } |
| 164 | |
| 165 | WebRtc_UWord8* bufferToSendPtr = (WebRtc_UWord8*)data; |
| 166 | WebRtc_Word32 bufferLength = len; |
| 167 | |
| 168 | // Dump the RTP packet to a file (if RTP dump is enabled). |
| 169 | if (_rtpDumpOut.DumpPacket((const WebRtc_UWord8*)data, len) == -1) |
| 170 | { |
| 171 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 172 | VoEId(_instanceId,_channelId), |
| 173 | "Channel::SendPacket() RTP dump to output file failed"); |
| 174 | } |
| 175 | |
| 176 | // SRTP or External encryption |
| 177 | if (_encrypting) |
| 178 | { |
| 179 | CriticalSectionScoped cs(&_callbackCritSect); |
| 180 | |
| 181 | if (_encryptionPtr) |
| 182 | { |
| 183 | if (!_encryptionRTPBufferPtr) |
| 184 | { |
| 185 | // Allocate memory for encryption buffer one time only |
| 186 | _encryptionRTPBufferPtr = |
| 187 | new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes]; |
xians@webrtc.org | bc53c40 | 2012-10-25 13:58:02 +0000 | [diff] [blame] | 188 | memset(_encryptionRTPBufferPtr, 0, |
| 189 | kVoiceEngineMaxIpPacketSizeBytes); |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 190 | } |
| 191 | |
| 192 | // Perform encryption (SRTP or external) |
| 193 | WebRtc_Word32 encryptedBufferLength = 0; |
| 194 | _encryptionPtr->encrypt(_channelId, |
| 195 | bufferToSendPtr, |
| 196 | _encryptionRTPBufferPtr, |
| 197 | bufferLength, |
| 198 | (int*)&encryptedBufferLength); |
| 199 | if (encryptedBufferLength <= 0) |
| 200 | { |
| 201 | _engineStatisticsPtr->SetLastError( |
| 202 | VE_ENCRYPTION_FAILED, |
| 203 | kTraceError, "Channel::SendPacket() encryption failed"); |
| 204 | return -1; |
| 205 | } |
| 206 | |
| 207 | // Replace default data buffer with encrypted buffer |
| 208 | bufferToSendPtr = _encryptionRTPBufferPtr; |
| 209 | bufferLength = encryptedBufferLength; |
| 210 | } |
| 211 | } |
| 212 | |
| 213 | // Packet transmission using WebRtc socket transport |
| 214 | if (!_externalTransport) |
| 215 | { |
| 216 | int n = _transportPtr->SendPacket(channel, bufferToSendPtr, |
| 217 | bufferLength); |
| 218 | if (n < 0) |
| 219 | { |
| 220 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 221 | VoEId(_instanceId,_channelId), |
| 222 | "Channel::SendPacket() RTP transmission using WebRtc" |
| 223 | " sockets failed"); |
| 224 | return -1; |
| 225 | } |
| 226 | return n; |
| 227 | } |
| 228 | |
| 229 | // Packet transmission using external transport transport |
| 230 | { |
| 231 | CriticalSectionScoped cs(&_callbackCritSect); |
| 232 | |
| 233 | int n = _transportPtr->SendPacket(channel, |
| 234 | bufferToSendPtr, |
| 235 | bufferLength); |
| 236 | if (n < 0) |
| 237 | { |
| 238 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 239 | VoEId(_instanceId,_channelId), |
| 240 | "Channel::SendPacket() RTP transmission using external" |
| 241 | " transport failed"); |
| 242 | return -1; |
| 243 | } |
| 244 | return n; |
| 245 | } |
| 246 | } |
| 247 | |
| 248 | int |
| 249 | Channel::SendRTCPPacket(int channel, const void *data, int len) |
| 250 | { |
| 251 | channel = VoEChannelId(channel); |
| 252 | assert(channel == _channelId); |
| 253 | |
| 254 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 255 | "Channel::SendRTCPPacket(channel=%d, len=%d)", channel, len); |
| 256 | |
| 257 | { |
| 258 | CriticalSectionScoped cs(&_callbackCritSect); |
| 259 | if (_transportPtr == NULL) |
| 260 | { |
| 261 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 262 | VoEId(_instanceId,_channelId), |
| 263 | "Channel::SendRTCPPacket() failed to send RTCP packet" |
| 264 | " due to invalid transport object"); |
| 265 | return -1; |
| 266 | } |
| 267 | } |
| 268 | |
| 269 | WebRtc_UWord8* bufferToSendPtr = (WebRtc_UWord8*)data; |
| 270 | WebRtc_Word32 bufferLength = len; |
| 271 | |
| 272 | // Dump the RTCP packet to a file (if RTP dump is enabled). |
| 273 | if (_rtpDumpOut.DumpPacket((const WebRtc_UWord8*)data, len) == -1) |
| 274 | { |
| 275 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 276 | VoEId(_instanceId,_channelId), |
| 277 | "Channel::SendPacket() RTCP dump to output file failed"); |
| 278 | } |
| 279 | |
| 280 | // SRTP or External encryption |
| 281 | if (_encrypting) |
| 282 | { |
| 283 | CriticalSectionScoped cs(&_callbackCritSect); |
| 284 | |
| 285 | if (_encryptionPtr) |
| 286 | { |
| 287 | if (!_encryptionRTCPBufferPtr) |
| 288 | { |
| 289 | // Allocate memory for encryption buffer one time only |
| 290 | _encryptionRTCPBufferPtr = |
| 291 | new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes]; |
| 292 | } |
| 293 | |
| 294 | // Perform encryption (SRTP or external). |
| 295 | WebRtc_Word32 encryptedBufferLength = 0; |
| 296 | _encryptionPtr->encrypt_rtcp(_channelId, |
| 297 | bufferToSendPtr, |
| 298 | _encryptionRTCPBufferPtr, |
| 299 | bufferLength, |
| 300 | (int*)&encryptedBufferLength); |
| 301 | if (encryptedBufferLength <= 0) |
| 302 | { |
| 303 | _engineStatisticsPtr->SetLastError( |
| 304 | VE_ENCRYPTION_FAILED, kTraceError, |
| 305 | "Channel::SendRTCPPacket() encryption failed"); |
| 306 | return -1; |
| 307 | } |
| 308 | |
| 309 | // Replace default data buffer with encrypted buffer |
| 310 | bufferToSendPtr = _encryptionRTCPBufferPtr; |
| 311 | bufferLength = encryptedBufferLength; |
| 312 | } |
| 313 | } |
| 314 | |
| 315 | // Packet transmission using WebRtc socket transport |
| 316 | if (!_externalTransport) |
| 317 | { |
| 318 | int n = _transportPtr->SendRTCPPacket(channel, |
| 319 | bufferToSendPtr, |
| 320 | bufferLength); |
| 321 | if (n < 0) |
| 322 | { |
| 323 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 324 | VoEId(_instanceId,_channelId), |
| 325 | "Channel::SendRTCPPacket() transmission using WebRtc" |
| 326 | " sockets failed"); |
| 327 | return -1; |
| 328 | } |
| 329 | return n; |
| 330 | } |
| 331 | |
| 332 | // Packet transmission using external transport transport |
| 333 | { |
| 334 | CriticalSectionScoped cs(&_callbackCritSect); |
henrike@webrtc.org | 03a161e | 2012-11-18 18:49:13 +0000 | [diff] [blame] | 335 | if (_transportPtr == NULL) |
| 336 | { |
| 337 | return -1; |
| 338 | } |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 339 | int n = _transportPtr->SendRTCPPacket(channel, |
| 340 | bufferToSendPtr, |
| 341 | bufferLength); |
| 342 | if (n < 0) |
| 343 | { |
| 344 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 345 | VoEId(_instanceId,_channelId), |
| 346 | "Channel::SendRTCPPacket() transmission using external" |
| 347 | " transport failed"); |
| 348 | return -1; |
| 349 | } |
| 350 | return n; |
| 351 | } |
| 352 | |
| 353 | return len; |
| 354 | } |
| 355 | |
| 356 | void |
| 357 | Channel::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket, |
| 358 | const WebRtc_Word32 rtpPacketLength, |
| 359 | const char* fromIP, |
| 360 | const WebRtc_UWord16 fromPort) |
| 361 | { |
| 362 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 363 | "Channel::IncomingRTPPacket(rtpPacketLength=%d," |
| 364 | " fromIP=%s, fromPort=%u)", |
| 365 | rtpPacketLength, fromIP, fromPort); |
| 366 | |
| 367 | // Store playout timestamp for the received RTP packet |
| 368 | // to be used for upcoming delay estimations |
| 369 | WebRtc_UWord32 playoutTimestamp(0); |
| 370 | if (GetPlayoutTimeStamp(playoutTimestamp) == 0) |
| 371 | { |
| 372 | _playoutTimeStampRTP = playoutTimestamp; |
| 373 | } |
| 374 | |
| 375 | WebRtc_UWord8* rtpBufferPtr = (WebRtc_UWord8*)incomingRtpPacket; |
| 376 | WebRtc_Word32 rtpBufferLength = rtpPacketLength; |
| 377 | |
| 378 | // SRTP or External decryption |
| 379 | if (_decrypting) |
| 380 | { |
| 381 | CriticalSectionScoped cs(&_callbackCritSect); |
| 382 | |
| 383 | if (_encryptionPtr) |
| 384 | { |
| 385 | if (!_decryptionRTPBufferPtr) |
| 386 | { |
| 387 | // Allocate memory for decryption buffer one time only |
| 388 | _decryptionRTPBufferPtr = |
| 389 | new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes]; |
| 390 | } |
| 391 | |
| 392 | // Perform decryption (SRTP or external) |
| 393 | WebRtc_Word32 decryptedBufferLength = 0; |
| 394 | _encryptionPtr->decrypt(_channelId, |
| 395 | rtpBufferPtr, |
| 396 | _decryptionRTPBufferPtr, |
| 397 | rtpBufferLength, |
| 398 | (int*)&decryptedBufferLength); |
| 399 | if (decryptedBufferLength <= 0) |
| 400 | { |
| 401 | _engineStatisticsPtr->SetLastError( |
| 402 | VE_DECRYPTION_FAILED, kTraceError, |
| 403 | "Channel::IncomingRTPPacket() decryption failed"); |
| 404 | return; |
| 405 | } |
| 406 | |
| 407 | // Replace default data buffer with decrypted buffer |
| 408 | rtpBufferPtr = _decryptionRTPBufferPtr; |
| 409 | rtpBufferLength = decryptedBufferLength; |
| 410 | } |
| 411 | } |
| 412 | |
| 413 | // Dump the RTP packet to a file (if RTP dump is enabled). |
| 414 | if (_rtpDumpIn.DumpPacket(rtpBufferPtr, |
| 415 | (WebRtc_UWord16)rtpBufferLength) == -1) |
| 416 | { |
| 417 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 418 | VoEId(_instanceId,_channelId), |
| 419 | "Channel::SendPacket() RTP dump to input file failed"); |
| 420 | } |
| 421 | |
| 422 | // Deliver RTP packet to RTP/RTCP module for parsing |
| 423 | // The packet will be pushed back to the channel thru the |
| 424 | // OnReceivedPayloadData callback so we don't push it to the ACM here |
| 425 | if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)rtpBufferPtr, |
| 426 | (WebRtc_UWord16)rtpBufferLength) == -1) |
| 427 | { |
| 428 | _engineStatisticsPtr->SetLastError( |
| 429 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning, |
| 430 | "Channel::IncomingRTPPacket() RTP packet is invalid"); |
| 431 | return; |
| 432 | } |
| 433 | } |
| 434 | |
| 435 | void |
| 436 | Channel::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket, |
| 437 | const WebRtc_Word32 rtcpPacketLength, |
| 438 | const char* fromIP, |
| 439 | const WebRtc_UWord16 fromPort) |
| 440 | { |
| 441 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 442 | "Channel::IncomingRTCPPacket(rtcpPacketLength=%d, fromIP=%s," |
| 443 | " fromPort=%u)", |
| 444 | rtcpPacketLength, fromIP, fromPort); |
| 445 | |
| 446 | // Temporary buffer pointer and size for decryption |
| 447 | WebRtc_UWord8* rtcpBufferPtr = (WebRtc_UWord8*)incomingRtcpPacket; |
| 448 | WebRtc_Word32 rtcpBufferLength = rtcpPacketLength; |
| 449 | |
| 450 | // Store playout timestamp for the received RTCP packet |
| 451 | // which will be read by the GetRemoteRTCPData API |
| 452 | WebRtc_UWord32 playoutTimestamp(0); |
| 453 | if (GetPlayoutTimeStamp(playoutTimestamp) == 0) |
| 454 | { |
| 455 | _playoutTimeStampRTCP = playoutTimestamp; |
| 456 | } |
| 457 | |
| 458 | // SRTP or External decryption |
| 459 | if (_decrypting) |
| 460 | { |
| 461 | CriticalSectionScoped cs(&_callbackCritSect); |
| 462 | |
| 463 | if (_encryptionPtr) |
| 464 | { |
| 465 | if (!_decryptionRTCPBufferPtr) |
| 466 | { |
| 467 | // Allocate memory for decryption buffer one time only |
| 468 | _decryptionRTCPBufferPtr = |
| 469 | new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes]; |
| 470 | } |
| 471 | |
| 472 | // Perform decryption (SRTP or external). |
| 473 | WebRtc_Word32 decryptedBufferLength = 0; |
| 474 | _encryptionPtr->decrypt_rtcp(_channelId, |
| 475 | rtcpBufferPtr, |
| 476 | _decryptionRTCPBufferPtr, |
| 477 | rtcpBufferLength, |
| 478 | (int*)&decryptedBufferLength); |
| 479 | if (decryptedBufferLength <= 0) |
| 480 | { |
| 481 | _engineStatisticsPtr->SetLastError( |
| 482 | VE_DECRYPTION_FAILED, kTraceError, |
| 483 | "Channel::IncomingRTCPPacket() decryption failed"); |
| 484 | return; |
| 485 | } |
| 486 | |
| 487 | // Replace default data buffer with decrypted buffer |
| 488 | rtcpBufferPtr = _decryptionRTCPBufferPtr; |
| 489 | rtcpBufferLength = decryptedBufferLength; |
| 490 | } |
| 491 | } |
| 492 | |
| 493 | // Dump the RTCP packet to a file (if RTP dump is enabled). |
| 494 | if (_rtpDumpIn.DumpPacket(rtcpBufferPtr, |
| 495 | (WebRtc_UWord16)rtcpBufferLength) == -1) |
| 496 | { |
| 497 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 498 | VoEId(_instanceId,_channelId), |
| 499 | "Channel::SendPacket() RTCP dump to input file failed"); |
| 500 | } |
| 501 | |
| 502 | // Deliver RTCP packet to RTP/RTCP module for parsing |
| 503 | if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)rtcpBufferPtr, |
| 504 | (WebRtc_UWord16)rtcpBufferLength) == -1) |
| 505 | { |
| 506 | _engineStatisticsPtr->SetLastError( |
| 507 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning, |
| 508 | "Channel::IncomingRTPPacket() RTCP packet is invalid"); |
| 509 | return; |
| 510 | } |
| 511 | } |
| 512 | |
| 513 | void |
| 514 | Channel::OnReceivedTelephoneEvent(const WebRtc_Word32 id, |
| 515 | const WebRtc_UWord8 event, |
| 516 | const bool endOfEvent) |
| 517 | { |
| 518 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 519 | "Channel::OnReceivedTelephoneEvent(id=%d, event=%u," |
| 520 | " endOfEvent=%d)", id, event, endOfEvent); |
| 521 | |
| 522 | #ifdef WEBRTC_DTMF_DETECTION |
| 523 | if (_outOfBandTelephoneEventDetecion) |
| 524 | { |
| 525 | CriticalSectionScoped cs(&_callbackCritSect); |
| 526 | |
| 527 | if (_telephoneEventDetectionPtr) |
| 528 | { |
| 529 | _telephoneEventDetectionPtr->OnReceivedTelephoneEventOutOfBand( |
| 530 | _channelId, event, endOfEvent); |
| 531 | } |
| 532 | } |
| 533 | #endif |
| 534 | } |
| 535 | |
| 536 | void |
| 537 | Channel::OnPlayTelephoneEvent(const WebRtc_Word32 id, |
| 538 | const WebRtc_UWord8 event, |
| 539 | const WebRtc_UWord16 lengthMs, |
| 540 | const WebRtc_UWord8 volume) |
| 541 | { |
| 542 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 543 | "Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u," |
| 544 | " volume=%u)", id, event, lengthMs, volume); |
| 545 | |
| 546 | if (!_playOutbandDtmfEvent || (event > 15)) |
| 547 | { |
| 548 | // Ignore callback since feedback is disabled or event is not a |
| 549 | // Dtmf tone event. |
| 550 | return; |
| 551 | } |
| 552 | |
| 553 | assert(_outputMixerPtr != NULL); |
| 554 | |
| 555 | // Start playing out the Dtmf tone (if playout is enabled). |
| 556 | // Reduce length of tone with 80ms to the reduce risk of echo. |
| 557 | _outputMixerPtr->PlayDtmfTone(event, lengthMs - 80, volume); |
| 558 | } |
| 559 | |
| 560 | void |
| 561 | Channel::OnIncomingSSRCChanged(const WebRtc_Word32 id, |
| 562 | const WebRtc_UWord32 SSRC) |
| 563 | { |
| 564 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 565 | "Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)", |
| 566 | id, SSRC); |
| 567 | |
| 568 | WebRtc_Word32 channel = VoEChannelId(id); |
| 569 | assert(channel == _channelId); |
| 570 | |
| 571 | // Reset RTP-module counters since a new incoming RTP stream is detected |
| 572 | _rtpRtcpModule->ResetReceiveDataCountersRTP(); |
| 573 | _rtpRtcpModule->ResetStatisticsRTP(); |
| 574 | |
| 575 | if (_rtpObserver) |
| 576 | { |
| 577 | CriticalSectionScoped cs(&_callbackCritSect); |
| 578 | |
| 579 | if (_rtpObserverPtr) |
| 580 | { |
| 581 | // Send new SSRC to registered observer using callback |
| 582 | _rtpObserverPtr->OnIncomingSSRCChanged(channel, SSRC); |
| 583 | } |
| 584 | } |
| 585 | } |
| 586 | |
| 587 | void Channel::OnIncomingCSRCChanged(const WebRtc_Word32 id, |
| 588 | const WebRtc_UWord32 CSRC, |
| 589 | const bool added) |
| 590 | { |
| 591 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 592 | "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)", |
| 593 | id, CSRC, added); |
| 594 | |
| 595 | WebRtc_Word32 channel = VoEChannelId(id); |
| 596 | assert(channel == _channelId); |
| 597 | |
| 598 | if (_rtpObserver) |
| 599 | { |
| 600 | CriticalSectionScoped cs(&_callbackCritSect); |
| 601 | |
| 602 | if (_rtpObserverPtr) |
| 603 | { |
| 604 | _rtpObserverPtr->OnIncomingCSRCChanged(channel, CSRC, added); |
| 605 | } |
| 606 | } |
| 607 | } |
| 608 | |
| 609 | void |
| 610 | Channel::OnApplicationDataReceived(const WebRtc_Word32 id, |
| 611 | const WebRtc_UWord8 subType, |
| 612 | const WebRtc_UWord32 name, |
| 613 | const WebRtc_UWord16 length, |
| 614 | const WebRtc_UWord8* data) |
| 615 | { |
| 616 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 617 | "Channel::OnApplicationDataReceived(id=%d, subType=%u," |
| 618 | " name=%u, length=%u)", |
| 619 | id, subType, name, length); |
| 620 | |
| 621 | WebRtc_Word32 channel = VoEChannelId(id); |
| 622 | assert(channel == _channelId); |
| 623 | |
| 624 | if (_rtcpObserver) |
| 625 | { |
| 626 | CriticalSectionScoped cs(&_callbackCritSect); |
| 627 | |
| 628 | if (_rtcpObserverPtr) |
| 629 | { |
| 630 | _rtcpObserverPtr->OnApplicationDataReceived(channel, |
| 631 | subType, |
| 632 | name, |
| 633 | data, |
| 634 | length); |
| 635 | } |
| 636 | } |
| 637 | } |
| 638 | |
| 639 | WebRtc_Word32 |
| 640 | Channel::OnInitializeDecoder( |
| 641 | const WebRtc_Word32 id, |
| 642 | const WebRtc_Word8 payloadType, |
| 643 | const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
| 644 | const int frequency, |
| 645 | const WebRtc_UWord8 channels, |
| 646 | const WebRtc_UWord32 rate) |
| 647 | { |
| 648 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 649 | "Channel::OnInitializeDecoder(id=%d, payloadType=%d, " |
| 650 | "payloadName=%s, frequency=%u, channels=%u, rate=%u)", |
| 651 | id, payloadType, payloadName, frequency, channels, rate); |
| 652 | |
| 653 | assert(VoEChannelId(id) == _channelId); |
| 654 | |
| 655 | CodecInst receiveCodec = {0}; |
| 656 | CodecInst dummyCodec = {0}; |
| 657 | |
| 658 | receiveCodec.pltype = payloadType; |
| 659 | receiveCodec.plfreq = frequency; |
| 660 | receiveCodec.channels = channels; |
| 661 | receiveCodec.rate = rate; |
| 662 | strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); |
andrew@webrtc.org | d468236 | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 663 | |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 664 | _audioCodingModule.Codec(payloadName, &dummyCodec, frequency, channels); |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 665 | receiveCodec.pacsize = dummyCodec.pacsize; |
| 666 | |
| 667 | // Register the new codec to the ACM |
| 668 | if (_audioCodingModule.RegisterReceiveCodec(receiveCodec) == -1) |
| 669 | { |
| 670 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 671 | VoEId(_instanceId, _channelId), |
| 672 | "Channel::OnInitializeDecoder() invalid codec (" |
| 673 | "pt=%d, name=%s) received - 1", payloadType, payloadName); |
| 674 | _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR); |
| 675 | return -1; |
| 676 | } |
| 677 | |
| 678 | return 0; |
| 679 | } |
| 680 | |
| 681 | void |
| 682 | Channel::OnPacketTimeout(const WebRtc_Word32 id) |
| 683 | { |
| 684 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 685 | "Channel::OnPacketTimeout(id=%d)", id); |
| 686 | |
| 687 | CriticalSectionScoped cs(_callbackCritSectPtr); |
| 688 | if (_voiceEngineObserverPtr) |
| 689 | { |
| 690 | if (_receiving || _externalTransport) |
| 691 | { |
| 692 | WebRtc_Word32 channel = VoEChannelId(id); |
| 693 | assert(channel == _channelId); |
| 694 | // Ensure that next OnReceivedPacket() callback will trigger |
| 695 | // a VE_PACKET_RECEIPT_RESTARTED callback. |
| 696 | _rtpPacketTimedOut = true; |
| 697 | // Deliver callback to the observer |
| 698 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 699 | VoEId(_instanceId,_channelId), |
| 700 | "Channel::OnPacketTimeout() => " |
| 701 | "CallbackOnError(VE_RECEIVE_PACKET_TIMEOUT)"); |
| 702 | _voiceEngineObserverPtr->CallbackOnError(channel, |
| 703 | VE_RECEIVE_PACKET_TIMEOUT); |
| 704 | } |
| 705 | } |
| 706 | } |
| 707 | |
| 708 | void |
| 709 | Channel::OnReceivedPacket(const WebRtc_Word32 id, |
| 710 | const RtpRtcpPacketType packetType) |
| 711 | { |
| 712 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 713 | "Channel::OnReceivedPacket(id=%d, packetType=%d)", |
| 714 | id, packetType); |
| 715 | |
| 716 | assert(VoEChannelId(id) == _channelId); |
| 717 | |
| 718 | // Notify only for the case when we have restarted an RTP session. |
| 719 | if (_rtpPacketTimedOut && (kPacketRtp == packetType)) |
| 720 | { |
| 721 | CriticalSectionScoped cs(_callbackCritSectPtr); |
| 722 | if (_voiceEngineObserverPtr) |
| 723 | { |
| 724 | WebRtc_Word32 channel = VoEChannelId(id); |
| 725 | assert(channel == _channelId); |
| 726 | // Reset timeout mechanism |
| 727 | _rtpPacketTimedOut = false; |
| 728 | // Deliver callback to the observer |
| 729 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 730 | VoEId(_instanceId,_channelId), |
| 731 | "Channel::OnPacketTimeout() =>" |
| 732 | " CallbackOnError(VE_PACKET_RECEIPT_RESTARTED)"); |
| 733 | _voiceEngineObserverPtr->CallbackOnError( |
| 734 | channel, |
| 735 | VE_PACKET_RECEIPT_RESTARTED); |
| 736 | } |
| 737 | } |
| 738 | } |
| 739 | |
| 740 | void |
| 741 | Channel::OnPeriodicDeadOrAlive(const WebRtc_Word32 id, |
| 742 | const RTPAliveType alive) |
| 743 | { |
| 744 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 745 | "Channel::OnPeriodicDeadOrAlive(id=%d, alive=%d)", id, alive); |
| 746 | |
| 747 | if (!_connectionObserver) |
| 748 | return; |
| 749 | |
| 750 | WebRtc_Word32 channel = VoEChannelId(id); |
| 751 | assert(channel == _channelId); |
| 752 | |
| 753 | // Use Alive as default to limit risk of false Dead detections |
| 754 | bool isAlive(true); |
| 755 | |
| 756 | // Always mark the connection as Dead when the module reports kRtpDead |
| 757 | if (kRtpDead == alive) |
| 758 | { |
| 759 | isAlive = false; |
| 760 | } |
| 761 | |
| 762 | // It is possible that the connection is alive even if no RTP packet has |
| 763 | // been received for a long time since the other side might use VAD/DTX |
| 764 | // and a low SID-packet update rate. |
| 765 | if ((kRtpNoRtp == alive) && _playing) |
| 766 | { |
| 767 | // Detect Alive for all NetEQ states except for the case when we are |
| 768 | // in PLC_CNG state. |
| 769 | // PLC_CNG <=> background noise only due to long expand or error. |
| 770 | // Note that, the case where the other side stops sending during CNG |
| 771 | // state will be detected as Alive. Dead is is not set until after |
| 772 | // missing RTCP packets for at least twelve seconds (handled |
| 773 | // internally by the RTP/RTCP module). |
| 774 | isAlive = (_outputSpeechType != AudioFrame::kPLCCNG); |
| 775 | } |
| 776 | |
| 777 | UpdateDeadOrAliveCounters(isAlive); |
| 778 | |
| 779 | // Send callback to the registered observer |
| 780 | if (_connectionObserver) |
| 781 | { |
| 782 | CriticalSectionScoped cs(&_callbackCritSect); |
| 783 | if (_connectionObserverPtr) |
| 784 | { |
| 785 | _connectionObserverPtr->OnPeriodicDeadOrAlive(channel, isAlive); |
| 786 | } |
| 787 | } |
| 788 | } |
| 789 | |
| 790 | WebRtc_Word32 |
| 791 | Channel::OnReceivedPayloadData(const WebRtc_UWord8* payloadData, |
| 792 | const WebRtc_UWord16 payloadSize, |
| 793 | const WebRtcRTPHeader* rtpHeader) |
| 794 | { |
| 795 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 796 | "Channel::OnReceivedPayloadData(payloadSize=%d," |
| 797 | " payloadType=%u, audioChannel=%u)", |
| 798 | payloadSize, |
| 799 | rtpHeader->header.payloadType, |
| 800 | rtpHeader->type.Audio.channel); |
| 801 | |
roosa@google.com | ca77149 | 2012-12-12 21:31:41 +0000 | [diff] [blame] | 802 | _lastRemoteTimeStamp = rtpHeader->header.timestamp; |
| 803 | |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 804 | if (!_playing) |
| 805 | { |
| 806 | // Avoid inserting into NetEQ when we are not playing. Count the |
| 807 | // packet as discarded. |
| 808 | WEBRTC_TRACE(kTraceStream, kTraceVoice, |
| 809 | VoEId(_instanceId, _channelId), |
| 810 | "received packet is discarded since playing is not" |
| 811 | " activated"); |
| 812 | _numberOfDiscardedPackets++; |
| 813 | return 0; |
| 814 | } |
| 815 | |
| 816 | // Push the incoming payload (parsed and ready for decoding) into the ACM |
| 817 | if (_audioCodingModule.IncomingPacket(payloadData, |
| 818 | payloadSize, |
| 819 | *rtpHeader) != 0) |
| 820 | { |
| 821 | _engineStatisticsPtr->SetLastError( |
| 822 | VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning, |
| 823 | "Channel::OnReceivedPayloadData() unable to push data to the ACM"); |
| 824 | return -1; |
| 825 | } |
| 826 | |
| 827 | // Update the packet delay |
| 828 | UpdatePacketDelay(rtpHeader->header.timestamp, |
| 829 | rtpHeader->header.sequenceNumber); |
| 830 | |
| 831 | return 0; |
| 832 | } |
| 833 | |
| 834 | WebRtc_Word32 Channel::GetAudioFrame(const WebRtc_Word32 id, |
| 835 | AudioFrame& audioFrame) |
| 836 | { |
| 837 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 838 | "Channel::GetAudioFrame(id=%d)", id); |
| 839 | |
| 840 | // Get 10ms raw PCM data from the ACM (mixer limits output frequency) |
| 841 | if (_audioCodingModule.PlayoutData10Ms(audioFrame.sample_rate_hz_, |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 842 | &audioFrame) == -1) |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 843 | { |
| 844 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 845 | VoEId(_instanceId,_channelId), |
| 846 | "Channel::GetAudioFrame() PlayoutData10Ms() failed!"); |
| 847 | // In all likelihood, the audio in this frame is garbage. We return an |
| 848 | // error so that the audio mixer module doesn't add it to the mix. As |
| 849 | // a result, it won't be played out and the actions skipped here are |
| 850 | // irrelevant. |
| 851 | return -1; |
| 852 | } |
| 853 | |
| 854 | if (_RxVadDetection) |
| 855 | { |
| 856 | UpdateRxVadDetection(audioFrame); |
| 857 | } |
| 858 | |
| 859 | // Convert module ID to internal VoE channel ID |
| 860 | audioFrame.id_ = VoEChannelId(audioFrame.id_); |
| 861 | // Store speech type for dead-or-alive detection |
| 862 | _outputSpeechType = audioFrame.speech_type_; |
| 863 | |
| 864 | // Perform far-end AudioProcessing module processing on the received signal |
| 865 | if (_rxApmIsEnabled) |
| 866 | { |
| 867 | ApmProcessRx(audioFrame); |
| 868 | } |
| 869 | |
| 870 | // Output volume scaling |
| 871 | if (_outputGain < 0.99f || _outputGain > 1.01f) |
| 872 | { |
| 873 | AudioFrameOperations::ScaleWithSat(_outputGain, audioFrame); |
| 874 | } |
| 875 | |
| 876 | // Scale left and/or right channel(s) if stereo and master balance is |
| 877 | // active |
| 878 | |
| 879 | if (_panLeft != 1.0f || _panRight != 1.0f) |
| 880 | { |
| 881 | if (audioFrame.num_channels_ == 1) |
| 882 | { |
| 883 | // Emulate stereo mode since panning is active. |
| 884 | // The mono signal is copied to both left and right channels here. |
| 885 | AudioFrameOperations::MonoToStereo(&audioFrame); |
| 886 | } |
| 887 | // For true stereo mode (when we are receiving a stereo signal), no |
| 888 | // action is needed. |
| 889 | |
| 890 | // Do the panning operation (the audio frame contains stereo at this |
| 891 | // stage) |
| 892 | AudioFrameOperations::Scale(_panLeft, _panRight, audioFrame); |
| 893 | } |
| 894 | |
| 895 | // Mix decoded PCM output with file if file mixing is enabled |
| 896 | if (_outputFilePlaying) |
| 897 | { |
| 898 | MixAudioWithFile(audioFrame, audioFrame.sample_rate_hz_); |
| 899 | } |
| 900 | |
| 901 | // Place channel in on-hold state (~muted) if on-hold is activated |
| 902 | if (_outputIsOnHold) |
| 903 | { |
| 904 | AudioFrameOperations::Mute(audioFrame); |
| 905 | } |
| 906 | |
| 907 | // External media |
| 908 | if (_outputExternalMedia) |
| 909 | { |
| 910 | CriticalSectionScoped cs(&_callbackCritSect); |
| 911 | const bool isStereo = (audioFrame.num_channels_ == 2); |
| 912 | if (_outputExternalMediaCallbackPtr) |
| 913 | { |
| 914 | _outputExternalMediaCallbackPtr->Process( |
| 915 | _channelId, |
| 916 | kPlaybackPerChannel, |
| 917 | (WebRtc_Word16*)audioFrame.data_, |
| 918 | audioFrame.samples_per_channel_, |
| 919 | audioFrame.sample_rate_hz_, |
| 920 | isStereo); |
| 921 | } |
| 922 | } |
| 923 | |
| 924 | // Record playout if enabled |
| 925 | { |
| 926 | CriticalSectionScoped cs(&_fileCritSect); |
| 927 | |
| 928 | if (_outputFileRecording && _outputFileRecorderPtr) |
| 929 | { |
| 930 | _outputFileRecorderPtr->RecordAudioToFile(audioFrame); |
| 931 | } |
| 932 | } |
| 933 | |
| 934 | // Measure audio level (0-9) |
| 935 | _outputAudioLevel.ComputeLevel(audioFrame); |
| 936 | |
| 937 | return 0; |
| 938 | } |
| 939 | |
| 940 | WebRtc_Word32 |
| 941 | Channel::NeededFrequency(const WebRtc_Word32 id) |
| 942 | { |
| 943 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 944 | "Channel::NeededFrequency(id=%d)", id); |
| 945 | |
| 946 | int highestNeeded = 0; |
| 947 | |
| 948 | // Determine highest needed receive frequency |
| 949 | WebRtc_Word32 receiveFrequency = _audioCodingModule.ReceiveFrequency(); |
| 950 | |
| 951 | // Return the bigger of playout and receive frequency in the ACM. |
| 952 | if (_audioCodingModule.PlayoutFrequency() > receiveFrequency) |
| 953 | { |
| 954 | highestNeeded = _audioCodingModule.PlayoutFrequency(); |
| 955 | } |
| 956 | else |
| 957 | { |
| 958 | highestNeeded = receiveFrequency; |
| 959 | } |
| 960 | |
| 961 | // Special case, if we're playing a file on the playout side |
| 962 | // we take that frequency into consideration as well |
| 963 | // This is not needed on sending side, since the codec will |
| 964 | // limit the spectrum anyway. |
| 965 | if (_outputFilePlaying) |
| 966 | { |
| 967 | CriticalSectionScoped cs(&_fileCritSect); |
| 968 | if (_outputFilePlayerPtr && _outputFilePlaying) |
| 969 | { |
| 970 | if(_outputFilePlayerPtr->Frequency()>highestNeeded) |
| 971 | { |
| 972 | highestNeeded=_outputFilePlayerPtr->Frequency(); |
| 973 | } |
| 974 | } |
| 975 | } |
| 976 | |
| 977 | return(highestNeeded); |
| 978 | } |
| 979 | |
| 980 | WebRtc_Word32 |
| 981 | Channel::CreateChannel(Channel*& channel, |
| 982 | const WebRtc_Word32 channelId, |
| 983 | const WebRtc_UWord32 instanceId) |
| 984 | { |
| 985 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId), |
| 986 | "Channel::CreateChannel(channelId=%d, instanceId=%d)", |
| 987 | channelId, instanceId); |
| 988 | |
| 989 | channel = new Channel(channelId, instanceId); |
| 990 | if (channel == NULL) |
| 991 | { |
| 992 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, |
| 993 | VoEId(instanceId,channelId), |
| 994 | "Channel::CreateChannel() unable to allocate memory for" |
| 995 | " channel"); |
| 996 | return -1; |
| 997 | } |
| 998 | return 0; |
| 999 | } |
| 1000 | |
| 1001 | void |
