niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1 | /* |
henrika@webrtc.org | 2919e95 | 2012-01-31 08:45:03 +0000 | [diff] [blame] | 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 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 | 6388c3e | 2013-02-12 21:42:18 +0000 | [diff] [blame] | 11 | #include "webrtc/voice_engine/channel.h" |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 12 | |
turaj@webrtc.org | 6388c3e | 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" |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 28 | |
| 29 | #if defined(_WIN32) |
| 30 | #include <Qos.h> |
| 31 | #endif |
| 32 | |
andrew@webrtc.org | 50419b0 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 33 | namespace webrtc { |
| 34 | namespace voe { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 35 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 36 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 37 | Channel::SendData(FrameType frameType, |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 38 | uint8_t payloadType, |
| 39 | uint32_t timeStamp, |
| 40 | const uint8_t* payloadData, |
| 41 | uint16_t payloadSize, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 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 | { |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 51 | assert(_rtpAudioProc.get() != NULL); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 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 |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 55 | _rtpRtcpModule->SetAudioLevel(_rtpAudioProc->level_estimator()->RMS()); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 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. |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 61 | if (_rtpRtcpModule->SendOutgoingData((FrameType&)frameType, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 62 | payloadType, |
| 63 | timeStamp, |
stefan@webrtc.org | ddfdfed | 2012-07-03 13:21:22 +0000 | [diff] [blame] | 64 | // Leaving the time when this frame was |
| 65 | // received from the capture device as |
| 66 | // undefined for voice for now. |
| 67 | -1, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 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 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 84 | int32_t |
| 85 | Channel::InFrameType(int16_t frameType) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 86 | { |
| 87 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 88 | "Channel::InFrameType(frameType=%d)", frameType); |
| 89 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 90 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 91 | // 1 indicates speech |
| 92 | _sendFrameType = (frameType == 1) ? 1 : 0; |
| 93 | return 0; |
| 94 | } |
| 95 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 96 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 97 | Channel::OnRxVadDetected(const int vadDecision) |
| 98 | { |
| 99 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 100 | "Channel::OnRxVadDetected(vadDecision=%d)", vadDecision); |
| 101 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 102 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 103 | if (_rxVadObserverPtr) |
| 104 | { |
| 105 | _rxVadObserverPtr->OnRxVad(_channelId, vadDecision); |
| 106 | } |
| 107 | |
| 108 | return 0; |
| 109 | } |
| 110 | |
| 111 | int |
| 112 | Channel::SendPacket(int channel, const void *data, int len) |
| 113 | { |
| 114 | channel = VoEChannelId(channel); |
| 115 | assert(channel == _channelId); |
| 116 | |
| 117 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 118 | "Channel::SendPacket(channel=%d, len=%d)", channel, len); |
| 119 | |
| 120 | if (_transportPtr == NULL) |
| 121 | { |
| 122 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 123 | "Channel::SendPacket() failed to send RTP packet due to" |
| 124 | " invalid transport object"); |
| 125 | return -1; |
| 126 | } |
| 127 | |
| 128 | // Insert extra RTP packet using if user has called the InsertExtraRTPPacket |
| 129 | // API |
| 130 | if (_insertExtraRTPPacket) |
| 131 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 132 | uint8_t* rtpHdr = (uint8_t*)data; |
| 133 | uint8_t M_PT(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 134 | if (_extraMarkerBit) |
| 135 | { |
| 136 | M_PT = 0x80; // set the M-bit |
| 137 | } |
| 138 | M_PT += _extraPayloadType; // set the payload type |
| 139 | *(++rtpHdr) = M_PT; // modify the M|PT-byte within the RTP header |
| 140 | _insertExtraRTPPacket = false; // insert one packet only |
| 141 | } |
| 142 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 143 | uint8_t* bufferToSendPtr = (uint8_t*)data; |
| 144 | int32_t bufferLength = len; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 145 | |
| 146 | // Dump the RTP packet to a file (if RTP dump is enabled). |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 147 | if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 148 | { |
| 149 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 150 | VoEId(_instanceId,_channelId), |
| 151 | "Channel::SendPacket() RTP dump to output file failed"); |
| 152 | } |
| 153 | |
| 154 | // SRTP or External encryption |
| 155 | if (_encrypting) |
| 156 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 157 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 158 | |
| 159 | if (_encryptionPtr) |
| 160 | { |
| 161 | if (!_encryptionRTPBufferPtr) |
| 162 | { |
| 163 | // Allocate memory for encryption buffer one time only |
| 164 | _encryptionRTPBufferPtr = |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 165 | new uint8_t[kVoiceEngineMaxIpPacketSizeBytes]; |
xians@webrtc.org | 5125350 | 2012-10-25 13:58:02 +0000 | [diff] [blame] | 166 | memset(_encryptionRTPBufferPtr, 0, |
| 167 | kVoiceEngineMaxIpPacketSizeBytes); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 168 | } |
| 169 | |
| 170 | // Perform encryption (SRTP or external) |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 171 | int32_t encryptedBufferLength = 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 172 | _encryptionPtr->encrypt(_channelId, |
| 173 | bufferToSendPtr, |
| 174 | _encryptionRTPBufferPtr, |
| 175 | bufferLength, |
| 176 | (int*)&encryptedBufferLength); |
| 177 | if (encryptedBufferLength <= 0) |
| 178 | { |
| 179 | _engineStatisticsPtr->SetLastError( |
| 180 | VE_ENCRYPTION_FAILED, |
| 181 | kTraceError, "Channel::SendPacket() encryption failed"); |
| 182 | return -1; |
| 183 | } |
| 184 | |
| 185 | // Replace default data buffer with encrypted buffer |
| 186 | bufferToSendPtr = _encryptionRTPBufferPtr; |
| 187 | bufferLength = encryptedBufferLength; |
| 188 | } |
| 189 | } |
| 190 | |
| 191 | // Packet transmission using WebRtc socket transport |
| 192 | if (!_externalTransport) |
| 193 | { |
| 194 | int n = _transportPtr->SendPacket(channel, bufferToSendPtr, |
| 195 | bufferLength); |
| 196 | if (n < 0) |
| 197 | { |
| 198 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 199 | VoEId(_instanceId,_channelId), |
| 200 | "Channel::SendPacket() RTP transmission using WebRtc" |
| 201 | " sockets failed"); |
| 202 | return -1; |
| 203 | } |
| 204 | return n; |
| 205 | } |
| 206 | |
| 207 | // Packet transmission using external transport transport |
| 208 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 209 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 210 | |
| 211 | int n = _transportPtr->SendPacket(channel, |
| 212 | bufferToSendPtr, |
| 213 | bufferLength); |
| 214 | if (n < 0) |
| 215 | { |
| 216 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 217 | VoEId(_instanceId,_channelId), |
| 218 | "Channel::SendPacket() RTP transmission using external" |
| 219 | " transport failed"); |
| 220 | return -1; |
| 221 | } |
| 222 | return n; |
| 223 | } |
| 224 | } |
| 225 | |
| 226 | int |
| 227 | Channel::SendRTCPPacket(int channel, const void *data, int len) |
| 228 | { |
| 229 | channel = VoEChannelId(channel); |
| 230 | assert(channel == _channelId); |
| 231 | |
| 232 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 233 | "Channel::SendRTCPPacket(channel=%d, len=%d)", channel, len); |
| 234 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 235 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 236 | CriticalSectionScoped cs(&_callbackCritSect); |
xians@webrtc.org | 83661f5 | 2011-11-25 10:58:15 +0000 | [diff] [blame] | 237 | if (_transportPtr == NULL) |
| 238 | { |
| 239 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 240 | VoEId(_instanceId,_channelId), |
| 241 | "Channel::SendRTCPPacket() failed to send RTCP packet" |
| 242 | " due to invalid transport object"); |
| 243 | return -1; |
| 244 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 245 | } |
| 246 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 247 | uint8_t* bufferToSendPtr = (uint8_t*)data; |
| 248 | int32_t bufferLength = len; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 249 | |
| 250 | // Dump the RTCP packet to a file (if RTP dump is enabled). |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 251 | if (_rtpDumpOut.DumpPacket((const uint8_t*)data, len) == -1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 252 | { |
| 253 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 254 | VoEId(_instanceId,_channelId), |
| 255 | "Channel::SendPacket() RTCP dump to output file failed"); |
| 256 | } |
| 257 | |
| 258 | // SRTP or External encryption |
| 259 | if (_encrypting) |
| 260 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 261 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 262 | |
| 263 | if (_encryptionPtr) |
| 264 | { |
| 265 | if (!_encryptionRTCPBufferPtr) |
| 266 | { |
| 267 | // Allocate memory for encryption buffer one time only |
| 268 | _encryptionRTCPBufferPtr = |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 269 | new uint8_t[kVoiceEngineMaxIpPacketSizeBytes]; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 270 | } |
| 271 | |
| 272 | // Perform encryption (SRTP or external). |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 273 | int32_t encryptedBufferLength = 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 274 | _encryptionPtr->encrypt_rtcp(_channelId, |
| 275 | bufferToSendPtr, |
| 276 | _encryptionRTCPBufferPtr, |
| 277 | bufferLength, |
| 278 | (int*)&encryptedBufferLength); |
| 279 | if (encryptedBufferLength <= 0) |
| 280 | { |
| 281 | _engineStatisticsPtr->SetLastError( |
| 282 | VE_ENCRYPTION_FAILED, kTraceError, |
| 283 | "Channel::SendRTCPPacket() encryption failed"); |
| 284 | return -1; |
| 285 | } |
| 286 | |
| 287 | // Replace default data buffer with encrypted buffer |
| 288 | bufferToSendPtr = _encryptionRTCPBufferPtr; |
| 289 | bufferLength = encryptedBufferLength; |
| 290 | } |
| 291 | } |
| 292 | |
| 293 | // Packet transmission using WebRtc socket transport |
| 294 | if (!_externalTransport) |
| 295 | { |
| 296 | int n = _transportPtr->SendRTCPPacket(channel, |
| 297 | bufferToSendPtr, |
| 298 | bufferLength); |
| 299 | if (n < 0) |
| 300 | { |
| 301 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 302 | VoEId(_instanceId,_channelId), |
| 303 | "Channel::SendRTCPPacket() transmission using WebRtc" |