| 1002 | Channel::PlayNotification(const WebRtc_Word32 id, |
| 1003 | const WebRtc_UWord32 durationMs) |
| 1004 | { |
| 1005 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1006 | "Channel::PlayNotification(id=%d, durationMs=%d)", |
| 1007 | id, durationMs); |
| 1008 | |
| 1009 | // Not implement yet |
| 1010 | } |
| 1011 | |
| 1012 | void |
| 1013 | Channel::RecordNotification(const WebRtc_Word32 id, |
| 1014 | const WebRtc_UWord32 durationMs) |
| 1015 | { |
| 1016 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1017 | "Channel::RecordNotification(id=%d, durationMs=%d)", |
| 1018 | id, durationMs); |
| 1019 | |
| 1020 | // Not implement yet |
| 1021 | } |
| 1022 | |
| 1023 | void |
| 1024 | Channel::PlayFileEnded(const WebRtc_Word32 id) |
| 1025 | { |
| 1026 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1027 | "Channel::PlayFileEnded(id=%d)", id); |
| 1028 | |
| 1029 | if (id == _inputFilePlayerId) |
| 1030 | { |
| 1031 | CriticalSectionScoped cs(&_fileCritSect); |
| 1032 | |
| 1033 | _inputFilePlaying = false; |
| 1034 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 1035 | VoEId(_instanceId,_channelId), |
| 1036 | "Channel::PlayFileEnded() => input file player module is" |
| 1037 | " shutdown"); |
| 1038 | } |
| 1039 | else if (id == _outputFilePlayerId) |
| 1040 | { |
| 1041 | CriticalSectionScoped cs(&_fileCritSect); |
| 1042 | |
| 1043 | _outputFilePlaying = false; |
| 1044 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 1045 | VoEId(_instanceId,_channelId), |
| 1046 | "Channel::PlayFileEnded() => output file player module is" |
| 1047 | " shutdown"); |
| 1048 | } |
| 1049 | } |
| 1050 | |
| 1051 | void |
| 1052 | Channel::RecordFileEnded(const WebRtc_Word32 id) |
| 1053 | { |
| 1054 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1055 | "Channel::RecordFileEnded(id=%d)", id); |
| 1056 | |
| 1057 | assert(id == _outputFileRecorderId); |
| 1058 | |
| 1059 | CriticalSectionScoped cs(&_fileCritSect); |
| 1060 | |
| 1061 | _outputFileRecording = false; |
| 1062 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 1063 | VoEId(_instanceId,_channelId), |
| 1064 | "Channel::RecordFileEnded() => output file recorder module is" |
| 1065 | " shutdown"); |
| 1066 | } |
| 1067 | |
| 1068 | Channel::Channel(const WebRtc_Word32 channelId, |
| 1069 | const WebRtc_UWord32 instanceId) : |
| 1070 | _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| 1071 | _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| 1072 | _instanceId(instanceId), |
| 1073 | _channelId(channelId), |
| 1074 | _audioCodingModule(*AudioCodingModule::Create( |
| 1075 | VoEModuleId(instanceId, channelId))), |
| 1076 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1077 | _numSocketThreads(KNumSocketThreads), |
| 1078 | _socketTransportModule(*UdpTransport::Create( |
| 1079 | VoEModuleId(instanceId, channelId), _numSocketThreads)), |
| 1080 | #endif |
| 1081 | #ifdef WEBRTC_SRTP |
| 1082 | _srtpModule(*SrtpModule::CreateSrtpModule(VoEModuleId(instanceId, |
| 1083 | channelId))), |
| 1084 | #endif |
| 1085 | _rtpDumpIn(*RtpDump::CreateRtpDump()), |
| 1086 | _rtpDumpOut(*RtpDump::CreateRtpDump()), |
| 1087 | _outputAudioLevel(), |
| 1088 | _externalTransport(false), |
| 1089 | _inputFilePlayerPtr(NULL), |
| 1090 | _outputFilePlayerPtr(NULL), |
| 1091 | _outputFileRecorderPtr(NULL), |
| 1092 | // Avoid conflict with other channels by adding 1024 - 1026, |
| 1093 | // won't use as much as 1024 channels. |
| 1094 | _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024), |
| 1095 | _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025), |
| 1096 | _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026), |
| 1097 | _inputFilePlaying(false), |
| 1098 | _outputFilePlaying(false), |
| 1099 | _outputFileRecording(false), |
| 1100 | _inbandDtmfQueue(VoEModuleId(instanceId, channelId)), |
| 1101 | _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)), |
| 1102 | _inputExternalMedia(false), |
| 1103 | _outputExternalMedia(false), |
| 1104 | _inputExternalMediaCallbackPtr(NULL), |
| 1105 | _outputExternalMediaCallbackPtr(NULL), |
| 1106 | _encryptionRTPBufferPtr(NULL), |
| 1107 | _decryptionRTPBufferPtr(NULL), |
| 1108 | _encryptionRTCPBufferPtr(NULL), |
| 1109 | _decryptionRTCPBufferPtr(NULL), |
| 1110 | _timeStamp(0), // This is just an offset, RTP module will add it's own random offset |
| 1111 | _sendTelephoneEventPayloadType(106), |
| 1112 | _playoutTimeStampRTP(0), |
| 1113 | _playoutTimeStampRTCP(0), |
| 1114 | _numberOfDiscardedPackets(0), |
| 1115 | _engineStatisticsPtr(NULL), |
| 1116 | _outputMixerPtr(NULL), |
| 1117 | _transmitMixerPtr(NULL), |
| 1118 | _moduleProcessThreadPtr(NULL), |
| 1119 | _audioDeviceModulePtr(NULL), |
| 1120 | _voiceEngineObserverPtr(NULL), |
| 1121 | _callbackCritSectPtr(NULL), |
| 1122 | _transportPtr(NULL), |
| 1123 | _encryptionPtr(NULL), |
| 1124 | _rtpAudioProc(NULL), |
| 1125 | _rxAudioProcessingModulePtr(NULL), |
| 1126 | #ifdef WEBRTC_DTMF_DETECTION |
| 1127 | _telephoneEventDetectionPtr(NULL), |
| 1128 | #endif |
| 1129 | _rxVadObserverPtr(NULL), |
| 1130 | _oldVadDecision(-1), |
| 1131 | _sendFrameType(0), |
| 1132 | _rtpObserverPtr(NULL), |
| 1133 | _rtcpObserverPtr(NULL), |
| 1134 | _outputIsOnHold(false), |
| 1135 | _externalPlayout(false), |
roosa@google.com | b9e3afc | 2012-12-12 23:00:29 +0000 | [diff] [blame] | 1136 | _externalMixing(false), |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 1137 | _inputIsOnHold(false), |
| 1138 | _playing(false), |
| 1139 | _sending(false), |
| 1140 | _receiving(false), |
| 1141 | _mixFileWithMicrophone(false), |
| 1142 | _rtpObserver(false), |
| 1143 | _rtcpObserver(false), |
| 1144 | _mute(false), |
| 1145 | _panLeft(1.0f), |
| 1146 | _panRight(1.0f), |
| 1147 | _outputGain(1.0f), |
| 1148 | _encrypting(false), |
| 1149 | _decrypting(false), |
| 1150 | _playOutbandDtmfEvent(false), |
| 1151 | _playInbandDtmfEvent(false), |
| 1152 | _inbandTelephoneEventDetection(false), |
| 1153 | _outOfBandTelephoneEventDetecion(false), |
| 1154 | _extraPayloadType(0), |
| 1155 | _insertExtraRTPPacket(false), |
| 1156 | _extraMarkerBit(false), |
| 1157 | _lastLocalTimeStamp(0), |
roosa@google.com | ca77149 | 2012-12-12 21:31:41 +0000 | [diff] [blame] | 1158 | _lastRemoteTimeStamp(0), |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 1159 | _lastPayloadType(0), |
| 1160 | _includeAudioLevelIndication(false), |
| 1161 | _rtpPacketTimedOut(false), |
| 1162 | _rtpPacketTimeOutIsEnabled(false), |
| 1163 | _rtpTimeOutSeconds(0), |
| 1164 | _connectionObserver(false), |
| 1165 | _connectionObserverPtr(NULL), |
| 1166 | _countAliveDetections(0), |
| 1167 | _countDeadDetections(0), |
| 1168 | _outputSpeechType(AudioFrame::kNormalSpeech), |
| 1169 | _averageDelayMs(0), |
| 1170 | _previousSequenceNumber(0), |
| 1171 | _previousTimestamp(0), |
| 1172 | _recPacketDelayMs(20), |
| 1173 | _RxVadDetection(false), |
| 1174 | _rxApmIsEnabled(false), |
| 1175 | _rxAgcIsEnabled(false), |
| 1176 | _rxNsIsEnabled(false) |
| 1177 | { |
| 1178 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1179 | "Channel::Channel() - ctor"); |
| 1180 | _inbandDtmfQueue.ResetDtmf(); |
| 1181 | _inbandDtmfGenerator.Init(); |
| 1182 | _outputAudioLevel.Clear(); |
| 1183 | |
| 1184 | RtpRtcp::Configuration configuration; |
| 1185 | configuration.id = VoEModuleId(instanceId, channelId); |
| 1186 | configuration.audio = true; |
| 1187 | configuration.incoming_data = this; |
| 1188 | configuration.incoming_messages = this; |
| 1189 | configuration.outgoing_transport = this; |
| 1190 | configuration.rtcp_feedback = this; |
| 1191 | configuration.audio_messages = this; |
| 1192 | |
| 1193 | _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration)); |
| 1194 | |
| 1195 | // Create far end AudioProcessing Module |
| 1196 | _rxAudioProcessingModulePtr = AudioProcessing::Create( |
| 1197 | VoEModuleId(instanceId, channelId)); |
| 1198 | } |
| 1199 | |
| 1200 | Channel::~Channel() |
| 1201 | { |
| 1202 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1203 | "Channel::~Channel() - dtor"); |
| 1204 | |
| 1205 | if (_outputExternalMedia) |
| 1206 | { |
| 1207 | DeRegisterExternalMediaProcessing(kPlaybackPerChannel); |
| 1208 | } |
| 1209 | if (_inputExternalMedia) |
| 1210 | { |
| 1211 | DeRegisterExternalMediaProcessing(kRecordingPerChannel); |
| 1212 | } |
| 1213 | StopSend(); |
| 1214 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1215 | StopReceiving(); |
| 1216 | // De-register packet callback to ensure we're not in a callback when |
| 1217 | // deleting channel state, avoids race condition and deadlock. |
| 1218 | if (_socketTransportModule.InitializeReceiveSockets(NULL, 0, NULL, NULL, 0) |
| 1219 | != 0) |
| 1220 | { |
| 1221 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1222 | VoEId(_instanceId, _channelId), |
| 1223 | "~Channel() failed to de-register receive callback"); |
| 1224 | } |
| 1225 | #endif |
| 1226 | StopPlayout(); |
| 1227 | |
| 1228 | { |
| 1229 | CriticalSectionScoped cs(&_fileCritSect); |
| 1230 | if (_inputFilePlayerPtr) |
| 1231 | { |
| 1232 | _inputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 1233 | _inputFilePlayerPtr->StopPlayingFile(); |
| 1234 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 1235 | _inputFilePlayerPtr = NULL; |
| 1236 | } |
| 1237 | if (_outputFilePlayerPtr) |
| 1238 | { |
| 1239 | _outputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 1240 | _outputFilePlayerPtr->StopPlayingFile(); |
| 1241 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 1242 | _outputFilePlayerPtr = NULL; |
| 1243 | } |
| 1244 | if (_outputFileRecorderPtr) |
| 1245 | { |
| 1246 | _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 1247 | _outputFileRecorderPtr->StopRecording(); |
| 1248 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 1249 | _outputFileRecorderPtr = NULL; |
| 1250 | } |
| 1251 | } |
| 1252 | |
| 1253 | // The order to safely shutdown modules in a channel is: |
| 1254 | // 1. De-register callbacks in modules |
| 1255 | // 2. De-register modules in process thread |
| 1256 | // 3. Destroy modules |
| 1257 | if (_audioCodingModule.RegisterTransportCallback(NULL) == -1) |
| 1258 | { |
| 1259 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1260 | VoEId(_instanceId,_channelId), |
| 1261 | "~Channel() failed to de-register transport callback" |
| 1262 | " (Audio coding module)"); |
| 1263 | } |
| 1264 | if (_audioCodingModule.RegisterVADCallback(NULL) == -1) |
| 1265 | { |
| 1266 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1267 | VoEId(_instanceId,_channelId), |
| 1268 | "~Channel() failed to de-register VAD callback" |
| 1269 | " (Audio coding module)"); |
| 1270 | } |
| 1271 | #ifdef WEBRTC_DTMF_DETECTION |
| 1272 | if (_audioCodingModule.RegisterIncomingMessagesCallback(NULL) == -1) |
| 1273 | { |
| 1274 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1275 | VoEId(_instanceId,_channelId), |
| 1276 | "~Channel() failed to de-register incoming messages " |
| 1277 | "callback (Audio coding module)"); |
| 1278 | } |
| 1279 | #endif |
| 1280 | // De-register modules in process thread |
| 1281 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1282 | if (_moduleProcessThreadPtr->DeRegisterModule(&_socketTransportModule) |
| 1283 | == -1) |
| 1284 | { |
| 1285 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 1286 | VoEId(_instanceId,_channelId), |
| 1287 | "~Channel() failed to deregister socket module"); |
| 1288 | } |
| 1289 | #endif |
| 1290 | if (_moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get()) == -1) |
| 1291 | { |
| 1292 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 1293 | VoEId(_instanceId,_channelId), |
| 1294 | "~Channel() failed to deregister RTP/RTCP module"); |
| 1295 | } |
| 1296 | |
| 1297 | // Destroy modules |
| 1298 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1299 | UdpTransport::Destroy( |
| 1300 | &_socketTransportModule); |
| 1301 | #endif |
| 1302 | AudioCodingModule::Destroy(&_audioCodingModule); |
| 1303 | #ifdef WEBRTC_SRTP |
| 1304 | SrtpModule::DestroySrtpModule(&_srtpModule); |
| 1305 | #endif |
| 1306 | if (_rxAudioProcessingModulePtr != NULL) |
| 1307 | { |
| 1308 | AudioProcessing::Destroy(_rxAudioProcessingModulePtr); // far end APM |
| 1309 | _rxAudioProcessingModulePtr = NULL; |
| 1310 | } |
| 1311 | |
| 1312 | // End of modules shutdown |
| 1313 | |
| 1314 | // Delete other objects |
| 1315 | RtpDump::DestroyRtpDump(&_rtpDumpIn); |
| 1316 | RtpDump::DestroyRtpDump(&_rtpDumpOut); |
| 1317 | delete [] _encryptionRTPBufferPtr; |
| 1318 | delete [] _decryptionRTPBufferPtr; |
| 1319 | delete [] _encryptionRTCPBufferPtr; |
| 1320 | delete [] _decryptionRTCPBufferPtr; |
| 1321 | delete &_callbackCritSect; |
| 1322 | delete &_fileCritSect; |
| 1323 | } |
| 1324 | |
| 1325 | WebRtc_Word32 |
| 1326 | Channel::Init() |
| 1327 | { |
| 1328 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1329 | "Channel::Init()"); |
| 1330 | |
| 1331 | // --- Initial sanity |
| 1332 | |
| 1333 | if ((_engineStatisticsPtr == NULL) || |
| 1334 | (_moduleProcessThreadPtr == NULL)) |
| 1335 | { |
| 1336 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 1337 | VoEId(_instanceId,_channelId), |
| 1338 | "Channel::Init() must call SetEngineInformation() first"); |
| 1339 | return -1; |
| 1340 | } |
| 1341 | |
| 1342 | // --- Add modules to process thread (for periodic schedulation) |
| 1343 | |
| 1344 | const bool processThreadFail = |
| 1345 | ((_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get()) != 0) || |
| 1346 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1347 | (_moduleProcessThreadPtr->RegisterModule( |
| 1348 | &_socketTransportModule) != 0)); |
| 1349 | #else |
| 1350 | false); |
| 1351 | #endif |
| 1352 | if (processThreadFail) |
| 1353 | { |
| 1354 | _engineStatisticsPtr->SetLastError( |
| 1355 | VE_CANNOT_INIT_CHANNEL, kTraceError, |
| 1356 | "Channel::Init() modules not registered"); |
| 1357 | return -1; |
| 1358 | } |
| 1359 | // --- ACM initialization |
| 1360 | |
| 1361 | if ((_audioCodingModule.InitializeReceiver() == -1) || |
| 1362 | #ifdef WEBRTC_CODEC_AVT |
| 1363 | // out-of-band Dtmf tones are played out by default |
| 1364 | (_audioCodingModule.SetDtmfPlayoutStatus(true) == -1) || |
| 1365 | #endif |
| 1366 | (_audioCodingModule.InitializeSender() == -1)) |
| 1367 | { |
| 1368 | _engineStatisticsPtr->SetLastError( |
| 1369 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1370 | "Channel::Init() unable to initialize the ACM - 1"); |
| 1371 | return -1; |
| 1372 | } |
| 1373 | |
| 1374 | // --- RTP/RTCP module initialization |
| 1375 | |
| 1376 | // Ensure that RTCP is enabled by default for the created channel. |
| 1377 | // Note that, the module will keep generating RTCP until it is explicitly |
| 1378 | // disabled by the user. |
| 1379 | // After StopListen (when no sockets exists), RTCP packets will no longer |
| 1380 | // be transmitted since the Transport object will then be invalid. |
| 1381 | |
| 1382 | const bool rtpRtcpFail = |
| 1383 | ((_rtpRtcpModule->SetTelephoneEventStatus(false, true, true) == -1) || |
| 1384 | // RTCP is enabled by default |
| 1385 | (_rtpRtcpModule->SetRTCPStatus(kRtcpCompound) == -1)); |
| 1386 | if (rtpRtcpFail) |
| 1387 | { |
| 1388 | _engineStatisticsPtr->SetLastError( |
| 1389 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 1390 | "Channel::Init() RTP/RTCP module not initialized"); |
| 1391 | return -1; |
| 1392 | } |
| 1393 | |
| 1394 | // --- Register all permanent callbacks |
| 1395 | const bool fail = |
| 1396 | (_audioCodingModule.RegisterTransportCallback(this) == -1) || |
| 1397 | (_audioCodingModule.RegisterVADCallback(this) == -1); |
| 1398 | |
| 1399 | if (fail) |
| 1400 | { |
| 1401 | _engineStatisticsPtr->SetLastError( |
| 1402 | VE_CANNOT_INIT_CHANNEL, kTraceError, |
| 1403 | "Channel::Init() callbacks not registered"); |
| 1404 | return -1; |
| 1405 | } |
| 1406 | |
| 1407 | // --- Register all supported codecs to the receiving side of the |
| 1408 | // RTP/RTCP module |
| 1409 | |
| 1410 | CodecInst codec; |
| 1411 | const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs(); |
| 1412 | |
| 1413 | for (int idx = 0; idx < nSupportedCodecs; idx++) |
| 1414 | { |
| 1415 | // Open up the RTP/RTCP receiver for all supported codecs |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 1416 | if ((_audioCodingModule.Codec(idx, &codec) == -1) || |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 1417 | (_rtpRtcpModule->RegisterReceivePayload(codec) == -1)) |
| 1418 | { |
| 1419 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1420 | VoEId(_instanceId,_channelId), |
| 1421 | "Channel::Init() unable to register %s (%d/%d/%d/%d) " |
| 1422 | "to RTP/RTCP receiver", |
| 1423 | codec.plname, codec.pltype, codec.plfreq, |
| 1424 | codec.channels, codec.rate); |
| 1425 | } |
| 1426 | else |
| 1427 | { |
| 1428 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 1429 | VoEId(_instanceId,_channelId), |
| 1430 | "Channel::Init() %s (%d/%d/%d/%d) has been added to " |
| 1431 | "the RTP/RTCP receiver", |
| 1432 | codec.plname, codec.pltype, codec.plfreq, |
| 1433 | codec.channels, codec.rate); |
| 1434 | } |
| 1435 | |
| 1436 | // Ensure that PCMU is used as default codec on the sending side |
| 1437 | if (!STR_CASE_CMP(codec.plname, "PCMU") && (codec.channels == 1)) |
| 1438 | { |
| 1439 | SetSendCodec(codec); |
| 1440 | } |
| 1441 | |
| 1442 | // Register default PT for outband 'telephone-event' |
| 1443 | if (!STR_CASE_CMP(codec.plname, "telephone-event")) |
| 1444 | { |
| 1445 | if ((_rtpRtcpModule->RegisterSendPayload(codec) == -1) || |
| 1446 | (_audioCodingModule.RegisterReceiveCodec(codec) == -1)) |
| 1447 | { |
| 1448 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1449 | VoEId(_instanceId,_channelId), |
| 1450 | "Channel::Init() failed to register outband " |
| 1451 | "'telephone-event' (%d/%d) correctly", |
| 1452 | codec.pltype, codec.plfreq); |
| 1453 | } |
| 1454 | } |
| 1455 | |
| 1456 | if (!STR_CASE_CMP(codec.plname, "CN")) |
| 1457 | { |
| 1458 | if ((_audioCodingModule.RegisterSendCodec(codec) == -1) || |
| 1459 | (_audioCodingModule.RegisterReceiveCodec(codec) == -1) || |
| 1460 | (_rtpRtcpModule->RegisterSendPayload(codec) == -1)) |
| 1461 | { |
| 1462 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1463 | VoEId(_instanceId,_channelId), |
| 1464 | "Channel::Init() failed to register CN (%d/%d) " |
| 1465 | "correctly - 1", |
| 1466 | codec.pltype, codec.plfreq); |
| 1467 | } |
| 1468 | } |
| 1469 | #ifdef WEBRTC_CODEC_RED |
| 1470 | // Register RED to the receiving side of the ACM. |
| 1471 | // We will not receive an OnInitializeDecoder() callback for RED. |
| 1472 | if (!STR_CASE_CMP(codec.plname, "RED")) |
| 1473 | { |
| 1474 | if (_audioCodingModule.RegisterReceiveCodec(codec) == -1) |
| 1475 | { |
| 1476 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1477 | VoEId(_instanceId,_channelId), |
| 1478 | "Channel::Init() failed to register RED (%d/%d) " |
| 1479 | "correctly", |
| 1480 | codec.pltype, codec.plfreq); |
| 1481 | } |
| 1482 | } |
| 1483 | #endif |
| 1484 | } |
| 1485 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1486 | // Ensure that the WebRtcSocketTransport implementation is used as |
| 1487 | // Transport on the sending side |
| 1488 | { |
| 1489 | // A lock is needed here since users can call |
| 1490 | // RegisterExternalTransport() at the same time. |
| 1491 | CriticalSectionScoped cs(&_callbackCritSect); |
| 1492 | _transportPtr = &_socketTransportModule; |
| 1493 | } |
| 1494 | #endif |
| 1495 | |
| 1496 | // Initialize the far end AP module |
| 1497 | // Using 8 kHz as initial Fs, the same as in transmission. Might be |
| 1498 | // changed at the first receiving audio. |
| 1499 | if (_rxAudioProcessingModulePtr == NULL) |
| 1500 | { |
| 1501 | _engineStatisticsPtr->SetLastError( |
| 1502 | VE_NO_MEMORY, kTraceCritical, |
| 1503 | "Channel::Init() failed to create the far-end AudioProcessing" |
| 1504 | " module"); |
| 1505 | return -1; |
| 1506 | } |
| 1507 | |
| 1508 | if (_rxAudioProcessingModulePtr->set_sample_rate_hz(8000)) |
| 1509 | { |
| 1510 | _engineStatisticsPtr->SetLastError( |
| 1511 | VE_APM_ERROR, kTraceWarning, |
| 1512 | "Channel::Init() failed to set the sample rate to 8K for" |
| 1513 | " far-end AP module"); |
| 1514 | } |
| 1515 | |
| 1516 | if (_rxAudioProcessingModulePtr->set_num_channels(1, 1) != 0) |
| 1517 | { |
| 1518 | _engineStatisticsPtr->SetLastError( |
| 1519 | VE_SOUNDCARD_ERROR, kTraceWarning, |
| 1520 | "Init() failed to set channels for the primary audio stream"); |
| 1521 | } |
| 1522 | |
| 1523 | if (_rxAudioProcessingModulePtr->high_pass_filter()->Enable( |
| 1524 | WEBRTC_VOICE_ENGINE_RX_HP_DEFAULT_STATE) != 0) |
| 1525 | { |
| 1526 | _engineStatisticsPtr->SetLastError( |
| 1527 | VE_APM_ERROR, kTraceWarning, |
| 1528 | "Channel::Init() failed to set the high-pass filter for" |
| 1529 | " far-end AP module"); |
| 1530 | } |
| 1531 | |
| 1532 | if (_rxAudioProcessingModulePtr->noise_suppression()->set_level( |
| 1533 | (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE) != 0) |
| 1534 | { |
| 1535 | _engineStatisticsPtr->SetLastError( |
| 1536 | VE_APM_ERROR, kTraceWarning, |
| 1537 | "Init() failed to set noise reduction level for far-end" |
| 1538 | " AP module"); |
| 1539 | } |
| 1540 | if (_rxAudioProcessingModulePtr->noise_suppression()->Enable( |
| 1541 | WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE) != 0) |
| 1542 | { |
| 1543 | _engineStatisticsPtr->SetLastError( |
| 1544 | VE_APM_ERROR, kTraceWarning, |
| 1545 | "Init() failed to set noise reduction state for far-end" |
| 1546 | " AP module"); |
| 1547 | } |
| 1548 | |
| 1549 | if (_rxAudioProcessingModulePtr->gain_control()->set_mode( |
| 1550 | (GainControl::Mode)WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_MODE) != 0) |
| 1551 | { |
| 1552 | _engineStatisticsPtr->SetLastError( |
| 1553 | VE_APM_ERROR, kTraceWarning, |
| 1554 | "Init() failed to set AGC mode for far-end AP module"); |
| 1555 | } |
| 1556 | if (_rxAudioProcessingModulePtr->gain_control()->Enable( |
| 1557 | WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_STATE) != 0) |
| 1558 | { |
| 1559 | _engineStatisticsPtr->SetLastError( |
| 1560 | VE_APM_ERROR, kTraceWarning, |
| 1561 | "Init() failed to set AGC state for far-end AP module"); |
| 1562 | } |
| 1563 | |
| 1564 | return 0; |
| 1565 | } |
| 1566 | |
| 1567 | WebRtc_Word32 |
| 1568 | Channel::SetEngineInformation(Statistics& engineStatistics, |
| 1569 | OutputMixer& outputMixer, |
| 1570 | voe::TransmitMixer& transmitMixer, |
| 1571 | ProcessThread& moduleProcessThread, |
| 1572 | AudioDeviceModule& audioDeviceModule, |
| 1573 | VoiceEngineObserver* voiceEngineObserver, |
| 1574 | CriticalSectionWrapper* callbackCritSect) |
| 1575 | { |
| 1576 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1577 | "Channel::SetEngineInformation()"); |
| 1578 | _engineStatisticsPtr = &engineStatistics; |
| 1579 | _outputMixerPtr = &outputMixer; |
| 1580 | _transmitMixerPtr = &transmitMixer, |
| 1581 | _moduleProcessThreadPtr = &moduleProcessThread; |
| 1582 | _audioDeviceModulePtr = &audioDeviceModule; |
| 1583 | _voiceEngineObserverPtr = voiceEngineObserver; |
| 1584 | _callbackCritSectPtr = callbackCritSect; |
| 1585 | return 0; |
| 1586 | } |
| 1587 | |
| 1588 | WebRtc_Word32 |
| 1589 | Channel::UpdateLocalTimeStamp() |
| 1590 | { |
| 1591 | |
| 1592 | _timeStamp += _audioFrame.samples_per_channel_; |
| 1593 | return 0; |
| 1594 | } |
| 1595 | |
| 1596 | WebRtc_Word32 |
| 1597 | Channel::StartPlayout() |
| 1598 | { |
| 1599 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1600 | "Channel::StartPlayout()"); |
| 1601 | if (_playing) |
| 1602 | { |
| 1603 | return 0; |
| 1604 | } |
roosa@google.com | b9e3afc | 2012-12-12 23:00:29 +0000 | [diff] [blame] | 1605 | |
| 1606 | if (!_externalMixing) { |
| 1607 | // Add participant as candidates for mixing. |
| 1608 | if (_outputMixerPtr->SetMixabilityStatus(*this, true) != 0) |
| 1609 | { |
| 1610 | _engineStatisticsPtr->SetLastError( |
| 1611 | VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError, |
| 1612 | "StartPlayout() failed to add participant to mixer"); |
| 1613 | return -1; |
| 1614 | } |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 1615 | } |
| 1616 | |
| 1617 | _playing = true; |
| 1618 | |
| 1619 | if (RegisterFilePlayingToMixer() != 0) |
| 1620 | return -1; |
| 1621 | |
| 1622 | return 0; |
| 1623 | } |
| 1624 | |
| 1625 | WebRtc_Word32 |
| 1626 | Channel::StopPlayout() |
| 1627 | { |
| 1628 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1629 | "Channel::StopPlayout()"); |
| 1630 | if (!_playing) |
| 1631 | { |
| 1632 | return 0; |
| 1633 | } |
roosa@google.com | b9e3afc | 2012-12-12 23:00:29 +0000 | [diff] [blame] | 1634 | |
| 1635 | if (!_externalMixing) { |
| 1636 | // Remove participant as candidates for mixing |
| 1637 | if (_outputMixerPtr->SetMixabilityStatus(*this, false) != 0) |
| 1638 | { |
| 1639 | _engineStatisticsPtr->SetLastError( |
| 1640 | VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError, |
| 1641 | "StopPlayout() failed to remove participant from mixer"); |
| 1642 | return -1; |
| 1643 | } |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 1644 | } |
| 1645 | |
| 1646 | _playing = false; |
| 1647 | _outputAudioLevel.Clear(); |
| 1648 | |
| 1649 | return 0; |
| 1650 | } |
| 1651 | |
| 1652 | WebRtc_Word32 |
| 1653 | Channel::StartSend() |
| 1654 | { |
| 1655 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1656 | "Channel::StartSend()"); |
| 1657 | { |
| 1658 | // A lock is needed because |_sending| can be accessed or modified by |
| 1659 | // another thread at the same time. |
| 1660 | CriticalSectionScoped cs(&_callbackCritSect); |
| 1661 | |
| 1662 | if (_sending) |
| 1663 | { |
| 1664 | return 0; |
| 1665 | } |
| 1666 | _sending = true; |
| 1667 | } |
| 1668 | |
| 1669 | if (_rtpRtcpModule->SetSendingStatus(true) != 0) |
| 1670 | { |
| 1671 | _engineStatisticsPtr->SetLastError( |
| 1672 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 1673 | "StartSend() RTP/RTCP failed to start sending"); |
| 1674 | CriticalSectionScoped cs(&_callbackCritSect); |
| 1675 | _sending = false; |
| 1676 | return -1; |
| 1677 | } |
| 1678 | |
| 1679 | return 0; |
| 1680 | } |
| 1681 | |
| 1682 | WebRtc_Word32 |
| 1683 | Channel::StopSend() |
| 1684 | { |
| 1685 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1686 | "Channel::StopSend()"); |
| 1687 | { |
| 1688 | // A lock is needed because |_sending| can be accessed or modified by |
| 1689 | // another thread at the same time. |
| 1690 | CriticalSectionScoped cs(&_callbackCritSect); |
| 1691 | |
| 1692 | if (!_sending) |
| 1693 | { |
| 1694 | return 0; |
| 1695 | } |
| 1696 | _sending = false; |
| 1697 | } |
| 1698 | |
| 1699 | // Reset sending SSRC and sequence number and triggers direct transmission |
| 1700 | // of RTCP BYE |
| 1701 | if (_rtpRtcpModule->SetSendingStatus(false) == -1 || |
| 1702 | _rtpRtcpModule->ResetSendDataCountersRTP() == -1) |
| 1703 | { |
| 1704 | _engineStatisticsPtr->SetLastError( |
| 1705 | VE_RTP_RTCP_MODULE_ERROR, kTraceWarning, |
| 1706 | "StartSend() RTP/RTCP failed to stop sending"); |
| 1707 | } |
| 1708 | |
| 1709 | return 0; |
| 1710 | } |
| 1711 | |
| 1712 | WebRtc_Word32 |
| 1713 | Channel::StartReceiving() |
| 1714 | { |
| 1715 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1716 | "Channel::StartReceiving()"); |
| 1717 | if (_receiving) |
| 1718 | { |
| 1719 | return 0; |
| 1720 | } |
| 1721 | // If external transport is used, we will only initialize/set the variables |
| 1722 | // after this section, since we are not using the WebRtc transport but |
| 1723 | // still need to keep track of e.g. if we are receiving. |
| 1724 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1725 | if (!_externalTransport) |
| 1726 | { |
| 1727 | if (!_socketTransportModule.ReceiveSocketsInitialized()) |
| 1728 | { |
| 1729 | _engineStatisticsPtr->SetLastError( |
| 1730 | VE_SOCKETS_NOT_INITED, kTraceError, |
| 1731 | "StartReceive() must set local receiver first"); |
| 1732 | return -1; |
| 1733 | } |
| 1734 | if (_socketTransportModule.StartReceiving(KNumberOfSocketBuffers) != 0) |
| 1735 | { |
| 1736 | _engineStatisticsPtr->SetLastError( |
| 1737 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError, |
| 1738 | "StartReceiving() failed to start receiving"); |
| 1739 | return -1; |
| 1740 | } |
| 1741 | } |
| 1742 | #endif |
| 1743 | _receiving = true; |
| 1744 | _numberOfDiscardedPackets = 0; |
| 1745 | return 0; |
| 1746 | } |
| 1747 | |
| 1748 | WebRtc_Word32 |
| 1749 | Channel::StopReceiving() |
| 1750 | { |
| 1751 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1752 | "Channel::StopReceiving()"); |
| 1753 | if (!_receiving) |
| 1754 | { |
| 1755 | return 0; |
| 1756 | } |
| 1757 | |
| 1758 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1759 | if (!_externalTransport && |
| 1760 | _socketTransportModule.ReceiveSocketsInitialized()) |
| 1761 | { |
| 1762 | if (_socketTransportModule.StopReceiving() != 0) |
| 1763 | { |
| 1764 | _engineStatisticsPtr->SetLastError( |
| 1765 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError, |
| 1766 | "StopReceiving() failed to stop receiving."); |
| 1767 | return -1; |
| 1768 | } |
| 1769 | } |
| 1770 | #endif |
| 1771 | bool dtmfDetection = _rtpRtcpModule->TelephoneEvent(); |
| 1772 | // Recover DTMF detection status. |
| 1773 | WebRtc_Word32 ret = _rtpRtcpModule->SetTelephoneEventStatus(dtmfDetection, |
| 1774 | true, true); |
| 1775 | if (ret != 0) { |
| 1776 | _engineStatisticsPtr->SetLastError( |
| 1777 | VE_INVALID_OPERATION, kTraceWarning, |
| 1778 | "StopReceiving() failed to restore telephone-event status."); |
| 1779 | } |
| 1780 | RegisterReceiveCodecsToRTPModule(); |
| 1781 | _receiving = false; |
| 1782 | return 0; |
| 1783 | } |
| 1784 | |
| 1785 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1786 | WebRtc_Word32 |
| 1787 | Channel::SetLocalReceiver(const WebRtc_UWord16 rtpPort, |
| 1788 | const WebRtc_UWord16 rtcpPort, |
| 1789 | const char ipAddr[64], |
| 1790 | const char multicastIpAddr[64]) |
| 1791 | { |
| 1792 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1793 | "Channel::SetLocalReceiver()"); |
| 1794 | |
| 1795 | if (_externalTransport) |
| 1796 | { |
| 1797 | _engineStatisticsPtr->SetLastError( |
| 1798 | VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError, |
| 1799 | "SetLocalReceiver() conflict with external transport"); |
| 1800 | return -1; |
| 1801 | } |
| 1802 | |
| 1803 | if (_sending) |
| 1804 | { |
| 1805 | _engineStatisticsPtr->SetLastError( |
| 1806 | VE_ALREADY_SENDING, kTraceError, |
| 1807 | "SetLocalReceiver() already sending"); |
| 1808 | return -1; |
| 1809 | } |
| 1810 | if (_receiving) |
| 1811 | { |
| 1812 | _engineStatisticsPtr->SetLastError( |
| 1813 | VE_ALREADY_LISTENING, kTraceError, |
| 1814 | "SetLocalReceiver() already receiving"); |
| 1815 | return -1; |
| 1816 | } |
| 1817 | |
| 1818 | if (_socketTransportModule.InitializeReceiveSockets(this, |
| 1819 | rtpPort, |
| 1820 | ipAddr, |
| 1821 | multicastIpAddr, |
| 1822 | rtcpPort) != 0) |
| 1823 | { |
| 1824 | UdpTransport::ErrorCode lastSockError( |
| 1825 | _socketTransportModule.LastError()); |
| 1826 | switch (lastSockError) |
| 1827 | { |
| 1828 | case UdpTransport::kIpAddressInvalid: |
| 1829 | _engineStatisticsPtr->SetLastError( |
| 1830 | VE_INVALID_IP_ADDRESS, kTraceError, |
| 1831 | "SetLocalReceiver() invalid IP address"); |
| 1832 | break; |
| 1833 | case UdpTransport::kSocketInvalid: |
| 1834 | _engineStatisticsPtr->SetLastError( |
| 1835 | VE_SOCKET_ERROR, kTraceError, |
| 1836 | "SetLocalReceiver() invalid socket"); |
| 1837 | break; |
| 1838 | case UdpTransport::kPortInvalid: |
| 1839 | _engineStatisticsPtr->SetLastError( |
| 1840 | VE_INVALID_PORT_NMBR, kTraceError, |
| 1841 | "SetLocalReceiver() invalid port"); |
| 1842 | break; |
| 1843 | case UdpTransport::kFailedToBindPort: |
| 1844 | _engineStatisticsPtr->SetLastError( |
| 1845 | VE_BINDING_SOCKET_TO_LOCAL_ADDRESS_FAILED, kTraceError, |
| 1846 | "SetLocalReceiver() binding failed"); |
| 1847 | break; |
| 1848 | default: |
| 1849 | _engineStatisticsPtr->SetLastError( |
| 1850 | VE_SOCKET_ERROR, kTraceError, |
| 1851 | "SetLocalReceiver() undefined socket error"); |
| 1852 | break; |
| 1853 | } |
| 1854 | return -1; |
| 1855 | } |
| 1856 | return 0; |
| 1857 | } |
| 1858 | #endif |
| 1859 | |
| 1860 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1861 | WebRtc_Word32 |
| 1862 | Channel::GetLocalReceiver(int& port, int& RTCPport, char ipAddr[64]) |
| 1863 | { |
| 1864 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1865 | "Channel::GetLocalReceiver()"); |
| 1866 | |
| 1867 | if (_externalTransport) |
| 1868 | { |
| 1869 | _engineStatisticsPtr->SetLastError( |
| 1870 | VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError, |
| 1871 | "SetLocalReceiver() conflict with external transport"); |
| 1872 | return -1; |
| 1873 | } |
| 1874 | |
| 1875 | char ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0}; |
| 1876 | WebRtc_UWord16 rtpPort(0); |
| 1877 | WebRtc_UWord16 rtcpPort(0); |
| 1878 | char multicastIpAddr[UdpTransport::kIpAddressVersion6Length] = {0}; |
| 1879 | |
| 1880 | // Acquire socket information from the socket module |
| 1881 | if (_socketTransportModule.ReceiveSocketInformation(ipAddrTmp, |
| 1882 | rtpPort, |
| 1883 | rtcpPort, |
| 1884 | multicastIpAddr) != 0) |
| 1885 | { |
| 1886 | _engineStatisticsPtr->SetLastError( |
| 1887 | VE_CANNOT_GET_SOCKET_INFO, kTraceError, |
| 1888 | "GetLocalReceiver() unable to retrieve socket information"); |
| 1889 | return -1; |
| 1890 | } |
| 1891 | |
| 1892 | // Deliver valid results to the user |
| 1893 | port = static_cast<int> (rtpPort); |
| 1894 | RTCPport = static_cast<int> (rtcpPort); |
| 1895 | if (ipAddr != NULL) |
| 1896 | { |
| 1897 | strcpy(ipAddr, ipAddrTmp); |
| 1898 | } |
| 1899 | return 0; |
| 1900 | } |
| 1901 | #endif |
| 1902 | |
| 1903 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 1904 | WebRtc_Word32 |
| 1905 | Channel::SetSendDestination(const WebRtc_UWord16 rtpPort, |
| 1906 | const char ipAddr[64], |
| 1907 | const int sourcePort, |
| 1908 | const WebRtc_UWord16 rtcpPort) |
| 1909 | { |
| 1910 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1911 | "Channel::SetSendDestination()"); |
| 1912 | |
| 1913 | if (_externalTransport) |
| 1914 | { |
| 1915 | _engineStatisticsPtr->SetLastError( |
| 1916 | VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError, |
| 1917 | "SetSendDestination() conflict with external transport"); |
| 1918 | return -1; |
| 1919 | } |
| 1920 | |
| 1921 | // Initialize ports and IP address for the remote (destination) side. |
| 1922 | // By default, the sockets used for receiving are used for transmission as |
| 1923 | // well, hence the source ports for outgoing packets are the same as the |
| 1924 | // receiving ports specified in SetLocalReceiver. |
| 1925 | // If an extra send socket has been created, it will be utilized until a |
| 1926 | // new source port is specified or until the channel has been deleted and |
| 1927 | // recreated. If no socket exists, sockets will be created when the first |
| 1928 | // RTP and RTCP packets shall be transmitted (see e.g. |
| 1929 | // UdpTransportImpl::SendPacket()). |
| 1930 | // |
| 1931 | // NOTE: this function does not require that sockets exists; all it does is |
| 1932 | // to build send structures to be used with the sockets when they exist. |
| 1933 | // It is therefore possible to call this method before SetLocalReceiver. |
| 1934 | // However, sockets must exist if a multi-cast address is given as input. |
| 1935 | |
| 1936 | // Build send structures and enable QoS (if enabled and supported) |
| 1937 | if (_socketTransportModule.InitializeSendSockets( |
| 1938 | ipAddr, rtpPort, rtcpPort) != UdpTransport::kNoSocketError) |
| 1939 | { |
| 1940 | UdpTransport::ErrorCode lastSockError( |
| 1941 | _socketTransportModule.LastError()); |
| 1942 | switch (lastSockError) |
| 1943 | { |
| 1944 | case UdpTransport::kIpAddressInvalid: |
| 1945 | _engineStatisticsPtr->SetLastError( |
| 1946 | VE_INVALID_IP_ADDRESS, kTraceError, |
| 1947 | "SetSendDestination() invalid IP address 1"); |
| 1948 | break; |
| 1949 | case UdpTransport::kSocketInvalid: |
| 1950 | _engineStatisticsPtr->SetLastError( |
| 1951 | VE_SOCKET_ERROR, kTraceError, |
| 1952 | "SetSendDestination() invalid socket 1"); |
| 1953 | break; |
| 1954 | case UdpTransport::kQosError: |
| 1955 | _engineStatisticsPtr->SetLastError( |
| 1956 | VE_GQOS_ERROR, kTraceError, |
| 1957 | "SetSendDestination() failed to set QoS"); |
| 1958 | break; |
| 1959 | case UdpTransport::kMulticastAddressInvalid: |
| 1960 | _engineStatisticsPtr->SetLastError( |
| 1961 | VE_INVALID_MULTICAST_ADDRESS, kTraceError, |
| 1962 | "SetSendDestination() invalid multicast address"); |
| 1963 | break; |
| 1964 | default: |
| 1965 | _engineStatisticsPtr->SetLastError( |
| 1966 | VE_SOCKET_ERROR, kTraceError, |
| 1967 | "SetSendDestination() undefined socket error 1"); |
| 1968 | break; |
| 1969 | } |
| 1970 | return -1; |
| 1971 | } |
| 1972 | |
| 1973 | // Check if the user has specified a non-default source port different from |
| 1974 | // the local receive port. |
| 1975 | // If so, an extra local socket will be created unless the source port is |
| 1976 | // not unique. |
| 1977 | if (sourcePort != kVoEDefault) |
| 1978 | { |
| 1979 | WebRtc_UWord16 receiverRtpPort(0); |
| 1980 | WebRtc_UWord16 rtcpNA(0); |
| 1981 | if (_socketTransportModule.ReceiveSocketInformation(NULL, |
| 1982 | receiverRtpPort, |
| 1983 | rtcpNA, |
| 1984 | NULL) != 0) |
| 1985 | { |
| 1986 | _engineStatisticsPtr->SetLastError( |
| 1987 | VE_CANNOT_GET_SOCKET_INFO, kTraceError, |
| 1988 | "SetSendDestination() failed to retrieve socket information"); |
| 1989 | return -1; |
| 1990 | } |
| 1991 | |
| 1992 | WebRtc_UWord16 sourcePortUW16 = |
| 1993 | static_cast<WebRtc_UWord16> (sourcePort); |
| 1994 | |
| 1995 | // An extra socket will only be created if the specified source port |
| 1996 | // differs from the local receive port. |
| 1997 | if (sourcePortUW16 != receiverRtpPort) |
| 1998 | { |
| 1999 | // Initialize extra local socket to get a different source port |
| 2000 | // than the local |
| 2001 | // receiver port. Always use default source for RTCP. |
| 2002 | // Note that, this calls UdpTransport::CloseSendSockets(). |
| 2003 | if (_socketTransportModule.InitializeSourcePorts( |
| 2004 | sourcePortUW16, |
| 2005 | sourcePortUW16+1) != 0) |
| 2006 | { |
| 2007 | UdpTransport::ErrorCode lastSockError( |
| 2008 | _socketTransportModule.LastError()); |
| 2009 | switch (lastSockError) |
| 2010 | { |
| 2011 | case UdpTransport::kIpAddressInvalid: |
| 2012 | _engineStatisticsPtr->SetLastError( |
| 2013 | VE_INVALID_IP_ADDRESS, kTraceError, |
| 2014 | "SetSendDestination() invalid IP address 2"); |
| 2015 | break; |
| 2016 | case UdpTransport::kSocketInvalid: |
| 2017 | _engineStatisticsPtr->SetLastError( |
| 2018 | VE_SOCKET_ERROR, kTraceError, |
| 2019 | "SetSendDestination() invalid socket 2"); |
| 2020 | break; |
| 2021 | default: |
| 2022 | _engineStatisticsPtr->SetLastError( |
| 2023 | VE_SOCKET_ERROR, kTraceError, |
| 2024 | "SetSendDestination() undefined socket error 2"); |
| 2025 | break; |
| 2026 | } |
| 2027 | return -1; |
| 2028 | } |
| 2029 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 2030 | VoEId(_instanceId,_channelId), |
| 2031 | "SetSendDestination() extra local socket is created" |
| 2032 | " to facilitate unique source port"); |
| 2033 | } |
| 2034 | else |
| 2035 | { |
| 2036 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 2037 | VoEId(_instanceId,_channelId), |
| 2038 | "SetSendDestination() sourcePort equals the local" |
| 2039 | " receive port => no extra socket is created"); |
| 2040 | } |
| 2041 | } |
| 2042 | |
| 2043 | return 0; |
| 2044 | } |
| 2045 | #endif |
| 2046 | |
| 2047 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 2048 | WebRtc_Word32 |
| 2049 | Channel::GetSendDestination(int& port, |
| 2050 | char ipAddr[64], |
| 2051 | int& sourcePort, |
| 2052 | int& RTCPport) |
| 2053 | { |
| 2054 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2055 | "Channel::GetSendDestination()"); |
| 2056 | |
| 2057 | if (_externalTransport) |
| 2058 | { |
| 2059 | _engineStatisticsPtr->SetLastError( |
| 2060 | VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError, |
| 2061 | "GetSendDestination() conflict with external transport"); |
| 2062 | return -1; |
| 2063 | } |
| 2064 | |
| 2065 | char ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0}; |
| 2066 | WebRtc_UWord16 rtpPort(0); |
| 2067 | WebRtc_UWord16 rtcpPort(0); |
| 2068 | WebRtc_UWord16 rtpSourcePort(0); |
| 2069 | WebRtc_UWord16 rtcpSourcePort(0); |
| 2070 | |
| 2071 | // Acquire sending socket information from the socket module |
| 2072 | _socketTransportModule.SendSocketInformation(ipAddrTmp, rtpPort, rtcpPort); |
| 2073 | _socketTransportModule.SourcePorts(rtpSourcePort, rtcpSourcePort); |
| 2074 | |
| 2075 | // Deliver valid results to the user |
| 2076 | port = static_cast<int> (rtpPort); |
| 2077 | RTCPport = static_cast<int> (rtcpPort); |
| 2078 | sourcePort = static_cast<int> (rtpSourcePort); |
| 2079 | if (ipAddr != NULL) |
| 2080 | { |
| 2081 | strcpy(ipAddr, ipAddrTmp); |
| 2082 | } |
| 2083 | |
| 2084 | return 0; |
| 2085 | } |
| 2086 | #endif |
| 2087 | |
| 2088 | |
| 2089 | WebRtc_Word32 |
| 2090 | Channel::SetNetEQPlayoutMode(NetEqModes mode) |
| 2091 | { |
| 2092 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2093 | "Channel::SetNetEQPlayoutMode()"); |
| 2094 | AudioPlayoutMode playoutMode(voice); |
| 2095 | switch (mode) |
| 2096 | { |
| 2097 | case kNetEqDefault: |
| 2098 | playoutMode = voice; |
| 2099 | break; |
| 2100 | case kNetEqStreaming: |
| 2101 | playoutMode = streaming; |
| 2102 | break; |
| 2103 | case kNetEqFax: |
| 2104 | playoutMode = fax; |
| 2105 | break; |
roosa@google.com | 90d333e | 2012-12-12 21:59:14 +0000 | [diff] [blame] | 2106 | case kNetEqOff: |
| 2107 | playoutMode = off; |
| 2108 | break; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2109 | } |
| 2110 | if (_audioCodingModule.SetPlayoutMode(playoutMode) != 0) |
| 2111 | { |
| 2112 | _engineStatisticsPtr->SetLastError( |
| 2113 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2114 | "SetNetEQPlayoutMode() failed to set playout mode"); |
| 2115 | return -1; |
| 2116 | } |
| 2117 | return 0; |
| 2118 | } |
| 2119 | |
| 2120 | WebRtc_Word32 |
| 2121 | Channel::GetNetEQPlayoutMode(NetEqModes& mode) |
| 2122 | { |
| 2123 | const AudioPlayoutMode playoutMode = _audioCodingModule.PlayoutMode(); |
| 2124 | switch (playoutMode) |
| 2125 | { |
| 2126 | case voice: |
| 2127 | mode = kNetEqDefault; |
| 2128 | break; |
| 2129 | case streaming: |
| 2130 | mode = kNetEqStreaming; |
| 2131 | break; |
| 2132 | case fax: |
| 2133 | mode = kNetEqFax; |
| 2134 | break; |
roosa@google.com | 90d333e | 2012-12-12 21:59:14 +0000 | [diff] [blame] | 2135 | case off: |
| 2136 | mode = kNetEqOff; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2137 | } |
| 2138 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 2139 | VoEId(_instanceId,_channelId), |
| 2140 | "Channel::GetNetEQPlayoutMode() => mode=%u", mode); |
| 2141 | return 0; |
| 2142 | } |
| 2143 | |
| 2144 | WebRtc_Word32 |
| 2145 | Channel::SetNetEQBGNMode(NetEqBgnModes mode) |
| 2146 | { |
| 2147 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2148 | "Channel::SetNetEQPlayoutMode()"); |
| 2149 | ACMBackgroundNoiseMode noiseMode(On); |
| 2150 | switch (mode) |
| 2151 | { |
| 2152 | case kBgnOn: |
| 2153 | noiseMode = On; |
| 2154 | break; |
| 2155 | case kBgnFade: |
| 2156 | noiseMode = Fade; |
| 2157 | break; |
| 2158 | case kBgnOff: |
| 2159 | noiseMode = Off; |
| 2160 | break; |
| 2161 | } |
| 2162 | if (_audioCodingModule.SetBackgroundNoiseMode(noiseMode) != 0) |
| 2163 | { |
| 2164 | _engineStatisticsPtr->SetLastError( |
| 2165 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2166 | "SetBackgroundNoiseMode() failed to set noise mode"); |
| 2167 | return -1; |
| 2168 | } |
| 2169 | return 0; |
| 2170 | } |
| 2171 | |
| 2172 | WebRtc_Word32 |
| 2173 | Channel::SetOnHoldStatus(bool enable, OnHoldModes mode) |
| 2174 | { |
| 2175 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2176 | "Channel::SetOnHoldStatus()"); |
| 2177 | if (mode == kHoldSendAndPlay) |
| 2178 | { |
| 2179 | _outputIsOnHold = enable; |
| 2180 | _inputIsOnHold = enable; |
| 2181 | } |
| 2182 | else if (mode == kHoldPlayOnly) |
| 2183 | { |
| 2184 | _outputIsOnHold = enable; |
| 2185 | } |
| 2186 | if (mode == kHoldSendOnly) |
| 2187 | { |
| 2188 | _inputIsOnHold = enable; |
| 2189 | } |
| 2190 | return 0; |
| 2191 | } |
| 2192 | |
| 2193 | WebRtc_Word32 |
| 2194 | Channel::GetOnHoldStatus(bool& enabled, OnHoldModes& mode) |
| 2195 | { |
| 2196 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2197 | "Channel::GetOnHoldStatus()"); |
| 2198 | enabled = (_outputIsOnHold || _inputIsOnHold); |
| 2199 | if (_outputIsOnHold && _inputIsOnHold) |
| 2200 | { |
| 2201 | mode = kHoldSendAndPlay; |
| 2202 | } |
| 2203 | else if (_outputIsOnHold && !_inputIsOnHold) |
| 2204 | { |
| 2205 | mode = kHoldPlayOnly; |
| 2206 | } |
| 2207 | else if (!_outputIsOnHold && _inputIsOnHold) |
| 2208 | { |
| 2209 | mode = kHoldSendOnly; |
| 2210 | } |
| 2211 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2212 | "Channel::GetOnHoldStatus() => enabled=%d, mode=%d", |
| 2213 | enabled, mode); |
| 2214 | return 0; |
| 2215 | } |
| 2216 | |
| 2217 | WebRtc_Word32 |
| 2218 | Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) |
| 2219 | { |
| 2220 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2221 | "Channel::RegisterVoiceEngineObserver()"); |
| 2222 | CriticalSectionScoped cs(&_callbackCritSect); |
| 2223 | |
| 2224 | if (_voiceEngineObserverPtr) |
| 2225 | { |
| 2226 | _engineStatisticsPtr->SetLastError( |
| 2227 | VE_INVALID_OPERATION, kTraceError, |
| 2228 | "RegisterVoiceEngineObserver() observer already enabled"); |
| 2229 | return -1; |
| 2230 | } |
| 2231 | _voiceEngineObserverPtr = &observer; |
| 2232 | return 0; |
| 2233 | } |
| 2234 | |
| 2235 | WebRtc_Word32 |
| 2236 | Channel::DeRegisterVoiceEngineObserver() |
| 2237 | { |
| 2238 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2239 | "Channel::DeRegisterVoiceEngineObserver()"); |
| 2240 | CriticalSectionScoped cs(&_callbackCritSect); |
| 2241 | |
| 2242 | if (!_voiceEngineObserverPtr) |
| 2243 | { |
| 2244 | _engineStatisticsPtr->SetLastError( |
| 2245 | VE_INVALID_OPERATION, kTraceWarning, |
| 2246 | "DeRegisterVoiceEngineObserver() observer already disabled"); |
| 2247 | return 0; |
| 2248 | } |
| 2249 | _voiceEngineObserverPtr = NULL; |
| 2250 | return 0; |
| 2251 | } |
| 2252 | |
| 2253 | WebRtc_Word32 |
| 2254 | Channel::GetNetEQBGNMode(NetEqBgnModes& mode) |
| 2255 | { |
| 2256 | ACMBackgroundNoiseMode noiseMode(On); |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 2257 | _audioCodingModule.BackgroundNoiseMode(&noiseMode); |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2258 | switch (noiseMode) |
| 2259 | { |
| 2260 | case On: |
| 2261 | mode = kBgnOn; |
| 2262 | break; |
| 2263 | case Fade: |
| 2264 | mode = kBgnFade; |
| 2265 | break; |
| 2266 | case Off: |
| 2267 | mode = kBgnOff; |
| 2268 | break; |
| 2269 | } |
| 2270 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2271 | "Channel::GetNetEQBGNMode() => mode=%u", mode); |
| 2272 | return 0; |
| 2273 | } |
| 2274 | |
| 2275 | WebRtc_Word32 |
| 2276 | Channel::GetSendCodec(CodecInst& codec) |
| 2277 | { |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 2278 | return (_audioCodingModule.SendCodec(&codec)); |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2279 | } |
| 2280 | |
| 2281 | WebRtc_Word32 |
| 2282 | Channel::GetRecCodec(CodecInst& codec) |
| 2283 | { |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 2284 | return (_audioCodingModule.ReceiveCodec(&codec)); |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2285 | } |
| 2286 | |
| 2287 | WebRtc_Word32 |
| 2288 | Channel::SetSendCodec(const CodecInst& codec) |
| 2289 | { |
| 2290 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2291 | "Channel::SetSendCodec()"); |
| 2292 | |
| 2293 | if (_audioCodingModule.RegisterSendCodec(codec) != 0) |
| 2294 | { |
| 2295 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2296 | "SetSendCodec() failed to register codec to ACM"); |
| 2297 | return -1; |
| 2298 | } |
| 2299 | |
| 2300 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
| 2301 | { |
| 2302 | _rtpRtcpModule->DeRegisterSendPayload(codec.pltype); |
| 2303 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
| 2304 | { |
| 2305 | WEBRTC_TRACE( |
| 2306 | kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2307 | "SetSendCodec() failed to register codec to" |
| 2308 | " RTP/RTCP module"); |
| 2309 | return -1; |
| 2310 | } |
| 2311 | } |
| 2312 | |
| 2313 | if (_rtpRtcpModule->SetAudioPacketSize(codec.pacsize) != 0) |
| 2314 | { |
| 2315 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2316 | "SetSendCodec() failed to set audio packet size"); |
| 2317 | return -1; |
| 2318 | } |
| 2319 | |
| 2320 | return 0; |
| 2321 | } |
| 2322 | |
| 2323 | WebRtc_Word32 |
| 2324 | Channel::SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX) |
| 2325 | { |
| 2326 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2327 | "Channel::SetVADStatus(mode=%d)", mode); |
| 2328 | // To disable VAD, DTX must be disabled too |
| 2329 | disableDTX = ((enableVAD == false) ? true : disableDTX); |
| 2330 | if (_audioCodingModule.SetVAD(!disableDTX, enableVAD, mode) != 0) |
| 2331 | { |
| 2332 | _engineStatisticsPtr->SetLastError( |
| 2333 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2334 | "SetVADStatus() failed to set VAD"); |
| 2335 | return -1; |
| 2336 | } |
| 2337 | return 0; |
| 2338 | } |
| 2339 | |
| 2340 | WebRtc_Word32 |
| 2341 | Channel::GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX) |
| 2342 | { |
| 2343 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2344 | "Channel::GetVADStatus"); |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 2345 | if (_audioCodingModule.VAD(&disabledDTX, &enabledVAD, &mode) != 0) |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2346 | { |
| 2347 | _engineStatisticsPtr->SetLastError( |
| 2348 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2349 | "GetVADStatus() failed to get VAD status"); |
| 2350 | return -1; |
| 2351 | } |
| 2352 | disabledDTX = !disabledDTX; |
| 2353 | return 0; |
| 2354 | } |
| 2355 | |
| 2356 | WebRtc_Word32 |
| 2357 | Channel::SetRecPayloadType(const CodecInst& codec) |
| 2358 | { |
| 2359 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2360 | "Channel::SetRecPayloadType()"); |
| 2361 | |
| 2362 | if (_playing) |
| 2363 | { |
| 2364 | _engineStatisticsPtr->SetLastError( |
| 2365 | VE_ALREADY_PLAYING, kTraceError, |
| 2366 | "SetRecPayloadType() unable to set PT while playing"); |
| 2367 | return -1; |
| 2368 | } |
| 2369 | if (_receiving) |
| 2370 | { |
| 2371 | _engineStatisticsPtr->SetLastError( |
| 2372 | VE_ALREADY_LISTENING, kTraceError, |
| 2373 | "SetRecPayloadType() unable to set PT while listening"); |
| 2374 | return -1; |
| 2375 | } |
| 2376 | |
| 2377 | if (codec.pltype == -1) |
| 2378 | { |
| 2379 | // De-register the selected codec (RTP/RTCP module and ACM) |
| 2380 | |
| 2381 | WebRtc_Word8 pltype(-1); |
| 2382 | CodecInst rxCodec = codec; |
| 2383 | |
| 2384 | // Get payload type for the given codec |
| 2385 | _rtpRtcpModule->ReceivePayloadType(rxCodec, &pltype); |
| 2386 | rxCodec.pltype = pltype; |
| 2387 | |
| 2388 | if (_rtpRtcpModule->DeRegisterReceivePayload(pltype) != 0) |
| 2389 | { |
| 2390 | _engineStatisticsPtr->SetLastError( |
| 2391 | VE_RTP_RTCP_MODULE_ERROR, |
| 2392 | kTraceError, |
| 2393 | "SetRecPayloadType() RTP/RTCP-module deregistration " |
| 2394 | "failed"); |
| 2395 | return -1; |
| 2396 | } |
| 2397 | if (_audioCodingModule.UnregisterReceiveCodec(rxCodec.pltype) != 0) |
| 2398 | { |
| 2399 | _engineStatisticsPtr->SetLastError( |
| 2400 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2401 | "SetRecPayloadType() ACM deregistration failed - 1"); |
| 2402 | return -1; |
| 2403 | } |
| 2404 | return 0; |
| 2405 | } |
| 2406 | |
| 2407 | if (_rtpRtcpModule->RegisterReceivePayload(codec) != 0) |
| 2408 | { |
| 2409 | // First attempt to register failed => de-register and try again |
| 2410 | _rtpRtcpModule->DeRegisterReceivePayload(codec.pltype); |
| 2411 | if (_rtpRtcpModule->RegisterReceivePayload(codec) != 0) |
| 2412 | { |
| 2413 | _engineStatisticsPtr->SetLastError( |
| 2414 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 2415 | "SetRecPayloadType() RTP/RTCP-module registration failed"); |
| 2416 | return -1; |
| 2417 | } |
| 2418 | } |
| 2419 | if (_audioCodingModule.RegisterReceiveCodec(codec) != 0) |
| 2420 | { |
| 2421 | _audioCodingModule.UnregisterReceiveCodec(codec.pltype); |
| 2422 | if (_audioCodingModule.RegisterReceiveCodec(codec) != 0) |
| 2423 | { |
| 2424 | _engineStatisticsPtr->SetLastError( |
| 2425 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2426 | "SetRecPayloadType() ACM registration failed - 1"); |
| 2427 | return -1; |
| 2428 | } |
| 2429 | } |
| 2430 | return 0; |
| 2431 | } |
| 2432 | |
| 2433 | WebRtc_Word32 |
| 2434 | Channel::GetRecPayloadType(CodecInst& codec) |
| 2435 | { |
| 2436 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2437 | "Channel::GetRecPayloadType()"); |
| 2438 | WebRtc_Word8 payloadType(-1); |
| 2439 | if (_rtpRtcpModule->ReceivePayloadType(codec, &payloadType) != 0) |
| 2440 | { |
| 2441 | _engineStatisticsPtr->SetLastError( |
| 2442 | VE_RTP_RTCP_MODULE_ERROR, kTraceWarning, |
| 2443 | "GetRecPayloadType() failed to retrieve RX payload type"); |
| 2444 | return -1; |
| 2445 | } |
| 2446 | codec.pltype = payloadType; |
| 2447 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2448 | "Channel::GetRecPayloadType() => pltype=%u", codec.pltype); |
| 2449 | return 0; |
| 2450 | } |
| 2451 | |
| 2452 | WebRtc_Word32 |
| 2453 | Channel::SetAMREncFormat(AmrMode mode) |
| 2454 | { |
| 2455 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2456 | "Channel::SetAMREncFormat()"); |
| 2457 | |
| 2458 | // ACM doesn't support AMR |
| 2459 | return -1; |
| 2460 | } |
| 2461 | |
| 2462 | WebRtc_Word32 |
| 2463 | Channel::SetAMRDecFormat(AmrMode mode) |
| 2464 | { |
| 2465 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2466 | "Channel::SetAMRDecFormat()"); |
| 2467 | |
| 2468 | // ACM doesn't support AMR |
| 2469 | return -1; |
| 2470 | } |
| 2471 | |
| 2472 | WebRtc_Word32 |
| 2473 | Channel::SetAMRWbEncFormat(AmrMode mode) |
| 2474 | { |
| 2475 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2476 | "Channel::SetAMRWbEncFormat()"); |
| 2477 | |
| 2478 | // ACM doesn't support AMR |
| 2479 | return -1; |
| 2480 | |
| 2481 | } |
| 2482 | |
| 2483 | WebRtc_Word32 |
| 2484 | Channel::SetAMRWbDecFormat(AmrMode mode) |
| 2485 | { |
| 2486 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2487 | "Channel::SetAMRWbDecFormat()"); |
| 2488 | |
| 2489 | // ACM doesn't support AMR |
| 2490 | return -1; |
| 2491 | } |
| 2492 | |
| 2493 | WebRtc_Word32 |
| 2494 | Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency) |
| 2495 | { |
| 2496 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2497 | "Channel::SetSendCNPayloadType()"); |
| 2498 | |
| 2499 | CodecInst codec; |
| 2500 | WebRtc_Word32 samplingFreqHz(-1); |
| 2501 | const int kMono = 1; |
| 2502 | if (frequency == kFreq32000Hz) |
| 2503 | samplingFreqHz = 32000; |
| 2504 | else if (frequency == kFreq16000Hz) |
| 2505 | samplingFreqHz = 16000; |
| 2506 | |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 2507 | if (_audioCodingModule.Codec("CN", &codec, samplingFreqHz, kMono) == -1) |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2508 | { |
| 2509 | _engineStatisticsPtr->SetLastError( |
| 2510 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2511 | "SetSendCNPayloadType() failed to retrieve default CN codec " |
| 2512 | "settings"); |
| 2513 | return -1; |
| 2514 | } |
| 2515 | |
| 2516 | // Modify the payload type (must be set to dynamic range) |
| 2517 | codec.pltype = type; |
| 2518 | |
| 2519 | if (_audioCodingModule.RegisterSendCodec(codec) != 0) |
| 2520 | { |
| 2521 | _engineStatisticsPtr->SetLastError( |
| 2522 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2523 | "SetSendCNPayloadType() failed to register CN to ACM"); |
| 2524 | return -1; |
| 2525 | } |
| 2526 | |
| 2527 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
| 2528 | { |
| 2529 | _rtpRtcpModule->DeRegisterSendPayload(codec.pltype); |
| 2530 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
| 2531 | { |
| 2532 | _engineStatisticsPtr->SetLastError( |
| 2533 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 2534 | "SetSendCNPayloadType() failed to register CN to RTP/RTCP " |
| 2535 | "module"); |
| 2536 | return -1; |
| 2537 | } |
| 2538 | } |
| 2539 | return 0; |
| 2540 | } |
| 2541 | |
| 2542 | WebRtc_Word32 |
| 2543 | Channel::SetISACInitTargetRate(int rateBps, bool useFixedFrameSize) |
| 2544 | { |
| 2545 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2546 | "Channel::SetISACInitTargetRate()"); |
| 2547 | |
| 2548 | CodecInst sendCodec; |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 2549 | if (_audioCodingModule.SendCodec(&sendCodec) == -1) |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2550 | { |
| 2551 | _engineStatisticsPtr->SetLastError( |
| 2552 | VE_CODEC_ERROR, kTraceError, |
| 2553 | "SetISACInitTargetRate() failed to retrieve send codec"); |
| 2554 | return -1; |
| 2555 | } |
| 2556 | if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0) |
| 2557 | { |
| 2558 | // This API is only valid if iSAC is setup to run in channel-adaptive |
| 2559 | // mode. |
| 2560 | // We do not validate the adaptive mode here. It is done later in the |
| 2561 | // ConfigISACBandwidthEstimator() API. |
| 2562 | _engineStatisticsPtr->SetLastError( |
| 2563 | VE_CODEC_ERROR, kTraceError, |
| 2564 | "SetISACInitTargetRate() send codec is not iSAC"); |
| 2565 | return -1; |
| 2566 | } |
| 2567 | |
| 2568 | WebRtc_UWord8 initFrameSizeMsec(0); |
| 2569 | if (16000 == sendCodec.plfreq) |
| 2570 | { |
| 2571 | // Note that 0 is a valid and corresponds to "use default |
| 2572 | if ((rateBps != 0 && |
| 2573 | rateBps < kVoiceEngineMinIsacInitTargetRateBpsWb) || |
| 2574 | (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsWb)) |
| 2575 | { |
| 2576 | _engineStatisticsPtr->SetLastError( |
| 2577 | VE_INVALID_ARGUMENT, kTraceError, |
| 2578 | "SetISACInitTargetRate() invalid target rate - 1"); |
| 2579 | return -1; |
| 2580 | } |
| 2581 | // 30 or 60ms |
| 2582 | initFrameSizeMsec = (WebRtc_UWord8)(sendCodec.pacsize / 16); |
| 2583 | } |
| 2584 | else if (32000 == sendCodec.plfreq) |
| 2585 | { |
| 2586 | if ((rateBps != 0 && |
| 2587 | rateBps < kVoiceEngineMinIsacInitTargetRateBpsSwb) || |
| 2588 | (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsSwb)) |
| 2589 | { |
| 2590 | _engineStatisticsPtr->SetLastError( |
| 2591 | VE_INVALID_ARGUMENT, kTraceError, |
| 2592 | "SetISACInitTargetRate() invalid target rate - 2"); |
| 2593 | return -1; |
| 2594 | } |
| 2595 | initFrameSizeMsec = (WebRtc_UWord8)(sendCodec.pacsize / 32); // 30ms |
| 2596 | } |
| 2597 | |
| 2598 | if (_audioCodingModule.ConfigISACBandwidthEstimator( |
| 2599 | initFrameSizeMsec, rateBps, useFixedFrameSize) == -1) |
| 2600 | { |
| 2601 | _engineStatisticsPtr->SetLastError( |
| 2602 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2603 | "SetISACInitTargetRate() iSAC BWE config failed"); |
| 2604 | return -1; |
| 2605 | } |
| 2606 | |
| 2607 | return 0; |
| 2608 | } |
| 2609 | |
| 2610 | WebRtc_Word32 |
| 2611 | Channel::SetISACMaxRate(int rateBps) |
| 2612 | { |
| 2613 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2614 | "Channel::SetISACMaxRate()"); |
| 2615 | |
| 2616 | CodecInst sendCodec; |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 2617 | if (_audioCodingModule.SendCodec(&sendCodec) == -1) |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2618 | { |
| 2619 | _engineStatisticsPtr->SetLastError( |
| 2620 | VE_CODEC_ERROR, kTraceError, |
| 2621 | "SetISACMaxRate() failed to retrieve send codec"); |
| 2622 | return -1; |
| 2623 | } |