| 304 | " sockets failed"); |
| 305 | return -1; |
| 306 | } |
| 307 | return n; |
| 308 | } |
| 309 | |
| 310 | // Packet transmission using external transport transport |
| 311 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 312 | CriticalSectionScoped cs(&_callbackCritSect); |
henrike@webrtc.org | de727ab | 2012-11-18 18:49:13 +0000 | [diff] [blame] | 313 | if (_transportPtr == NULL) |
| 314 | { |
| 315 | return -1; |
| 316 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 317 | int n = _transportPtr->SendRTCPPacket(channel, |
| 318 | bufferToSendPtr, |
| 319 | bufferLength); |
| 320 | if (n < 0) |
| 321 | { |
| 322 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 323 | VoEId(_instanceId,_channelId), |
| 324 | "Channel::SendRTCPPacket() transmission using external" |
| 325 | " transport failed"); |
| 326 | return -1; |
| 327 | } |
| 328 | return n; |
| 329 | } |
| 330 | |
| 331 | return len; |
| 332 | } |
| 333 | |
| 334 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 335 | Channel::OnPlayTelephoneEvent(const int32_t id, |
| 336 | const uint8_t event, |
| 337 | const uint16_t lengthMs, |
| 338 | const uint8_t volume) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 339 | { |
| 340 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 341 | "Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u," |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 342 | " volume=%u)", id, event, lengthMs, volume); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 343 | |
| 344 | if (!_playOutbandDtmfEvent || (event > 15)) |
| 345 | { |
| 346 | // Ignore callback since feedback is disabled or event is not a |
| 347 | // Dtmf tone event. |
| 348 | return; |
| 349 | } |
| 350 | |
| 351 | assert(_outputMixerPtr != NULL); |
| 352 | |
| 353 | // Start playing out the Dtmf tone (if playout is enabled). |
| 354 | // Reduce length of tone with 80ms to the reduce risk of echo. |
| 355 | _outputMixerPtr->PlayDtmfTone(event, lengthMs - 80, volume); |
| 356 | } |
| 357 | |
| 358 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 359 | Channel::OnIncomingSSRCChanged(const int32_t id, |
| 360 | const uint32_t SSRC) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 361 | { |
| 362 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 363 | "Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)", |
| 364 | id, SSRC); |
| 365 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 366 | int32_t channel = VoEChannelId(id); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 367 | assert(channel == _channelId); |
| 368 | |
| 369 | // Reset RTP-module counters since a new incoming RTP stream is detected |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 370 | _rtpRtcpModule->ResetReceiveDataCountersRTP(); |
| 371 | _rtpRtcpModule->ResetStatisticsRTP(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 372 | |
| 373 | if (_rtpObserver) |
| 374 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 375 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 376 | |
| 377 | if (_rtpObserverPtr) |
| 378 | { |
| 379 | // Send new SSRC to registered observer using callback |
| 380 | _rtpObserverPtr->OnIncomingSSRCChanged(channel, SSRC); |
| 381 | } |
| 382 | } |
| 383 | } |
| 384 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 385 | void Channel::OnIncomingCSRCChanged(const int32_t id, |
| 386 | const uint32_t CSRC, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 387 | const bool added) |
| 388 | { |
| 389 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 390 | "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)", |
| 391 | id, CSRC, added); |
| 392 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 393 | int32_t channel = VoEChannelId(id); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 394 | assert(channel == _channelId); |
| 395 | |
| 396 | if (_rtpObserver) |
| 397 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 398 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 399 | |
| 400 | if (_rtpObserverPtr) |
| 401 | { |
| 402 | _rtpObserverPtr->OnIncomingCSRCChanged(channel, CSRC, added); |
| 403 | } |
| 404 | } |
| 405 | } |
| 406 | |
| 407 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 408 | Channel::OnApplicationDataReceived(const int32_t id, |
| 409 | const uint8_t subType, |
| 410 | const uint32_t name, |
| 411 | const uint16_t length, |
| 412 | const uint8_t* data) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 413 | { |
| 414 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 415 | "Channel::OnApplicationDataReceived(id=%d, subType=%u," |
| 416 | " name=%u, length=%u)", |
| 417 | id, subType, name, length); |
| 418 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 419 | int32_t channel = VoEChannelId(id); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 420 | assert(channel == _channelId); |
| 421 | |
| 422 | if (_rtcpObserver) |
| 423 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 424 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 425 | |
| 426 | if (_rtcpObserverPtr) |
| 427 | { |
| 428 | _rtcpObserverPtr->OnApplicationDataReceived(channel, |
| 429 | subType, |
| 430 | name, |
| 431 | data, |
| 432 | length); |
| 433 | } |
| 434 | } |
| 435 | } |
| 436 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 437 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 438 | Channel::OnInitializeDecoder( |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 439 | const int32_t id, |
| 440 | const int8_t payloadType, |
leozwang@webrtc.org | 813e4b0 | 2012-03-01 18:34:25 +0000 | [diff] [blame] | 441 | const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
xians@google.com | 0b0665a | 2011-08-08 08:18:44 +0000 | [diff] [blame] | 442 | const int frequency, |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 443 | const uint8_t channels, |
| 444 | const uint32_t rate) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 445 | { |
| 446 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 447 | "Channel::OnInitializeDecoder(id=%d, payloadType=%d, " |
| 448 | "payloadName=%s, frequency=%u, channels=%u, rate=%u)", |
| 449 | id, payloadType, payloadName, frequency, channels, rate); |
| 450 | |
andrew@webrtc.org | ceb148c | 2011-08-23 17:53:54 +0000 | [diff] [blame] | 451 | assert(VoEChannelId(id) == _channelId); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 452 | |
henrika@webrtc.org | f75901f | 2012-01-16 08:45:42 +0000 | [diff] [blame] | 453 | CodecInst receiveCodec = {0}; |
| 454 | CodecInst dummyCodec = {0}; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 455 | |
| 456 | receiveCodec.pltype = payloadType; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 457 | receiveCodec.plfreq = frequency; |
| 458 | receiveCodec.channels = channels; |
| 459 | receiveCodec.rate = rate; |
henrika@webrtc.org | f75901f | 2012-01-16 08:45:42 +0000 | [diff] [blame] | 460 | strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); |
andrew@webrtc.org | ae1a58b | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 461 | |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 462 | _audioCodingModule.Codec(payloadName, &dummyCodec, frequency, channels); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 463 | receiveCodec.pacsize = dummyCodec.pacsize; |
| 464 | |
| 465 | // Register the new codec to the ACM |
| 466 | if (_audioCodingModule.RegisterReceiveCodec(receiveCodec) == -1) |
| 467 | { |
| 468 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
andrew@webrtc.org | ceb148c | 2011-08-23 17:53:54 +0000 | [diff] [blame] | 469 | VoEId(_instanceId, _channelId), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 470 | "Channel::OnInitializeDecoder() invalid codec (" |
| 471 | "pt=%d, name=%s) received - 1", payloadType, payloadName); |
| 472 | _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR); |
| 473 | return -1; |
| 474 | } |
| 475 | |
| 476 | return 0; |
| 477 | } |
| 478 | |
| 479 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 480 | Channel::OnPacketTimeout(const int32_t id) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 481 | { |
| 482 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 483 | "Channel::OnPacketTimeout(id=%d)", id); |
| 484 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 485 | CriticalSectionScoped cs(_callbackCritSectPtr); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 486 | if (_voiceEngineObserverPtr) |
| 487 | { |
| 488 | if (_receiving || _externalTransport) |
| 489 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 490 | int32_t channel = VoEChannelId(id); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 491 | assert(channel == _channelId); |
| 492 | // Ensure that next OnReceivedPacket() callback will trigger |
| 493 | // a VE_PACKET_RECEIPT_RESTARTED callback. |
| 494 | _rtpPacketTimedOut = true; |
| 495 | // Deliver callback to the observer |
| 496 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 497 | VoEId(_instanceId,_channelId), |
| 498 | "Channel::OnPacketTimeout() => " |
| 499 | "CallbackOnError(VE_RECEIVE_PACKET_TIMEOUT)"); |
| 500 | _voiceEngineObserverPtr->CallbackOnError(channel, |
| 501 | VE_RECEIVE_PACKET_TIMEOUT); |
| 502 | } |
| 503 | } |
| 504 | } |
| 505 | |
| 506 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 507 | Channel::OnReceivedPacket(const int32_t id, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 508 | const RtpRtcpPacketType packetType) |
| 509 | { |
| 510 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 511 | "Channel::OnReceivedPacket(id=%d, packetType=%d)", |
| 512 | id, packetType); |
| 513 | |
andrew@webrtc.org | ceb148c | 2011-08-23 17:53:54 +0000 | [diff] [blame] | 514 | assert(VoEChannelId(id) == _channelId); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 515 | |
| 516 | // Notify only for the case when we have restarted an RTP session. |
| 517 | if (_rtpPacketTimedOut && (kPacketRtp == packetType)) |
| 518 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 519 | CriticalSectionScoped cs(_callbackCritSectPtr); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 520 | if (_voiceEngineObserverPtr) |
| 521 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 522 | int32_t channel = VoEChannelId(id); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 523 | assert(channel == _channelId); |
| 524 | // Reset timeout mechanism |
| 525 | _rtpPacketTimedOut = false; |
| 526 | // Deliver callback to the observer |
| 527 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 528 | VoEId(_instanceId,_channelId), |
| 529 | "Channel::OnPacketTimeout() =>" |
| 530 | " CallbackOnError(VE_PACKET_RECEIPT_RESTARTED)"); |
| 531 | _voiceEngineObserverPtr->CallbackOnError( |
| 532 | channel, |
| 533 | VE_PACKET_RECEIPT_RESTARTED); |
| 534 | } |
| 535 | } |
| 536 | } |
| 537 | |
| 538 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 539 | Channel::OnPeriodicDeadOrAlive(const int32_t id, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 540 | const RTPAliveType alive) |
| 541 | { |
| 542 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 543 | "Channel::OnPeriodicDeadOrAlive(id=%d, alive=%d)", id, alive); |
| 544 | |
henrika@webrtc.org | 19da719 | 2013-04-05 14:34:57 +0000 | [diff] [blame] | 545 | { |