| 2624 | if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0) |
| 2625 | { |
| 2626 | // This API is only valid if iSAC is selected as sending codec. |
| 2627 | _engineStatisticsPtr->SetLastError( |
| 2628 | VE_CODEC_ERROR, kTraceError, |
| 2629 | "SetISACMaxRate() send codec is not iSAC"); |
| 2630 | return -1; |
| 2631 | } |
| 2632 | if (16000 == sendCodec.plfreq) |
| 2633 | { |
| 2634 | if ((rateBps < kVoiceEngineMinIsacMaxRateBpsWb) || |
| 2635 | (rateBps > kVoiceEngineMaxIsacMaxRateBpsWb)) |
| 2636 | { |
| 2637 | _engineStatisticsPtr->SetLastError( |
| 2638 | VE_INVALID_ARGUMENT, kTraceError, |
| 2639 | "SetISACMaxRate() invalid max rate - 1"); |
| 2640 | return -1; |
| 2641 | } |
| 2642 | } |
| 2643 | else if (32000 == sendCodec.plfreq) |
| 2644 | { |
| 2645 | if ((rateBps < kVoiceEngineMinIsacMaxRateBpsSwb) || |
| 2646 | (rateBps > kVoiceEngineMaxIsacMaxRateBpsSwb)) |
| 2647 | { |
| 2648 | _engineStatisticsPtr->SetLastError( |
| 2649 | VE_INVALID_ARGUMENT, kTraceError, |
| 2650 | "SetISACMaxRate() invalid max rate - 2"); |
| 2651 | return -1; |
| 2652 | } |
| 2653 | } |
| 2654 | if (_sending) |
| 2655 | { |
| 2656 | _engineStatisticsPtr->SetLastError( |
| 2657 | VE_SENDING, kTraceError, |
| 2658 | "SetISACMaxRate() unable to set max rate while sending"); |
| 2659 | return -1; |
| 2660 | } |
| 2661 | |
| 2662 | // Set the maximum instantaneous rate of iSAC (works for both adaptive |
| 2663 | // and non-adaptive mode) |
| 2664 | if (_audioCodingModule.SetISACMaxRate(rateBps) == -1) |
| 2665 | { |
| 2666 | _engineStatisticsPtr->SetLastError( |
| 2667 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2668 | "SetISACMaxRate() failed to set max rate"); |
| 2669 | return -1; |
| 2670 | } |
| 2671 | |
| 2672 | return 0; |
| 2673 | } |
| 2674 | |
| 2675 | WebRtc_Word32 |
| 2676 | Channel::SetISACMaxPayloadSize(int sizeBytes) |
| 2677 | { |
| 2678 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2679 | "Channel::SetISACMaxPayloadSize()"); |
| 2680 | CodecInst sendCodec; |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 2681 | if (_audioCodingModule.SendCodec(&sendCodec) == -1) |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 2682 | { |
| 2683 | _engineStatisticsPtr->SetLastError( |
| 2684 | VE_CODEC_ERROR, kTraceError, |
| 2685 | "SetISACMaxPayloadSize() failed to retrieve send codec"); |
| 2686 | return -1; |
| 2687 | } |
| 2688 | if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0) |
| 2689 | { |
| 2690 | _engineStatisticsPtr->SetLastError( |
| 2691 | VE_CODEC_ERROR, kTraceError, |
| 2692 | "SetISACMaxPayloadSize() send codec is not iSAC"); |
| 2693 | return -1; |
| 2694 | } |
| 2695 | if (16000 == sendCodec.plfreq) |
| 2696 | { |
| 2697 | if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesWb) || |
| 2698 | (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesWb)) |
| 2699 | { |
| 2700 | _engineStatisticsPtr->SetLastError( |
| 2701 | VE_INVALID_ARGUMENT, kTraceError, |
| 2702 | "SetISACMaxPayloadSize() invalid max payload - 1"); |
| 2703 | return -1; |
| 2704 | } |
| 2705 | } |
| 2706 | else if (32000 == sendCodec.plfreq) |
| 2707 | { |
| 2708 | if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesSwb) || |
| 2709 | (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb)) |
| 2710 | { |
| 2711 | _engineStatisticsPtr->SetLastError( |
| 2712 | VE_INVALID_ARGUMENT, kTraceError, |
| 2713 | "SetISACMaxPayloadSize() invalid max payload - 2"); |
| 2714 | return -1; |
| 2715 | } |
| 2716 | } |
| 2717 | if (_sending) |
| 2718 | { |
| 2719 | _engineStatisticsPtr->SetLastError( |
| 2720 | VE_SENDING, kTraceError, |
| 2721 | "SetISACMaxPayloadSize() unable to set max rate while sending"); |
| 2722 | return -1; |
| 2723 | } |
| 2724 | |
| 2725 | if (_audioCodingModule.SetISACMaxPayloadSize(sizeBytes) == -1) |
| 2726 | { |
| 2727 | _engineStatisticsPtr->SetLastError( |
| 2728 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2729 | "SetISACMaxPayloadSize() failed to set max payload size"); |
| 2730 | return -1; |
| 2731 | } |
| 2732 | return 0; |
| 2733 | } |
| 2734 | |
| 2735 | WebRtc_Word32 Channel::RegisterExternalTransport(Transport& transport) |
| 2736 | { |
| 2737 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 2738 | "Channel::RegisterExternalTransport()"); |
| 2739 | |
| 2740 | CriticalSectionScoped cs(&_callbackCritSect); |
| 2741 | |
| 2742 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 2743 | // Sanity checks for default (non external transport) to avoid conflict with |
| 2744 | // WebRtc sockets. |
| 2745 | if (_socketTransportModule.SendSocketsInitialized()) |
| 2746 | { |
| 2747 | _engineStatisticsPtr->SetLastError(VE_SEND_SOCKETS_CONFLICT, |
| 2748 | kTraceError, |
| 2749 | "RegisterExternalTransport() send sockets already initialized"); |
| 2750 | return -1; |
| 2751 | } |
| 2752 | if (_socketTransportModule.ReceiveSocketsInitialized()) |
| 2753 | { |
| 2754 | _engineStatisticsPtr->SetLastError(VE_RECEIVE_SOCKETS_CONFLICT, |
| 2755 | kTraceError, |
| 2756 | "RegisterExternalTransport() receive sockets already initialized"); |
| 2757 | return -1; |
| 2758 | } |
| 2759 | #endif |
| 2760 | if (_externalTransport) |
| 2761 | { |
| 2762 | _engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION, |
| 2763 | kTraceError, |
| 2764 | "RegisterExternalTransport() external transport already enabled"); |
| 2765 | return -1; |
| 2766 | } |
| 2767 | _externalTransport = true; |
| 2768 | _transportPtr = &transport; |
| 2769 | return 0; |
| 2770 | } |
| 2771 | |
| 2772 | WebRtc_Word32 |
| 2773 | Channel::DeRegisterExternalTransport() |
| 2774 | { |
| 2775 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2776 | "Channel::DeRegisterExternalTransport()"); |
| 2777 | |
| 2778 | CriticalSectionScoped cs(&_callbackCritSect); |
| 2779 | |
| 2780 | if (!_transportPtr) |
| 2781 | { |
| 2782 | _engineStatisticsPtr->SetLastError( |
| 2783 | VE_INVALID_OPERATION, kTraceWarning, |
| 2784 | "DeRegisterExternalTransport() external transport already " |
| 2785 | "disabled"); |
| 2786 | return 0; |
| 2787 | } |
| 2788 | _externalTransport = false; |
| 2789 | #ifdef WEBRTC_EXTERNAL_TRANSPORT |
| 2790 | _transportPtr = NULL; |
| 2791 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2792 | "DeRegisterExternalTransport() all transport is disabled"); |
| 2793 | #else |
| 2794 | _transportPtr = &_socketTransportModule; |
| 2795 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2796 | "DeRegisterExternalTransport() internal Transport is enabled"); |
| 2797 | #endif |
| 2798 | return 0; |
| 2799 | } |
| 2800 | |
| 2801 | WebRtc_Word32 |
| 2802 | Channel::ReceivedRTPPacket(const WebRtc_Word8* data, WebRtc_Word32 length) |
| 2803 | { |
| 2804 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2805 | "Channel::ReceivedRTPPacket()"); |
| 2806 | const char dummyIP[] = "127.0.0.1"; |
| 2807 | IncomingRTPPacket(data, length, dummyIP, 0); |
| 2808 | return 0; |
| 2809 | } |
| 2810 | |
| 2811 | WebRtc_Word32 |
| 2812 | Channel::ReceivedRTCPPacket(const WebRtc_Word8* data, WebRtc_Word32 length) |
| 2813 | { |
| 2814 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2815 | "Channel::ReceivedRTCPPacket()"); |
| 2816 | const char dummyIP[] = "127.0.0.1"; |
| 2817 | IncomingRTCPPacket(data, length, dummyIP, 0); |
| 2818 | return 0; |
| 2819 | } |
| 2820 | |
| 2821 | #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| 2822 | WebRtc_Word32 |
| 2823 | Channel::GetSourceInfo(int& rtpPort, int& rtcpPort, char ipAddr[64]) |
| 2824 | { |
| 2825 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2826 | "Channel::GetSourceInfo()"); |
| 2827 | |
| 2828 | WebRtc_UWord16 rtpPortModule; |
| 2829 | WebRtc_UWord16 rtcpPortModule; |
| 2830 | char ipaddr[UdpTransport::kIpAddressVersion6Length] = {0}; |
| 2831 | |
| 2832 | if (_socketTransportModule.RemoteSocketInformation(ipaddr, |
| 2833 | rtpPortModule, |
| 2834 | rtcpPortModule) != 0) |
| 2835 | { |
| 2836 | _engineStatisticsPtr->SetLastError( |
| 2837 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError, |
| 2838 | "GetSourceInfo() failed to retrieve remote socket information"); |
| 2839 | return -1; |
| 2840 | } |
| 2841 | strcpy(ipAddr, ipaddr); |
| 2842 | rtpPort = rtpPortModule; |
| 2843 | rtcpPort = rtcpPortModule; |
| 2844 | |
| 2845 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2846 | "GetSourceInfo() => rtpPort=%d, rtcpPort=%d, ipAddr=%s", |
| 2847 | rtpPort, rtcpPort, ipAddr); |
| 2848 | return 0; |
| 2849 | } |
| 2850 | |
| 2851 | WebRtc_Word32 |
| 2852 | Channel::EnableIPv6() |
| 2853 | { |
| 2854 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2855 | "Channel::EnableIPv6()"); |
| 2856 | if (_socketTransportModule.ReceiveSocketsInitialized() || |
| 2857 | _socketTransportModule.SendSocketsInitialized()) |
| 2858 | { |
| 2859 | _engineStatisticsPtr->SetLastError( |
| 2860 | VE_INVALID_OPERATION, kTraceError, |
| 2861 | "EnableIPv6() socket layer is already initialized"); |
| 2862 | return -1; |
| 2863 | } |
| 2864 | if (_socketTransportModule.EnableIpV6() != 0) |
| 2865 | { |
| 2866 | _engineStatisticsPtr->SetLastError( |
| 2867 | VE_SOCKET_ERROR, kTraceError, |
| 2868 | "EnableIPv6() failed to enable IPv6"); |
| 2869 | const UdpTransport::ErrorCode lastError = |
| 2870 | _socketTransportModule.LastError(); |
| 2871 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2872 | "UdpTransport::LastError() => %d", lastError); |
| 2873 | return -1; |
| 2874 | } |
| 2875 | return 0; |
| 2876 | } |
| 2877 | |
| 2878 | bool |
| 2879 | Channel::IPv6IsEnabled() const |
| 2880 | { |
| 2881 | bool isEnabled = _socketTransportModule.IpV6Enabled(); |
| 2882 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2883 | "IPv6IsEnabled() => %d", isEnabled); |
| 2884 | return isEnabled; |
| 2885 | } |
| 2886 | |
| 2887 | WebRtc_Word32 |
| 2888 | Channel::SetSourceFilter(int rtpPort, int rtcpPort, const char ipAddr[64]) |
| 2889 | { |
| 2890 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2891 | "Channel::SetSourceFilter()"); |
| 2892 | if (_socketTransportModule.SetFilterPorts( |
| 2893 | static_cast<WebRtc_UWord16>(rtpPort), |
| 2894 | static_cast<WebRtc_UWord16>(rtcpPort)) != 0) |
| 2895 | { |
| 2896 | _engineStatisticsPtr->SetLastError( |
| 2897 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError, |
| 2898 | "SetSourceFilter() failed to set filter ports"); |
| 2899 | const UdpTransport::ErrorCode lastError = |
| 2900 | _socketTransportModule.LastError(); |
| 2901 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2902 | "UdpTransport::LastError() => %d", |
| 2903 | lastError); |
| 2904 | return -1; |
| 2905 | } |
| 2906 | const char* filterIpAddress = ipAddr; |
| 2907 | if (_socketTransportModule.SetFilterIP(filterIpAddress) != 0) |
| 2908 | { |
| 2909 | _engineStatisticsPtr->SetLastError( |
| 2910 | VE_INVALID_IP_ADDRESS, kTraceError, |
| 2911 | "SetSourceFilter() failed to set filter IP address"); |
| 2912 | const UdpTransport::ErrorCode lastError = |
| 2913 | _socketTransportModule.LastError(); |
| 2914 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2915 | "UdpTransport::LastError() => %d", lastError); |
| 2916 | return -1; |
| 2917 | } |
| 2918 | return 0; |
| 2919 | } |
| 2920 | |
| 2921 | WebRtc_Word32 |
| 2922 | Channel::GetSourceFilter(int& rtpPort, int& rtcpPort, char ipAddr[64]) |
| 2923 | { |
| 2924 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2925 | "Channel::GetSourceFilter()"); |
| 2926 | WebRtc_UWord16 rtpFilterPort(0); |
| 2927 | WebRtc_UWord16 rtcpFilterPort(0); |
| 2928 | if (_socketTransportModule.FilterPorts(rtpFilterPort, rtcpFilterPort) != 0) |
| 2929 | { |
| 2930 | _engineStatisticsPtr->SetLastError( |
| 2931 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning, |
| 2932 | "GetSourceFilter() failed to retrieve filter ports"); |
| 2933 | } |
| 2934 | char ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0}; |
| 2935 | if (_socketTransportModule.FilterIP(ipAddrTmp) != 0) |
| 2936 | { |
| 2937 | // no filter has been configured (not seen as an error) |
| 2938 | memset(ipAddrTmp, |
| 2939 | 0, UdpTransport::kIpAddressVersion6Length); |
| 2940 | } |
| 2941 | rtpPort = static_cast<int> (rtpFilterPort); |
| 2942 | rtcpPort = static_cast<int> (rtcpFilterPort); |
| 2943 | strcpy(ipAddr, ipAddrTmp); |
| 2944 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2945 | "GetSourceFilter() => rtpPort=%d, rtcpPort=%d, ipAddr=%s", |
| 2946 | rtpPort, rtcpPort, ipAddr); |
| 2947 | return 0; |
| 2948 | } |
| 2949 | |
| 2950 | WebRtc_Word32 |
| 2951 | Channel::SetSendTOS(int DSCP, int priority, bool useSetSockopt) |
| 2952 | { |
| 2953 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2954 | "Channel::SetSendTOS(DSCP=%d, useSetSockopt=%d)", |
| 2955 | DSCP, (int)useSetSockopt); |
| 2956 | |
| 2957 | // Set TOS value and possibly try to force usage of setsockopt() |
| 2958 | if (_socketTransportModule.SetToS(DSCP, useSetSockopt) != 0) |
| 2959 | { |
| 2960 | UdpTransport::ErrorCode lastSockError( |
| 2961 | _socketTransportModule.LastError()); |
| 2962 | switch (lastSockError) |
| 2963 | { |
| 2964 | case UdpTransport::kTosError: |
| 2965 | _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError, |
| 2966 | "SetSendTOS() TOS error"); |
| 2967 | break; |
| 2968 | case UdpTransport::kQosError: |
| 2969 | _engineStatisticsPtr->SetLastError( |
| 2970 | VE_TOS_GQOS_CONFLICT, kTraceError, |
| 2971 | "SetSendTOS() GQOS error"); |
| 2972 | break; |
| 2973 | case UdpTransport::kTosInvalid: |
| 2974 | // can't switch SetSockOpt method without disabling TOS first, or |
| 2975 | // SetSockopt() call failed |
| 2976 | _engineStatisticsPtr->SetLastError(VE_TOS_INVALID, kTraceError, |
| 2977 | "SetSendTOS() invalid TOS"); |
| 2978 | break; |
| 2979 | case UdpTransport::kSocketInvalid: |
| 2980 | _engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError, |
| 2981 | "SetSendTOS() invalid Socket"); |
| 2982 | break; |
| 2983 | default: |
| 2984 | _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError, |
| 2985 | "SetSendTOS() TOS error"); |
| 2986 | break; |
| 2987 | } |
| 2988 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2989 | "UdpTransport => lastError = %d", |
| 2990 | lastSockError); |
| 2991 | return -1; |
| 2992 | } |
| 2993 | |
| 2994 | // Set priority (PCP) value, -1 means don't change |
| 2995 | if (-1 != priority) |
| 2996 | { |
| 2997 | if (_socketTransportModule.SetPCP(priority) != 0) |
| 2998 | { |
| 2999 | UdpTransport::ErrorCode lastSockError( |
| 3000 | _socketTransportModule.LastError()); |
| 3001 | switch (lastSockError) |
| 3002 | { |
| 3003 | case UdpTransport::kPcpError: |
| 3004 | _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError, |
| 3005 | "SetSendTOS() PCP error"); |
| 3006 | break; |
| 3007 | case UdpTransport::kQosError: |
| 3008 | _engineStatisticsPtr->SetLastError( |
| 3009 | VE_TOS_GQOS_CONFLICT, kTraceError, |
| 3010 | "SetSendTOS() GQOS conflict"); |
| 3011 | break; |
| 3012 | case UdpTransport::kSocketInvalid: |
| 3013 | _engineStatisticsPtr->SetLastError( |
| 3014 | VE_SOCKET_ERROR, kTraceError, |
| 3015 | "SetSendTOS() invalid Socket"); |
| 3016 | break; |
| 3017 | default: |
| 3018 | _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError, |
| 3019 | "SetSendTOS() PCP error"); |
| 3020 | break; |
| 3021 | } |
| 3022 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 3023 | VoEId(_instanceId,_channelId), |
| 3024 | "UdpTransport => lastError = %d", |
| 3025 | lastSockError); |
| 3026 | return -1; |
| 3027 | } |
| 3028 | } |
| 3029 | |
| 3030 | return 0; |
| 3031 | } |
| 3032 | |
| 3033 | WebRtc_Word32 |
| 3034 | Channel::GetSendTOS(int &DSCP, int& priority, bool &useSetSockopt) |
| 3035 | { |
| 3036 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3037 | "Channel::GetSendTOS(DSCP=?, useSetSockopt=?)"); |
| 3038 | WebRtc_Word32 dscp(0), prio(0); |
| 3039 | bool setSockopt(false); |
| 3040 | if (_socketTransportModule.ToS(dscp, setSockopt) != 0) |
| 3041 | { |
| 3042 | _engineStatisticsPtr->SetLastError( |
| 3043 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError, |
| 3044 | "GetSendTOS() failed to get TOS info"); |
| 3045 | return -1; |
| 3046 | } |
| 3047 | if (_socketTransportModule.PCP(prio) != 0) |
| 3048 | { |
| 3049 | _engineStatisticsPtr->SetLastError( |
| 3050 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError, |
| 3051 | "GetSendTOS() failed to get PCP info"); |
| 3052 | return -1; |
| 3053 | } |
| 3054 | DSCP = static_cast<int> (dscp); |
| 3055 | priority = static_cast<int> (prio); |
| 3056 | useSetSockopt = setSockopt; |
| 3057 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1), |
| 3058 | "GetSendTOS() => DSCP=%d, priority=%d, useSetSockopt=%d", |
| 3059 | DSCP, priority, (int)useSetSockopt); |
| 3060 | return 0; |
| 3061 | } |
| 3062 | |
| 3063 | #if defined(_WIN32) |
| 3064 | WebRtc_Word32 |
| 3065 | Channel::SetSendGQoS(bool enable, int serviceType, int overrideDSCP) |
| 3066 | { |
| 3067 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3068 | "Channel::SetSendGQoS(enable=%d, serviceType=%d, " |
| 3069 | "overrideDSCP=%d)", |
| 3070 | (int)enable, serviceType, overrideDSCP); |
| 3071 | if(!_socketTransportModule.ReceiveSocketsInitialized()) |
| 3072 | { |
| 3073 | _engineStatisticsPtr->SetLastError( |
| 3074 | VE_SOCKETS_NOT_INITED, kTraceError, |
| 3075 | "SetSendGQoS() GQoS state must be set after sockets are created"); |
| 3076 | return -1; |
| 3077 | } |
| 3078 | if(!_socketTransportModule.SendSocketsInitialized()) |
| 3079 | { |
| 3080 | _engineStatisticsPtr->SetLastError( |
| 3081 | VE_DESTINATION_NOT_INITED, kTraceError, |
| 3082 | "SetSendGQoS() GQoS state must be set after sending side is " |
| 3083 | "initialized"); |
| 3084 | return -1; |
| 3085 | } |
| 3086 | if (enable && |
| 3087 | (serviceType != SERVICETYPE_BESTEFFORT) && |
| 3088 | (serviceType != SERVICETYPE_CONTROLLEDLOAD) && |
| 3089 | (serviceType != SERVICETYPE_GUARANTEED) && |
| 3090 | (serviceType != SERVICETYPE_QUALITATIVE)) |
| 3091 | { |
| 3092 | _engineStatisticsPtr->SetLastError( |
| 3093 | VE_INVALID_ARGUMENT, kTraceError, |
| 3094 | "SetSendGQoS() Invalid service type"); |
| 3095 | return -1; |
| 3096 | } |
| 3097 | if (enable && ((overrideDSCP < 0) || (overrideDSCP > 63))) |
| 3098 | { |
| 3099 | _engineStatisticsPtr->SetLastError( |
| 3100 | VE_INVALID_ARGUMENT, kTraceError, |
| 3101 | "SetSendGQoS() Invalid overrideDSCP value"); |
| 3102 | return -1; |
| 3103 | } |
| 3104 | |
| 3105 | // Avoid GQoS/ToS conflict when user wants to override the default DSCP |
| 3106 | // mapping |
| 3107 | bool QoS(false); |
| 3108 | WebRtc_Word32 sType(0); |
| 3109 | WebRtc_Word32 ovrDSCP(0); |
| 3110 | if (_socketTransportModule.QoS(QoS, sType, ovrDSCP)) |
| 3111 | { |
| 3112 | _engineStatisticsPtr->SetLastError( |
| 3113 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError, |
| 3114 | "SetSendGQoS() failed to get QOS info"); |
| 3115 | return -1; |
| 3116 | } |
| 3117 | if (QoS && ovrDSCP == 0 && overrideDSCP != 0) |
| 3118 | { |
| 3119 | _engineStatisticsPtr->SetLastError( |
| 3120 | VE_TOS_GQOS_CONFLICT, kTraceError, |
| 3121 | "SetSendGQoS() QOS is already enabled and overrideDSCP differs," |
| 3122 | " not allowed"); |
| 3123 | return -1; |
| 3124 | } |
| 3125 | const WebRtc_Word32 maxBitrate(0); |
| 3126 | if (_socketTransportModule.SetQoS(enable, |
| 3127 | static_cast<WebRtc_Word32>(serviceType), |
| 3128 | maxBitrate, |
| 3129 | static_cast<WebRtc_Word32>(overrideDSCP), |
| 3130 | true)) |
| 3131 | { |
| 3132 | UdpTransport::ErrorCode lastSockError( |
| 3133 | _socketTransportModule.LastError()); |
| 3134 | switch (lastSockError) |
| 3135 | { |
| 3136 | case UdpTransport::kQosError: |
| 3137 | _engineStatisticsPtr->SetLastError(VE_GQOS_ERROR, kTraceError, |
| 3138 | "SetSendGQoS() QOS error"); |
| 3139 | break; |
| 3140 | default: |
| 3141 | _engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError, |
| 3142 | "SetSendGQoS() Socket error"); |
| 3143 | break; |
| 3144 | } |
| 3145 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3146 | "UdpTransport() => lastError = %d", |
| 3147 | lastSockError); |
| 3148 | return -1; |
| 3149 | } |
| 3150 | return 0; |
| 3151 | } |
| 3152 | #endif |
| 3153 | |
| 3154 | #if defined(_WIN32) |
| 3155 | WebRtc_Word32 |
| 3156 | Channel::GetSendGQoS(bool &enabled, int &serviceType, int &overrideDSCP) |
| 3157 | { |
| 3158 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3159 | "Channel::GetSendGQoS(enable=?, serviceType=?, " |
| 3160 | "overrideDSCP=?)"); |
| 3161 | |
| 3162 | bool QoS(false); |
| 3163 | WebRtc_Word32 serviceTypeModule(0); |
| 3164 | WebRtc_Word32 overrideDSCPModule(0); |
| 3165 | _socketTransportModule.QoS(QoS, serviceTypeModule, overrideDSCPModule); |
| 3166 | |
| 3167 | enabled = QoS; |
| 3168 | serviceType = static_cast<int> (serviceTypeModule); |
| 3169 | overrideDSCP = static_cast<int> (overrideDSCPModule); |
| 3170 | |
| 3171 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3172 | "GetSendGQoS() => enabled=%d, serviceType=%d, overrideDSCP=%d", |
| 3173 | (int)enabled, serviceType, overrideDSCP); |
| 3174 | return 0; |
| 3175 | } |
| 3176 | #endif |
| 3177 | #endif |
| 3178 | |
| 3179 | WebRtc_Word32 |
| 3180 | Channel::SetPacketTimeoutNotification(bool enable, int timeoutSeconds) |
| 3181 | { |
| 3182 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3183 | "Channel::SetPacketTimeoutNotification()"); |
| 3184 | if (enable) |
| 3185 | { |
| 3186 | const WebRtc_UWord32 RTPtimeoutMS = 1000*timeoutSeconds; |
| 3187 | const WebRtc_UWord32 RTCPtimeoutMS = 0; |
| 3188 | _rtpRtcpModule->SetPacketTimeout(RTPtimeoutMS, RTCPtimeoutMS); |
| 3189 | _rtpPacketTimeOutIsEnabled = true; |
| 3190 | _rtpTimeOutSeconds = timeoutSeconds; |
| 3191 | } |
| 3192 | else |
| 3193 | { |
| 3194 | _rtpRtcpModule->SetPacketTimeout(0, 0); |
| 3195 | _rtpPacketTimeOutIsEnabled = false; |
| 3196 | _rtpTimeOutSeconds = 0; |
| 3197 | } |
| 3198 | return 0; |
| 3199 | } |
| 3200 | |
| 3201 | WebRtc_Word32 |
| 3202 | Channel::GetPacketTimeoutNotification(bool& enabled, int& timeoutSeconds) |
| 3203 | { |
| 3204 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3205 | "Channel::GetPacketTimeoutNotification()"); |
| 3206 | enabled = _rtpPacketTimeOutIsEnabled; |
| 3207 | if (enabled) |
| 3208 | { |
| 3209 | timeoutSeconds = _rtpTimeOutSeconds; |
| 3210 | } |
| 3211 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1), |
| 3212 | "GetPacketTimeoutNotification() => enabled=%d," |
| 3213 | " timeoutSeconds=%d", |
| 3214 | enabled, timeoutSeconds); |
| 3215 | return 0; |
| 3216 | } |
| 3217 | |
| 3218 | WebRtc_Word32 |
| 3219 | Channel::RegisterDeadOrAliveObserver(VoEConnectionObserver& observer) |
| 3220 | { |
| 3221 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3222 | "Channel::RegisterDeadOrAliveObserver()"); |
| 3223 | CriticalSectionScoped cs(&_callbackCritSect); |
| 3224 | |
| 3225 | if (_connectionObserverPtr) |
| 3226 | { |
| 3227 | _engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION, kTraceError, |
| 3228 | "RegisterDeadOrAliveObserver() observer already enabled"); |
| 3229 | return -1; |
| 3230 | } |
| 3231 | |
| 3232 | _connectionObserverPtr = &observer; |
| 3233 | _connectionObserver = true; |
| 3234 | |
| 3235 | return 0; |
| 3236 | } |
| 3237 | |
| 3238 | WebRtc_Word32 |
| 3239 | Channel::DeRegisterDeadOrAliveObserver() |
| 3240 | { |
| 3241 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3242 | "Channel::DeRegisterDeadOrAliveObserver()"); |
| 3243 | CriticalSectionScoped cs(&_callbackCritSect); |
| 3244 | |
| 3245 | if (!_connectionObserverPtr) |
| 3246 | { |
| 3247 | _engineStatisticsPtr->SetLastError( |
| 3248 | VE_INVALID_OPERATION, kTraceWarning, |
| 3249 | "DeRegisterDeadOrAliveObserver() observer already disabled"); |
| 3250 | return 0; |
| 3251 | } |
| 3252 | |
| 3253 | _connectionObserver = false; |
| 3254 | _connectionObserverPtr = NULL; |
| 3255 | |
| 3256 | return 0; |
| 3257 | } |
| 3258 | |
| 3259 | WebRtc_Word32 |
| 3260 | Channel::SetPeriodicDeadOrAliveStatus(bool enable, int sampleTimeSeconds) |
| 3261 | { |
| 3262 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3263 | "Channel::SetPeriodicDeadOrAliveStatus()"); |
| 3264 | if (!_connectionObserverPtr) |
| 3265 | { |
| 3266 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3267 | "SetPeriodicDeadOrAliveStatus() connection observer has" |
| 3268 | " not been registered"); |
| 3269 | } |
| 3270 | if (enable) |
| 3271 | { |
| 3272 | ResetDeadOrAliveCounters(); |
| 3273 | } |
| 3274 | bool enabled(false); |
| 3275 | WebRtc_UWord8 currentSampleTimeSec(0); |
| 3276 | // Store last state (will be used later if dead-or-alive is disabled). |
| 3277 | _rtpRtcpModule->PeriodicDeadOrAliveStatus(enabled, currentSampleTimeSec); |
| 3278 | // Update the dead-or-alive state. |
| 3279 | if (_rtpRtcpModule->SetPeriodicDeadOrAliveStatus( |
| 3280 | enable, (WebRtc_UWord8)sampleTimeSeconds) != 0) |
| 3281 | { |
| 3282 | _engineStatisticsPtr->SetLastError( |
| 3283 | VE_RTP_RTCP_MODULE_ERROR, |
| 3284 | kTraceError, |
| 3285 | "SetPeriodicDeadOrAliveStatus() failed to set dead-or-alive " |
| 3286 | "status"); |
| 3287 | return -1; |
| 3288 | } |
| 3289 | if (!enable) |
| 3290 | { |
| 3291 | // Restore last utilized sample time. |
| 3292 | // Without this, the sample time would always be reset to default |
| 3293 | // (2 sec), each time dead-or-alived was disabled without sample-time |
| 3294 | // parameter. |
| 3295 | _rtpRtcpModule->SetPeriodicDeadOrAliveStatus(enable, |
| 3296 | currentSampleTimeSec); |
| 3297 | } |
| 3298 | return 0; |
| 3299 | } |
| 3300 | |
| 3301 | WebRtc_Word32 |
| 3302 | Channel::GetPeriodicDeadOrAliveStatus(bool& enabled, int& sampleTimeSeconds) |
| 3303 | { |
| 3304 | _rtpRtcpModule->PeriodicDeadOrAliveStatus( |
| 3305 | enabled, |
| 3306 | (WebRtc_UWord8&)sampleTimeSeconds); |
| 3307 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1), |
| 3308 | "GetPeriodicDeadOrAliveStatus() => enabled=%d," |
| 3309 | " sampleTimeSeconds=%d", |
| 3310 | enabled, sampleTimeSeconds); |
| 3311 | return 0; |
| 3312 | } |
| 3313 | |
| 3314 | WebRtc_Word32 |
| 3315 | Channel::SendUDPPacket(const void* data, |
| 3316 | unsigned int length, |
| 3317 | int& transmittedBytes, |
| 3318 | bool useRtcpSocket) |
| 3319 | { |
| 3320 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3321 | "Channel::SendUDPPacket()"); |
| 3322 | if (_externalTransport) |
| 3323 | { |
| 3324 | _engineStatisticsPtr->SetLastError( |
| 3325 | VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError, |
| 3326 | "SendUDPPacket() external transport is enabled"); |
| 3327 | return -1; |
| 3328 | } |
| 3329 | if (useRtcpSocket && !_rtpRtcpModule->RTCP()) |
| 3330 | { |
| 3331 | _engineStatisticsPtr->SetLastError( |
| 3332 | VE_RTCP_ERROR, kTraceError, |
| 3333 | "SendUDPPacket() RTCP is disabled"); |
| 3334 | return -1; |
| 3335 | } |
| 3336 | if (!_sending) |
| 3337 | { |
| 3338 | _engineStatisticsPtr->SetLastError( |
| 3339 | VE_NOT_SENDING, kTraceError, |
| 3340 | "SendUDPPacket() not sending"); |
| 3341 | return -1; |
| 3342 | } |
| 3343 | |
| 3344 | char* dataC = new char[length]; |
| 3345 | if (NULL == dataC) |
| 3346 | { |
| 3347 | _engineStatisticsPtr->SetLastError( |
| 3348 | VE_NO_MEMORY, kTraceError, |
| 3349 | "SendUDPPacket() memory allocation failed"); |
| 3350 | return -1; |
| 3351 | } |
| 3352 | memcpy(dataC, data, length); |
| 3353 | |
| 3354 | transmittedBytes = SendPacketRaw(dataC, length, useRtcpSocket); |
| 3355 | |
| 3356 | delete [] dataC; |
| 3357 | dataC = NULL; |
| 3358 | |
| 3359 | if (transmittedBytes <= 0) |
| 3360 | { |
| 3361 | _engineStatisticsPtr->SetLastError( |
| 3362 | VE_SEND_ERROR, kTraceError, |
| 3363 | "SendUDPPacket() transmission failed"); |
| 3364 | transmittedBytes = 0; |
| 3365 | return -1; |
| 3366 | } |
| 3367 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3368 | "SendUDPPacket() => transmittedBytes=%d", transmittedBytes); |
| 3369 | return 0; |
| 3370 | } |
| 3371 | |
| 3372 | |
| 3373 | int Channel::StartPlayingFileLocally(const char* fileName, |
| 3374 | const bool loop, |
| 3375 | const FileFormats format, |
| 3376 | const int startPosition, |
| 3377 | const float volumeScaling, |
| 3378 | const int stopPosition, |
| 3379 | const CodecInst* codecInst) |
| 3380 | { |
| 3381 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3382 | "Channel::StartPlayingFileLocally(fileNameUTF8[]=%s, loop=%d," |
| 3383 | " format=%d, volumeScaling=%5.3f, startPosition=%d, " |
| 3384 | "stopPosition=%d)", fileName, loop, format, volumeScaling, |
| 3385 | startPosition, stopPosition); |
| 3386 | |
| 3387 | if (_outputFilePlaying) |
| 3388 | { |
| 3389 | _engineStatisticsPtr->SetLastError( |
| 3390 | VE_ALREADY_PLAYING, kTraceError, |
| 3391 | "StartPlayingFileLocally() is already playing"); |
| 3392 | return -1; |
| 3393 | } |
| 3394 | |
| 3395 | { |
| 3396 | CriticalSectionScoped cs(&_fileCritSect); |
| 3397 | |
| 3398 | if (_outputFilePlayerPtr) |
| 3399 | { |
| 3400 | _outputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 3401 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 3402 | _outputFilePlayerPtr = NULL; |
| 3403 | } |
| 3404 | |
| 3405 | _outputFilePlayerPtr = FilePlayer::CreateFilePlayer( |
| 3406 | _outputFilePlayerId, (const FileFormats)format); |
| 3407 | |
| 3408 | if (_outputFilePlayerPtr == NULL) |
| 3409 | { |
| 3410 | _engineStatisticsPtr->SetLastError( |
| 3411 | VE_INVALID_ARGUMENT, kTraceError, |
| 3412 | "StartPlayingFileLocally() filePlayer format is not correct"); |
| 3413 | return -1; |
| 3414 | } |
| 3415 | |
| 3416 | const WebRtc_UWord32 notificationTime(0); |
| 3417 | |
| 3418 | if (_outputFilePlayerPtr->StartPlayingFile( |
| 3419 | fileName, |
| 3420 | loop, |
| 3421 | startPosition, |
| 3422 | volumeScaling, |
| 3423 | notificationTime, |
| 3424 | stopPosition, |
| 3425 | (const CodecInst*)codecInst) != 0) |
| 3426 | { |
| 3427 | _engineStatisticsPtr->SetLastError( |
| 3428 | VE_BAD_FILE, kTraceError, |
| 3429 | "StartPlayingFile() failed to start file playout"); |
| 3430 | _outputFilePlayerPtr->StopPlayingFile(); |
| 3431 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 3432 | _outputFilePlayerPtr = NULL; |
| 3433 | return -1; |
| 3434 | } |
| 3435 | _outputFilePlayerPtr->RegisterModuleFileCallback(this); |
| 3436 | _outputFilePlaying = true; |
| 3437 | } |
| 3438 | |
| 3439 | if (RegisterFilePlayingToMixer() != 0) |
| 3440 | return -1; |
| 3441 | |
| 3442 | return 0; |
| 3443 | } |
| 3444 | |
| 3445 | int Channel::StartPlayingFileLocally(InStream* stream, |
| 3446 | const FileFormats format, |
| 3447 | const int startPosition, |
| 3448 | const float volumeScaling, |
| 3449 | const int stopPosition, |
| 3450 | const CodecInst* codecInst) |
| 3451 | { |
| 3452 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3453 | "Channel::StartPlayingFileLocally(format=%d," |
| 3454 | " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)", |
| 3455 | format, volumeScaling, startPosition, stopPosition); |
| 3456 | |
| 3457 | if(stream == NULL) |
| 3458 | { |
| 3459 | _engineStatisticsPtr->SetLastError( |
| 3460 | VE_BAD_FILE, kTraceError, |
| 3461 | "StartPlayingFileLocally() NULL as input stream"); |
| 3462 | return -1; |
| 3463 | } |
| 3464 | |
| 3465 | |
| 3466 | if (_outputFilePlaying) |
| 3467 | { |
| 3468 | _engineStatisticsPtr->SetLastError( |
| 3469 | VE_ALREADY_PLAYING, kTraceError, |
| 3470 | "StartPlayingFileLocally() is already playing"); |
| 3471 | return -1; |
| 3472 | } |
| 3473 | |
| 3474 | { |
| 3475 | CriticalSectionScoped cs(&_fileCritSect); |
| 3476 | |
| 3477 | // Destroy the old instance |
| 3478 | if (_outputFilePlayerPtr) |
| 3479 | { |
| 3480 | _outputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 3481 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 3482 | _outputFilePlayerPtr = NULL; |
| 3483 | } |
| 3484 | |
| 3485 | // Create the instance |
| 3486 | _outputFilePlayerPtr = FilePlayer::CreateFilePlayer( |
| 3487 | _outputFilePlayerId, |
| 3488 | (const FileFormats)format); |
| 3489 | |
| 3490 | if (_outputFilePlayerPtr == NULL) |
| 3491 | { |
| 3492 | _engineStatisticsPtr->SetLastError( |
| 3493 | VE_INVALID_ARGUMENT, kTraceError, |
| 3494 | "StartPlayingFileLocally() filePlayer format isnot correct"); |
| 3495 | return -1; |
| 3496 | } |
| 3497 | |
| 3498 | const WebRtc_UWord32 notificationTime(0); |
| 3499 | |
| 3500 | if (_outputFilePlayerPtr->StartPlayingFile(*stream, startPosition, |
| 3501 | volumeScaling, |
| 3502 | notificationTime, |
| 3503 | stopPosition, codecInst) != 0) |
| 3504 | { |
| 3505 | _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| 3506 | "StartPlayingFile() failed to " |
| 3507 | "start file playout"); |
| 3508 | _outputFilePlayerPtr->StopPlayingFile(); |
| 3509 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 3510 | _outputFilePlayerPtr = NULL; |
| 3511 | return -1; |
| 3512 | } |
| 3513 | _outputFilePlayerPtr->RegisterModuleFileCallback(this); |
| 3514 | _outputFilePlaying = true; |
| 3515 | } |
| 3516 | |
| 3517 | if (RegisterFilePlayingToMixer() != 0) |
| 3518 | return -1; |
| 3519 | |
| 3520 | return 0; |
| 3521 | } |
| 3522 | |
| 3523 | int Channel::StopPlayingFileLocally() |
| 3524 | { |
| 3525 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3526 | "Channel::StopPlayingFileLocally()"); |
| 3527 | |
| 3528 | if (!_outputFilePlaying) |
| 3529 | { |
| 3530 | _engineStatisticsPtr->SetLastError( |
| 3531 | VE_INVALID_OPERATION, kTraceWarning, |
| 3532 | "StopPlayingFileLocally() isnot playing"); |
| 3533 | return 0; |
| 3534 | } |
| 3535 | |
| 3536 | { |
| 3537 | CriticalSectionScoped cs(&_fileCritSect); |
| 3538 | |
| 3539 | if (_outputFilePlayerPtr->StopPlayingFile() != 0) |
| 3540 | { |
| 3541 | _engineStatisticsPtr->SetLastError( |
| 3542 | VE_STOP_RECORDING_FAILED, kTraceError, |
| 3543 | "StopPlayingFile() could not stop playing"); |
| 3544 | return -1; |
| 3545 | } |
| 3546 | _outputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 3547 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 3548 | _outputFilePlayerPtr = NULL; |
| 3549 | _outputFilePlaying = false; |
| 3550 | } |
| 3551 | // _fileCritSect cannot be taken while calling |
| 3552 | // SetAnonymousMixibilityStatus. Refer to comments in |
| 3553 | // StartPlayingFileLocally(const char* ...) for more details. |
| 3554 | if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, false) != 0) |
| 3555 | { |
| 3556 | _engineStatisticsPtr->SetLastError( |
| 3557 | VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError, |
| 3558 | "StopPlayingFile() failed to stop participant from playing as" |
| 3559 | "file in the mixer"); |
| 3560 | return -1; |
| 3561 | } |
| 3562 | |
| 3563 | return 0; |
| 3564 | } |
| 3565 | |
| 3566 | int Channel::IsPlayingFileLocally() const |
| 3567 | { |
| 3568 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3569 | "Channel::IsPlayingFileLocally()"); |
| 3570 | |
| 3571 | return (WebRtc_Word32)_outputFilePlaying; |
| 3572 | } |
| 3573 | |
| 3574 | int Channel::RegisterFilePlayingToMixer() |
| 3575 | { |
| 3576 | // Return success for not registering for file playing to mixer if: |
| 3577 | // 1. playing file before playout is started on that channel. |
| 3578 | // 2. starting playout without file playing on that channel. |
| 3579 | if (!_playing || !_outputFilePlaying) |
| 3580 | { |
| 3581 | return 0; |
| 3582 | } |
| 3583 | |
| 3584 | // |_fileCritSect| cannot be taken while calling |
| 3585 | // SetAnonymousMixabilityStatus() since as soon as the participant is added |
| 3586 | // frames can be pulled by the mixer. Since the frames are generated from |
| 3587 | // the file, _fileCritSect will be taken. This would result in a deadlock. |
| 3588 | if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0) |
| 3589 | { |
| 3590 | CriticalSectionScoped cs(&_fileCritSect); |
| 3591 | _outputFilePlaying = false; |
| 3592 | _engineStatisticsPtr->SetLastError( |
| 3593 | VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError, |
| 3594 | "StartPlayingFile() failed to add participant as file to mixer"); |
| 3595 | _outputFilePlayerPtr->StopPlayingFile(); |
| 3596 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 3597 | _outputFilePlayerPtr = NULL; |
| 3598 | return -1; |
| 3599 | } |
| 3600 | |
| 3601 | return 0; |
| 3602 | } |
| 3603 | |
| 3604 | int Channel::ScaleLocalFilePlayout(const float scale) |
| 3605 | { |
| 3606 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3607 | "Channel::ScaleLocalFilePlayout(scale=%5.3f)", scale); |
| 3608 | |
| 3609 | CriticalSectionScoped cs(&_fileCritSect); |
| 3610 | |
| 3611 | if (!_outputFilePlaying) |
| 3612 | { |
| 3613 | _engineStatisticsPtr->SetLastError( |
| 3614 | VE_INVALID_OPERATION, kTraceError, |
| 3615 | "ScaleLocalFilePlayout() isnot playing"); |
| 3616 | return -1; |
| 3617 | } |
| 3618 | if ((_outputFilePlayerPtr == NULL) || |
| 3619 | (_outputFilePlayerPtr->SetAudioScaling(scale) != 0)) |
| 3620 | { |
| 3621 | _engineStatisticsPtr->SetLastError( |
| 3622 | VE_BAD_ARGUMENT, kTraceError, |
| 3623 | "SetAudioScaling() failed to scale the playout"); |
| 3624 | return -1; |
| 3625 | } |
| 3626 | |
| 3627 | return 0; |
| 3628 | } |
| 3629 | |
| 3630 | int Channel::GetLocalPlayoutPosition(int& positionMs) |
| 3631 | { |
| 3632 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3633 | "Channel::GetLocalPlayoutPosition(position=?)"); |
| 3634 | |
| 3635 | WebRtc_UWord32 position; |
| 3636 | |
| 3637 | CriticalSectionScoped cs(&_fileCritSect); |
| 3638 | |
| 3639 | if (_outputFilePlayerPtr == NULL) |
| 3640 | { |
| 3641 | _engineStatisticsPtr->SetLastError( |
| 3642 | VE_INVALID_OPERATION, kTraceError, |
| 3643 | "GetLocalPlayoutPosition() filePlayer instance doesnot exist"); |
| 3644 | return -1; |
| 3645 | } |
| 3646 | |
| 3647 | if (_outputFilePlayerPtr->GetPlayoutPosition(position) != 0) |
| 3648 | { |
| 3649 | _engineStatisticsPtr->SetLastError( |
| 3650 | VE_BAD_FILE, kTraceError, |
| 3651 | "GetLocalPlayoutPosition() failed"); |
| 3652 | return -1; |
| 3653 | } |
| 3654 | positionMs = position; |
| 3655 | |
| 3656 | return 0; |
| 3657 | } |
| 3658 | |
| 3659 | int Channel::StartPlayingFileAsMicrophone(const char* fileName, |
| 3660 | const bool loop, |
| 3661 | const FileFormats format, |
| 3662 | const int startPosition, |
| 3663 | const float volumeScaling, |
| 3664 | const int stopPosition, |
| 3665 | const CodecInst* codecInst) |
| 3666 | { |
| 3667 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3668 | "Channel::StartPlayingFileAsMicrophone(fileNameUTF8[]=%s, " |
| 3669 | "loop=%d, format=%d, volumeScaling=%5.3f, startPosition=%d, " |
| 3670 | "stopPosition=%d)", fileName, loop, format, volumeScaling, |
| 3671 | startPosition, stopPosition); |
| 3672 | |
| 3673 | if (_inputFilePlaying) |
| 3674 | { |
| 3675 | _engineStatisticsPtr->SetLastError( |
| 3676 | VE_ALREADY_PLAYING, kTraceWarning, |
| 3677 | "StartPlayingFileAsMicrophone() filePlayer is playing"); |
| 3678 | return 0; |
| 3679 | } |
| 3680 | |
| 3681 | CriticalSectionScoped cs(&_fileCritSect); |
| 3682 | |
| 3683 | // Destroy the old instance |
| 3684 | if (_inputFilePlayerPtr) |
| 3685 | { |
| 3686 | _inputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 3687 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 3688 | _inputFilePlayerPtr = NULL; |
| 3689 | } |
| 3690 | |
| 3691 | // Create the instance |
| 3692 | _inputFilePlayerPtr = FilePlayer::CreateFilePlayer( |
| 3693 | _inputFilePlayerId, (const FileFormats)format); |
| 3694 | |
| 3695 | if (_inputFilePlayerPtr == NULL) |
| 3696 | { |
| 3697 | _engineStatisticsPtr->SetLastError( |
| 3698 | VE_INVALID_ARGUMENT, kTraceError, |
| 3699 | "StartPlayingFileAsMicrophone() filePlayer format isnot correct"); |
| 3700 | return -1; |
| 3701 | } |
| 3702 | |
| 3703 | const WebRtc_UWord32 notificationTime(0); |
| 3704 | |
| 3705 | if (_inputFilePlayerPtr->StartPlayingFile( |
| 3706 | fileName, |
| 3707 | loop, |
| 3708 | startPosition, |
| 3709 | volumeScaling, |
| 3710 | notificationTime, |
| 3711 | stopPosition, |
| 3712 | (const CodecInst*)codecInst) != 0) |
| 3713 | { |
| 3714 | _engineStatisticsPtr->SetLastError( |
| 3715 | VE_BAD_FILE, kTraceError, |
| 3716 | "StartPlayingFile() failed to start file playout"); |
| 3717 | _inputFilePlayerPtr->StopPlayingFile(); |
| 3718 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 3719 | _inputFilePlayerPtr = NULL; |
| 3720 | return -1; |
| 3721 | } |
| 3722 | _inputFilePlayerPtr->RegisterModuleFileCallback(this); |
| 3723 | _inputFilePlaying = true; |
| 3724 | |
| 3725 | return 0; |
| 3726 | } |
| 3727 | |
| 3728 | int Channel::StartPlayingFileAsMicrophone(InStream* stream, |
| 3729 | const FileFormats format, |
| 3730 | const int startPosition, |
| 3731 | const float volumeScaling, |
| 3732 | const int stopPosition, |
| 3733 | const CodecInst* codecInst) |
| 3734 | { |
| 3735 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3736 | "Channel::StartPlayingFileAsMicrophone(format=%d, " |
| 3737 | "volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)", |
| 3738 | format, volumeScaling, startPosition, stopPosition); |
| 3739 | |
| 3740 | if(stream == NULL) |
| 3741 | { |
| 3742 | _engineStatisticsPtr->SetLastError( |
| 3743 | VE_BAD_FILE, kTraceError, |
| 3744 | "StartPlayingFileAsMicrophone NULL as input stream"); |
| 3745 | return -1; |
| 3746 | } |
| 3747 | |
| 3748 | if (_inputFilePlaying) |
| 3749 | { |
| 3750 | _engineStatisticsPtr->SetLastError( |
| 3751 | VE_ALREADY_PLAYING, kTraceWarning, |
| 3752 | "StartPlayingFileAsMicrophone() is playing"); |
| 3753 | return 0; |
| 3754 | } |
| 3755 | |
| 3756 | CriticalSectionScoped cs(&_fileCritSect); |
| 3757 | |
| 3758 | // Destroy the old instance |
| 3759 | if (_inputFilePlayerPtr) |
| 3760 | { |
| 3761 | _inputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 3762 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 3763 | _inputFilePlayerPtr = NULL; |
| 3764 | } |
| 3765 | |
| 3766 | // Create the instance |
| 3767 | _inputFilePlayerPtr = FilePlayer::CreateFilePlayer( |
| 3768 | _inputFilePlayerId, (const FileFormats)format); |
| 3769 | |
| 3770 | if (_inputFilePlayerPtr == NULL) |
| 3771 | { |
| 3772 | _engineStatisticsPtr->SetLastError( |
| 3773 | VE_INVALID_ARGUMENT, kTraceError, |
| 3774 | "StartPlayingInputFile() filePlayer format isnot correct"); |
| 3775 | return -1; |
| 3776 | } |
| 3777 | |
| 3778 | const WebRtc_UWord32 notificationTime(0); |
| 3779 | |
| 3780 | if (_inputFilePlayerPtr->StartPlayingFile(*stream, startPosition, |
| 3781 | volumeScaling, notificationTime, |
| 3782 | stopPosition, codecInst) != 0) |
| 3783 | { |
| 3784 | _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| 3785 | "StartPlayingFile() failed to start " |
| 3786 | "file playout"); |
| 3787 | _inputFilePlayerPtr->StopPlayingFile(); |
| 3788 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 3789 | _inputFilePlayerPtr = NULL; |
| 3790 | return -1; |
| 3791 | } |
andrew@webrtc.org | d468236 | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 3792 | |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 3793 | _inputFilePlayerPtr->RegisterModuleFileCallback(this); |
| 3794 | _inputFilePlaying = true; |
| 3795 | |
| 3796 | return 0; |
| 3797 | } |
| 3798 | |
| 3799 | int Channel::StopPlayingFileAsMicrophone() |
| 3800 | { |
| 3801 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3802 | "Channel::StopPlayingFileAsMicrophone()"); |
| 3803 | |
| 3804 | if (!_inputFilePlaying) |
| 3805 | { |
| 3806 | _engineStatisticsPtr->SetLastError( |
| 3807 | VE_INVALID_OPERATION, kTraceWarning, |
| 3808 | "StopPlayingFileAsMicrophone() isnot playing"); |
| 3809 | return 0; |
| 3810 | } |
| 3811 | |
| 3812 | CriticalSectionScoped cs(&_fileCritSect); |
| 3813 | if (_inputFilePlayerPtr->StopPlayingFile() != 0) |
| 3814 | { |
| 3815 | _engineStatisticsPtr->SetLastError( |
| 3816 | VE_STOP_RECORDING_FAILED, kTraceError, |
| 3817 | "StopPlayingFile() could not stop playing"); |
| 3818 | return -1; |
| 3819 | } |
| 3820 | _inputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 3821 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 3822 | _inputFilePlayerPtr = NULL; |
| 3823 | _inputFilePlaying = false; |
| 3824 | |
| 3825 | return 0; |
| 3826 | } |
| 3827 | |
| 3828 | int Channel::IsPlayingFileAsMicrophone() const |
| 3829 | { |
| 3830 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3831 | "Channel::IsPlayingFileAsMicrophone()"); |
| 3832 | |
| 3833 | return _inputFilePlaying; |
| 3834 | } |
| 3835 | |
| 3836 | int Channel::ScaleFileAsMicrophonePlayout(const float scale) |
| 3837 | { |
| 3838 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3839 | "Channel::ScaleFileAsMicrophonePlayout(scale=%5.3f)", scale); |
| 3840 | |
| 3841 | CriticalSectionScoped cs(&_fileCritSect); |
| 3842 | |
| 3843 | if (!_inputFilePlaying) |
| 3844 | { |
| 3845 | _engineStatisticsPtr->SetLastError( |
| 3846 | VE_INVALID_OPERATION, kTraceError, |
| 3847 | "ScaleFileAsMicrophonePlayout() isnot playing"); |
| 3848 | return -1; |
| 3849 | } |
| 3850 | |
| 3851 | if ((_inputFilePlayerPtr == NULL) || |
| 3852 | (_inputFilePlayerPtr->SetAudioScaling(scale) != 0)) |
| 3853 | { |
| 3854 | _engineStatisticsPtr->SetLastError( |
| 3855 | VE_BAD_ARGUMENT, kTraceError, |
| 3856 | "SetAudioScaling() failed to scale playout"); |
| 3857 | return -1; |
| 3858 | } |
| 3859 | |
| 3860 | return 0; |
| 3861 | } |
| 3862 | |
| 3863 | int Channel::StartRecordingPlayout(const char* fileName, |
| 3864 | const CodecInst* codecInst) |
| 3865 | { |
| 3866 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3867 | "Channel::StartRecordingPlayout(fileName=%s)", fileName); |
| 3868 | |
| 3869 | if (_outputFileRecording) |
| 3870 | { |
| 3871 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1), |
| 3872 | "StartRecordingPlayout() is already recording"); |
| 3873 | return 0; |
| 3874 | } |
| 3875 | |
| 3876 | FileFormats format; |
| 3877 | const WebRtc_UWord32 notificationTime(0); // Not supported in VoE |
| 3878 | CodecInst dummyCodec={100,"L16",16000,320,1,320000}; |
| 3879 | |
| 3880 | if ((codecInst != NULL) && |
| 3881 | ((codecInst->channels < 1) || (codecInst->channels > 2))) |
| 3882 | { |
| 3883 | _engineStatisticsPtr->SetLastError( |
| 3884 | VE_BAD_ARGUMENT, kTraceError, |
| 3885 | "StartRecordingPlayout() invalid compression"); |
| 3886 | return(-1); |
| 3887 | } |
| 3888 | if(codecInst == NULL) |
| 3889 | { |
| 3890 | format = kFileFormatPcm16kHzFile; |
| 3891 | codecInst=&dummyCodec; |
| 3892 | } |
| 3893 | else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| 3894 | (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| 3895 | (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| 3896 | { |
| 3897 | format = kFileFormatWavFile; |
| 3898 | } |
| 3899 | else |
| 3900 | { |
| 3901 | format = kFileFormatCompressedFile; |
| 3902 | } |
| 3903 | |
| 3904 | CriticalSectionScoped cs(&_fileCritSect); |
| 3905 | |
| 3906 | // Destroy the old instance |
| 3907 | if (_outputFileRecorderPtr) |
| 3908 | { |
| 3909 | _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 3910 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 3911 | _outputFileRecorderPtr = NULL; |
| 3912 | } |
| 3913 | |
| 3914 | _outputFileRecorderPtr = FileRecorder::CreateFileRecorder( |
| 3915 | _outputFileRecorderId, (const FileFormats)format); |
| 3916 | if (_outputFileRecorderPtr == NULL) |
| 3917 | { |
| 3918 | _engineStatisticsPtr->SetLastError( |
| 3919 | VE_INVALID_ARGUMENT, kTraceError, |
| 3920 | "StartRecordingPlayout() fileRecorder format isnot correct"); |
| 3921 | return -1; |
| 3922 | } |
| 3923 | |
| 3924 | if (_outputFileRecorderPtr->StartRecordingAudioFile( |
| 3925 | fileName, (const CodecInst&)*codecInst, notificationTime) != 0) |
| 3926 | { |
| 3927 | _engineStatisticsPtr->SetLastError( |
| 3928 | VE_BAD_FILE, kTraceError, |
| 3929 | "StartRecordingAudioFile() failed to start file recording"); |
| 3930 | _outputFileRecorderPtr->StopRecording(); |
| 3931 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 3932 | _outputFileRecorderPtr = NULL; |
| 3933 | return -1; |
| 3934 | } |
| 3935 | _outputFileRecorderPtr->RegisterModuleFileCallback(this); |
| 3936 | _outputFileRecording = true; |
| 3937 | |
| 3938 | return 0; |
| 3939 | } |
| 3940 | |
| 3941 | int Channel::StartRecordingPlayout(OutStream* stream, |
| 3942 | const CodecInst* codecInst) |
| 3943 | { |
| 3944 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3945 | "Channel::StartRecordingPlayout()"); |
| 3946 | |
| 3947 | if (_outputFileRecording) |
| 3948 | { |
| 3949 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1), |
| 3950 | "StartRecordingPlayout() is already recording"); |
| 3951 | return 0; |
| 3952 | } |
| 3953 | |
| 3954 | FileFormats format; |
| 3955 | const WebRtc_UWord32 notificationTime(0); // Not supported in VoE |
| 3956 | CodecInst dummyCodec={100,"L16",16000,320,1,320000}; |
| 3957 | |
| 3958 | if (codecInst != NULL && codecInst->channels != 1) |
| 3959 | { |
| 3960 | _engineStatisticsPtr->SetLastError( |
| 3961 | VE_BAD_ARGUMENT, kTraceError, |
| 3962 | "StartRecordingPlayout() invalid compression"); |
| 3963 | return(-1); |
| 3964 | } |
| 3965 | if(codecInst == NULL) |
| 3966 | { |
| 3967 | format = kFileFormatPcm16kHzFile; |
| 3968 | codecInst=&dummyCodec; |
| 3969 | } |
| 3970 | else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| 3971 | (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| 3972 | (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| 3973 | { |
| 3974 | format = kFileFormatWavFile; |
| 3975 | } |
| 3976 | else |
| 3977 | { |
| 3978 | format = kFileFormatCompressedFile; |
| 3979 | } |
| 3980 | |
| 3981 | CriticalSectionScoped cs(&_fileCritSect); |
| 3982 | |
| 3983 | // Destroy the old instance |
| 3984 | if (_outputFileRecorderPtr) |
| 3985 | { |
| 3986 | _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 3987 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 3988 | _outputFileRecorderPtr = NULL; |
| 3989 | } |
| 3990 | |
| 3991 | _outputFileRecorderPtr = FileRecorder::CreateFileRecorder( |
| 3992 | _outputFileRecorderId, (const FileFormats)format); |
| 3993 | if (_outputFileRecorderPtr == NULL) |
| 3994 | { |
| 3995 | _engineStatisticsPtr->SetLastError( |
| 3996 | VE_INVALID_ARGUMENT, kTraceError, |
| 3997 | "StartRecordingPlayout() fileRecorder format isnot correct"); |
| 3998 | return -1; |
| 3999 | } |
| 4000 | |
| 4001 | if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, *codecInst, |
| 4002 | notificationTime) != 0) |
| 4003 | { |
| 4004 | _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| 4005 | "StartRecordingPlayout() failed to " |
| 4006 | "start file recording"); |
| 4007 | _outputFileRecorderPtr->StopRecording(); |
| 4008 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 4009 | _outputFileRecorderPtr = NULL; |
| 4010 | return -1; |
| 4011 | } |
andrew@webrtc.org | d468236 | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 4012 | |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 4013 | _outputFileRecorderPtr->RegisterModuleFileCallback(this); |
| 4014 | _outputFileRecording = true; |
| 4015 | |
| 4016 | return 0; |
| 4017 | } |
| 4018 | |
| 4019 | int Channel::StopRecordingPlayout() |
| 4020 | { |
| 4021 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| 4022 | "Channel::StopRecordingPlayout()"); |
| 4023 | |
| 4024 | if (!_outputFileRecording) |
| 4025 | { |
| 4026 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1), |
| 4027 | "StopRecordingPlayout() isnot recording"); |
| 4028 | return -1; |
| 4029 | } |
| 4030 | |
| 4031 | |
| 4032 | CriticalSectionScoped cs(&_fileCritSect); |
| 4033 | |
| 4034 | if (_outputFileRecorderPtr->StopRecording() != 0) |
| 4035 | { |
| 4036 | _engineStatisticsPtr->SetLastError( |
| 4037 | VE_STOP_RECORDING_FAILED, kTraceError, |
| 4038 | "StopRecording() could not stop recording"); |
| 4039 | return(-1); |
| 4040 | } |
| 4041 | _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 4042 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 4043 | _outputFileRecorderPtr = NULL; |
| 4044 | _outputFileRecording = false; |
| 4045 | |
| 4046 | return 0; |
| 4047 | } |
| 4048 | |
| 4049 | void |
| 4050 | Channel::SetMixWithMicStatus(bool mix) |
| 4051 | { |
| 4052 | _mixFileWithMicrophone=mix; |
| 4053 | } |
| 4054 | |
| 4055 | int |
| 4056 | Channel::GetSpeechOutputLevel(WebRtc_UWord32& level) const |
| 4057 | { |
| 4058 | WebRtc_Word8 currentLevel = _outputAudioLevel.Level(); |
| 4059 | level = static_cast<WebRtc_Word32> (currentLevel); |
| 4060 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4061 | VoEId(_instanceId,_channelId), |
| 4062 | "GetSpeechOutputLevel() => level=%u", level); |
| 4063 | return 0; |
| 4064 | } |
| 4065 | |
| 4066 | int |
| 4067 | Channel::GetSpeechOutputLevelFullRange(WebRtc_UWord32& level) const |
| 4068 | { |
| 4069 | WebRtc_Word16 currentLevel = _outputAudioLevel.LevelFullRange(); |
| 4070 | level = static_cast<WebRtc_Word32> (currentLevel); |
| 4071 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4072 | VoEId(_instanceId,_channelId), |
| 4073 | "GetSpeechOutputLevelFullRange() => level=%u", level); |
| 4074 | return 0; |
| 4075 | } |
| 4076 | |
| 4077 | int |
| 4078 | Channel::SetMute(bool enable) |
| 4079 | { |
| 4080 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4081 | "Channel::SetMute(enable=%d)", enable); |
| 4082 | _mute = enable; |
| 4083 | return 0; |
| 4084 | } |
| 4085 | |
| 4086 | bool |
| 4087 | Channel::Mute() const |
| 4088 | { |
| 4089 | return _mute; |
| 4090 | } |
| 4091 | |
| 4092 | int |
| 4093 | Channel::SetOutputVolumePan(float left, float right) |
| 4094 | { |
| 4095 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4096 | "Channel::SetOutputVolumePan()"); |
| 4097 | _panLeft = left; |
| 4098 | _panRight = right; |
| 4099 | return 0; |
| 4100 | } |
| 4101 | |
| 4102 | int |
| 4103 | Channel::GetOutputVolumePan(float& left, float& right) const |
| 4104 | { |
| 4105 | left = _panLeft; |
| 4106 | right = _panRight; |
| 4107 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4108 | VoEId(_instanceId,_channelId), |
| 4109 | "GetOutputVolumePan() => left=%3.2f, right=%3.2f", left, right); |
| 4110 | return 0; |
| 4111 | } |
| 4112 | |
| 4113 | int |
| 4114 | Channel::SetChannelOutputVolumeScaling(float scaling) |
| 4115 | { |
| 4116 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4117 | "Channel::SetChannelOutputVolumeScaling()"); |
| 4118 | _outputGain = scaling; |
| 4119 | return 0; |
| 4120 | } |
| 4121 | |
| 4122 | int |
| 4123 | Channel::GetChannelOutputVolumeScaling(float& scaling) const |
| 4124 | { |
| 4125 | scaling = _outputGain; |
| 4126 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4127 | VoEId(_instanceId,_channelId), |
| 4128 | "GetChannelOutputVolumeScaling() => scaling=%3.2f", scaling); |
| 4129 | return 0; |
| 4130 | } |
| 4131 | |
| 4132 | #ifdef WEBRTC_SRTP |
| 4133 | |
| 4134 | int |
| 4135 | Channel::EnableSRTPSend( |
| 4136 | CipherTypes cipherType, |
| 4137 | int cipherKeyLength, |
| 4138 | AuthenticationTypes authType, |
| 4139 | int authKeyLength, |
| 4140 | int authTagLength, |
| 4141 | SecurityLevels level, |
| 4142 | const unsigned char key[kVoiceEngineMaxSrtpKeyLength], |
| 4143 | bool useForRTCP) |
| 4144 | { |
| 4145 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4146 | "Channel::EnableSRTPSend()"); |
| 4147 | |
| 4148 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4149 | |
| 4150 | if (_encrypting) |
| 4151 | { |
| 4152 | _engineStatisticsPtr->SetLastError( |
| 4153 | VE_INVALID_OPERATION, kTraceWarning, |
| 4154 | "EnableSRTPSend() encryption already enabled"); |
| 4155 | return -1; |
| 4156 | } |
| 4157 | |
| 4158 | if (key == NULL) |
| 4159 | { |
| 4160 | _engineStatisticsPtr->SetLastError( |
| 4161 | VE_INVALID_ARGUMENT, kTraceWarning, |
| 4162 | "EnableSRTPSend() invalid key string"); |
| 4163 | return -1; |
| 4164 | } |
| 4165 | |
| 4166 | if (((kEncryption == level || |
| 4167 | kEncryptionAndAuthentication == level) && |
| 4168 | (cipherKeyLength < kVoiceEngineMinSrtpEncryptLength || |
| 4169 | cipherKeyLength > kVoiceEngineMaxSrtpEncryptLength)) || |
| 4170 | ((kAuthentication == level || |
| 4171 | kEncryptionAndAuthentication == level) && |
| 4172 | kAuthHmacSha1 == authType && |
| 4173 | (authKeyLength > kVoiceEngineMaxSrtpAuthSha1Length || |
| 4174 | authTagLength > kVoiceEngineMaxSrtpAuthSha1Length)) || |
| 4175 | ((kAuthentication == level || |
| 4176 | kEncryptionAndAuthentication == level) && |
| 4177 | kAuthNull == authType && |
| 4178 | (authKeyLength > kVoiceEngineMaxSrtpKeyAuthNullLength || |
| 4179 | authTagLength > kVoiceEngineMaxSrtpTagAuthNullLength))) |
| 4180 | { |
| 4181 | _engineStatisticsPtr->SetLastError( |
| 4182 | VE_INVALID_ARGUMENT, kTraceError, |
| 4183 | "EnableSRTPSend() invalid key length(s)"); |
| 4184 | return -1; |
| 4185 | } |
| 4186 | |
| 4187 | |
| 4188 | if (_srtpModule.EnableSRTPEncrypt( |
| 4189 | !useForRTCP, |
| 4190 | (SrtpModule::CipherTypes)cipherType, |
| 4191 | cipherKeyLength, |
| 4192 | (SrtpModule::AuthenticationTypes)authType, |
| 4193 | authKeyLength, authTagLength, |
| 4194 | (SrtpModule::SecurityLevels)level, |
| 4195 | key) == -1) |
| 4196 | { |
| 4197 | _engineStatisticsPtr->SetLastError( |
| 4198 | VE_SRTP_ERROR, kTraceError, |
| 4199 | "EnableSRTPSend() failed to enable SRTP encryption"); |
| 4200 | return -1; |
| 4201 | } |
| 4202 | |
| 4203 | if (_encryptionPtr == NULL) |
| 4204 | { |
| 4205 | _encryptionPtr = &_srtpModule; |
| 4206 | } |
| 4207 | _encrypting = true; |
| 4208 | |
| 4209 | return 0; |
| 4210 | } |
| 4211 | |
| 4212 | int |
| 4213 | Channel::DisableSRTPSend() |
| 4214 | { |
| 4215 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4216 | "Channel::DisableSRTPSend()"); |
| 4217 | |
| 4218 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4219 | |
| 4220 | if (!_encrypting) |
| 4221 | { |
| 4222 | _engineStatisticsPtr->SetLastError( |
| 4223 | VE_INVALID_OPERATION, kTraceWarning, |
| 4224 | "DisableSRTPSend() SRTP encryption already disabled"); |
| 4225 | return 0; |
| 4226 | } |
| 4227 | |
| 4228 | _encrypting = false; |
| 4229 | |
| 4230 | if (_srtpModule.DisableSRTPEncrypt() == -1) |
| 4231 | { |
| 4232 | _engineStatisticsPtr->SetLastError( |
| 4233 | VE_SRTP_ERROR, kTraceError, |
| 4234 | "DisableSRTPSend() failed to disable SRTP encryption"); |
| 4235 | return -1; |
| 4236 | } |
| 4237 | |
| 4238 | if (!_srtpModule.SRTPDecrypt() && !_srtpModule.SRTPEncrypt()) |
| 4239 | { |
| 4240 | // Both directions are disabled |
| 4241 | _encryptionPtr = NULL; |
| 4242 | } |
| 4243 | |
| 4244 | return 0; |
| 4245 | } |
| 4246 | |
| 4247 | int |
| 4248 | Channel::EnableSRTPReceive( |
| 4249 | CipherTypes cipherType, |
| 4250 | int cipherKeyLength, |
| 4251 | AuthenticationTypes authType, |
| 4252 | int authKeyLength, |
| 4253 | int authTagLength, |
| 4254 | SecurityLevels level, |
| 4255 | const unsigned char key[kVoiceEngineMaxSrtpKeyLength], |
| 4256 | bool useForRTCP) |
| 4257 | { |
| 4258 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4259 | "Channel::EnableSRTPReceive()"); |
| 4260 | |
| 4261 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4262 | |
| 4263 | if (_decrypting) |
| 4264 | { |
| 4265 | _engineStatisticsPtr->SetLastError( |
| 4266 | VE_INVALID_OPERATION, kTraceWarning, |
| 4267 | "EnableSRTPReceive() SRTP decryption already enabled"); |
| 4268 | return -1; |
| 4269 | } |
| 4270 | |
| 4271 | if (key == NULL) |
| 4272 | { |
| 4273 | _engineStatisticsPtr->SetLastError( |
| 4274 | VE_INVALID_ARGUMENT, kTraceWarning, |
| 4275 | "EnableSRTPReceive() invalid key string"); |
| 4276 | return -1; |
| 4277 | } |
| 4278 | |
| 4279 | if ((((kEncryption == level) || |
| 4280 | (kEncryptionAndAuthentication == level)) && |
| 4281 | ((cipherKeyLength < kVoiceEngineMinSrtpEncryptLength) || |
| 4282 | (cipherKeyLength > kVoiceEngineMaxSrtpEncryptLength))) || |
| 4283 | (((kAuthentication == level) || |
| 4284 | (kEncryptionAndAuthentication == level)) && |
| 4285 | (kAuthHmacSha1 == authType) && |
| 4286 | ((authKeyLength > kVoiceEngineMaxSrtpAuthSha1Length) || |
| 4287 | (authTagLength > kVoiceEngineMaxSrtpAuthSha1Length))) || |
| 4288 | (((kAuthentication == level) || |
| 4289 | (kEncryptionAndAuthentication == level)) && |
| 4290 | (kAuthNull == authType) && |
| 4291 | ((authKeyLength > kVoiceEngineMaxSrtpKeyAuthNullLength) || |
| 4292 | (authTagLength > kVoiceEngineMaxSrtpTagAuthNullLength)))) |
| 4293 | { |
| 4294 | _engineStatisticsPtr->SetLastError( |
| 4295 | VE_INVALID_ARGUMENT, kTraceError, |
| 4296 | "EnableSRTPReceive() invalid key length(s)"); |