| 546 | CriticalSectionScoped cs(&_callbackCritSect); |
| 547 | if (!_connectionObserver) |
| 548 | return; |
| 549 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 550 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 551 | int32_t channel = VoEChannelId(id); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 552 | assert(channel == _channelId); |
| 553 | |
| 554 | // Use Alive as default to limit risk of false Dead detections |
| 555 | bool isAlive(true); |
| 556 | |
| 557 | // Always mark the connection as Dead when the module reports kRtpDead |
| 558 | if (kRtpDead == alive) |
| 559 | { |
| 560 | isAlive = false; |
| 561 | } |
| 562 | |
| 563 | // It is possible that the connection is alive even if no RTP packet has |
| 564 | // been received for a long time since the other side might use VAD/DTX |
| 565 | // and a low SID-packet update rate. |
| 566 | if ((kRtpNoRtp == alive) && _playing) |
| 567 | { |
| 568 | // Detect Alive for all NetEQ states except for the case when we are |
| 569 | // in PLC_CNG state. |
| 570 | // PLC_CNG <=> background noise only due to long expand or error. |
| 571 | // Note that, the case where the other side stops sending during CNG |
| 572 | // state will be detected as Alive. Dead is is not set until after |
| 573 | // missing RTCP packets for at least twelve seconds (handled |
| 574 | // internally by the RTP/RTCP module). |
| 575 | isAlive = (_outputSpeechType != AudioFrame::kPLCCNG); |
| 576 | } |
| 577 | |
| 578 | UpdateDeadOrAliveCounters(isAlive); |
| 579 | |
| 580 | // Send callback to the registered observer |
| 581 | if (_connectionObserver) |
| 582 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 583 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 584 | if (_connectionObserverPtr) |
| 585 | { |
| 586 | _connectionObserverPtr->OnPeriodicDeadOrAlive(channel, isAlive); |
| 587 | } |
| 588 | } |
| 589 | } |
| 590 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 591 | int32_t |
| 592 | Channel::OnReceivedPayloadData(const uint8_t* payloadData, |
| 593 | const uint16_t payloadSize, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 594 | const WebRtcRTPHeader* rtpHeader) |
| 595 | { |
| 596 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 597 | "Channel::OnReceivedPayloadData(payloadSize=%d," |
| 598 | " payloadType=%u, audioChannel=%u)", |
| 599 | payloadSize, |
| 600 | rtpHeader->header.payloadType, |
| 601 | rtpHeader->type.Audio.channel); |
| 602 | |
roosa@google.com | 0870f02 | 2012-12-12 21:31:41 +0000 | [diff] [blame] | 603 | _lastRemoteTimeStamp = rtpHeader->header.timestamp; |
| 604 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 605 | if (!_playing) |
| 606 | { |
| 607 | // Avoid inserting into NetEQ when we are not playing. Count the |
| 608 | // packet as discarded. |
| 609 | WEBRTC_TRACE(kTraceStream, kTraceVoice, |
| 610 | VoEId(_instanceId, _channelId), |
| 611 | "received packet is discarded since playing is not" |
| 612 | " activated"); |
| 613 | _numberOfDiscardedPackets++; |
| 614 | return 0; |
| 615 | } |
| 616 | |
| 617 | // Push the incoming payload (parsed and ready for decoding) into the ACM |
tina.legrand@webrtc.org | 16b6b90 | 2012-04-12 11:02:38 +0000 | [diff] [blame] | 618 | if (_audioCodingModule.IncomingPacket(payloadData, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 619 | payloadSize, |
| 620 | *rtpHeader) != 0) |
| 621 | { |
| 622 | _engineStatisticsPtr->SetLastError( |
| 623 | VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning, |
| 624 | "Channel::OnReceivedPayloadData() unable to push data to the ACM"); |
| 625 | return -1; |
| 626 | } |
| 627 | |
| 628 | // Update the packet delay |
| 629 | UpdatePacketDelay(rtpHeader->header.timestamp, |
| 630 | rtpHeader->header.sequenceNumber); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 631 | return 0; |
| 632 | } |
| 633 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 634 | int32_t Channel::GetAudioFrame(const int32_t id, AudioFrame& audioFrame) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 635 | { |
| 636 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 637 | "Channel::GetAudioFrame(id=%d)", id); |
| 638 | |
| 639 | // Get 10ms raw PCM data from the ACM (mixer limits output frequency) |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 640 | if (_audioCodingModule.PlayoutData10Ms(audioFrame.sample_rate_hz_, |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 641 | &audioFrame) == -1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 642 | { |
| 643 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 644 | VoEId(_instanceId,_channelId), |
| 645 | "Channel::GetAudioFrame() PlayoutData10Ms() failed!"); |
andrew@webrtc.org | 7859e10 | 2012-01-13 00:30:11 +0000 | [diff] [blame] | 646 | // In all likelihood, the audio in this frame is garbage. We return an |
| 647 | // error so that the audio mixer module doesn't add it to the mix. As |
| 648 | // a result, it won't be played out and the actions skipped here are |
| 649 | // irrelevant. |
| 650 | return -1; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 651 | } |
| 652 | |
| 653 | if (_RxVadDetection) |
| 654 | { |
| 655 | UpdateRxVadDetection(audioFrame); |
| 656 | } |
| 657 | |
| 658 | // Convert module ID to internal VoE channel ID |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 659 | audioFrame.id_ = VoEChannelId(audioFrame.id_); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 660 | // Store speech type for dead-or-alive detection |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 661 | _outputSpeechType = audioFrame.speech_type_; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 662 | |
| 663 | // Perform far-end AudioProcessing module processing on the received signal |
| 664 | if (_rxApmIsEnabled) |
| 665 | { |
| 666 | ApmProcessRx(audioFrame); |
| 667 | } |
| 668 | |
| 669 | // Output volume scaling |
| 670 | if (_outputGain < 0.99f || _outputGain > 1.01f) |
| 671 | { |
| 672 | AudioFrameOperations::ScaleWithSat(_outputGain, audioFrame); |
| 673 | } |
| 674 | |
| 675 | // Scale left and/or right channel(s) if stereo and master balance is |
| 676 | // active |
| 677 | |
| 678 | if (_panLeft != 1.0f || _panRight != 1.0f) |
| 679 | { |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 680 | if (audioFrame.num_channels_ == 1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 681 | { |
| 682 | // Emulate stereo mode since panning is active. |
| 683 | // The mono signal is copied to both left and right channels here. |
andrew@webrtc.org | 4ecea3e | 2012-06-27 03:25:31 +0000 | [diff] [blame] | 684 | AudioFrameOperations::MonoToStereo(&audioFrame); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 685 | } |
| 686 | // For true stereo mode (when we are receiving a stereo signal), no |
| 687 | // action is needed. |
| 688 | |
| 689 | // Do the panning operation (the audio frame contains stereo at this |
| 690 | // stage) |
| 691 | AudioFrameOperations::Scale(_panLeft, _panRight, audioFrame); |
| 692 | } |
| 693 | |
| 694 | // Mix decoded PCM output with file if file mixing is enabled |
| 695 | if (_outputFilePlaying) |
| 696 | { |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 697 | MixAudioWithFile(audioFrame, audioFrame.sample_rate_hz_); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 698 | } |
| 699 | |
| 700 | // Place channel in on-hold state (~muted) if on-hold is activated |
| 701 | if (_outputIsOnHold) |
| 702 | { |
| 703 | AudioFrameOperations::Mute(audioFrame); |
| 704 | } |
| 705 | |
| 706 | // External media |
| 707 | if (_outputExternalMedia) |
| 708 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 709 | CriticalSectionScoped cs(&_callbackCritSect); |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 710 | const bool isStereo = (audioFrame.num_channels_ == 2); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 711 | if (_outputExternalMediaCallbackPtr) |
| 712 | { |
| 713 | _outputExternalMediaCallbackPtr->Process( |
| 714 | _channelId, |
| 715 | kPlaybackPerChannel, |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 716 | (int16_t*)audioFrame.data_, |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 717 | audioFrame.samples_per_channel_, |
| 718 | audioFrame.sample_rate_hz_, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 719 | isStereo); |
| 720 | } |
| 721 | } |
| 722 | |
| 723 | // Record playout if enabled |
| 724 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 725 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 726 | |
| 727 | if (_outputFileRecording && _outputFileRecorderPtr) |
| 728 | { |
niklas.enbom@webrtc.org | 5398d95 | 2012-03-26 08:11:25 +0000 | [diff] [blame] | 729 | _outputFileRecorderPtr->RecordAudioToFile(audioFrame); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 730 | } |
| 731 | } |
| 732 | |
| 733 | // Measure audio level (0-9) |
| 734 | _outputAudioLevel.ComputeLevel(audioFrame); |
| 735 | |
| 736 | return 0; |
| 737 | } |
| 738 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 739 | int32_t |
| 740 | Channel::NeededFrequency(const int32_t id) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 741 | { |
| 742 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 743 | "Channel::NeededFrequency(id=%d)", id); |
| 744 | |
| 745 | int highestNeeded = 0; |
| 746 | |
| 747 | // Determine highest needed receive frequency |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 748 | int32_t receiveFrequency = _audioCodingModule.ReceiveFrequency(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 749 | |
| 750 | // Return the bigger of playout and receive frequency in the ACM. |
| 751 | if (_audioCodingModule.PlayoutFrequency() > receiveFrequency) |
| 752 | { |
| 753 | highestNeeded = _audioCodingModule.PlayoutFrequency(); |
| 754 | } |
| 755 | else |
| 756 | { |
| 757 | highestNeeded = receiveFrequency; |
| 758 | } |
| 759 | |
| 760 | // Special case, if we're playing a file on the playout side |
| 761 | // we take that frequency into consideration as well |
| 762 | // This is not needed on sending side, since the codec will |
| 763 | // limit the spectrum anyway. |
| 764 | if (_outputFilePlaying) |
| 765 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 766 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 767 | if (_outputFilePlayerPtr && _outputFilePlaying) |
| 768 | { |
| 769 | if(_outputFilePlayerPtr->Frequency()>highestNeeded) |
| 770 | { |
| 771 | highestNeeded=_outputFilePlayerPtr->Frequency(); |
| 772 | } |
| 773 | } |
| 774 | } |
| 775 | |
| 776 | return(highestNeeded); |
| 777 | } |
| 778 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 779 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 780 | Channel::CreateChannel(Channel*& channel, |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 781 | const int32_t channelId, |
| 782 | const uint32_t instanceId) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 783 | { |
| 784 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId), |
| 785 | "Channel::CreateChannel(channelId=%d, instanceId=%d)", |
| 786 | channelId, instanceId); |
| 787 | |
| 788 | channel = new Channel(channelId, instanceId); |
| 789 | if (channel == NULL) |
| 790 | { |
| 791 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, |
| 792 | VoEId(instanceId,channelId), |
| 793 | "Channel::CreateChannel() unable to allocate memory for" |
| 794 | " channel"); |
| 795 | return -1; |
| 796 | } |
| 797 | return 0; |
| 798 | } |
| 799 | |
| 800 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 801 | Channel::PlayNotification(const int32_t id, const uint32_t durationMs) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 802 | { |
| 803 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 804 | "Channel::PlayNotification(id=%d, durationMs=%d)", |
| 805 | id, durationMs); |
| 806 | |
| 807 | // Not implement yet |
| 808 | } |
| 809 | |
| 810 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 811 | Channel::RecordNotification(const int32_t id, const uint32_t durationMs) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 812 | { |
| 813 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 814 | "Channel::RecordNotification(id=%d, durationMs=%d)", |
| 815 | id, durationMs); |
| 816 | |
| 817 | // Not implement yet |
| 818 | } |
| 819 | |
| 820 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 821 | Channel::PlayFileEnded(const int32_t id) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 822 | { |
| 823 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 824 | "Channel::PlayFileEnded(id=%d)", id); |
| 825 | |
| 826 | if (id == _inputFilePlayerId) |
| 827 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 828 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 829 | |
| 830 | _inputFilePlaying = false; |
| 831 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 832 | VoEId(_instanceId,_channelId), |
| 833 | "Channel::PlayFileEnded() => input file player module is" |
| 834 | " shutdown"); |
| 835 | } |
| 836 | else if (id == _outputFilePlayerId) |
| 837 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 838 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 839 | |
| 840 | _outputFilePlaying = false; |
| 841 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 842 | VoEId(_instanceId,_channelId), |
| 843 | "Channel::PlayFileEnded() => output file player module is" |
| 844 | " shutdown"); |
| 845 | } |
| 846 | } |
| 847 | |
| 848 | void |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 849 | Channel::RecordFileEnded(const int32_t id) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 850 | { |
| 851 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 852 | "Channel::RecordFileEnded(id=%d)", id); |
| 853 | |
| 854 | assert(id == _outputFileRecorderId); |
| 855 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 856 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 857 | |
| 858 | _outputFileRecording = false; |
| 859 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 860 | VoEId(_instanceId,_channelId), |
| 861 | "Channel::RecordFileEnded() => output file recorder module is" |
| 862 | " shutdown"); |
| 863 | } |
| 864 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 865 | Channel::Channel(const int32_t channelId, |
| 866 | const uint32_t instanceId) : |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 867 | _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| 868 | _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 869 | _instanceId(instanceId), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 870 | _channelId(channelId), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 871 | _audioCodingModule(*AudioCodingModule::Create( |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 872 | VoEModuleId(instanceId, channelId))), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 873 | _rtpDumpIn(*RtpDump::CreateRtpDump()), |
| 874 | _rtpDumpOut(*RtpDump::CreateRtpDump()), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 875 | _outputAudioLevel(), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 876 | _externalTransport(false), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 877 | _inputFilePlayerPtr(NULL), |
| 878 | _outputFilePlayerPtr(NULL), |
| 879 | _outputFileRecorderPtr(NULL), |
| 880 | // Avoid conflict with other channels by adding 1024 - 1026, |
| 881 | // won't use as much as 1024 channels. |
| 882 | _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024), |
| 883 | _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025), |
| 884 | _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026), |
| 885 | _inputFilePlaying(false), |
| 886 | _outputFilePlaying(false), |
| 887 | _outputFileRecording(false), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 888 | _inbandDtmfQueue(VoEModuleId(instanceId, channelId)), |
| 889 | _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 890 | _inputExternalMedia(false), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 891 | _outputExternalMedia(false), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 892 | _inputExternalMediaCallbackPtr(NULL), |
| 893 | _outputExternalMediaCallbackPtr(NULL), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 894 | _encryptionRTPBufferPtr(NULL), |
| 895 | _decryptionRTPBufferPtr(NULL), |
| 896 | _encryptionRTCPBufferPtr(NULL), |
| 897 | _decryptionRTCPBufferPtr(NULL), |
| 898 | _timeStamp(0), // This is just an offset, RTP module will add it's own random offset |
| 899 | _sendTelephoneEventPayloadType(106), |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 900 | playout_timestamp_rtp_(0), |
| 901 | playout_timestamp_rtcp_(0), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 902 | _numberOfDiscardedPackets(0), |
| 903 | _engineStatisticsPtr(NULL), |
henrika@webrtc.org | 2919e95 | 2012-01-31 08:45:03 +0000 | [diff] [blame] | 904 | _outputMixerPtr(NULL), |
| 905 | _transmitMixerPtr(NULL), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 906 | _moduleProcessThreadPtr(NULL), |
| 907 | _audioDeviceModulePtr(NULL), |
| 908 | _voiceEngineObserverPtr(NULL), |
| 909 | _callbackCritSectPtr(NULL), |
| 910 | _transportPtr(NULL), |
| 911 | _encryptionPtr(NULL), |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 912 | _rtpAudioProc(NULL), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 913 | _rxAudioProcessingModulePtr(NULL), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 914 | _rxVadObserverPtr(NULL), |
| 915 | _oldVadDecision(-1), |
| 916 | _sendFrameType(0), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 917 | _rtpObserverPtr(NULL), |
| 918 | _rtcpObserverPtr(NULL), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 919 | _outputIsOnHold(false), |
| 920 | _externalPlayout(false), |
roosa@google.com | 1b60ceb | 2012-12-12 23:00:29 +0000 | [diff] [blame] | 921 | _externalMixing(false), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 922 | _inputIsOnHold(false), |
| 923 | _playing(false), |
| 924 | _sending(false), |
| 925 | _receiving(false), |
| 926 | _mixFileWithMicrophone(false), |
| 927 | _rtpObserver(false), |
| 928 | _rtcpObserver(false), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 929 | _mute(false), |
| 930 | _panLeft(1.0f), |
| 931 | _panRight(1.0f), |
| 932 | _outputGain(1.0f), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 933 | _encrypting(false), |
| 934 | _decrypting(false), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 935 | _playOutbandDtmfEvent(false), |
| 936 | _playInbandDtmfEvent(false), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 937 | _extraPayloadType(0), |
| 938 | _insertExtraRTPPacket(false), |
| 939 | _extraMarkerBit(false), |
| 940 | _lastLocalTimeStamp(0), |
roosa@google.com | 0870f02 | 2012-12-12 21:31:41 +0000 | [diff] [blame] | 941 | _lastRemoteTimeStamp(0), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 942 | _lastPayloadType(0), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 943 | _includeAudioLevelIndication(false), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 944 | _rtpPacketTimedOut(false), |
| 945 | _rtpPacketTimeOutIsEnabled(false), |
| 946 | _rtpTimeOutSeconds(0), |
| 947 | _connectionObserver(false), |
| 948 | _connectionObserverPtr(NULL), |
| 949 | _countAliveDetections(0), |
| 950 | _countDeadDetections(0), |
| 951 | _outputSpeechType(AudioFrame::kNormalSpeech), |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 952 | _average_jitter_buffer_delay_us(0), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 953 | _previousTimestamp(0), |
| 954 | _recPacketDelayMs(20), |
| 955 | _RxVadDetection(false), |
| 956 | _rxApmIsEnabled(false), |
| 957 | _rxAgcIsEnabled(false), |
xians@google.com | 22963ab | 2011-08-03 12:40:23 +0000 | [diff] [blame] | 958 | _rxNsIsEnabled(false) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 959 | { |
| 960 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId), |
| 961 | "Channel::Channel() - ctor"); |
| 962 | _inbandDtmfQueue.ResetDtmf(); |
| 963 | _inbandDtmfGenerator.Init(); |
| 964 | _outputAudioLevel.Clear(); |
| 965 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 966 | RtpRtcp::Configuration configuration; |
| 967 | configuration.id = VoEModuleId(instanceId, channelId); |
| 968 | configuration.audio = true; |
| 969 | configuration.incoming_data = this; |
| 970 | configuration.incoming_messages = this; |
| 971 | configuration.outgoing_transport = this; |
| 972 | configuration.rtcp_feedback = this; |
| 973 | configuration.audio_messages = this; |
| 974 | |
| 975 | _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration)); |
| 976 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 977 | // Create far end AudioProcessing Module |
| 978 | _rxAudioProcessingModulePtr = AudioProcessing::Create( |
| 979 | VoEModuleId(instanceId, channelId)); |
| 980 | } |
| 981 | |
| 982 | Channel::~Channel() |
| 983 | { |
| 984 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId), |
| 985 | "Channel::~Channel() - dtor"); |
| 986 | |
| 987 | if (_outputExternalMedia) |
| 988 | { |
| 989 | DeRegisterExternalMediaProcessing(kPlaybackPerChannel); |
| 990 | } |
| 991 | if (_inputExternalMedia) |
| 992 | { |
| 993 | DeRegisterExternalMediaProcessing(kRecordingPerChannel); |
| 994 | } |
| 995 | StopSend(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 996 | StopPlayout(); |
| 997 | |
| 998 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 999 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1000 | if (_inputFilePlayerPtr) |
| 1001 | { |
| 1002 | _inputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 1003 | _inputFilePlayerPtr->StopPlayingFile(); |
| 1004 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 1005 | _inputFilePlayerPtr = NULL; |
| 1006 | } |
| 1007 | if (_outputFilePlayerPtr) |
| 1008 | { |
| 1009 | _outputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 1010 | _outputFilePlayerPtr->StopPlayingFile(); |
| 1011 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 1012 | _outputFilePlayerPtr = NULL; |
| 1013 | } |
| 1014 | if (_outputFileRecorderPtr) |
| 1015 | { |
| 1016 | _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 1017 | _outputFileRecorderPtr->StopRecording(); |
| 1018 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 1019 | _outputFileRecorderPtr = NULL; |
| 1020 | } |
| 1021 | } |
| 1022 | |
| 1023 | // The order to safely shutdown modules in a channel is: |
| 1024 | // 1. De-register callbacks in modules |
| 1025 | // 2. De-register modules in process thread |