| 4297 | return -1; |
| 4298 | } |
| 4299 | |
| 4300 | if (_srtpModule.EnableSRTPDecrypt( |
| 4301 | !useForRTCP, |
| 4302 | (SrtpModule::CipherTypes)cipherType, |
| 4303 | cipherKeyLength, |
| 4304 | (SrtpModule::AuthenticationTypes)authType, |
| 4305 | authKeyLength, |
| 4306 | authTagLength, |
| 4307 | (SrtpModule::SecurityLevels)level, |
| 4308 | key) == -1) |
| 4309 | { |
| 4310 | _engineStatisticsPtr->SetLastError( |
| 4311 | VE_SRTP_ERROR, kTraceError, |
| 4312 | "EnableSRTPReceive() failed to enable SRTP decryption"); |
| 4313 | return -1; |
| 4314 | } |
| 4315 | |
| 4316 | if (_encryptionPtr == NULL) |
| 4317 | { |
| 4318 | _encryptionPtr = &_srtpModule; |
| 4319 | } |
| 4320 | |
| 4321 | _decrypting = true; |
| 4322 | |
| 4323 | return 0; |
| 4324 | } |
| 4325 | |
| 4326 | int |
| 4327 | Channel::DisableSRTPReceive() |
| 4328 | { |
| 4329 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4330 | "Channel::DisableSRTPReceive()"); |
| 4331 | |
| 4332 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4333 | |
| 4334 | if (!_decrypting) |
| 4335 | { |
| 4336 | _engineStatisticsPtr->SetLastError( |
| 4337 | VE_INVALID_OPERATION, kTraceWarning, |
| 4338 | "DisableSRTPReceive() SRTP decryption already disabled"); |
| 4339 | return 0; |
| 4340 | } |
| 4341 | |
| 4342 | _decrypting = false; |
| 4343 | |
| 4344 | if (_srtpModule.DisableSRTPDecrypt() == -1) |
| 4345 | { |
| 4346 | _engineStatisticsPtr->SetLastError( |
| 4347 | VE_SRTP_ERROR, kTraceError, |
| 4348 | "DisableSRTPReceive() failed to disable SRTP decryption"); |
| 4349 | return -1; |
| 4350 | } |
| 4351 | |
| 4352 | if (!_srtpModule.SRTPDecrypt() && !_srtpModule.SRTPEncrypt()) |
| 4353 | { |
| 4354 | _encryptionPtr = NULL; |
| 4355 | } |
| 4356 | |
| 4357 | return 0; |
| 4358 | } |
| 4359 | |
| 4360 | #endif |
| 4361 | |
| 4362 | int |
| 4363 | Channel::RegisterExternalEncryption(Encryption& encryption) |
| 4364 | { |
| 4365 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4366 | "Channel::RegisterExternalEncryption()"); |
| 4367 | |
| 4368 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4369 | |
| 4370 | if (_encryptionPtr) |
| 4371 | { |
| 4372 | _engineStatisticsPtr->SetLastError( |
| 4373 | VE_INVALID_OPERATION, kTraceError, |
| 4374 | "RegisterExternalEncryption() encryption already enabled"); |
| 4375 | return -1; |
| 4376 | } |
| 4377 | |
| 4378 | _encryptionPtr = &encryption; |
| 4379 | |
| 4380 | _decrypting = true; |
| 4381 | _encrypting = true; |
| 4382 | |
| 4383 | return 0; |
| 4384 | } |
| 4385 | |
| 4386 | int |
| 4387 | Channel::DeRegisterExternalEncryption() |
| 4388 | { |
| 4389 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4390 | "Channel::DeRegisterExternalEncryption()"); |
| 4391 | |
| 4392 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4393 | |
| 4394 | if (!_encryptionPtr) |
| 4395 | { |
| 4396 | _engineStatisticsPtr->SetLastError( |
| 4397 | VE_INVALID_OPERATION, kTraceWarning, |
| 4398 | "DeRegisterExternalEncryption() encryption already disabled"); |
| 4399 | return 0; |
| 4400 | } |
| 4401 | |
| 4402 | _decrypting = false; |
| 4403 | _encrypting = false; |
| 4404 | |
| 4405 | _encryptionPtr = NULL; |
| 4406 | |
| 4407 | return 0; |
| 4408 | } |
| 4409 | |
| 4410 | int Channel::SendTelephoneEventOutband(unsigned char eventCode, |
| 4411 | int lengthMs, int attenuationDb, |
| 4412 | bool playDtmfEvent) |
| 4413 | { |
| 4414 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 4415 | "Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)", |
| 4416 | playDtmfEvent); |
| 4417 | |
| 4418 | _playOutbandDtmfEvent = playDtmfEvent; |
| 4419 | |
| 4420 | if (_rtpRtcpModule->SendTelephoneEventOutband(eventCode, lengthMs, |
| 4421 | attenuationDb) != 0) |
| 4422 | { |
| 4423 | _engineStatisticsPtr->SetLastError( |
| 4424 | VE_SEND_DTMF_FAILED, |
| 4425 | kTraceWarning, |
| 4426 | "SendTelephoneEventOutband() failed to send event"); |
| 4427 | return -1; |
| 4428 | } |
| 4429 | return 0; |
| 4430 | } |
| 4431 | |
| 4432 | int Channel::SendTelephoneEventInband(unsigned char eventCode, |
| 4433 | int lengthMs, |
| 4434 | int attenuationDb, |
| 4435 | bool playDtmfEvent) |
| 4436 | { |
| 4437 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 4438 | "Channel::SendTelephoneEventInband(..., playDtmfEvent=%d)", |
| 4439 | playDtmfEvent); |
| 4440 | |
| 4441 | _playInbandDtmfEvent = playDtmfEvent; |
| 4442 | _inbandDtmfQueue.AddDtmf(eventCode, lengthMs, attenuationDb); |
| 4443 | |
| 4444 | return 0; |
| 4445 | } |
| 4446 | |
| 4447 | int |
| 4448 | Channel::SetDtmfPlayoutStatus(bool enable) |
| 4449 | { |
| 4450 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4451 | "Channel::SetDtmfPlayoutStatus()"); |
| 4452 | if (_audioCodingModule.SetDtmfPlayoutStatus(enable) != 0) |
| 4453 | { |
| 4454 | _engineStatisticsPtr->SetLastError( |
| 4455 | VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning, |
| 4456 | "SetDtmfPlayoutStatus() failed to set Dtmf playout"); |
| 4457 | return -1; |
| 4458 | } |
| 4459 | return 0; |
| 4460 | } |
| 4461 | |
| 4462 | bool |
| 4463 | Channel::DtmfPlayoutStatus() const |
| 4464 | { |
| 4465 | return _audioCodingModule.DtmfPlayoutStatus(); |
| 4466 | } |
| 4467 | |
| 4468 | int |
| 4469 | Channel::SetSendTelephoneEventPayloadType(unsigned char type) |
| 4470 | { |
| 4471 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4472 | "Channel::SetSendTelephoneEventPayloadType()"); |
| 4473 | if (type > 127) |
| 4474 | { |
| 4475 | _engineStatisticsPtr->SetLastError( |
| 4476 | VE_INVALID_ARGUMENT, kTraceError, |
| 4477 | "SetSendTelephoneEventPayloadType() invalid type"); |
| 4478 | return -1; |
| 4479 | } |
| 4480 | CodecInst codec; |
| 4481 | codec.plfreq = 8000; |
| 4482 | codec.pltype = type; |
| 4483 | memcpy(codec.plname, "telephone-event", 16); |
| 4484 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
| 4485 | { |
| 4486 | _engineStatisticsPtr->SetLastError( |
| 4487 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 4488 | "SetSendTelephoneEventPayloadType() failed to register send" |
| 4489 | "payload type"); |
| 4490 | return -1; |
| 4491 | } |
| 4492 | _sendTelephoneEventPayloadType = type; |
| 4493 | return 0; |
| 4494 | } |
| 4495 | |
| 4496 | int |
| 4497 | Channel::GetSendTelephoneEventPayloadType(unsigned char& type) |
| 4498 | { |
| 4499 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4500 | "Channel::GetSendTelephoneEventPayloadType()"); |
| 4501 | type = _sendTelephoneEventPayloadType; |
| 4502 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4503 | VoEId(_instanceId,_channelId), |
| 4504 | "GetSendTelephoneEventPayloadType() => type=%u", type); |
| 4505 | return 0; |
| 4506 | } |
| 4507 | |
| 4508 | #ifdef WEBRTC_DTMF_DETECTION |
| 4509 | |
| 4510 | WebRtc_Word32 |
| 4511 | Channel::RegisterTelephoneEventDetection( |
| 4512 | TelephoneEventDetectionMethods detectionMethod, |
| 4513 | VoETelephoneEventObserver& observer) |
| 4514 | { |
| 4515 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4516 | "Channel::RegisterTelephoneEventDetection()"); |
| 4517 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4518 | |
| 4519 | if (_telephoneEventDetectionPtr) |
| 4520 | { |
| 4521 | _engineStatisticsPtr->SetLastError( |
| 4522 | VE_INVALID_OPERATION, kTraceError, |
| 4523 | "RegisterTelephoneEventDetection() detection already enabled"); |
| 4524 | return -1; |
| 4525 | } |
| 4526 | |
| 4527 | _telephoneEventDetectionPtr = &observer; |
| 4528 | |
| 4529 | switch (detectionMethod) |
| 4530 | { |
| 4531 | case kInBand: |
| 4532 | _inbandTelephoneEventDetection = true; |
| 4533 | _outOfBandTelephoneEventDetecion = false; |
| 4534 | break; |
| 4535 | case kOutOfBand: |
| 4536 | _inbandTelephoneEventDetection = false; |
| 4537 | _outOfBandTelephoneEventDetecion = true; |
| 4538 | break; |
| 4539 | case kInAndOutOfBand: |
| 4540 | _inbandTelephoneEventDetection = true; |
| 4541 | _outOfBandTelephoneEventDetecion = true; |
| 4542 | break; |
| 4543 | default: |
| 4544 | _engineStatisticsPtr->SetLastError( |
| 4545 | VE_INVALID_ARGUMENT, kTraceError, |
| 4546 | "RegisterTelephoneEventDetection() invalid detection method"); |
| 4547 | return -1; |
| 4548 | } |
| 4549 | |
| 4550 | if (_inbandTelephoneEventDetection) |
| 4551 | { |
| 4552 | // Enable in-band Dtmf detectin in the ACM. |
| 4553 | if (_audioCodingModule.RegisterIncomingMessagesCallback(this) != 0) |
| 4554 | { |
| 4555 | _engineStatisticsPtr->SetLastError( |
| 4556 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 4557 | "RegisterTelephoneEventDetection() failed to enable Dtmf " |
| 4558 | "detection"); |
| 4559 | } |
| 4560 | } |
| 4561 | |
| 4562 | // Enable/disable out-of-band detection of received telephone-events. |
| 4563 | // When enabled, RtpAudioFeedback::OnReceivedTelephoneEvent() will be |
| 4564 | // called two times by the RTP/RTCP module (start & end). |
| 4565 | const bool forwardToDecoder = |
| 4566 | _rtpRtcpModule->TelephoneEventForwardToDecoder(); |
| 4567 | const bool detectEndOfTone = true; |
| 4568 | _rtpRtcpModule->SetTelephoneEventStatus(_outOfBandTelephoneEventDetecion, |
| 4569 | forwardToDecoder, |
| 4570 | detectEndOfTone); |
| 4571 | |
| 4572 | return 0; |
| 4573 | } |
| 4574 | |
| 4575 | int |
| 4576 | Channel::DeRegisterTelephoneEventDetection() |
| 4577 | { |
| 4578 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 4579 | "Channel::DeRegisterTelephoneEventDetection()"); |
| 4580 | |
| 4581 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4582 | |
| 4583 | if (!_telephoneEventDetectionPtr) |
| 4584 | { |
| 4585 | _engineStatisticsPtr->SetLastError( |
| 4586 | VE_INVALID_OPERATION, |
| 4587 | kTraceWarning, |
| 4588 | "DeRegisterTelephoneEventDetection() detection already disabled"); |
| 4589 | return 0; |
| 4590 | } |
| 4591 | |
| 4592 | // Disable out-of-band event detection |
| 4593 | const bool forwardToDecoder = |
| 4594 | _rtpRtcpModule->TelephoneEventForwardToDecoder(); |
| 4595 | _rtpRtcpModule->SetTelephoneEventStatus(false, forwardToDecoder); |
| 4596 | |
| 4597 | // Disable in-band Dtmf detection |
| 4598 | _audioCodingModule.RegisterIncomingMessagesCallback(NULL); |
| 4599 | |
| 4600 | _inbandTelephoneEventDetection = false; |
| 4601 | _outOfBandTelephoneEventDetecion = false; |
| 4602 | _telephoneEventDetectionPtr = NULL; |
| 4603 | |
| 4604 | return 0; |
| 4605 | } |
| 4606 | |
| 4607 | int |
| 4608 | Channel::GetTelephoneEventDetectionStatus( |
| 4609 | bool& enabled, |
| 4610 | TelephoneEventDetectionMethods& detectionMethod) |
| 4611 | { |
| 4612 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 4613 | "Channel::GetTelephoneEventDetectionStatus()"); |
| 4614 | |
| 4615 | { |
| 4616 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4617 | enabled = (_telephoneEventDetectionPtr != NULL); |
| 4618 | } |
| 4619 | |
| 4620 | if (enabled) |
| 4621 | { |
| 4622 | if (_inbandTelephoneEventDetection && !_outOfBandTelephoneEventDetecion) |
| 4623 | detectionMethod = kInBand; |
| 4624 | else if (!_inbandTelephoneEventDetection |
| 4625 | && _outOfBandTelephoneEventDetecion) |
| 4626 | detectionMethod = kOutOfBand; |
| 4627 | else if (_inbandTelephoneEventDetection |
| 4628 | && _outOfBandTelephoneEventDetecion) |
| 4629 | detectionMethod = kInAndOutOfBand; |
| 4630 | else |
| 4631 | { |
| 4632 | assert(false); |
| 4633 | return -1; |
| 4634 | } |
| 4635 | } |
| 4636 | |
| 4637 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4638 | VoEId(_instanceId, _channelId), |
| 4639 | "GetTelephoneEventDetectionStatus() => enabled=%d," |
| 4640 | "detectionMethod=%d", enabled, detectionMethod); |
| 4641 | return 0; |
| 4642 | } |
| 4643 | |
| 4644 | #endif // #ifdef WEBRTC_DTMF_DETECTION |
| 4645 | |
| 4646 | int |
| 4647 | Channel::UpdateRxVadDetection(AudioFrame& audioFrame) |
| 4648 | { |
| 4649 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4650 | "Channel::UpdateRxVadDetection()"); |
| 4651 | |
| 4652 | int vadDecision = 1; |
| 4653 | |
| 4654 | vadDecision = (audioFrame.vad_activity_ == AudioFrame::kVadActive)? 1 : 0; |
| 4655 | |
| 4656 | if ((vadDecision != _oldVadDecision) && _rxVadObserverPtr) |
| 4657 | { |
| 4658 | OnRxVadDetected(vadDecision); |
| 4659 | _oldVadDecision = vadDecision; |
| 4660 | } |
| 4661 | |
| 4662 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4663 | "Channel::UpdateRxVadDetection() => vadDecision=%d", |
| 4664 | vadDecision); |
| 4665 | return 0; |
| 4666 | } |
| 4667 | |
| 4668 | int |
| 4669 | Channel::RegisterRxVadObserver(VoERxVadCallback &observer) |
| 4670 | { |
| 4671 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4672 | "Channel::RegisterRxVadObserver()"); |
| 4673 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4674 | |
| 4675 | if (_rxVadObserverPtr) |
| 4676 | { |
| 4677 | _engineStatisticsPtr->SetLastError( |
| 4678 | VE_INVALID_OPERATION, kTraceError, |
| 4679 | "RegisterRxVadObserver() observer already enabled"); |
| 4680 | return -1; |
| 4681 | } |
| 4682 | _rxVadObserverPtr = &observer; |
| 4683 | _RxVadDetection = true; |
| 4684 | return 0; |
| 4685 | } |
| 4686 | |
| 4687 | int |
| 4688 | Channel::DeRegisterRxVadObserver() |
| 4689 | { |
| 4690 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4691 | "Channel::DeRegisterRxVadObserver()"); |
| 4692 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4693 | |
| 4694 | if (!_rxVadObserverPtr) |
| 4695 | { |
| 4696 | _engineStatisticsPtr->SetLastError( |
| 4697 | VE_INVALID_OPERATION, kTraceWarning, |
| 4698 | "DeRegisterRxVadObserver() observer already disabled"); |
| 4699 | return 0; |
| 4700 | } |
| 4701 | _rxVadObserverPtr = NULL; |
| 4702 | _RxVadDetection = false; |
| 4703 | return 0; |
| 4704 | } |
| 4705 | |
| 4706 | int |
| 4707 | Channel::VoiceActivityIndicator(int &activity) |
| 4708 | { |
| 4709 | activity = _sendFrameType; |
| 4710 | |
| 4711 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4712 | "Channel::VoiceActivityIndicator(indicator=%d)", activity); |
| 4713 | return 0; |
| 4714 | } |
| 4715 | |
| 4716 | #ifdef WEBRTC_VOICE_ENGINE_AGC |
| 4717 | |
| 4718 | int |
| 4719 | Channel::SetRxAgcStatus(const bool enable, const AgcModes mode) |
| 4720 | { |
| 4721 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4722 | "Channel::SetRxAgcStatus(enable=%d, mode=%d)", |
| 4723 | (int)enable, (int)mode); |
| 4724 | |
| 4725 | GainControl::Mode agcMode(GainControl::kFixedDigital); |
| 4726 | switch (mode) |
| 4727 | { |
| 4728 | case kAgcDefault: |
| 4729 | agcMode = GainControl::kAdaptiveDigital; |
| 4730 | break; |
| 4731 | case kAgcUnchanged: |
| 4732 | agcMode = _rxAudioProcessingModulePtr->gain_control()->mode(); |
| 4733 | break; |
| 4734 | case kAgcFixedDigital: |
| 4735 | agcMode = GainControl::kFixedDigital; |
| 4736 | break; |
| 4737 | case kAgcAdaptiveDigital: |
| 4738 | agcMode =GainControl::kAdaptiveDigital; |
| 4739 | break; |
| 4740 | default: |
| 4741 | _engineStatisticsPtr->SetLastError( |
| 4742 | VE_INVALID_ARGUMENT, kTraceError, |
| 4743 | "SetRxAgcStatus() invalid Agc mode"); |
| 4744 | return -1; |
| 4745 | } |
| 4746 | |
| 4747 | if (_rxAudioProcessingModulePtr->gain_control()->set_mode(agcMode) != 0) |
| 4748 | { |
| 4749 | _engineStatisticsPtr->SetLastError( |
| 4750 | VE_APM_ERROR, kTraceError, |
| 4751 | "SetRxAgcStatus() failed to set Agc mode"); |
| 4752 | return -1; |
| 4753 | } |
| 4754 | if (_rxAudioProcessingModulePtr->gain_control()->Enable(enable) != 0) |
| 4755 | { |
| 4756 | _engineStatisticsPtr->SetLastError( |
| 4757 | VE_APM_ERROR, kTraceError, |
| 4758 | "SetRxAgcStatus() failed to set Agc state"); |
| 4759 | return -1; |
| 4760 | } |
| 4761 | |
| 4762 | _rxAgcIsEnabled = enable; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 4763 | _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true)); |
| 4764 | |
| 4765 | return 0; |
| 4766 | } |
| 4767 | |
| 4768 | int |
| 4769 | Channel::GetRxAgcStatus(bool& enabled, AgcModes& mode) |
| 4770 | { |
| 4771 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4772 | "Channel::GetRxAgcStatus(enable=?, mode=?)"); |
| 4773 | |
| 4774 | bool enable = _rxAudioProcessingModulePtr->gain_control()->is_enabled(); |
| 4775 | GainControl::Mode agcMode = |
| 4776 | _rxAudioProcessingModulePtr->gain_control()->mode(); |
| 4777 | |
| 4778 | enabled = enable; |
| 4779 | |
| 4780 | switch (agcMode) |
| 4781 | { |
| 4782 | case GainControl::kFixedDigital: |
| 4783 | mode = kAgcFixedDigital; |
| 4784 | break; |
| 4785 | case GainControl::kAdaptiveDigital: |
| 4786 | mode = kAgcAdaptiveDigital; |
| 4787 | break; |
| 4788 | default: |
| 4789 | _engineStatisticsPtr->SetLastError( |
| 4790 | VE_APM_ERROR, kTraceError, |
| 4791 | "GetRxAgcStatus() invalid Agc mode"); |
| 4792 | return -1; |
| 4793 | } |
| 4794 | |
| 4795 | return 0; |
| 4796 | } |
| 4797 | |
| 4798 | int |
| 4799 | Channel::SetRxAgcConfig(const AgcConfig config) |
| 4800 | { |
| 4801 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4802 | "Channel::SetRxAgcConfig()"); |
| 4803 | |
| 4804 | if (_rxAudioProcessingModulePtr->gain_control()->set_target_level_dbfs( |
| 4805 | config.targetLeveldBOv) != 0) |
| 4806 | { |
| 4807 | _engineStatisticsPtr->SetLastError( |
| 4808 | VE_APM_ERROR, kTraceError, |
| 4809 | "SetRxAgcConfig() failed to set target peak |level|" |
| 4810 | "(or envelope) of the Agc"); |
| 4811 | return -1; |
| 4812 | } |
| 4813 | if (_rxAudioProcessingModulePtr->gain_control()->set_compression_gain_db( |
| 4814 | config.digitalCompressionGaindB) != 0) |
| 4815 | { |
| 4816 | _engineStatisticsPtr->SetLastError( |
| 4817 | VE_APM_ERROR, kTraceError, |
| 4818 | "SetRxAgcConfig() failed to set the range in |gain| the" |
| 4819 | " digital compression stage may apply"); |
| 4820 | return -1; |
| 4821 | } |
| 4822 | if (_rxAudioProcessingModulePtr->gain_control()->enable_limiter( |
| 4823 | config.limiterEnable) != 0) |
| 4824 | { |
| 4825 | _engineStatisticsPtr->SetLastError( |
| 4826 | VE_APM_ERROR, kTraceError, |
| 4827 | "SetRxAgcConfig() failed to set hard limiter to the signal"); |
| 4828 | return -1; |
| 4829 | } |
| 4830 | |
| 4831 | return 0; |
| 4832 | } |
| 4833 | |
| 4834 | int |
| 4835 | Channel::GetRxAgcConfig(AgcConfig& config) |
| 4836 | { |
| 4837 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4838 | "Channel::GetRxAgcConfig(config=%?)"); |
| 4839 | |
| 4840 | config.targetLeveldBOv = |
| 4841 | _rxAudioProcessingModulePtr->gain_control()->target_level_dbfs(); |
| 4842 | config.digitalCompressionGaindB = |
| 4843 | _rxAudioProcessingModulePtr->gain_control()->compression_gain_db(); |
| 4844 | config.limiterEnable = |
| 4845 | _rxAudioProcessingModulePtr->gain_control()->is_limiter_enabled(); |
| 4846 | |
| 4847 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4848 | VoEId(_instanceId,_channelId), "GetRxAgcConfig() => " |
| 4849 | "targetLeveldBOv=%u, digitalCompressionGaindB=%u," |
| 4850 | " limiterEnable=%d", |
| 4851 | config.targetLeveldBOv, |
| 4852 | config.digitalCompressionGaindB, |
| 4853 | config.limiterEnable); |
| 4854 | |
| 4855 | return 0; |
| 4856 | } |
| 4857 | |
| 4858 | #endif // #ifdef WEBRTC_VOICE_ENGINE_AGC |
| 4859 | |
| 4860 | #ifdef WEBRTC_VOICE_ENGINE_NR |
| 4861 | |
| 4862 | int |
| 4863 | Channel::SetRxNsStatus(const bool enable, const NsModes mode) |
| 4864 | { |
| 4865 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4866 | "Channel::SetRxNsStatus(enable=%d, mode=%d)", |
| 4867 | (int)enable, (int)mode); |
| 4868 | |
| 4869 | NoiseSuppression::Level nsLevel( |
| 4870 | (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE); |
| 4871 | switch (mode) |
| 4872 | { |
| 4873 | |
| 4874 | case kNsDefault: |
| 4875 | nsLevel = (NoiseSuppression::Level) |
| 4876 | WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE; |
| 4877 | break; |
| 4878 | case kNsUnchanged: |
| 4879 | nsLevel = _rxAudioProcessingModulePtr->noise_suppression()->level(); |
| 4880 | break; |
| 4881 | case kNsConference: |
| 4882 | nsLevel = NoiseSuppression::kHigh; |
| 4883 | break; |
| 4884 | case kNsLowSuppression: |
| 4885 | nsLevel = NoiseSuppression::kLow; |
| 4886 | break; |
| 4887 | case kNsModerateSuppression: |
| 4888 | nsLevel = NoiseSuppression::kModerate; |
| 4889 | break; |
| 4890 | case kNsHighSuppression: |
| 4891 | nsLevel = NoiseSuppression::kHigh; |
| 4892 | break; |
| 4893 | case kNsVeryHighSuppression: |
| 4894 | nsLevel = NoiseSuppression::kVeryHigh; |
| 4895 | break; |
| 4896 | } |
| 4897 | |
| 4898 | if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(nsLevel) |
| 4899 | != 0) |
| 4900 | { |
| 4901 | _engineStatisticsPtr->SetLastError( |
| 4902 | VE_APM_ERROR, kTraceError, |
| 4903 | "SetRxAgcStatus() failed to set Ns level"); |
| 4904 | return -1; |
| 4905 | } |
| 4906 | if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(enable) != 0) |
| 4907 | { |
| 4908 | _engineStatisticsPtr->SetLastError( |
| 4909 | VE_APM_ERROR, kTraceError, |
| 4910 | "SetRxAgcStatus() failed to set Agc state"); |
| 4911 | return -1; |
| 4912 | } |
| 4913 | |
| 4914 | _rxNsIsEnabled = enable; |
| 4915 | _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true)); |
| 4916 | |
| 4917 | return 0; |
| 4918 | } |
| 4919 | |
| 4920 | int |
| 4921 | Channel::GetRxNsStatus(bool& enabled, NsModes& mode) |
| 4922 | { |
| 4923 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4924 | "Channel::GetRxNsStatus(enable=?, mode=?)"); |
| 4925 | |
| 4926 | bool enable = |
| 4927 | _rxAudioProcessingModulePtr->noise_suppression()->is_enabled(); |
| 4928 | NoiseSuppression::Level ncLevel = |
| 4929 | _rxAudioProcessingModulePtr->noise_suppression()->level(); |
| 4930 | |
| 4931 | enabled = enable; |
| 4932 | |
| 4933 | switch (ncLevel) |
| 4934 | { |
| 4935 | case NoiseSuppression::kLow: |
| 4936 | mode = kNsLowSuppression; |
| 4937 | break; |
| 4938 | case NoiseSuppression::kModerate: |
| 4939 | mode = kNsModerateSuppression; |
| 4940 | break; |
| 4941 | case NoiseSuppression::kHigh: |
| 4942 | mode = kNsHighSuppression; |
| 4943 | break; |
| 4944 | case NoiseSuppression::kVeryHigh: |
| 4945 | mode = kNsVeryHighSuppression; |
| 4946 | break; |
| 4947 | } |
| 4948 | |
| 4949 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4950 | VoEId(_instanceId,_channelId), |
| 4951 | "GetRxNsStatus() => enabled=%d, mode=%d", enabled, mode); |
| 4952 | return 0; |
| 4953 | } |
| 4954 | |
| 4955 | #endif // #ifdef WEBRTC_VOICE_ENGINE_NR |
| 4956 | |
| 4957 | int |
| 4958 | Channel::RegisterRTPObserver(VoERTPObserver& observer) |
| 4959 | { |
| 4960 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 4961 | "Channel::RegisterRTPObserver()"); |
| 4962 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4963 | |
| 4964 | if (_rtpObserverPtr) |
| 4965 | { |
| 4966 | _engineStatisticsPtr->SetLastError( |
| 4967 | VE_INVALID_OPERATION, kTraceError, |
| 4968 | "RegisterRTPObserver() observer already enabled"); |
| 4969 | return -1; |
| 4970 | } |
| 4971 | |
| 4972 | _rtpObserverPtr = &observer; |
| 4973 | _rtpObserver = true; |
| 4974 | |
| 4975 | return 0; |
| 4976 | } |
| 4977 | |
| 4978 | int |
| 4979 | Channel::DeRegisterRTPObserver() |
| 4980 | { |
| 4981 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4982 | "Channel::DeRegisterRTPObserver()"); |
| 4983 | CriticalSectionScoped cs(&_callbackCritSect); |
| 4984 | |
| 4985 | if (!_rtpObserverPtr) |
| 4986 | { |
| 4987 | _engineStatisticsPtr->SetLastError( |
| 4988 | VE_INVALID_OPERATION, kTraceWarning, |
| 4989 | "DeRegisterRTPObserver() observer already disabled"); |
| 4990 | return 0; |
| 4991 | } |
| 4992 | |
| 4993 | _rtpObserver = false; |
| 4994 | _rtpObserverPtr = NULL; |
| 4995 | |
| 4996 | return 0; |
| 4997 | } |
| 4998 | |
| 4999 | int |
| 5000 | Channel::RegisterRTCPObserver(VoERTCPObserver& observer) |
| 5001 | { |
| 5002 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5003 | "Channel::RegisterRTCPObserver()"); |
| 5004 | CriticalSectionScoped cs(&_callbackCritSect); |
| 5005 | |
| 5006 | if (_rtcpObserverPtr) |
| 5007 | { |
| 5008 | _engineStatisticsPtr->SetLastError( |
| 5009 | VE_INVALID_OPERATION, kTraceError, |
| 5010 | "RegisterRTCPObserver() observer already enabled"); |
| 5011 | return -1; |
| 5012 | } |
| 5013 | |
| 5014 | _rtcpObserverPtr = &observer; |
| 5015 | _rtcpObserver = true; |
| 5016 | |
| 5017 | return 0; |
| 5018 | } |
| 5019 | |
| 5020 | int |
| 5021 | Channel::DeRegisterRTCPObserver() |
| 5022 | { |
| 5023 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 5024 | "Channel::DeRegisterRTCPObserver()"); |
| 5025 | CriticalSectionScoped cs(&_callbackCritSect); |
| 5026 | |
| 5027 | if (!_rtcpObserverPtr) |
| 5028 | { |
| 5029 | _engineStatisticsPtr->SetLastError( |
| 5030 | VE_INVALID_OPERATION, kTraceWarning, |
| 5031 | "DeRegisterRTCPObserver() observer already disabled"); |
| 5032 | return 0; |
| 5033 | } |
| 5034 | |
| 5035 | _rtcpObserver = false; |
| 5036 | _rtcpObserverPtr = NULL; |
| 5037 | |
| 5038 | return 0; |
| 5039 | } |
| 5040 | |
| 5041 | int |
| 5042 | Channel::SetLocalSSRC(unsigned int ssrc) |
| 5043 | { |
| 5044 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 5045 | "Channel::SetLocalSSRC()"); |
| 5046 | if (_sending) |
| 5047 | { |
| 5048 | _engineStatisticsPtr->SetLastError( |
| 5049 | VE_ALREADY_SENDING, kTraceError, |
| 5050 | "SetLocalSSRC() already sending"); |
| 5051 | return -1; |
| 5052 | } |
| 5053 | if (_rtpRtcpModule->SetSSRC(ssrc) != 0) |
| 5054 | { |
| 5055 | _engineStatisticsPtr->SetLastError( |
| 5056 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5057 | "SetLocalSSRC() failed to set SSRC"); |
| 5058 | return -1; |
| 5059 | } |
| 5060 | return 0; |
| 5061 | } |
| 5062 | |
| 5063 | int |
| 5064 | Channel::GetLocalSSRC(unsigned int& ssrc) |
| 5065 | { |
| 5066 | ssrc = _rtpRtcpModule->SSRC(); |
| 5067 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5068 | VoEId(_instanceId,_channelId), |
| 5069 | "GetLocalSSRC() => ssrc=%lu", ssrc); |
| 5070 | return 0; |
| 5071 | } |
| 5072 | |
| 5073 | int |
| 5074 | Channel::GetRemoteSSRC(unsigned int& ssrc) |
| 5075 | { |
| 5076 | ssrc = _rtpRtcpModule->RemoteSSRC(); |
| 5077 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5078 | VoEId(_instanceId,_channelId), |
| 5079 | "GetRemoteSSRC() => ssrc=%lu", ssrc); |
| 5080 | return 0; |
| 5081 | } |
| 5082 | |
| 5083 | int |
| 5084 | Channel::GetRemoteCSRCs(unsigned int arrCSRC[15]) |
| 5085 | { |
| 5086 | if (arrCSRC == NULL) |
| 5087 | { |
| 5088 | _engineStatisticsPtr->SetLastError( |
| 5089 | VE_INVALID_ARGUMENT, kTraceError, |
| 5090 | "GetRemoteCSRCs() invalid array argument"); |
| 5091 | return -1; |
| 5092 | } |
| 5093 | WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]; |
| 5094 | WebRtc_Word32 CSRCs(0); |
| 5095 | CSRCs = _rtpRtcpModule->CSRCs(arrOfCSRC); |
| 5096 | if (CSRCs > 0) |
| 5097 | { |
| 5098 | memcpy(arrCSRC, arrOfCSRC, CSRCs * sizeof(WebRtc_UWord32)); |
| 5099 | for (int i = 0; i < (int) CSRCs; i++) |
| 5100 | { |
| 5101 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5102 | VoEId(_instanceId, _channelId), |
| 5103 | "GetRemoteCSRCs() => arrCSRC[%d]=%lu", i, arrCSRC[i]); |
| 5104 | } |
| 5105 | } else |
| 5106 | { |
| 5107 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5108 | VoEId(_instanceId, _channelId), |
| 5109 | "GetRemoteCSRCs() => list is empty!"); |
| 5110 | } |
| 5111 | return CSRCs; |
| 5112 | } |
| 5113 | |
| 5114 | int |
| 5115 | Channel::SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID) |
| 5116 | { |
| 5117 | if (_rtpAudioProc.get() == NULL) |
| 5118 | { |
| 5119 | _rtpAudioProc.reset(AudioProcessing::Create(VoEModuleId(_instanceId, |
| 5120 | _channelId))); |
| 5121 | if (_rtpAudioProc.get() == NULL) |
| 5122 | { |
| 5123 | _engineStatisticsPtr->SetLastError(VE_NO_MEMORY, kTraceCritical, |
| 5124 | "Failed to create AudioProcessing"); |
| 5125 | return -1; |
| 5126 | } |
| 5127 | } |
| 5128 | |
| 5129 | if (_rtpAudioProc->level_estimator()->Enable(enable) != |
| 5130 | AudioProcessing::kNoError) |
| 5131 | { |
| 5132 | _engineStatisticsPtr->SetLastError(VE_APM_ERROR, kTraceWarning, |
| 5133 | "Failed to enable AudioProcessing::level_estimator()"); |
| 5134 | } |
| 5135 | |
| 5136 | _includeAudioLevelIndication = enable; |
| 5137 | return _rtpRtcpModule->SetRTPAudioLevelIndicationStatus(enable, ID); |
| 5138 | } |
| 5139 | int |
| 5140 | Channel::GetRTPAudioLevelIndicationStatus(bool& enabled, unsigned char& ID) |
| 5141 | { |
| 5142 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5143 | VoEId(_instanceId,_channelId), |
| 5144 | "GetRTPAudioLevelIndicationStatus() => enabled=%d, ID=%u", |
| 5145 | enabled, ID); |
| 5146 | return _rtpRtcpModule->GetRTPAudioLevelIndicationStatus(enabled, ID); |
| 5147 | } |
| 5148 | |
| 5149 | int |
| 5150 | Channel::SetRTCPStatus(bool enable) |
| 5151 | { |
| 5152 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5153 | "Channel::SetRTCPStatus()"); |
| 5154 | if (_rtpRtcpModule->SetRTCPStatus(enable ? |
| 5155 | kRtcpCompound : kRtcpOff) != 0) |
| 5156 | { |
| 5157 | _engineStatisticsPtr->SetLastError( |
| 5158 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5159 | "SetRTCPStatus() failed to set RTCP status"); |
| 5160 | return -1; |
| 5161 | } |
| 5162 | return 0; |
| 5163 | } |
| 5164 | |
| 5165 | int |
| 5166 | Channel::GetRTCPStatus(bool& enabled) |
| 5167 | { |
| 5168 | RTCPMethod method = _rtpRtcpModule->RTCP(); |
| 5169 | enabled = (method != kRtcpOff); |
| 5170 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5171 | VoEId(_instanceId,_channelId), |
| 5172 | "GetRTCPStatus() => enabled=%d", enabled); |
| 5173 | return 0; |
| 5174 | } |
| 5175 | |
| 5176 | int |
| 5177 | Channel::SetRTCP_CNAME(const char cName[256]) |
| 5178 | { |
| 5179 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 5180 | "Channel::SetRTCP_CNAME()"); |
| 5181 | if (_rtpRtcpModule->SetCNAME(cName) != 0) |
| 5182 | { |
| 5183 | _engineStatisticsPtr->SetLastError( |
| 5184 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5185 | "SetRTCP_CNAME() failed to set RTCP CNAME"); |
| 5186 | return -1; |
| 5187 | } |
| 5188 | return 0; |
| 5189 | } |
| 5190 | |
| 5191 | int |
| 5192 | Channel::GetRTCP_CNAME(char cName[256]) |
| 5193 | { |
| 5194 | if (_rtpRtcpModule->CNAME(cName) != 0) |
| 5195 | { |
| 5196 | _engineStatisticsPtr->SetLastError( |
| 5197 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5198 | "GetRTCP_CNAME() failed to retrieve RTCP CNAME"); |