| 1026 | // 3. Destroy modules |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1027 | if (_audioCodingModule.RegisterTransportCallback(NULL) == -1) |
| 1028 | { |
| 1029 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1030 | VoEId(_instanceId,_channelId), |
| 1031 | "~Channel() failed to de-register transport callback" |
| 1032 | " (Audio coding module)"); |
| 1033 | } |
| 1034 | if (_audioCodingModule.RegisterVADCallback(NULL) == -1) |
| 1035 | { |
| 1036 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1037 | VoEId(_instanceId,_channelId), |
| 1038 | "~Channel() failed to de-register VAD callback" |
| 1039 | " (Audio coding module)"); |
| 1040 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1041 | // De-register modules in process thread |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1042 | if (_moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get()) == -1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1043 | { |
| 1044 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 1045 | VoEId(_instanceId,_channelId), |
| 1046 | "~Channel() failed to deregister RTP/RTCP module"); |
| 1047 | } |
| 1048 | |
| 1049 | // Destroy modules |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1050 | AudioCodingModule::Destroy(&_audioCodingModule); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1051 | if (_rxAudioProcessingModulePtr != NULL) |
| 1052 | { |
| 1053 | AudioProcessing::Destroy(_rxAudioProcessingModulePtr); // far end APM |
| 1054 | _rxAudioProcessingModulePtr = NULL; |
| 1055 | } |
| 1056 | |
| 1057 | // End of modules shutdown |
| 1058 | |
| 1059 | // Delete other objects |
| 1060 | RtpDump::DestroyRtpDump(&_rtpDumpIn); |
| 1061 | RtpDump::DestroyRtpDump(&_rtpDumpOut); |
| 1062 | delete [] _encryptionRTPBufferPtr; |
| 1063 | delete [] _decryptionRTPBufferPtr; |
| 1064 | delete [] _encryptionRTCPBufferPtr; |
| 1065 | delete [] _decryptionRTCPBufferPtr; |
| 1066 | delete &_callbackCritSect; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1067 | delete &_fileCritSect; |
| 1068 | } |
| 1069 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1070 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1071 | Channel::Init() |
| 1072 | { |
| 1073 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1074 | "Channel::Init()"); |
| 1075 | |
| 1076 | // --- Initial sanity |
| 1077 | |
| 1078 | if ((_engineStatisticsPtr == NULL) || |
| 1079 | (_moduleProcessThreadPtr == NULL)) |
| 1080 | { |
| 1081 | WEBRTC_TRACE(kTraceError, kTraceVoice, |
| 1082 | VoEId(_instanceId,_channelId), |
| 1083 | "Channel::Init() must call SetEngineInformation() first"); |
| 1084 | return -1; |
| 1085 | } |
| 1086 | |
| 1087 | // --- Add modules to process thread (for periodic schedulation) |
| 1088 | |
| 1089 | const bool processThreadFail = |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1090 | ((_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get()) != 0) || |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1091 | false); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1092 | if (processThreadFail) |
| 1093 | { |
| 1094 | _engineStatisticsPtr->SetLastError( |
| 1095 | VE_CANNOT_INIT_CHANNEL, kTraceError, |
| 1096 | "Channel::Init() modules not registered"); |
| 1097 | return -1; |
| 1098 | } |
pwestin@webrtc.org | c450a19 | 2012-01-04 15:00:12 +0000 | [diff] [blame] | 1099 | // --- ACM initialization |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1100 | |
| 1101 | if ((_audioCodingModule.InitializeReceiver() == -1) || |
| 1102 | #ifdef WEBRTC_CODEC_AVT |
| 1103 | // out-of-band Dtmf tones are played out by default |
| 1104 | (_audioCodingModule.SetDtmfPlayoutStatus(true) == -1) || |
| 1105 | #endif |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1106 | (_audioCodingModule.InitializeSender() == -1)) |
| 1107 | { |
| 1108 | _engineStatisticsPtr->SetLastError( |
| 1109 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1110 | "Channel::Init() unable to initialize the ACM - 1"); |
| 1111 | return -1; |
| 1112 | } |
| 1113 | |
| 1114 | // --- RTP/RTCP module initialization |
| 1115 | |
| 1116 | // Ensure that RTCP is enabled by default for the created channel. |
| 1117 | // Note that, the module will keep generating RTCP until it is explicitly |
| 1118 | // disabled by the user. |
| 1119 | // After StopListen (when no sockets exists), RTCP packets will no longer |
| 1120 | // be transmitted since the Transport object will then be invalid. |
| 1121 | |
| 1122 | const bool rtpRtcpFail = |
turaj@webrtc.org | b7edd06 | 2013-03-12 22:27:27 +0000 | [diff] [blame] | 1123 | ((_rtpRtcpModule->SetTelephoneEventForwardToDecoder(true) == -1) || |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1124 | // RTCP is enabled by default |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1125 | (_rtpRtcpModule->SetRTCPStatus(kRtcpCompound) == -1)); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1126 | if (rtpRtcpFail) |
| 1127 | { |
| 1128 | _engineStatisticsPtr->SetLastError( |
| 1129 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 1130 | "Channel::Init() RTP/RTCP module not initialized"); |
| 1131 | return -1; |
| 1132 | } |
| 1133 | |
| 1134 | // --- Register all permanent callbacks |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1135 | const bool fail = |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1136 | (_audioCodingModule.RegisterTransportCallback(this) == -1) || |
| 1137 | (_audioCodingModule.RegisterVADCallback(this) == -1); |
| 1138 | |
| 1139 | if (fail) |
| 1140 | { |
| 1141 | _engineStatisticsPtr->SetLastError( |
| 1142 | VE_CANNOT_INIT_CHANNEL, kTraceError, |
| 1143 | "Channel::Init() callbacks not registered"); |
| 1144 | return -1; |
| 1145 | } |
| 1146 | |
| 1147 | // --- Register all supported codecs to the receiving side of the |
| 1148 | // RTP/RTCP module |
| 1149 | |
| 1150 | CodecInst codec; |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1151 | const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1152 | |
| 1153 | for (int idx = 0; idx < nSupportedCodecs; idx++) |
| 1154 | { |
| 1155 | // Open up the RTP/RTCP receiver for all supported codecs |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 1156 | if ((_audioCodingModule.Codec(idx, &codec) == -1) || |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1157 | (_rtpRtcpModule->RegisterReceivePayload(codec) == -1)) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1158 | { |
| 1159 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1160 | VoEId(_instanceId,_channelId), |
| 1161 | "Channel::Init() unable to register %s (%d/%d/%d/%d) " |
| 1162 | "to RTP/RTCP receiver", |
| 1163 | codec.plname, codec.pltype, codec.plfreq, |
| 1164 | codec.channels, codec.rate); |
| 1165 | } |
| 1166 | else |
| 1167 | { |
| 1168 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| 1169 | VoEId(_instanceId,_channelId), |
| 1170 | "Channel::Init() %s (%d/%d/%d/%d) has been added to " |
| 1171 | "the RTP/RTCP receiver", |
| 1172 | codec.plname, codec.pltype, codec.plfreq, |
| 1173 | codec.channels, codec.rate); |
| 1174 | } |
| 1175 | |
| 1176 | // Ensure that PCMU is used as default codec on the sending side |
tina.legrand@webrtc.org | 4517585 | 2012-06-01 09:27:35 +0000 | [diff] [blame] | 1177 | if (!STR_CASE_CMP(codec.plname, "PCMU") && (codec.channels == 1)) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1178 | { |
| 1179 | SetSendCodec(codec); |
| 1180 | } |
| 1181 | |
| 1182 | // Register default PT for outband 'telephone-event' |
| 1183 | if (!STR_CASE_CMP(codec.plname, "telephone-event")) |
| 1184 | { |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1185 | if ((_rtpRtcpModule->RegisterSendPayload(codec) == -1) || |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1186 | (_audioCodingModule.RegisterReceiveCodec(codec) == -1)) |
| 1187 | { |
| 1188 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1189 | VoEId(_instanceId,_channelId), |
| 1190 | "Channel::Init() failed to register outband " |
| 1191 | "'telephone-event' (%d/%d) correctly", |
| 1192 | codec.pltype, codec.plfreq); |
| 1193 | } |
| 1194 | } |
| 1195 | |
| 1196 | if (!STR_CASE_CMP(codec.plname, "CN")) |
| 1197 | { |
| 1198 | if ((_audioCodingModule.RegisterSendCodec(codec) == -1) || |
| 1199 | (_audioCodingModule.RegisterReceiveCodec(codec) == -1) || |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1200 | (_rtpRtcpModule->RegisterSendPayload(codec) == -1)) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1201 | { |
| 1202 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1203 | VoEId(_instanceId,_channelId), |
| 1204 | "Channel::Init() failed to register CN (%d/%d) " |
| 1205 | "correctly - 1", |
| 1206 | codec.pltype, codec.plfreq); |
| 1207 | } |
| 1208 | } |
| 1209 | #ifdef WEBRTC_CODEC_RED |
| 1210 | // Register RED to the receiving side of the ACM. |
| 1211 | // We will not receive an OnInitializeDecoder() callback for RED. |
| 1212 | if (!STR_CASE_CMP(codec.plname, "RED")) |
| 1213 | { |
| 1214 | if (_audioCodingModule.RegisterReceiveCodec(codec) == -1) |
| 1215 | { |
| 1216 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1217 | VoEId(_instanceId,_channelId), |
| 1218 | "Channel::Init() failed to register RED (%d/%d) " |
| 1219 | "correctly", |
| 1220 | codec.pltype, codec.plfreq); |
| 1221 | } |
| 1222 | } |
| 1223 | #endif |
| 1224 | } |
pwestin@webrtc.org | 684f057 | 2013-03-13 23:20:57 +0000 | [diff] [blame] | 1225 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1226 | // Initialize the far end AP module |
| 1227 | // Using 8 kHz as initial Fs, the same as in transmission. Might be |
| 1228 | // changed at the first receiving audio. |
| 1229 | if (_rxAudioProcessingModulePtr == NULL) |
| 1230 | { |
| 1231 | _engineStatisticsPtr->SetLastError( |
| 1232 | VE_NO_MEMORY, kTraceCritical, |
| 1233 | "Channel::Init() failed to create the far-end AudioProcessing" |
| 1234 | " module"); |
| 1235 | return -1; |
| 1236 | } |
| 1237 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1238 | if (_rxAudioProcessingModulePtr->set_sample_rate_hz(8000)) |
| 1239 | { |
| 1240 | _engineStatisticsPtr->SetLastError( |
| 1241 | VE_APM_ERROR, kTraceWarning, |
| 1242 | "Channel::Init() failed to set the sample rate to 8K for" |
| 1243 | " far-end AP module"); |
| 1244 | } |
| 1245 | |
| 1246 | if (_rxAudioProcessingModulePtr->set_num_channels(1, 1) != 0) |
| 1247 | { |
| 1248 | _engineStatisticsPtr->SetLastError( |
| 1249 | VE_SOUNDCARD_ERROR, kTraceWarning, |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 1250 | "Init() failed to set channels for the primary audio stream"); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1251 | } |
| 1252 | |
| 1253 | if (_rxAudioProcessingModulePtr->high_pass_filter()->Enable( |
| 1254 | WEBRTC_VOICE_ENGINE_RX_HP_DEFAULT_STATE) != 0) |
| 1255 | { |
| 1256 | _engineStatisticsPtr->SetLastError( |
| 1257 | VE_APM_ERROR, kTraceWarning, |
| 1258 | "Channel::Init() failed to set the high-pass filter for" |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 1259 | " far-end AP module"); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1260 | } |
| 1261 | |
| 1262 | if (_rxAudioProcessingModulePtr->noise_suppression()->set_level( |
| 1263 | (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE) != 0) |
| 1264 | { |
| 1265 | _engineStatisticsPtr->SetLastError( |
| 1266 | VE_APM_ERROR, kTraceWarning, |
| 1267 | "Init() failed to set noise reduction level for far-end" |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 1268 | " AP module"); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1269 | } |
| 1270 | if (_rxAudioProcessingModulePtr->noise_suppression()->Enable( |
| 1271 | WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE) != 0) |
| 1272 | { |
| 1273 | _engineStatisticsPtr->SetLastError( |
| 1274 | VE_APM_ERROR, kTraceWarning, |
| 1275 | "Init() failed to set noise reduction state for far-end" |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 1276 | " AP module"); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1277 | } |
| 1278 | |
| 1279 | if (_rxAudioProcessingModulePtr->gain_control()->set_mode( |
| 1280 | (GainControl::Mode)WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_MODE) != 0) |
| 1281 | { |
| 1282 | _engineStatisticsPtr->SetLastError( |
| 1283 | VE_APM_ERROR, kTraceWarning, |
| 1284 | "Init() failed to set AGC mode for far-end AP module"); |
| 1285 | } |
| 1286 | if (_rxAudioProcessingModulePtr->gain_control()->Enable( |
| 1287 | WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_STATE) != 0) |
| 1288 | { |
| 1289 | _engineStatisticsPtr->SetLastError( |
| 1290 | VE_APM_ERROR, kTraceWarning, |
| 1291 | "Init() failed to set AGC state for far-end AP module"); |
| 1292 | } |
| 1293 | |
| 1294 | return 0; |
| 1295 | } |
| 1296 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1297 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1298 | Channel::SetEngineInformation(Statistics& engineStatistics, |
| 1299 | OutputMixer& outputMixer, |
| 1300 | voe::TransmitMixer& transmitMixer, |
| 1301 | ProcessThread& moduleProcessThread, |
| 1302 | AudioDeviceModule& audioDeviceModule, |
| 1303 | VoiceEngineObserver* voiceEngineObserver, |
| 1304 | CriticalSectionWrapper* callbackCritSect) |
| 1305 | { |
| 1306 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1307 | "Channel::SetEngineInformation()"); |
| 1308 | _engineStatisticsPtr = &engineStatistics; |
| 1309 | _outputMixerPtr = &outputMixer; |
| 1310 | _transmitMixerPtr = &transmitMixer, |
| 1311 | _moduleProcessThreadPtr = &moduleProcessThread; |
| 1312 | _audioDeviceModulePtr = &audioDeviceModule; |
| 1313 | _voiceEngineObserverPtr = voiceEngineObserver; |
| 1314 | _callbackCritSectPtr = callbackCritSect; |
| 1315 | return 0; |
| 1316 | } |
| 1317 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1318 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1319 | Channel::UpdateLocalTimeStamp() |
| 1320 | { |
| 1321 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 1322 | _timeStamp += _audioFrame.samples_per_channel_; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1323 | return 0; |
| 1324 | } |
| 1325 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1326 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1327 | Channel::StartPlayout() |
| 1328 | { |
| 1329 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1330 | "Channel::StartPlayout()"); |
| 1331 | if (_playing) |
| 1332 | { |
| 1333 | return 0; |
| 1334 | } |
roosa@google.com | 1b60ceb | 2012-12-12 23:00:29 +0000 | [diff] [blame] | 1335 | |
| 1336 | if (!_externalMixing) { |
| 1337 | // Add participant as candidates for mixing. |
| 1338 | if (_outputMixerPtr->SetMixabilityStatus(*this, true) != 0) |
| 1339 | { |
| 1340 | _engineStatisticsPtr->SetLastError( |
| 1341 | VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError, |
| 1342 | "StartPlayout() failed to add participant to mixer"); |
| 1343 | return -1; |
| 1344 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1345 | } |
| 1346 | |
| 1347 | _playing = true; |
braveyao@webrtc.org | ab12990 | 2012-06-04 03:26:39 +0000 | [diff] [blame] | 1348 | |
| 1349 | if (RegisterFilePlayingToMixer() != 0) |
| 1350 | return -1; |
| 1351 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1352 | return 0; |
| 1353 | } |
| 1354 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1355 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1356 | Channel::StopPlayout() |
| 1357 | { |
| 1358 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1359 | "Channel::StopPlayout()"); |
| 1360 | if (!_playing) |
| 1361 | { |
| 1362 | return 0; |
| 1363 | } |
roosa@google.com | 1b60ceb | 2012-12-12 23:00:29 +0000 | [diff] [blame] | 1364 | |
| 1365 | if (!_externalMixing) { |
| 1366 | // Remove participant as candidates for mixing |
| 1367 | if (_outputMixerPtr->SetMixabilityStatus(*this, false) != 0) |
| 1368 | { |
| 1369 | _engineStatisticsPtr->SetLastError( |
| 1370 | VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError, |
| 1371 | "StopPlayout() failed to remove participant from mixer"); |
| 1372 | return -1; |
| 1373 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1374 | } |
| 1375 | |
| 1376 | _playing = false; |
| 1377 | _outputAudioLevel.Clear(); |
| 1378 | |
| 1379 | return 0; |
| 1380 | } |
| 1381 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1382 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1383 | Channel::StartSend() |
| 1384 | { |
| 1385 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1386 | "Channel::StartSend()"); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1387 | { |
xians@webrtc.org | e07247a | 2011-11-28 16:31:28 +0000 | [diff] [blame] | 1388 | // A lock is needed because |_sending| can be accessed or modified by |
| 1389 | // another thread at the same time. |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 1390 | CriticalSectionScoped cs(&_callbackCritSect); |
xians@webrtc.org | e07247a | 2011-11-28 16:31:28 +0000 | [diff] [blame] | 1391 | |
| 1392 | if (_sending) |
| 1393 | { |
| 1394 | return 0; |
| 1395 | } |
| 1396 | _sending = true; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1397 | } |
xians@webrtc.org | e07247a | 2011-11-28 16:31:28 +0000 | [diff] [blame] | 1398 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1399 | if (_rtpRtcpModule->SetSendingStatus(true) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1400 | { |
| 1401 | _engineStatisticsPtr->SetLastError( |
| 1402 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 1403 | "StartSend() RTP/RTCP failed to start sending"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 1404 | CriticalSectionScoped cs(&_callbackCritSect); |
xians@webrtc.org | e07247a | 2011-11-28 16:31:28 +0000 | [diff] [blame] | 1405 | _sending = false; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1406 | return -1; |
| 1407 | } |
xians@webrtc.org | e07247a | 2011-11-28 16:31:28 +0000 | [diff] [blame] | 1408 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1409 | return 0; |
| 1410 | } |
| 1411 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1412 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1413 | Channel::StopSend() |
| 1414 | { |
| 1415 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1416 | "Channel::StopSend()"); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1417 | { |
xians@webrtc.org | e07247a | 2011-11-28 16:31:28 +0000 | [diff] [blame] | 1418 | // A lock is needed because |_sending| can be accessed or modified by |
| 1419 | // another thread at the same time. |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 1420 | CriticalSectionScoped cs(&_callbackCritSect); |
xians@webrtc.org | e07247a | 2011-11-28 16:31:28 +0000 | [diff] [blame] | 1421 | |
| 1422 | if (!_sending) |
| 1423 | { |
| 1424 | return 0; |
| 1425 | } |
| 1426 | _sending = false; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1427 | } |
xians@webrtc.org | e07247a | 2011-11-28 16:31:28 +0000 | [diff] [blame] | 1428 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1429 | // Reset sending SSRC and sequence number and triggers direct transmission |
| 1430 | // of RTCP BYE |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1431 | if (_rtpRtcpModule->SetSendingStatus(false) == -1 || |
| 1432 | _rtpRtcpModule->ResetSendDataCountersRTP() == -1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1433 | { |
| 1434 | _engineStatisticsPtr->SetLastError( |
| 1435 | VE_RTP_RTCP_MODULE_ERROR, kTraceWarning, |
| 1436 | "StartSend() RTP/RTCP failed to stop sending"); |
| 1437 | } |
| 1438 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1439 | return 0; |
| 1440 | } |
| 1441 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1442 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1443 | Channel::StartReceiving() |
| 1444 | { |
| 1445 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1446 | "Channel::StartReceiving()"); |
| 1447 | if (_receiving) |
| 1448 | { |
| 1449 | return 0; |
| 1450 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1451 | _receiving = true; |
| 1452 | _numberOfDiscardedPackets = 0; |
| 1453 | return 0; |
| 1454 | } |
| 1455 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1456 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1457 | Channel::StopReceiving() |
| 1458 | { |
| 1459 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1460 | "Channel::StopReceiving()"); |
| 1461 | if (!_receiving) |
| 1462 | { |
| 1463 | return 0; |
| 1464 | } |
pwestin@webrtc.org | 684f057 | 2013-03-13 23:20:57 +0000 | [diff] [blame] | 1465 | |
henrika@webrtc.org | af71f0e | 2011-12-05 07:02:22 +0000 | [diff] [blame] | 1466 | // Recover DTMF detection status. |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1467 | int32_t ret = _rtpRtcpModule->SetTelephoneEventForwardToDecoder(true); |
henrika@webrtc.org | af71f0e | 2011-12-05 07:02:22 +0000 | [diff] [blame] | 1468 | if (ret != 0) { |
| 1469 | _engineStatisticsPtr->SetLastError( |
| 1470 | VE_INVALID_OPERATION, kTraceWarning, |
| 1471 | "StopReceiving() failed to restore telephone-event status."); |
| 1472 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1473 | RegisterReceiveCodecsToRTPModule(); |
| 1474 | _receiving = false; |
| 1475 | return 0; |
| 1476 | } |
| 1477 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1478 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1479 | Channel::SetNetEQPlayoutMode(NetEqModes mode) |
| 1480 | { |
| 1481 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1482 | "Channel::SetNetEQPlayoutMode()"); |
| 1483 | AudioPlayoutMode playoutMode(voice); |
| 1484 | switch (mode) |
| 1485 | { |
| 1486 | case kNetEqDefault: |
| 1487 | playoutMode = voice; |
| 1488 | break; |
| 1489 | case kNetEqStreaming: |
| 1490 | playoutMode = streaming; |
| 1491 | break; |
| 1492 | case kNetEqFax: |
| 1493 | playoutMode = fax; |
| 1494 | break; |
roosa@google.com | b718619 | 2012-12-12 21:59:14 +0000 | [diff] [blame] | 1495 | case kNetEqOff: |
| 1496 | playoutMode = off; |
| 1497 | break; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1498 | } |
| 1499 | if (_audioCodingModule.SetPlayoutMode(playoutMode) != 0) |
| 1500 | { |
| 1501 | _engineStatisticsPtr->SetLastError( |
| 1502 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1503 | "SetNetEQPlayoutMode() failed to set playout mode"); |
| 1504 | return -1; |
| 1505 | } |
| 1506 | return 0; |
| 1507 | } |