| 5199 | return -1; |
| 5200 | } |
| 5201 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5202 | VoEId(_instanceId, _channelId), |
| 5203 | "GetRTCP_CNAME() => cName=%s", cName); |
| 5204 | return 0; |
| 5205 | } |
| 5206 | |
| 5207 | int |
| 5208 | Channel::GetRemoteRTCP_CNAME(char cName[256]) |
| 5209 | { |
| 5210 | if (cName == NULL) |
| 5211 | { |
| 5212 | _engineStatisticsPtr->SetLastError( |
| 5213 | VE_INVALID_ARGUMENT, kTraceError, |
| 5214 | "GetRemoteRTCP_CNAME() invalid CNAME input buffer"); |
| 5215 | return -1; |
| 5216 | } |
| 5217 | char cname[RTCP_CNAME_SIZE]; |
| 5218 | const WebRtc_UWord32 remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
| 5219 | if (_rtpRtcpModule->RemoteCNAME(remoteSSRC, cname) != 0) |
| 5220 | { |
| 5221 | _engineStatisticsPtr->SetLastError( |
| 5222 | VE_CANNOT_RETRIEVE_CNAME, kTraceError, |
| 5223 | "GetRemoteRTCP_CNAME() failed to retrieve remote RTCP CNAME"); |
| 5224 | return -1; |
| 5225 | } |
| 5226 | strcpy(cName, cname); |
| 5227 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5228 | VoEId(_instanceId, _channelId), |
| 5229 | "GetRemoteRTCP_CNAME() => cName=%s", cName); |
| 5230 | return 0; |
| 5231 | } |
| 5232 | |
| 5233 | int |
| 5234 | Channel::GetRemoteRTCPData( |
| 5235 | unsigned int& NTPHigh, |
| 5236 | unsigned int& NTPLow, |
| 5237 | unsigned int& timestamp, |
| 5238 | unsigned int& playoutTimestamp, |
| 5239 | unsigned int* jitter, |
| 5240 | unsigned short* fractionLost) |
| 5241 | { |
| 5242 | // --- Information from sender info in received Sender Reports |
| 5243 | |
| 5244 | RTCPSenderInfo senderInfo; |
| 5245 | if (_rtpRtcpModule->RemoteRTCPStat(&senderInfo) != 0) |
| 5246 | { |
| 5247 | _engineStatisticsPtr->SetLastError( |
| 5248 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5249 | "GetRemoteRTCPData() failed to retrieve sender info for remote " |
| 5250 | "side"); |
| 5251 | return -1; |
| 5252 | } |
| 5253 | |
| 5254 | // We only utilize 12 out of 20 bytes in the sender info (ignores packet |
| 5255 | // and octet count) |
| 5256 | NTPHigh = senderInfo.NTPseconds; |
| 5257 | NTPLow = senderInfo.NTPfraction; |
| 5258 | timestamp = senderInfo.RTPtimeStamp; |
| 5259 | |
| 5260 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5261 | VoEId(_instanceId, _channelId), |
| 5262 | "GetRemoteRTCPData() => NTPHigh=%lu, NTPLow=%lu, " |
| 5263 | "timestamp=%lu", |
| 5264 | NTPHigh, NTPLow, timestamp); |
| 5265 | |
| 5266 | // --- Locally derived information |
| 5267 | |
| 5268 | // This value is updated on each incoming RTCP packet (0 when no packet |
| 5269 | // has been received) |
| 5270 | playoutTimestamp = _playoutTimeStampRTCP; |
| 5271 | |
| 5272 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5273 | VoEId(_instanceId, _channelId), |
| 5274 | "GetRemoteRTCPData() => playoutTimestamp=%lu", |
| 5275 | _playoutTimeStampRTCP); |
| 5276 | |
| 5277 | if (NULL != jitter || NULL != fractionLost) |
| 5278 | { |
| 5279 | // Get all RTCP receiver report blocks that have been received on this |
| 5280 | // channel. If we receive RTP packets from a remote source we know the |
| 5281 | // remote SSRC and use the report block from him. |
| 5282 | // Otherwise use the first report block. |
| 5283 | std::vector<RTCPReportBlock> remote_stats; |
| 5284 | if (_rtpRtcpModule->RemoteRTCPStat(&remote_stats) != 0 || |
| 5285 | remote_stats.empty()) { |
| 5286 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 5287 | VoEId(_instanceId, _channelId), |
| 5288 | "GetRemoteRTCPData() failed to measure statistics due" |
| 5289 | " to lack of received RTP and/or RTCP packets"); |
| 5290 | return -1; |
| 5291 | } |
| 5292 | |
| 5293 | WebRtc_UWord32 remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
| 5294 | std::vector<RTCPReportBlock>::const_iterator it = remote_stats.begin(); |
| 5295 | for (; it != remote_stats.end(); ++it) { |
| 5296 | if (it->remoteSSRC == remoteSSRC) |
| 5297 | break; |
| 5298 | } |
| 5299 | |
| 5300 | if (it == remote_stats.end()) { |
| 5301 | // If we have not received any RTCP packets from this SSRC it probably |
| 5302 | // means that we have not received any RTP packets. |
| 5303 | // Use the first received report block instead. |
| 5304 | it = remote_stats.begin(); |
| 5305 | remoteSSRC = it->remoteSSRC; |
| 5306 | } |
| 5307 | |
| 5308 | if (jitter) { |
| 5309 | *jitter = it->jitter; |
| 5310 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5311 | VoEId(_instanceId, _channelId), |
| 5312 | "GetRemoteRTCPData() => jitter = %lu", *jitter); |
| 5313 | } |
| 5314 | |
| 5315 | if (fractionLost) { |
| 5316 | *fractionLost = it->fractionLost; |
| 5317 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5318 | VoEId(_instanceId, _channelId), |
| 5319 | "GetRemoteRTCPData() => fractionLost = %lu", |
| 5320 | *fractionLost); |
| 5321 | } |
| 5322 | } |
| 5323 | return 0; |
| 5324 | } |
| 5325 | |
| 5326 | int |
| 5327 | Channel::SendApplicationDefinedRTCPPacket(const unsigned char subType, |
| 5328 | unsigned int name, |
| 5329 | const char* data, |
| 5330 | unsigned short dataLengthInBytes) |
| 5331 | { |
| 5332 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 5333 | "Channel::SendApplicationDefinedRTCPPacket()"); |
| 5334 | if (!_sending) |
| 5335 | { |
| 5336 | _engineStatisticsPtr->SetLastError( |
| 5337 | VE_NOT_SENDING, kTraceError, |
| 5338 | "SendApplicationDefinedRTCPPacket() not sending"); |
| 5339 | return -1; |
| 5340 | } |
| 5341 | if (NULL == data) |
| 5342 | { |
| 5343 | _engineStatisticsPtr->SetLastError( |
| 5344 | VE_INVALID_ARGUMENT, kTraceError, |
| 5345 | "SendApplicationDefinedRTCPPacket() invalid data value"); |
| 5346 | return -1; |
| 5347 | } |
| 5348 | if (dataLengthInBytes % 4 != 0) |
| 5349 | { |
| 5350 | _engineStatisticsPtr->SetLastError( |
| 5351 | VE_INVALID_ARGUMENT, kTraceError, |
| 5352 | "SendApplicationDefinedRTCPPacket() invalid length value"); |
| 5353 | return -1; |
| 5354 | } |
| 5355 | RTCPMethod status = _rtpRtcpModule->RTCP(); |
| 5356 | if (status == kRtcpOff) |
| 5357 | { |
| 5358 | _engineStatisticsPtr->SetLastError( |
| 5359 | VE_RTCP_ERROR, kTraceError, |
| 5360 | "SendApplicationDefinedRTCPPacket() RTCP is disabled"); |
| 5361 | return -1; |
| 5362 | } |
| 5363 | |
| 5364 | // Create and schedule the RTCP APP packet for transmission |
| 5365 | if (_rtpRtcpModule->SetRTCPApplicationSpecificData( |
| 5366 | subType, |
| 5367 | name, |
| 5368 | (const unsigned char*) data, |
| 5369 | dataLengthInBytes) != 0) |
| 5370 | { |
| 5371 | _engineStatisticsPtr->SetLastError( |
| 5372 | VE_SEND_ERROR, kTraceError, |
| 5373 | "SendApplicationDefinedRTCPPacket() failed to send RTCP packet"); |
| 5374 | return -1; |
| 5375 | } |
| 5376 | return 0; |
| 5377 | } |
| 5378 | |
| 5379 | int |
| 5380 | Channel::GetRTPStatistics( |
| 5381 | unsigned int& averageJitterMs, |
| 5382 | unsigned int& maxJitterMs, |
| 5383 | unsigned int& discardedPackets) |
| 5384 | { |
| 5385 | WebRtc_UWord8 fraction_lost(0); |
| 5386 | WebRtc_UWord32 cum_lost(0); |
| 5387 | WebRtc_UWord32 ext_max(0); |
| 5388 | WebRtc_UWord32 jitter(0); |
| 5389 | WebRtc_UWord32 max_jitter(0); |
| 5390 | |
| 5391 | // The jitter statistics is updated for each received RTP packet and is |
| 5392 | // based on received packets. |
| 5393 | if (_rtpRtcpModule->StatisticsRTP(&fraction_lost, |
| 5394 | &cum_lost, |
| 5395 | &ext_max, |
| 5396 | &jitter, |
| 5397 | &max_jitter) != 0) |
| 5398 | { |
| 5399 | _engineStatisticsPtr->SetLastError( |
| 5400 | VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning, |
| 5401 | "GetRTPStatistics() failed to read RTP statistics from the " |
| 5402 | "RTP/RTCP module"); |
| 5403 | } |
| 5404 | |
| 5405 | const WebRtc_Word32 playoutFrequency = |
| 5406 | _audioCodingModule.PlayoutFrequency(); |
| 5407 | if (playoutFrequency > 0) |
| 5408 | { |
| 5409 | // Scale RTP statistics given the current playout frequency |
| 5410 | maxJitterMs = max_jitter / (playoutFrequency / 1000); |
| 5411 | averageJitterMs = jitter / (playoutFrequency / 1000); |
| 5412 | } |
| 5413 | |
| 5414 | discardedPackets = _numberOfDiscardedPackets; |
| 5415 | |
| 5416 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5417 | VoEId(_instanceId, _channelId), |
| 5418 | "GetRTPStatistics() => averageJitterMs = %lu, maxJitterMs = %lu," |
| 5419 | " discardedPackets = %lu)", |
| 5420 | averageJitterMs, maxJitterMs, discardedPackets); |
| 5421 | return 0; |
| 5422 | } |
| 5423 | |
| 5424 | int Channel::GetRemoteRTCPSenderInfo(SenderInfo* sender_info) { |
| 5425 | if (sender_info == NULL) { |
| 5426 | _engineStatisticsPtr->SetLastError(VE_INVALID_ARGUMENT, kTraceError, |
| 5427 | "GetRemoteRTCPSenderInfo() invalid sender_info."); |
| 5428 | return -1; |
| 5429 | } |
| 5430 | |
| 5431 | // Get the sender info from the latest received RTCP Sender Report. |
| 5432 | RTCPSenderInfo rtcp_sender_info; |
| 5433 | if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_sender_info) != 0) { |
| 5434 | _engineStatisticsPtr->SetLastError(VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5435 | "GetRemoteRTCPSenderInfo() failed to read RTCP SR sender info."); |
| 5436 | return -1; |
| 5437 | } |
| 5438 | |
| 5439 | sender_info->NTP_timestamp_high = rtcp_sender_info.NTPseconds; |
| 5440 | sender_info->NTP_timestamp_low = rtcp_sender_info.NTPfraction; |
| 5441 | sender_info->RTP_timestamp = rtcp_sender_info.RTPtimeStamp; |
| 5442 | sender_info->sender_packet_count = rtcp_sender_info.sendPacketCount; |
| 5443 | sender_info->sender_octet_count = rtcp_sender_info.sendOctetCount; |
| 5444 | return 0; |
| 5445 | } |
| 5446 | |
| 5447 | int Channel::GetRemoteRTCPReportBlocks( |
| 5448 | std::vector<ReportBlock>* report_blocks) { |
| 5449 | if (report_blocks == NULL) { |
| 5450 | _engineStatisticsPtr->SetLastError(VE_INVALID_ARGUMENT, kTraceError, |
| 5451 | "GetRemoteRTCPReportBlock()s invalid report_blocks."); |
| 5452 | return -1; |
| 5453 | } |
| 5454 | |
| 5455 | // Get the report blocks from the latest received RTCP Sender or Receiver |
| 5456 | // Report. Each element in the vector contains the sender's SSRC and a |
| 5457 | // report block according to RFC 3550. |
| 5458 | std::vector<RTCPReportBlock> rtcp_report_blocks; |
| 5459 | if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks) != 0) { |
| 5460 | _engineStatisticsPtr->SetLastError(VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5461 | "GetRemoteRTCPReportBlocks() failed to read RTCP SR/RR report block."); |
| 5462 | return -1; |
| 5463 | } |
| 5464 | |
| 5465 | if (rtcp_report_blocks.empty()) |
| 5466 | return 0; |
| 5467 | |
| 5468 | std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin(); |
| 5469 | for (; it != rtcp_report_blocks.end(); ++it) { |
| 5470 | ReportBlock report_block; |
| 5471 | report_block.sender_SSRC = it->remoteSSRC; |
| 5472 | report_block.source_SSRC = it->sourceSSRC; |
| 5473 | report_block.fraction_lost = it->fractionLost; |
| 5474 | report_block.cumulative_num_packets_lost = it->cumulativeLost; |
| 5475 | report_block.extended_highest_sequence_number = it->extendedHighSeqNum; |
| 5476 | report_block.interarrival_jitter = it->jitter; |
| 5477 | report_block.last_SR_timestamp = it->lastSR; |
| 5478 | report_block.delay_since_last_SR = it->delaySinceLastSR; |
| 5479 | report_blocks->push_back(report_block); |
| 5480 | } |
| 5481 | return 0; |
| 5482 | } |
| 5483 | |
| 5484 | int |
| 5485 | Channel::GetRTPStatistics(CallStatistics& stats) |
| 5486 | { |
| 5487 | WebRtc_UWord8 fraction_lost(0); |
| 5488 | WebRtc_UWord32 cum_lost(0); |
| 5489 | WebRtc_UWord32 ext_max(0); |
| 5490 | WebRtc_UWord32 jitter(0); |
| 5491 | WebRtc_UWord32 max_jitter(0); |
| 5492 | |
| 5493 | // --- Part one of the final structure (four values) |
| 5494 | |
| 5495 | // The jitter statistics is updated for each received RTP packet and is |
| 5496 | // based on received packets. |
| 5497 | if (_rtpRtcpModule->StatisticsRTP(&fraction_lost, |
| 5498 | &cum_lost, |
| 5499 | &ext_max, |
| 5500 | &jitter, |
| 5501 | &max_jitter) != 0) |
| 5502 | { |
| 5503 | _engineStatisticsPtr->SetLastError( |
| 5504 | VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning, |
| 5505 | "GetRTPStatistics() failed to read RTP statistics from the " |
| 5506 | "RTP/RTCP module"); |
| 5507 | } |
| 5508 | |
| 5509 | stats.fractionLost = fraction_lost; |
| 5510 | stats.cumulativeLost = cum_lost; |
| 5511 | stats.extendedMax = ext_max; |
| 5512 | stats.jitterSamples = jitter; |
| 5513 | |
| 5514 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5515 | VoEId(_instanceId, _channelId), |
| 5516 | "GetRTPStatistics() => fractionLost=%lu, cumulativeLost=%lu," |
| 5517 | " extendedMax=%lu, jitterSamples=%li)", |
| 5518 | stats.fractionLost, stats.cumulativeLost, stats.extendedMax, |
| 5519 | stats.jitterSamples); |
| 5520 | |
| 5521 | // --- Part two of the final structure (one value) |
| 5522 | |
| 5523 | WebRtc_UWord16 RTT(0); |
| 5524 | RTCPMethod method = _rtpRtcpModule->RTCP(); |
| 5525 | if (method == kRtcpOff) |
| 5526 | { |
| 5527 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 5528 | VoEId(_instanceId, _channelId), |
| 5529 | "GetRTPStatistics() RTCP is disabled => valid RTT " |
| 5530 | "measurements cannot be retrieved"); |
| 5531 | } else |
| 5532 | { |
| 5533 | // The remote SSRC will be zero if no RTP packet has been received. |
| 5534 | WebRtc_UWord32 remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
| 5535 | if (remoteSSRC > 0) |
| 5536 | { |
| 5537 | WebRtc_UWord16 avgRTT(0); |
| 5538 | WebRtc_UWord16 maxRTT(0); |
| 5539 | WebRtc_UWord16 minRTT(0); |
| 5540 | |
| 5541 | if (_rtpRtcpModule->RTT(remoteSSRC, &RTT, &avgRTT, &minRTT, &maxRTT) |
| 5542 | != 0) |
| 5543 | { |
| 5544 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 5545 | VoEId(_instanceId, _channelId), |
| 5546 | "GetRTPStatistics() failed to retrieve RTT from " |
| 5547 | "the RTP/RTCP module"); |
| 5548 | } |
| 5549 | } else |
| 5550 | { |
| 5551 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 5552 | VoEId(_instanceId, _channelId), |
| 5553 | "GetRTPStatistics() failed to measure RTT since no " |
| 5554 | "RTP packets have been received yet"); |
| 5555 | } |
| 5556 | } |
| 5557 | |
| 5558 | stats.rttMs = static_cast<int> (RTT); |
| 5559 | |
| 5560 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5561 | VoEId(_instanceId, _channelId), |
| 5562 | "GetRTPStatistics() => rttMs=%d", stats.rttMs); |
| 5563 | |
| 5564 | // --- Part three of the final structure (four values) |
| 5565 | |
| 5566 | WebRtc_UWord32 bytesSent(0); |
| 5567 | WebRtc_UWord32 packetsSent(0); |
| 5568 | WebRtc_UWord32 bytesReceived(0); |
| 5569 | WebRtc_UWord32 packetsReceived(0); |
| 5570 | |
| 5571 | if (_rtpRtcpModule->DataCountersRTP(&bytesSent, |
| 5572 | &packetsSent, |
| 5573 | &bytesReceived, |
| 5574 | &packetsReceived) != 0) |
| 5575 | { |
| 5576 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 5577 | VoEId(_instanceId, _channelId), |
| 5578 | "GetRTPStatistics() failed to retrieve RTP datacounters =>" |
| 5579 | " output will not be complete"); |
| 5580 | } |
| 5581 | |
| 5582 | stats.bytesSent = bytesSent; |
| 5583 | stats.packetsSent = packetsSent; |
| 5584 | stats.bytesReceived = bytesReceived; |
| 5585 | stats.packetsReceived = packetsReceived; |
| 5586 | |
| 5587 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5588 | VoEId(_instanceId, _channelId), |
| 5589 | "GetRTPStatistics() => bytesSent=%d, packetsSent=%d," |
| 5590 | " bytesReceived=%d, packetsReceived=%d)", |
| 5591 | stats.bytesSent, stats.packetsSent, stats.bytesReceived, |
| 5592 | stats.packetsReceived); |
| 5593 | |
| 5594 | return 0; |
| 5595 | } |
| 5596 | |
turaj@webrtc.org | 7db5290 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 5597 | int Channel::SetFECStatus(bool enable, int redPayloadtype) { |
| 5598 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 5599 | "Channel::SetFECStatus()"); |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 5600 | |
turaj@webrtc.org | 040f800 | 2013-01-31 18:20:17 +0000 | [diff] [blame] | 5601 | if (enable) { |
| 5602 | if (redPayloadtype < 0 || redPayloadtype > 127) { |
| 5603 | _engineStatisticsPtr->SetLastError( |
| 5604 | VE_PLTYPE_ERROR, kTraceError, |
| 5605 | "SetFECStatus() invalid RED payload type"); |
| 5606 | return -1; |
| 5607 | } |
| 5608 | |
| 5609 | if (SetRedPayloadType(redPayloadtype) < 0) { |
| 5610 | _engineStatisticsPtr->SetLastError( |
| 5611 | VE_CODEC_ERROR, kTraceError, |
| 5612 | "SetSecondarySendCodec() Failed to register RED ACM"); |
| 5613 | return -1; |
| 5614 | } |
turaj@webrtc.org | 7db5290 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 5615 | } |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 5616 | |
turaj@webrtc.org | 7db5290 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 5617 | if (_audioCodingModule.SetFECStatus(enable) != 0) { |
| 5618 | _engineStatisticsPtr->SetLastError( |
| 5619 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 5620 | "SetFECStatus() failed to set FEC state in the ACM"); |
| 5621 | return -1; |
| 5622 | } |
| 5623 | return 0; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 5624 | } |
| 5625 | |
| 5626 | int |
| 5627 | Channel::GetFECStatus(bool& enabled, int& redPayloadtype) |
| 5628 | { |
| 5629 | enabled = _audioCodingModule.FECStatus(); |
| 5630 | if (enabled) |
| 5631 | { |
| 5632 | WebRtc_Word8 payloadType(0); |
| 5633 | if (_rtpRtcpModule->SendREDPayloadType(payloadType) != 0) |
| 5634 | { |
| 5635 | _engineStatisticsPtr->SetLastError( |
| 5636 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5637 | "GetFECStatus() failed to retrieve RED PT from RTP/RTCP " |
| 5638 | "module"); |
| 5639 | return -1; |
| 5640 | } |
| 5641 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5642 | VoEId(_instanceId, _channelId), |
| 5643 | "GetFECStatus() => enabled=%d, redPayloadtype=%d", |
| 5644 | enabled, redPayloadtype); |
| 5645 | return 0; |
| 5646 | } |
| 5647 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 5648 | VoEId(_instanceId, _channelId), |
| 5649 | "GetFECStatus() => enabled=%d", enabled); |
| 5650 | return 0; |
| 5651 | } |
| 5652 | |
| 5653 | int |
| 5654 | Channel::StartRTPDump(const char fileNameUTF8[1024], |
| 5655 | RTPDirections direction) |
| 5656 | { |
| 5657 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 5658 | "Channel::StartRTPDump()"); |
| 5659 | if ((direction != kRtpIncoming) && (direction != kRtpOutgoing)) |
| 5660 | { |
| 5661 | _engineStatisticsPtr->SetLastError( |
| 5662 | VE_INVALID_ARGUMENT, kTraceError, |
| 5663 | "StartRTPDump() invalid RTP direction"); |
| 5664 | return -1; |
| 5665 | } |
| 5666 | RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ? |
| 5667 | &_rtpDumpIn : &_rtpDumpOut; |
| 5668 | if (rtpDumpPtr == NULL) |
| 5669 | { |
| 5670 | assert(false); |
| 5671 | return -1; |
| 5672 | } |
| 5673 | if (rtpDumpPtr->IsActive()) |
| 5674 | { |
| 5675 | rtpDumpPtr->Stop(); |
| 5676 | } |
| 5677 | if (rtpDumpPtr->Start(fileNameUTF8) != 0) |
| 5678 | { |
| 5679 | _engineStatisticsPtr->SetLastError( |
| 5680 | VE_BAD_FILE, kTraceError, |
| 5681 | "StartRTPDump() failed to create file"); |
| 5682 | return -1; |
| 5683 | } |
| 5684 | return 0; |
| 5685 | } |
| 5686 | |
| 5687 | int |
| 5688 | Channel::StopRTPDump(RTPDirections direction) |
| 5689 | { |
| 5690 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 5691 | "Channel::StopRTPDump()"); |
| 5692 | if ((direction != kRtpIncoming) && (direction != kRtpOutgoing)) |
| 5693 | { |
| 5694 | _engineStatisticsPtr->SetLastError( |
| 5695 | VE_INVALID_ARGUMENT, kTraceError, |
| 5696 | "StopRTPDump() invalid RTP direction"); |
| 5697 | return -1; |
| 5698 | } |
| 5699 | RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ? |
| 5700 | &_rtpDumpIn : &_rtpDumpOut; |
| 5701 | if (rtpDumpPtr == NULL) |
| 5702 | { |
| 5703 | assert(false); |
| 5704 | return -1; |
| 5705 | } |
| 5706 | if (!rtpDumpPtr->IsActive()) |
| 5707 | { |
| 5708 | return 0; |
| 5709 | } |
| 5710 | return rtpDumpPtr->Stop(); |
| 5711 | } |
| 5712 | |
| 5713 | bool |
| 5714 | Channel::RTPDumpIsActive(RTPDirections direction) |
| 5715 | { |
| 5716 | if ((direction != kRtpIncoming) && |
| 5717 | (direction != kRtpOutgoing)) |
| 5718 | { |
| 5719 | _engineStatisticsPtr->SetLastError( |
| 5720 | VE_INVALID_ARGUMENT, kTraceError, |
| 5721 | "RTPDumpIsActive() invalid RTP direction"); |
| 5722 | return false; |
| 5723 | } |
| 5724 | RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ? |
| 5725 | &_rtpDumpIn : &_rtpDumpOut; |
| 5726 | return rtpDumpPtr->IsActive(); |
| 5727 | } |
| 5728 | |
| 5729 | int |
| 5730 | Channel::InsertExtraRTPPacket(unsigned char payloadType, |
| 5731 | bool markerBit, |
| 5732 | const char* payloadData, |
| 5733 | unsigned short payloadSize) |
| 5734 | { |
| 5735 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 5736 | "Channel::InsertExtraRTPPacket()"); |
| 5737 | if (payloadType > 127) |
| 5738 | { |
| 5739 | _engineStatisticsPtr->SetLastError( |
| 5740 | VE_INVALID_PLTYPE, kTraceError, |
| 5741 | "InsertExtraRTPPacket() invalid payload type"); |
| 5742 | return -1; |
| 5743 | } |
| 5744 | if (payloadData == NULL) |
| 5745 | { |
| 5746 | _engineStatisticsPtr->SetLastError( |
| 5747 | VE_INVALID_ARGUMENT, kTraceError, |
| 5748 | "InsertExtraRTPPacket() invalid payload data"); |
| 5749 | return -1; |
| 5750 | } |
| 5751 | if (payloadSize > _rtpRtcpModule->MaxDataPayloadLength()) |
| 5752 | { |
| 5753 | _engineStatisticsPtr->SetLastError( |
| 5754 | VE_INVALID_ARGUMENT, kTraceError, |
| 5755 | "InsertExtraRTPPacket() invalid payload size"); |
| 5756 | return -1; |
| 5757 | } |
| 5758 | if (!_sending) |
| 5759 | { |
| 5760 | _engineStatisticsPtr->SetLastError( |
| 5761 | VE_NOT_SENDING, kTraceError, |
| 5762 | "InsertExtraRTPPacket() not sending"); |
| 5763 | return -1; |
| 5764 | } |
| 5765 | |
| 5766 | // Create extra RTP packet by calling RtpRtcp::SendOutgoingData(). |
| 5767 | // Transport::SendPacket() will be called by the module when the RTP packet |
| 5768 | // is created. |
| 5769 | // The call to SendOutgoingData() does *not* modify the timestamp and |
| 5770 | // payloadtype to ensure that the RTP module generates a valid RTP packet |
| 5771 | // (user might utilize a non-registered payload type). |
| 5772 | // The marker bit and payload type will be replaced just before the actual |
| 5773 | // transmission, i.e., the actual modification is done *after* the RTP |
| 5774 | // module has delivered its RTP packet back to the VoE. |
| 5775 | // We will use the stored values above when the packet is modified |
| 5776 | // (see Channel::SendPacket()). |
| 5777 | |
| 5778 | _extraPayloadType = payloadType; |
| 5779 | _extraMarkerBit = markerBit; |
| 5780 | _insertExtraRTPPacket = true; |
| 5781 | |
| 5782 | if (_rtpRtcpModule->SendOutgoingData(kAudioFrameSpeech, |
| 5783 | _lastPayloadType, |
| 5784 | _lastLocalTimeStamp, |
| 5785 | // Leaving the time when this frame was |
| 5786 | // received from the capture device as |
| 5787 | // undefined for voice for now. |
| 5788 | -1, |
| 5789 | (const WebRtc_UWord8*) payloadData, |
| 5790 | payloadSize) != 0) |
| 5791 | { |
| 5792 | _engineStatisticsPtr->SetLastError( |
| 5793 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5794 | "InsertExtraRTPPacket() failed to send extra RTP packet"); |
| 5795 | return -1; |
| 5796 | } |
| 5797 | |
| 5798 | return 0; |
| 5799 | } |
| 5800 | |
| 5801 | WebRtc_UWord32 |
| 5802 | Channel::Demultiplex(const AudioFrame& audioFrame) |
| 5803 | { |
| 5804 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5805 | "Channel::Demultiplex()"); |
andrew@webrtc.org | d468236 | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 5806 | _audioFrame.CopyFrom(audioFrame); |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 5807 | _audioFrame.id_ = _channelId; |
| 5808 | return 0; |
| 5809 | } |
| 5810 | |
| 5811 | WebRtc_UWord32 |
| 5812 | Channel::PrepareEncodeAndSend(int mixingFrequency) |
| 5813 | { |
| 5814 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5815 | "Channel::PrepareEncodeAndSend()"); |
| 5816 | |
| 5817 | if (_audioFrame.samples_per_channel_ == 0) |
| 5818 | { |
| 5819 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5820 | "Channel::PrepareEncodeAndSend() invalid audio frame"); |
| 5821 | return -1; |
| 5822 | } |
| 5823 | |
| 5824 | if (_inputFilePlaying) |
| 5825 | { |
| 5826 | MixOrReplaceAudioWithFile(mixingFrequency); |
| 5827 | } |
| 5828 | |
| 5829 | if (_mute) |
| 5830 | { |
| 5831 | AudioFrameOperations::Mute(_audioFrame); |
| 5832 | } |
| 5833 | |
| 5834 | if (_inputExternalMedia) |
| 5835 | { |
| 5836 | CriticalSectionScoped cs(&_callbackCritSect); |
| 5837 | const bool isStereo = (_audioFrame.num_channels_ == 2); |
| 5838 | if (_inputExternalMediaCallbackPtr) |
| 5839 | { |
| 5840 | _inputExternalMediaCallbackPtr->Process( |
| 5841 | _channelId, |
| 5842 | kRecordingPerChannel, |
| 5843 | (WebRtc_Word16*)_audioFrame.data_, |
| 5844 | _audioFrame.samples_per_channel_, |
| 5845 | _audioFrame.sample_rate_hz_, |
| 5846 | isStereo); |
| 5847 | } |
| 5848 | } |
| 5849 | |
| 5850 | InsertInbandDtmfTone(); |
| 5851 | |
| 5852 | if (_includeAudioLevelIndication) |
| 5853 | { |
| 5854 | assert(_rtpAudioProc.get() != NULL); |
| 5855 | |
| 5856 | // Check if settings need to be updated. |
| 5857 | if (_rtpAudioProc->sample_rate_hz() != _audioFrame.sample_rate_hz_) |
| 5858 | { |
| 5859 | if (_rtpAudioProc->set_sample_rate_hz(_audioFrame.sample_rate_hz_) != |
| 5860 | AudioProcessing::kNoError) |
| 5861 | { |
| 5862 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 5863 | VoEId(_instanceId, _channelId), |
| 5864 | "Error setting AudioProcessing sample rate"); |
| 5865 | return -1; |
| 5866 | } |
| 5867 | } |
| 5868 | |
| 5869 | if (_rtpAudioProc->num_input_channels() != _audioFrame.num_channels_) |
| 5870 | { |
| 5871 | if (_rtpAudioProc->set_num_channels(_audioFrame.num_channels_, |
| 5872 | _audioFrame.num_channels_) |
| 5873 | != AudioProcessing::kNoError) |
| 5874 | { |
| 5875 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 5876 | VoEId(_instanceId, _channelId), |
| 5877 | "Error setting AudioProcessing channels"); |
| 5878 | return -1; |
| 5879 | } |
| 5880 | } |
| 5881 | |
| 5882 | // Performs level analysis only; does not affect the signal. |
| 5883 | _rtpAudioProc->ProcessStream(&_audioFrame); |
| 5884 | } |
| 5885 | |
| 5886 | return 0; |
| 5887 | } |
| 5888 | |
| 5889 | WebRtc_UWord32 |
| 5890 | Channel::EncodeAndSend() |
| 5891 | { |
| 5892 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5893 | "Channel::EncodeAndSend()"); |
| 5894 | |
| 5895 | assert(_audioFrame.num_channels_ <= 2); |
| 5896 | if (_audioFrame.samples_per_channel_ == 0) |
| 5897 | { |
| 5898 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5899 | "Channel::EncodeAndSend() invalid audio frame"); |
| 5900 | return -1; |
| 5901 | } |
| 5902 | |
| 5903 | _audioFrame.id_ = _channelId; |
| 5904 | |
| 5905 | // --- Add 10ms of raw (PCM) audio data to the encoder @ 32kHz. |
| 5906 | |
| 5907 | // The ACM resamples internally. |
| 5908 | _audioFrame.timestamp_ = _timeStamp; |
| 5909 | if (_audioCodingModule.Add10MsData((AudioFrame&)_audioFrame) != 0) |
| 5910 | { |
| 5911 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5912 | "Channel::EncodeAndSend() ACM encoding failed"); |
| 5913 | return -1; |
| 5914 | } |
| 5915 | |
| 5916 | _timeStamp += _audioFrame.samples_per_channel_; |
| 5917 | |
| 5918 | // --- Encode if complete frame is ready |
| 5919 | |
| 5920 | // This call will trigger AudioPacketizationCallback::SendData if encoding |
| 5921 | // is done and payload is ready for packetization and transmission. |
| 5922 | return _audioCodingModule.Process(); |
| 5923 | } |
| 5924 | |
| 5925 | int Channel::RegisterExternalMediaProcessing( |
| 5926 | ProcessingTypes type, |
| 5927 | VoEMediaProcess& processObject) |
| 5928 | { |
| 5929 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5930 | "Channel::RegisterExternalMediaProcessing()"); |
| 5931 | |
| 5932 | CriticalSectionScoped cs(&_callbackCritSect); |
| 5933 | |
| 5934 | if (kPlaybackPerChannel == type) |
| 5935 | { |
| 5936 | if (_outputExternalMediaCallbackPtr) |
| 5937 | { |
| 5938 | _engineStatisticsPtr->SetLastError( |
| 5939 | VE_INVALID_OPERATION, kTraceError, |
| 5940 | "Channel::RegisterExternalMediaProcessing() " |
| 5941 | "output external media already enabled"); |
| 5942 | return -1; |
| 5943 | } |
| 5944 | _outputExternalMediaCallbackPtr = &processObject; |
| 5945 | _outputExternalMedia = true; |
| 5946 | } |
| 5947 | else if (kRecordingPerChannel == type) |
| 5948 | { |
| 5949 | if (_inputExternalMediaCallbackPtr) |
| 5950 | { |
| 5951 | _engineStatisticsPtr->SetLastError( |
| 5952 | VE_INVALID_OPERATION, kTraceError, |
| 5953 | "Channel::RegisterExternalMediaProcessing() " |
| 5954 | "output external media already enabled"); |
| 5955 | return -1; |
| 5956 | } |
| 5957 | _inputExternalMediaCallbackPtr = &processObject; |
| 5958 | _inputExternalMedia = true; |
| 5959 | } |
| 5960 | return 0; |
| 5961 | } |
| 5962 | |
| 5963 | int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type) |
| 5964 | { |
| 5965 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5966 | "Channel::DeRegisterExternalMediaProcessing()"); |
| 5967 | |
| 5968 | CriticalSectionScoped cs(&_callbackCritSect); |
| 5969 | |
| 5970 | if (kPlaybackPerChannel == type) |
| 5971 | { |
| 5972 | if (!_outputExternalMediaCallbackPtr) |
| 5973 | { |
| 5974 | _engineStatisticsPtr->SetLastError( |
| 5975 | VE_INVALID_OPERATION, kTraceWarning, |
| 5976 | "Channel::DeRegisterExternalMediaProcessing() " |
| 5977 | "output external media already disabled"); |
| 5978 | return 0; |
| 5979 | } |
| 5980 | _outputExternalMedia = false; |
| 5981 | _outputExternalMediaCallbackPtr = NULL; |
| 5982 | } |
| 5983 | else if (kRecordingPerChannel == type) |
| 5984 | { |
| 5985 | if (!_inputExternalMediaCallbackPtr) |
| 5986 | { |
| 5987 | _engineStatisticsPtr->SetLastError( |
| 5988 | VE_INVALID_OPERATION, kTraceWarning, |
| 5989 | "Channel::DeRegisterExternalMediaProcessing() " |
| 5990 | "input external media already disabled"); |
| 5991 | return 0; |
| 5992 | } |
| 5993 | _inputExternalMedia = false; |
| 5994 | _inputExternalMediaCallbackPtr = NULL; |
| 5995 | } |
| 5996 | |