| 1508 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1509 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1510 | Channel::GetNetEQPlayoutMode(NetEqModes& mode) |
| 1511 | { |
| 1512 | const AudioPlayoutMode playoutMode = _audioCodingModule.PlayoutMode(); |
| 1513 | switch (playoutMode) |
| 1514 | { |
| 1515 | case voice: |
| 1516 | mode = kNetEqDefault; |
| 1517 | break; |
| 1518 | case streaming: |
| 1519 | mode = kNetEqStreaming; |
| 1520 | break; |
| 1521 | case fax: |
| 1522 | mode = kNetEqFax; |
| 1523 | break; |
roosa@google.com | b718619 | 2012-12-12 21:59:14 +0000 | [diff] [blame] | 1524 | case off: |
| 1525 | mode = kNetEqOff; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1526 | } |
| 1527 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 1528 | VoEId(_instanceId,_channelId), |
| 1529 | "Channel::GetNetEQPlayoutMode() => mode=%u", mode); |
| 1530 | return 0; |
| 1531 | } |
| 1532 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1533 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1534 | Channel::SetOnHoldStatus(bool enable, OnHoldModes mode) |
| 1535 | { |
| 1536 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1537 | "Channel::SetOnHoldStatus()"); |
| 1538 | if (mode == kHoldSendAndPlay) |
| 1539 | { |
| 1540 | _outputIsOnHold = enable; |
| 1541 | _inputIsOnHold = enable; |
| 1542 | } |
| 1543 | else if (mode == kHoldPlayOnly) |
| 1544 | { |
| 1545 | _outputIsOnHold = enable; |
| 1546 | } |
| 1547 | if (mode == kHoldSendOnly) |
| 1548 | { |
| 1549 | _inputIsOnHold = enable; |
| 1550 | } |
| 1551 | return 0; |
| 1552 | } |
| 1553 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1554 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1555 | Channel::GetOnHoldStatus(bool& enabled, OnHoldModes& mode) |
| 1556 | { |
| 1557 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1558 | "Channel::GetOnHoldStatus()"); |
| 1559 | enabled = (_outputIsOnHold || _inputIsOnHold); |
| 1560 | if (_outputIsOnHold && _inputIsOnHold) |
| 1561 | { |
| 1562 | mode = kHoldSendAndPlay; |
| 1563 | } |
| 1564 | else if (_outputIsOnHold && !_inputIsOnHold) |
| 1565 | { |
| 1566 | mode = kHoldPlayOnly; |
| 1567 | } |
| 1568 | else if (!_outputIsOnHold && _inputIsOnHold) |
| 1569 | { |
| 1570 | mode = kHoldSendOnly; |
| 1571 | } |
| 1572 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1573 | "Channel::GetOnHoldStatus() => enabled=%d, mode=%d", |
| 1574 | enabled, mode); |
| 1575 | return 0; |
| 1576 | } |
| 1577 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1578 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1579 | Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) |
| 1580 | { |
| 1581 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1582 | "Channel::RegisterVoiceEngineObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 1583 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1584 | |
| 1585 | if (_voiceEngineObserverPtr) |
| 1586 | { |
| 1587 | _engineStatisticsPtr->SetLastError( |
| 1588 | VE_INVALID_OPERATION, kTraceError, |
| 1589 | "RegisterVoiceEngineObserver() observer already enabled"); |
| 1590 | return -1; |
| 1591 | } |
| 1592 | _voiceEngineObserverPtr = &observer; |
| 1593 | return 0; |
| 1594 | } |
| 1595 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1596 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1597 | Channel::DeRegisterVoiceEngineObserver() |
| 1598 | { |
| 1599 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1600 | "Channel::DeRegisterVoiceEngineObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 1601 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1602 | |
| 1603 | if (!_voiceEngineObserverPtr) |
| 1604 | { |
| 1605 | _engineStatisticsPtr->SetLastError( |
| 1606 | VE_INVALID_OPERATION, kTraceWarning, |
| 1607 | "DeRegisterVoiceEngineObserver() observer already disabled"); |
| 1608 | return 0; |
| 1609 | } |
| 1610 | _voiceEngineObserverPtr = NULL; |
| 1611 | return 0; |
| 1612 | } |
| 1613 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1614 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1615 | Channel::GetSendCodec(CodecInst& codec) |
| 1616 | { |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 1617 | return (_audioCodingModule.SendCodec(&codec)); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1618 | } |
| 1619 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1620 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1621 | Channel::GetRecCodec(CodecInst& codec) |
| 1622 | { |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 1623 | return (_audioCodingModule.ReceiveCodec(&codec)); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1624 | } |
| 1625 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1626 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1627 | Channel::SetSendCodec(const CodecInst& codec) |
| 1628 | { |
| 1629 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1630 | "Channel::SetSendCodec()"); |
| 1631 | |
| 1632 | if (_audioCodingModule.RegisterSendCodec(codec) != 0) |
| 1633 | { |
| 1634 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1635 | "SetSendCodec() failed to register codec to ACM"); |
| 1636 | return -1; |
| 1637 | } |
| 1638 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1639 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1640 | { |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1641 | _rtpRtcpModule->DeRegisterSendPayload(codec.pltype); |
| 1642 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1643 | { |
| 1644 | WEBRTC_TRACE( |
| 1645 | kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1646 | "SetSendCodec() failed to register codec to" |
| 1647 | " RTP/RTCP module"); |
| 1648 | return -1; |
| 1649 | } |
| 1650 | } |
| 1651 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1652 | if (_rtpRtcpModule->SetAudioPacketSize(codec.pacsize) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1653 | { |
| 1654 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1655 | "SetSendCodec() failed to set audio packet size"); |
| 1656 | return -1; |
| 1657 | } |
| 1658 | |
| 1659 | return 0; |
| 1660 | } |
| 1661 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1662 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1663 | Channel::SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX) |
| 1664 | { |
| 1665 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1666 | "Channel::SetVADStatus(mode=%d)", mode); |
| 1667 | // To disable VAD, DTX must be disabled too |
| 1668 | disableDTX = ((enableVAD == false) ? true : disableDTX); |
| 1669 | if (_audioCodingModule.SetVAD(!disableDTX, enableVAD, mode) != 0) |
| 1670 | { |
| 1671 | _engineStatisticsPtr->SetLastError( |
| 1672 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1673 | "SetVADStatus() failed to set VAD"); |
| 1674 | return -1; |
| 1675 | } |
| 1676 | return 0; |
| 1677 | } |
| 1678 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1679 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1680 | Channel::GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX) |
| 1681 | { |
| 1682 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1683 | "Channel::GetVADStatus"); |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 1684 | if (_audioCodingModule.VAD(&disabledDTX, &enabledVAD, &mode) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1685 | { |
| 1686 | _engineStatisticsPtr->SetLastError( |
| 1687 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1688 | "GetVADStatus() failed to get VAD status"); |
| 1689 | return -1; |
| 1690 | } |
| 1691 | disabledDTX = !disabledDTX; |
| 1692 | return 0; |
| 1693 | } |
| 1694 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1695 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1696 | Channel::SetRecPayloadType(const CodecInst& codec) |
| 1697 | { |
| 1698 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1699 | "Channel::SetRecPayloadType()"); |
| 1700 | |
| 1701 | if (_playing) |
| 1702 | { |
| 1703 | _engineStatisticsPtr->SetLastError( |
| 1704 | VE_ALREADY_PLAYING, kTraceError, |
| 1705 | "SetRecPayloadType() unable to set PT while playing"); |
| 1706 | return -1; |
| 1707 | } |
| 1708 | if (_receiving) |
| 1709 | { |
| 1710 | _engineStatisticsPtr->SetLastError( |
| 1711 | VE_ALREADY_LISTENING, kTraceError, |
| 1712 | "SetRecPayloadType() unable to set PT while listening"); |
| 1713 | return -1; |
| 1714 | } |
| 1715 | |
| 1716 | if (codec.pltype == -1) |
| 1717 | { |
| 1718 | // De-register the selected codec (RTP/RTCP module and ACM) |
| 1719 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1720 | int8_t pltype(-1); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1721 | CodecInst rxCodec = codec; |
| 1722 | |
| 1723 | // Get payload type for the given codec |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1724 | _rtpRtcpModule->ReceivePayloadType(rxCodec, &pltype); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1725 | rxCodec.pltype = pltype; |
| 1726 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1727 | if (_rtpRtcpModule->DeRegisterReceivePayload(pltype) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1728 | { |
| 1729 | _engineStatisticsPtr->SetLastError( |
| 1730 | VE_RTP_RTCP_MODULE_ERROR, |
| 1731 | kTraceError, |
| 1732 | "SetRecPayloadType() RTP/RTCP-module deregistration " |
| 1733 | "failed"); |
| 1734 | return -1; |
| 1735 | } |
| 1736 | if (_audioCodingModule.UnregisterReceiveCodec(rxCodec.pltype) != 0) |
| 1737 | { |
| 1738 | _engineStatisticsPtr->SetLastError( |
| 1739 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1740 | "SetRecPayloadType() ACM deregistration failed - 1"); |
| 1741 | return -1; |
| 1742 | } |
| 1743 | return 0; |
| 1744 | } |
| 1745 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1746 | if (_rtpRtcpModule->RegisterReceivePayload(codec) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1747 | { |
| 1748 | // First attempt to register failed => de-register and try again |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1749 | _rtpRtcpModule->DeRegisterReceivePayload(codec.pltype); |
| 1750 | if (_rtpRtcpModule->RegisterReceivePayload(codec) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1751 | { |
| 1752 | _engineStatisticsPtr->SetLastError( |
| 1753 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 1754 | "SetRecPayloadType() RTP/RTCP-module registration failed"); |
| 1755 | return -1; |
| 1756 | } |
| 1757 | } |
| 1758 | if (_audioCodingModule.RegisterReceiveCodec(codec) != 0) |
| 1759 | { |
| 1760 | _audioCodingModule.UnregisterReceiveCodec(codec.pltype); |
| 1761 | if (_audioCodingModule.RegisterReceiveCodec(codec) != 0) |
| 1762 | { |
| 1763 | _engineStatisticsPtr->SetLastError( |
| 1764 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1765 | "SetRecPayloadType() ACM registration failed - 1"); |
| 1766 | return -1; |
| 1767 | } |
| 1768 | } |
| 1769 | return 0; |
| 1770 | } |
| 1771 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1772 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1773 | Channel::GetRecPayloadType(CodecInst& codec) |
| 1774 | { |