| 5997 | return 0; |
| 5998 | } |
| 5999 | |
roosa@google.com | b9e3afc | 2012-12-12 23:00:29 +0000 | [diff] [blame] | 6000 | int Channel::SetExternalMixing(bool enabled) { |
| 6001 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6002 | "Channel::SetExternalMixing(enabled=%d)", enabled); |
| 6003 | |
| 6004 | if (_playing) |
| 6005 | { |
| 6006 | _engineStatisticsPtr->SetLastError( |
| 6007 | VE_INVALID_OPERATION, kTraceError, |
| 6008 | "Channel::SetExternalMixing() " |
| 6009 | "external mixing cannot be changed while playing."); |
| 6010 | return -1; |
| 6011 | } |
| 6012 | |
| 6013 | _externalMixing = enabled; |
| 6014 | |
| 6015 | return 0; |
| 6016 | } |
| 6017 | |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6018 | int |
| 6019 | Channel::ResetRTCPStatistics() |
| 6020 | { |
| 6021 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6022 | "Channel::ResetRTCPStatistics()"); |
| 6023 | WebRtc_UWord32 remoteSSRC(0); |
| 6024 | remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
| 6025 | return _rtpRtcpModule->ResetRTT(remoteSSRC); |
| 6026 | } |
| 6027 | |
| 6028 | int |
| 6029 | Channel::GetRoundTripTimeSummary(StatVal& delaysMs) const |
| 6030 | { |
| 6031 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6032 | "Channel::GetRoundTripTimeSummary()"); |
| 6033 | // Override default module outputs for the case when RTCP is disabled. |
| 6034 | // This is done to ensure that we are backward compatible with the |
| 6035 | // VoiceEngine where we did not use RTP/RTCP module. |
| 6036 | if (!_rtpRtcpModule->RTCP()) |
| 6037 | { |
| 6038 | delaysMs.min = -1; |
| 6039 | delaysMs.max = -1; |
| 6040 | delaysMs.average = -1; |
| 6041 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6042 | "Channel::GetRoundTripTimeSummary() RTCP is disabled =>" |
| 6043 | " valid RTT measurements cannot be retrieved"); |
| 6044 | return 0; |
| 6045 | } |
| 6046 | |
| 6047 | WebRtc_UWord32 remoteSSRC; |
| 6048 | WebRtc_UWord16 RTT; |
| 6049 | WebRtc_UWord16 avgRTT; |
| 6050 | WebRtc_UWord16 maxRTT; |
| 6051 | WebRtc_UWord16 minRTT; |
| 6052 | // The remote SSRC will be zero if no RTP packet has been received. |
| 6053 | remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
| 6054 | if (remoteSSRC == 0) |
| 6055 | { |
| 6056 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6057 | "Channel::GetRoundTripTimeSummary() unable to measure RTT" |
| 6058 | " since no RTP packet has been received yet"); |
| 6059 | } |
| 6060 | |
| 6061 | // Retrieve RTT statistics from the RTP/RTCP module for the specified |
| 6062 | // channel and SSRC. The SSRC is required to parse out the correct source |
| 6063 | // in conference scenarios. |
| 6064 | if (_rtpRtcpModule->RTT(remoteSSRC, &RTT, &avgRTT, &minRTT,&maxRTT) != 0) |
| 6065 | { |
| 6066 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6067 | "GetRoundTripTimeSummary unable to retrieve RTT values" |
| 6068 | " from the RTCP layer"); |
| 6069 | delaysMs.min = -1; delaysMs.max = -1; delaysMs.average = -1; |
| 6070 | } |
| 6071 | else |
| 6072 | { |
| 6073 | delaysMs.min = minRTT; |
| 6074 | delaysMs.max = maxRTT; |
| 6075 | delaysMs.average = avgRTT; |
| 6076 | } |
| 6077 | return 0; |
| 6078 | } |
| 6079 | |
| 6080 | int |
| 6081 | Channel::GetNetworkStatistics(NetworkStatistics& stats) |
| 6082 | { |
| 6083 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6084 | "Channel::GetNetworkStatistics()"); |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 6085 | ACMNetworkStatistics acm_stats; |
| 6086 | int return_value = _audioCodingModule.NetworkStatistics(&acm_stats); |
| 6087 | if (return_value >= 0) { |
| 6088 | memcpy(&stats, &acm_stats, sizeof(NetworkStatistics)); |
| 6089 | } |
| 6090 | return return_value; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6091 | } |
| 6092 | |
| 6093 | int |
| 6094 | Channel::GetDelayEstimate(int& delayMs) const |
| 6095 | { |
| 6096 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6097 | "Channel::GetDelayEstimate()"); |
| 6098 | delayMs = (_averageDelayMs + 5) / 10 + _recPacketDelayMs; |
| 6099 | return 0; |
| 6100 | } |
| 6101 | |
turaj@webrtc.org | ead8a5b | 2013-02-12 21:42:18 +0000 | [diff] [blame] | 6102 | int Channel::SetInitialPlayoutDelay(int delay_ms) |
| 6103 | { |
| 6104 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6105 | "Channel::SetInitialPlayoutDelay()"); |
| 6106 | if ((delay_ms < kVoiceEngineMinMinPlayoutDelayMs) || |
| 6107 | (delay_ms > kVoiceEngineMaxMinPlayoutDelayMs)) |
| 6108 | { |
| 6109 | _engineStatisticsPtr->SetLastError( |
| 6110 | VE_INVALID_ARGUMENT, kTraceError, |
| 6111 | "SetInitialPlayoutDelay() invalid min delay"); |
| 6112 | return -1; |
| 6113 | } |
| 6114 | if (_audioCodingModule.SetInitialPlayoutDelay(delay_ms) != 0) |
| 6115 | { |
| 6116 | _engineStatisticsPtr->SetLastError( |
| 6117 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 6118 | "SetInitialPlayoutDelay() failed to set min playout delay"); |
| 6119 | return -1; |
| 6120 | } |
| 6121 | return 0; |
| 6122 | } |
| 6123 | |
| 6124 | |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6125 | int |
| 6126 | Channel::SetMinimumPlayoutDelay(int delayMs) |
| 6127 | { |
| 6128 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6129 | "Channel::SetMinimumPlayoutDelay()"); |
| 6130 | if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) || |
| 6131 | (delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) |
| 6132 | { |
| 6133 | _engineStatisticsPtr->SetLastError( |
| 6134 | VE_INVALID_ARGUMENT, kTraceError, |
| 6135 | "SetMinimumPlayoutDelay() invalid min delay"); |
| 6136 | return -1; |
| 6137 | } |
| 6138 | if (_audioCodingModule.SetMinimumPlayoutDelay(delayMs) != 0) |
| 6139 | { |
| 6140 | _engineStatisticsPtr->SetLastError( |
| 6141 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 6142 | "SetMinimumPlayoutDelay() failed to set min playout delay"); |
| 6143 | return -1; |
| 6144 | } |
| 6145 | return 0; |
| 6146 | } |
| 6147 | |
| 6148 | int |
| 6149 | Channel::GetPlayoutTimestamp(unsigned int& timestamp) |
| 6150 | { |
| 6151 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6152 | "Channel::GetPlayoutTimestamp()"); |
| 6153 | WebRtc_UWord32 playoutTimestamp(0); |
| 6154 | if (GetPlayoutTimeStamp(playoutTimestamp) != 0) |
| 6155 | { |
| 6156 | _engineStatisticsPtr->SetLastError( |
| 6157 | VE_CANNOT_RETRIEVE_VALUE, kTraceError, |
| 6158 | "GetPlayoutTimestamp() failed to retrieve timestamp"); |
| 6159 | return -1; |
| 6160 | } |
| 6161 | timestamp = playoutTimestamp; |
| 6162 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 6163 | VoEId(_instanceId,_channelId), |
| 6164 | "GetPlayoutTimestamp() => timestamp=%u", timestamp); |
| 6165 | return 0; |
| 6166 | } |
| 6167 | |
| 6168 | int |
| 6169 | Channel::SetInitTimestamp(unsigned int timestamp) |
| 6170 | { |
| 6171 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6172 | "Channel::SetInitTimestamp()"); |
| 6173 | if (_sending) |
| 6174 | { |
| 6175 | _engineStatisticsPtr->SetLastError( |
| 6176 | VE_SENDING, kTraceError, "SetInitTimestamp() already sending"); |
| 6177 | return -1; |
| 6178 | } |
| 6179 | if (_rtpRtcpModule->SetStartTimestamp(timestamp) != 0) |
| 6180 | { |
| 6181 | _engineStatisticsPtr->SetLastError( |
| 6182 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 6183 | "SetInitTimestamp() failed to set timestamp"); |
| 6184 | return -1; |
| 6185 | } |
| 6186 | return 0; |
| 6187 | } |
| 6188 | |
| 6189 | int |
| 6190 | Channel::SetInitSequenceNumber(short sequenceNumber) |
| 6191 | { |
| 6192 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6193 | "Channel::SetInitSequenceNumber()"); |
| 6194 | if (_sending) |
| 6195 | { |
| 6196 | _engineStatisticsPtr->SetLastError( |
| 6197 | VE_SENDING, kTraceError, |
| 6198 | "SetInitSequenceNumber() already sending"); |
| 6199 | return -1; |
| 6200 | } |
| 6201 | if (_rtpRtcpModule->SetSequenceNumber(sequenceNumber) != 0) |
| 6202 | { |
| 6203 | _engineStatisticsPtr->SetLastError( |
| 6204 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 6205 | "SetInitSequenceNumber() failed to set sequence number"); |
| 6206 | return -1; |
| 6207 | } |
| 6208 | return 0; |
| 6209 | } |
| 6210 | |
| 6211 | int |
| 6212 | Channel::GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const |
| 6213 | { |
| 6214 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6215 | "Channel::GetRtpRtcp()"); |
| 6216 | rtpRtcpModule = _rtpRtcpModule.get(); |
| 6217 | return 0; |
| 6218 | } |
| 6219 | |
| 6220 | // TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use |
| 6221 | // a shared helper. |
| 6222 | WebRtc_Word32 |
| 6223 | Channel::MixOrReplaceAudioWithFile(const int mixingFrequency) |
| 6224 | { |
| 6225 | scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]); |
| 6226 | int fileSamples(0); |
| 6227 | |
| 6228 | { |
| 6229 | CriticalSectionScoped cs(&_fileCritSect); |
| 6230 | |
| 6231 | if (_inputFilePlayerPtr == NULL) |
| 6232 | { |
| 6233 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 6234 | VoEId(_instanceId, _channelId), |
| 6235 | "Channel::MixOrReplaceAudioWithFile() fileplayer" |
| 6236 | " doesnt exist"); |
| 6237 | return -1; |
| 6238 | } |
| 6239 | |
| 6240 | if (_inputFilePlayerPtr->Get10msAudioFromFile(fileBuffer.get(), |
| 6241 | fileSamples, |
| 6242 | mixingFrequency) == -1) |
| 6243 | { |
| 6244 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 6245 | VoEId(_instanceId, _channelId), |
| 6246 | "Channel::MixOrReplaceAudioWithFile() file mixing " |
| 6247 | "failed"); |
| 6248 | return -1; |
| 6249 | } |
| 6250 | if (fileSamples == 0) |
| 6251 | { |
| 6252 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 6253 | VoEId(_instanceId, _channelId), |
| 6254 | "Channel::MixOrReplaceAudioWithFile() file is ended"); |
| 6255 | return 0; |
| 6256 | } |
| 6257 | } |
| 6258 | |
| 6259 | assert(_audioFrame.samples_per_channel_ == fileSamples); |
| 6260 | |
| 6261 | if (_mixFileWithMicrophone) |
| 6262 | { |
| 6263 | // Currently file stream is always mono. |
| 6264 | // TODO(xians): Change the code when FilePlayer supports real stereo. |
| 6265 | Utility::MixWithSat(_audioFrame.data_, |
| 6266 | _audioFrame.num_channels_, |
| 6267 | fileBuffer.get(), |
| 6268 | 1, |
| 6269 | fileSamples); |
| 6270 | } |
| 6271 | else |
| 6272 | { |
| 6273 | // Replace ACM audio with file. |
| 6274 | // Currently file stream is always mono. |
| 6275 | // TODO(xians): Change the code when FilePlayer supports real stereo. |
| 6276 | _audioFrame.UpdateFrame(_channelId, |
| 6277 | -1, |
| 6278 | fileBuffer.get(), |
| 6279 | fileSamples, |
| 6280 | mixingFrequency, |
| 6281 | AudioFrame::kNormalSpeech, |
| 6282 | AudioFrame::kVadUnknown, |
| 6283 | 1); |
| 6284 | |
| 6285 | } |
| 6286 | return 0; |
| 6287 | } |
| 6288 | |
| 6289 | WebRtc_Word32 |
| 6290 | Channel::MixAudioWithFile(AudioFrame& audioFrame, |
| 6291 | const int mixingFrequency) |
| 6292 | { |
| 6293 | assert(mixingFrequency <= 32000); |
| 6294 | |
| 6295 | scoped_array<WebRtc_Word16> fileBuffer(new WebRtc_Word16[640]); |
| 6296 | int fileSamples(0); |
| 6297 | |
| 6298 | { |
| 6299 | CriticalSectionScoped cs(&_fileCritSect); |
| 6300 | |
| 6301 | if (_outputFilePlayerPtr == NULL) |
| 6302 | { |
| 6303 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 6304 | VoEId(_instanceId, _channelId), |
| 6305 | "Channel::MixAudioWithFile() file mixing failed"); |
| 6306 | return -1; |
| 6307 | } |
| 6308 | |
| 6309 | // We should get the frequency we ask for. |
| 6310 | if (_outputFilePlayerPtr->Get10msAudioFromFile(fileBuffer.get(), |
| 6311 | fileSamples, |
| 6312 | mixingFrequency) == -1) |
| 6313 | { |
| 6314 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 6315 | VoEId(_instanceId, _channelId), |
| 6316 | "Channel::MixAudioWithFile() file mixing failed"); |
| 6317 | return -1; |
| 6318 | } |
| 6319 | } |
| 6320 | |
| 6321 | if (audioFrame.samples_per_channel_ == fileSamples) |
| 6322 | { |
| 6323 | // Currently file stream is always mono. |
| 6324 | // TODO(xians): Change the code when FilePlayer supports real stereo. |
| 6325 | Utility::MixWithSat(audioFrame.data_, |
| 6326 | audioFrame.num_channels_, |
| 6327 | fileBuffer.get(), |
| 6328 | 1, |
| 6329 | fileSamples); |
| 6330 | } |
| 6331 | else |
| 6332 | { |
| 6333 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6334 | "Channel::MixAudioWithFile() samples_per_channel_(%d) != " |
| 6335 | "fileSamples(%d)", |
| 6336 | audioFrame.samples_per_channel_, fileSamples); |
| 6337 | return -1; |
| 6338 | } |
| 6339 | |
| 6340 | return 0; |
| 6341 | } |
| 6342 | |
| 6343 | int |
| 6344 | Channel::InsertInbandDtmfTone() |
| 6345 | { |
| 6346 | // Check if we should start a new tone. |
| 6347 | if (_inbandDtmfQueue.PendingDtmf() && |
| 6348 | !_inbandDtmfGenerator.IsAddingTone() && |
| 6349 | _inbandDtmfGenerator.DelaySinceLastTone() > |
| 6350 | kMinTelephoneEventSeparationMs) |
| 6351 | { |
| 6352 | WebRtc_Word8 eventCode(0); |
| 6353 | WebRtc_UWord16 lengthMs(0); |
| 6354 | WebRtc_UWord8 attenuationDb(0); |
| 6355 | |
| 6356 | eventCode = _inbandDtmfQueue.NextDtmf(&lengthMs, &attenuationDb); |
| 6357 | _inbandDtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb); |
| 6358 | if (_playInbandDtmfEvent) |
| 6359 | { |
| 6360 | // Add tone to output mixer using a reduced length to minimize |
| 6361 | // risk of echo. |
| 6362 | _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs - 80, |
| 6363 | attenuationDb); |
| 6364 | } |
| 6365 | } |
| 6366 | |
| 6367 | if (_inbandDtmfGenerator.IsAddingTone()) |
| 6368 | { |
| 6369 | WebRtc_UWord16 frequency(0); |
| 6370 | _inbandDtmfGenerator.GetSampleRate(frequency); |
| 6371 | |
| 6372 | if (frequency != _audioFrame.sample_rate_hz_) |
| 6373 | { |
| 6374 | // Update sample rate of Dtmf tone since the mixing frequency |
| 6375 | // has changed. |
| 6376 | _inbandDtmfGenerator.SetSampleRate( |
| 6377 | (WebRtc_UWord16) (_audioFrame.sample_rate_hz_)); |
| 6378 | // Reset the tone to be added taking the new sample rate into |
| 6379 | // account. |
| 6380 | _inbandDtmfGenerator.ResetTone(); |
| 6381 | } |
andrew@webrtc.org | d468236 | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 6382 | |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6383 | WebRtc_Word16 toneBuffer[320]; |
| 6384 | WebRtc_UWord16 toneSamples(0); |
| 6385 | // Get 10ms tone segment and set time since last tone to zero |
| 6386 | if (_inbandDtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1) |
| 6387 | { |
| 6388 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 6389 | VoEId(_instanceId, _channelId), |
| 6390 | "Channel::EncodeAndSend() inserting Dtmf failed"); |
| 6391 | return -1; |
| 6392 | } |
| 6393 | |
| 6394 | // Replace mixed audio with DTMF tone. |
andrew@webrtc.org | d468236 | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 6395 | for (int sample = 0; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6396 | sample < _audioFrame.samples_per_channel_; |
| 6397 | sample++) |
| 6398 | { |
andrew@webrtc.org | d468236 | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 6399 | for (int channel = 0; |
| 6400 | channel < _audioFrame.num_channels_; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6401 | channel++) |
| 6402 | { |
andrew@webrtc.org | d468236 | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 6403 | const int index = sample * _audioFrame.num_channels_ + channel; |
| 6404 | _audioFrame.data_[index] = toneBuffer[sample]; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6405 | } |
| 6406 | } |
andrew@webrtc.org | d468236 | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 6407 | |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6408 | assert(_audioFrame.samples_per_channel_ == toneSamples); |
| 6409 | } else |
| 6410 | { |
| 6411 | // Add 10ms to "delay-since-last-tone" counter |
| 6412 | _inbandDtmfGenerator.UpdateDelaySinceLastTone(); |
| 6413 | } |
| 6414 | return 0; |
| 6415 | } |
| 6416 | |
| 6417 | WebRtc_Word32 |
| 6418 | Channel::GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp) |
| 6419 | { |
| 6420 | WebRtc_UWord32 timestamp(0); |
| 6421 | CodecInst currRecCodec; |
| 6422 | |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 6423 | if (_audioCodingModule.PlayoutTimestamp(×tamp) == -1) |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6424 | { |
| 6425 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6426 | "Channel::GetPlayoutTimeStamp() failed to read playout" |
| 6427 | " timestamp from the ACM"); |
| 6428 | return -1; |
| 6429 | } |
| 6430 | |
| 6431 | WebRtc_UWord16 delayMS(0); |
| 6432 | if (_audioDeviceModulePtr->PlayoutDelay(&delayMS) == -1) |
| 6433 | { |
| 6434 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6435 | "Channel::GetPlayoutTimeStamp() failed to read playout" |
| 6436 | " delay from the ADM"); |
| 6437 | return -1; |
| 6438 | } |
| 6439 | |
| 6440 | WebRtc_Word32 playoutFrequency = _audioCodingModule.PlayoutFrequency(); |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 6441 | if (_audioCodingModule.ReceiveCodec(&currRecCodec) == 0) { |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6442 | if (STR_CASE_CMP("G722", currRecCodec.plname) == 0) { |
| 6443 | playoutFrequency = 8000; |
| 6444 | } else if (STR_CASE_CMP("opus", currRecCodec.plname) == 0) { |
| 6445 | playoutFrequency = 48000; |
| 6446 | } |
| 6447 | } |
| 6448 | timestamp -= (delayMS * (playoutFrequency/1000)); |
| 6449 | |
| 6450 | playoutTimestamp = timestamp; |
| 6451 | |
| 6452 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6453 | "Channel::GetPlayoutTimeStamp() => playoutTimestamp = %lu", |
| 6454 | playoutTimestamp); |
| 6455 | return 0; |
| 6456 | } |
| 6457 | |
| 6458 | void |
| 6459 | Channel::ResetDeadOrAliveCounters() |
| 6460 | { |
| 6461 | _countDeadDetections = 0; |
| 6462 | _countAliveDetections = 0; |
| 6463 | } |
| 6464 | |
| 6465 | void |
| 6466 | Channel::UpdateDeadOrAliveCounters(bool alive) |
| 6467 | { |
| 6468 | if (alive) |
| 6469 | _countAliveDetections++; |
| 6470 | else |
| 6471 | _countDeadDetections++; |
| 6472 | } |
| 6473 | |
| 6474 | int |
| 6475 | Channel::GetDeadOrAliveCounters(int& countDead, int& countAlive) const |
| 6476 | { |
| 6477 | bool enabled; |
| 6478 | WebRtc_UWord8 timeSec; |
| 6479 | |
| 6480 | _rtpRtcpModule->PeriodicDeadOrAliveStatus(enabled, timeSec); |
| 6481 | if (!enabled) |
| 6482 | return (-1); |
| 6483 | |
| 6484 | countDead = static_cast<int> (_countDeadDetections); |
| 6485 | countAlive = static_cast<int> (_countAliveDetections); |
| 6486 | return 0; |
| 6487 | } |
| 6488 | |
| 6489 | WebRtc_Word32 |
| 6490 | Channel::SendPacketRaw(const void *data, int len, bool RTCP) |
| 6491 | { |
| 6492 | if (_transportPtr == NULL) |
| 6493 | { |
| 6494 | return -1; |
| 6495 | } |
| 6496 | if (!RTCP) |
| 6497 | { |
| 6498 | return _transportPtr->SendPacket(_channelId, data, len); |
| 6499 | } |
| 6500 | else |
| 6501 | { |
| 6502 | return _transportPtr->SendRTCPPacket(_channelId, data, len); |
| 6503 | } |
| 6504 | } |
| 6505 | |
| 6506 | WebRtc_Word32 |
| 6507 | Channel::UpdatePacketDelay(const WebRtc_UWord32 timestamp, |
| 6508 | const WebRtc_UWord16 sequenceNumber) |
| 6509 | { |
| 6510 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6511 | "Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)", |
| 6512 | timestamp, sequenceNumber); |
| 6513 | |
| 6514 | WebRtc_Word32 rtpReceiveFrequency(0); |
| 6515 | |
| 6516 | // Get frequency of last received payload |
| 6517 | rtpReceiveFrequency = _audioCodingModule.ReceiveFrequency(); |
| 6518 | |
| 6519 | CodecInst currRecCodec; |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 6520 | if (_audioCodingModule.ReceiveCodec(&currRecCodec) == 0) { |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6521 | if (STR_CASE_CMP("G722", currRecCodec.plname) == 0) { |
| 6522 | // Even though the actual sampling rate for G.722 audio is |
| 6523 | // 16,000 Hz, the RTP clock rate for the G722 payload format is |
| 6524 | // 8,000 Hz because that value was erroneously assigned in |
| 6525 | // RFC 1890 and must remain unchanged for backward compatibility. |
| 6526 | rtpReceiveFrequency = 8000; |
| 6527 | } else if (STR_CASE_CMP("opus", currRecCodec.plname) == 0) { |
| 6528 | // We are resampling Opus internally to 32,000 Hz until all our |
| 6529 | // DSP routines can operate at 48,000 Hz, but the RTP clock |
| 6530 | // rate for the Opus payload format is standardized to 48,000 Hz, |
| 6531 | // because that is the maximum supported decoding sampling rate. |
| 6532 | rtpReceiveFrequency = 48000; |
| 6533 | } |
| 6534 | } |
| 6535 | |
| 6536 | const WebRtc_UWord32 timeStampDiff = timestamp - _playoutTimeStampRTP; |
| 6537 | WebRtc_UWord32 timeStampDiffMs(0); |
| 6538 | |
| 6539 | if (timeStampDiff > 0) |
| 6540 | { |
| 6541 | switch (rtpReceiveFrequency) { |
| 6542 | case 8000: |
| 6543 | timeStampDiffMs = static_cast<WebRtc_UWord32>(timeStampDiff >> 3); |
| 6544 | break; |
| 6545 | case 16000: |
| 6546 | timeStampDiffMs = static_cast<WebRtc_UWord32>(timeStampDiff >> 4); |
| 6547 | break; |
| 6548 | case 32000: |
| 6549 | timeStampDiffMs = static_cast<WebRtc_UWord32>(timeStampDiff >> 5); |
| 6550 | break; |
| 6551 | case 48000: |
| 6552 | timeStampDiffMs = static_cast<WebRtc_UWord32>(timeStampDiff / 48); |
| 6553 | break; |
| 6554 | default: |
| 6555 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 6556 | VoEId(_instanceId, _channelId), |
| 6557 | "Channel::UpdatePacketDelay() invalid sample rate"); |
| 6558 | timeStampDiffMs = 0; |
| 6559 | return -1; |
| 6560 | } |
niklas.enbom@webrtc.org | db32ab0 | 2013-01-17 22:25:49 +0000 | [diff] [blame] | 6561 | if (timeStampDiffMs > (2 * kVoiceEngineMaxMinPlayoutDelayMs)) |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6562 | { |
| 6563 | timeStampDiffMs = 0; |
| 6564 | } |
| 6565 | |
| 6566 | if (_averageDelayMs == 0) |
| 6567 | { |
niklas.enbom@webrtc.org | db32ab0 | 2013-01-17 22:25:49 +0000 | [diff] [blame] | 6568 | _averageDelayMs = timeStampDiffMs * 10; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6569 | } |
| 6570 | else |
| 6571 | { |
| 6572 | // Filter average delay value using exponential filter (alpha is |
| 6573 | // 7/8). We derive 10*_averageDelayMs here (reduces risk of |
| 6574 | // rounding error) and compensate for it in GetDelayEstimate() |
| 6575 | // later. Adding 4/8 results in correct rounding. |
| 6576 | _averageDelayMs = ((_averageDelayMs*7 + 10*timeStampDiffMs + 4)>>3); |
| 6577 | } |
| 6578 | |
| 6579 | if (sequenceNumber - _previousSequenceNumber == 1) |
| 6580 | { |
| 6581 | WebRtc_UWord16 packetDelayMs = 0; |
| 6582 | switch (rtpReceiveFrequency) { |
| 6583 | case 8000: |
| 6584 | packetDelayMs = static_cast<WebRtc_UWord16>( |
| 6585 | (timestamp - _previousTimestamp) >> 3); |
| 6586 | break; |
| 6587 | case 16000: |
| 6588 | packetDelayMs = static_cast<WebRtc_UWord16>( |
| 6589 | (timestamp - _previousTimestamp) >> 4); |
| 6590 | break; |
| 6591 | case 32000: |
| 6592 | packetDelayMs = static_cast<WebRtc_UWord16>( |
| 6593 | (timestamp - _previousTimestamp) >> 5); |
| 6594 | break; |
| 6595 | case 48000: |
| 6596 | packetDelayMs = static_cast<WebRtc_UWord16>( |
| 6597 | (timestamp - _previousTimestamp) / 48); |
| 6598 | break; |
| 6599 | } |
| 6600 | |
| 6601 | if (packetDelayMs >= 10 && packetDelayMs <= 60) |
| 6602 | _recPacketDelayMs = packetDelayMs; |
| 6603 | } |
| 6604 | } |
| 6605 | |
| 6606 | _previousSequenceNumber = sequenceNumber; |
| 6607 | _previousTimestamp = timestamp; |
| 6608 | |
| 6609 | return 0; |
| 6610 | } |
| 6611 | |
| 6612 | void |
| 6613 | Channel::RegisterReceiveCodecsToRTPModule() |
| 6614 | { |
| 6615 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 6616 | "Channel::RegisterReceiveCodecsToRTPModule()"); |
| 6617 | |
| 6618 | |
| 6619 | CodecInst codec; |
| 6620 | const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs(); |
| 6621 | |
| 6622 | for (int idx = 0; idx < nSupportedCodecs; idx++) |
| 6623 | { |
| 6624 | // Open up the RTP/RTCP receiver for all supported codecs |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 6625 | if ((_audioCodingModule.Codec(idx, &codec) == -1) || |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6626 | (_rtpRtcpModule->RegisterReceivePayload(codec) == -1)) |
| 6627 | { |
| 6628 | WEBRTC_TRACE( |
| 6629 | kTraceWarning, |
| 6630 | kTraceVoice, |
| 6631 | VoEId(_instanceId, _channelId), |
| 6632 | "Channel::RegisterReceiveCodecsToRTPModule() unable" |
| 6633 | " to register %s (%d/%d/%d/%d) to RTP/RTCP receiver", |
| 6634 | codec.plname, codec.pltype, codec.plfreq, |
| 6635 | codec.channels, codec.rate); |
| 6636 | } |
| 6637 | else |
| 6638 | { |
| 6639 | WEBRTC_TRACE( |
| 6640 | kTraceInfo, |
| 6641 | kTraceVoice, |
| 6642 | VoEId(_instanceId, _channelId), |
| 6643 | "Channel::RegisterReceiveCodecsToRTPModule() %s " |
| 6644 | "(%d/%d/%d/%d) has been added to the RTP/RTCP " |
| 6645 | "receiver", |
| 6646 | codec.plname, codec.pltype, codec.plfreq, |
| 6647 | codec.channels, codec.rate); |
| 6648 | } |
| 6649 | } |
| 6650 | } |
| 6651 | |
andrew@webrtc.org | d898c01 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 6652 | int Channel::ApmProcessRx(AudioFrame& frame) { |
| 6653 | AudioProcessing* audioproc = _rxAudioProcessingModulePtr; |
| 6654 | // Register the (possibly new) frame parameters. |
| 6655 | if (audioproc->set_sample_rate_hz(frame.sample_rate_hz_) != 0) { |
andrew@webrtc.org | bc687c5 | 2012-11-20 07:34:45 +0000 | [diff] [blame] | 6656 | LOG_FERR1(LS_WARNING, set_sample_rate_hz, frame.sample_rate_hz_); |
andrew@webrtc.org | d898c01 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 6657 | } |
| 6658 | if (audioproc->set_num_channels(frame.num_channels_, |
| 6659 | frame.num_channels_) != 0) { |
andrew@webrtc.org | bc687c5 | 2012-11-20 07:34:45 +0000 | [diff] [blame] | 6660 | LOG_FERR1(LS_WARNING, set_num_channels, frame.num_channels_); |
andrew@webrtc.org | d898c01 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 6661 | } |
| 6662 | if (audioproc->ProcessStream(&frame) != 0) { |
andrew@webrtc.org | bc687c5 | 2012-11-20 07:34:45 +0000 | [diff] [blame] | 6663 | LOG_FERR0(LS_WARNING, ProcessStream); |
andrew@webrtc.org | d898c01 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 6664 | } |
| 6665 | return 0; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6666 | } |
| 6667 | |
turaj@webrtc.org | 7db5290 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 6668 | int Channel::SetSecondarySendCodec(const CodecInst& codec, |
| 6669 | int red_payload_type) { |
turaj@webrtc.org | 040f800 | 2013-01-31 18:20:17 +0000 | [diff] [blame] | 6670 | // Sanity check for payload type. |
| 6671 | if (red_payload_type < 0 || red_payload_type > 127) { |
| 6672 | _engineStatisticsPtr->SetLastError( |
| 6673 | VE_PLTYPE_ERROR, kTraceError, |
| 6674 | "SetRedPayloadType() invalid RED payload type"); |
| 6675 | return -1; |
| 6676 | } |
| 6677 | |
turaj@webrtc.org | 7db5290 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 6678 | if (SetRedPayloadType(red_payload_type) < 0) { |
| 6679 | _engineStatisticsPtr->SetLastError( |
| 6680 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 6681 | "SetSecondarySendCodec() Failed to register RED ACM"); |
| 6682 | return -1; |
| 6683 | } |
| 6684 | if (_audioCodingModule.RegisterSecondarySendCodec(codec) < 0) { |
| 6685 | _engineStatisticsPtr->SetLastError( |
| 6686 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 6687 | "SetSecondarySendCodec() Failed to register secondary send codec in " |
| 6688 | "ACM"); |
| 6689 | return -1; |
| 6690 | } |
| 6691 | |
| 6692 | return 0; |
| 6693 | } |
| 6694 | |
| 6695 | void Channel::RemoveSecondarySendCodec() { |
| 6696 | _audioCodingModule.UnregisterSecondarySendCodec(); |
| 6697 | } |
| 6698 | |
| 6699 | int Channel::GetSecondarySendCodec(CodecInst* codec) { |
| 6700 | if (_audioCodingModule.SecondarySendCodec(codec) < 0) { |
| 6701 | _engineStatisticsPtr->SetLastError( |
| 6702 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 6703 | "GetSecondarySendCodec() Failed to get secondary sent codec from ACM"); |
| 6704 | return -1; |
| 6705 | } |
| 6706 | return 0; |
| 6707 | } |
| 6708 | |
turaj@webrtc.org | 040f800 | 2013-01-31 18:20:17 +0000 | [diff] [blame] | 6709 | // Assuming this method is called with valid payload type. |
turaj@webrtc.org | 7db5290 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 6710 | int Channel::SetRedPayloadType(int red_payload_type) { |
turaj@webrtc.org | 7db5290 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 6711 | CodecInst codec; |
| 6712 | bool found_red = false; |
| 6713 | |
| 6714 | // Get default RED settings from the ACM database |
| 6715 | const int num_codecs = AudioCodingModule::NumberOfCodecs(); |
| 6716 | for (int idx = 0; idx < num_codecs; idx++) { |
tina.legrand@webrtc.org | e9bb4e5 | 2013-02-21 10:27:48 +0000 | [diff] [blame^] | 6717 | _audioCodingModule.Codec(idx, &codec); |
turaj@webrtc.org | 7db5290 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 6718 | if (!STR_CASE_CMP(codec.plname, "RED")) { |
| 6719 | found_red = true; |
| 6720 | break; |
| 6721 | } |
| 6722 | } |
| 6723 | |
| 6724 | if (!found_red) { |
| 6725 | _engineStatisticsPtr->SetLastError( |
| 6726 | VE_CODEC_ERROR, kTraceError, |
| 6727 | "SetRedPayloadType() RED is not supported"); |
| 6728 | return -1; |
| 6729 | } |
| 6730 | |
turaj@webrtc.org | 2344ebe | 2013-01-31 18:34:19 +0000 | [diff] [blame] | 6731 | codec.pltype = red_payload_type; |
turaj@webrtc.org | 7db5290 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 6732 | if (_audioCodingModule.RegisterSendCodec(codec) < 0) { |
| 6733 | _engineStatisticsPtr->SetLastError( |
| 6734 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 6735 | "SetRedPayloadType() RED registration in ACM module failed"); |
| 6736 | return -1; |
| 6737 | } |
| 6738 | |
| 6739 | if (_rtpRtcpModule->SetSendREDPayloadType(red_payload_type) != 0) { |
| 6740 | _engineStatisticsPtr->SetLastError( |
| 6741 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 6742 | "SetRedPayloadType() RED registration in RTP/RTCP module failed"); |
| 6743 | return -1; |
| 6744 | } |
| 6745 | return 0; |
| 6746 | } |
| 6747 | |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6748 | } // namespace voe |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 6749 | } // namespace webrtc |