| 1775 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1776 | "Channel::GetRecPayloadType()"); |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1777 | int8_t payloadType(-1); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1778 | if (_rtpRtcpModule->ReceivePayloadType(codec, &payloadType) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1779 | { |
| 1780 | _engineStatisticsPtr->SetLastError( |
henrika@webrtc.org | 3719800 | 2012-06-18 11:00:12 +0000 | [diff] [blame] | 1781 | VE_RTP_RTCP_MODULE_ERROR, kTraceWarning, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1782 | "GetRecPayloadType() failed to retrieve RX payload type"); |
| 1783 | return -1; |
| 1784 | } |
| 1785 | codec.pltype = payloadType; |
| 1786 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1787 | "Channel::GetRecPayloadType() => pltype=%u", codec.pltype); |
| 1788 | return 0; |
| 1789 | } |
| 1790 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1791 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1792 | Channel::SetAMREncFormat(AmrMode mode) |
| 1793 | { |
| 1794 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1795 | "Channel::SetAMREncFormat()"); |
| 1796 | |
| 1797 | // ACM doesn't support AMR |
| 1798 | return -1; |
| 1799 | } |
| 1800 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1801 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1802 | Channel::SetAMRDecFormat(AmrMode mode) |
| 1803 | { |
| 1804 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1805 | "Channel::SetAMRDecFormat()"); |
| 1806 | |
| 1807 | // ACM doesn't support AMR |
| 1808 | return -1; |
| 1809 | } |
| 1810 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1811 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1812 | Channel::SetAMRWbEncFormat(AmrMode mode) |
| 1813 | { |
| 1814 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1815 | "Channel::SetAMRWbEncFormat()"); |
| 1816 | |
| 1817 | // ACM doesn't support AMR |
| 1818 | return -1; |
| 1819 | |
| 1820 | } |
| 1821 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1822 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1823 | Channel::SetAMRWbDecFormat(AmrMode mode) |
| 1824 | { |
| 1825 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1826 | "Channel::SetAMRWbDecFormat()"); |
| 1827 | |
| 1828 | // ACM doesn't support AMR |
| 1829 | return -1; |
| 1830 | } |
| 1831 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1832 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1833 | Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency) |
| 1834 | { |
| 1835 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1836 | "Channel::SetSendCNPayloadType()"); |
| 1837 | |
| 1838 | CodecInst codec; |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1839 | int32_t samplingFreqHz(-1); |
tina.legrand@webrtc.org | 4517585 | 2012-06-01 09:27:35 +0000 | [diff] [blame] | 1840 | const int kMono = 1; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1841 | if (frequency == kFreq32000Hz) |
| 1842 | samplingFreqHz = 32000; |
| 1843 | else if (frequency == kFreq16000Hz) |
| 1844 | samplingFreqHz = 16000; |
| 1845 | |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 1846 | if (_audioCodingModule.Codec("CN", &codec, samplingFreqHz, kMono) == -1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1847 | { |
| 1848 | _engineStatisticsPtr->SetLastError( |
| 1849 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1850 | "SetSendCNPayloadType() failed to retrieve default CN codec " |
| 1851 | "settings"); |
| 1852 | return -1; |
| 1853 | } |
| 1854 | |
| 1855 | // Modify the payload type (must be set to dynamic range) |
| 1856 | codec.pltype = type; |
| 1857 | |
| 1858 | if (_audioCodingModule.RegisterSendCodec(codec) != 0) |
| 1859 | { |
| 1860 | _engineStatisticsPtr->SetLastError( |
| 1861 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1862 | "SetSendCNPayloadType() failed to register CN to ACM"); |
| 1863 | return -1; |
| 1864 | } |
| 1865 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1866 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1867 | { |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 1868 | _rtpRtcpModule->DeRegisterSendPayload(codec.pltype); |
| 1869 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1870 | { |
| 1871 | _engineStatisticsPtr->SetLastError( |
| 1872 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 1873 | "SetSendCNPayloadType() failed to register CN to RTP/RTCP " |
| 1874 | "module"); |
| 1875 | return -1; |
| 1876 | } |
| 1877 | } |
| 1878 | return 0; |
| 1879 | } |
| 1880 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1881 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1882 | Channel::SetISACInitTargetRate(int rateBps, bool useFixedFrameSize) |
| 1883 | { |
| 1884 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1885 | "Channel::SetISACInitTargetRate()"); |
| 1886 | |
| 1887 | CodecInst sendCodec; |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 1888 | if (_audioCodingModule.SendCodec(&sendCodec) == -1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1889 | { |
| 1890 | _engineStatisticsPtr->SetLastError( |
| 1891 | VE_CODEC_ERROR, kTraceError, |
| 1892 | "SetISACInitTargetRate() failed to retrieve send codec"); |
| 1893 | return -1; |
| 1894 | } |
| 1895 | if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0) |
| 1896 | { |
| 1897 | // This API is only valid if iSAC is setup to run in channel-adaptive |
| 1898 | // mode. |
| 1899 | // We do not validate the adaptive mode here. It is done later in the |
| 1900 | // ConfigISACBandwidthEstimator() API. |
| 1901 | _engineStatisticsPtr->SetLastError( |
| 1902 | VE_CODEC_ERROR, kTraceError, |
| 1903 | "SetISACInitTargetRate() send codec is not iSAC"); |
| 1904 | return -1; |
| 1905 | } |
| 1906 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1907 | uint8_t initFrameSizeMsec(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1908 | if (16000 == sendCodec.plfreq) |
| 1909 | { |
| 1910 | // Note that 0 is a valid and corresponds to "use default |
| 1911 | if ((rateBps != 0 && |
| 1912 | rateBps < kVoiceEngineMinIsacInitTargetRateBpsWb) || |
| 1913 | (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsWb)) |
| 1914 | { |
| 1915 | _engineStatisticsPtr->SetLastError( |
| 1916 | VE_INVALID_ARGUMENT, kTraceError, |
| 1917 | "SetISACInitTargetRate() invalid target rate - 1"); |
| 1918 | return -1; |
| 1919 | } |
| 1920 | // 30 or 60ms |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1921 | initFrameSizeMsec = (uint8_t)(sendCodec.pacsize / 16); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1922 | } |
| 1923 | else if (32000 == sendCodec.plfreq) |
| 1924 | { |
| 1925 | if ((rateBps != 0 && |
| 1926 | rateBps < kVoiceEngineMinIsacInitTargetRateBpsSwb) || |
| 1927 | (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsSwb)) |
| 1928 | { |
| 1929 | _engineStatisticsPtr->SetLastError( |
| 1930 | VE_INVALID_ARGUMENT, kTraceError, |
| 1931 | "SetISACInitTargetRate() invalid target rate - 2"); |
| 1932 | return -1; |
| 1933 | } |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1934 | initFrameSizeMsec = (uint8_t)(sendCodec.pacsize / 32); // 30ms |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1935 | } |
| 1936 | |
| 1937 | if (_audioCodingModule.ConfigISACBandwidthEstimator( |
| 1938 | initFrameSizeMsec, rateBps, useFixedFrameSize) == -1) |
| 1939 | { |
| 1940 | _engineStatisticsPtr->SetLastError( |
| 1941 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 1942 | "SetISACInitTargetRate() iSAC BWE config failed"); |
| 1943 | return -1; |
| 1944 | } |
| 1945 | |
| 1946 | return 0; |
| 1947 | } |
| 1948 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 1949 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1950 | Channel::SetISACMaxRate(int rateBps) |
| 1951 | { |
| 1952 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 1953 | "Channel::SetISACMaxRate()"); |
| 1954 | |
| 1955 | CodecInst sendCodec; |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 1956 | if (_audioCodingModule.SendCodec(&sendCodec) == -1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 1957 | { |
| 1958 | _engineStatisticsPtr->SetLastError( |
| 1959 | VE_CODEC_ERROR, kTraceError, |
| 1960 | "SetISACMaxRate() failed to retrieve send codec"); |
| 1961 | return -1; |
| 1962 | } |
| 1963 | if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0) |
| 1964 | { |
| 1965 | // This API is only valid if iSAC is selected as sending codec. |
| 1966 | _engineStatisticsPtr->SetLastError( |
| 1967 | VE_CODEC_ERROR, kTraceError, |
| 1968 | "SetISACMaxRate() send codec is not iSAC"); |
| 1969 | return -1; |
| 1970 | } |
| 1971 | if (16000 == sendCodec.plfreq) |
| 1972 | { |
| 1973 | if ((rateBps < kVoiceEngineMinIsacMaxRateBpsWb) || |
| 1974 | (rateBps > kVoiceEngineMaxIsacMaxRateBpsWb)) |
| 1975 | { |
| 1976 | _engineStatisticsPtr->SetLastError( |
| 1977 | VE_INVALID_ARGUMENT, kTraceError, |
| 1978 | "SetISACMaxRate() invalid max rate - 1"); |
| 1979 | return -1; |
| 1980 | } |
| 1981 | } |
| 1982 | else if (32000 == sendCodec.plfreq) |
| 1983 | { |
| 1984 | if ((rateBps < kVoiceEngineMinIsacMaxRateBpsSwb) || |
| 1985 | (rateBps > kVoiceEngineMaxIsacMaxRateBpsSwb)) |
| 1986 | { |
| 1987 | _engineStatisticsPtr->SetLastError( |
| 1988 | VE_INVALID_ARGUMENT, kTraceError, |
| 1989 | "SetISACMaxRate() invalid max rate - 2"); |
| 1990 | return -1; |
| 1991 | } |
| 1992 | } |
| 1993 | if (_sending) |
| 1994 | { |
| 1995 | _engineStatisticsPtr->SetLastError( |
| 1996 | VE_SENDING, kTraceError, |
| 1997 | "SetISACMaxRate() unable to set max rate while sending"); |
| 1998 | return -1; |
| 1999 | } |
| 2000 | |
| 2001 | // Set the maximum instantaneous rate of iSAC (works for both adaptive |
| 2002 | // and non-adaptive mode) |
| 2003 | if (_audioCodingModule.SetISACMaxRate(rateBps) == -1) |
| 2004 | { |
| 2005 | _engineStatisticsPtr->SetLastError( |
| 2006 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2007 | "SetISACMaxRate() failed to set max rate"); |
| 2008 | return -1; |
| 2009 | } |
| 2010 | |
| 2011 | return 0; |
| 2012 | } |
| 2013 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2014 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2015 | Channel::SetISACMaxPayloadSize(int sizeBytes) |
| 2016 | { |
| 2017 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2018 | "Channel::SetISACMaxPayloadSize()"); |
| 2019 | CodecInst sendCodec; |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 2020 | if (_audioCodingModule.SendCodec(&sendCodec) == -1) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2021 | { |
| 2022 | _engineStatisticsPtr->SetLastError( |
| 2023 | VE_CODEC_ERROR, kTraceError, |
| 2024 | "SetISACMaxPayloadSize() failed to retrieve send codec"); |
| 2025 | return -1; |
| 2026 | } |
| 2027 | if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0) |
| 2028 | { |
| 2029 | _engineStatisticsPtr->SetLastError( |
| 2030 | VE_CODEC_ERROR, kTraceError, |
| 2031 | "SetISACMaxPayloadSize() send codec is not iSAC"); |
| 2032 | return -1; |
| 2033 | } |
| 2034 | if (16000 == sendCodec.plfreq) |
| 2035 | { |
| 2036 | if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesWb) || |
| 2037 | (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesWb)) |
| 2038 | { |
| 2039 | _engineStatisticsPtr->SetLastError( |
| 2040 | VE_INVALID_ARGUMENT, kTraceError, |
| 2041 | "SetISACMaxPayloadSize() invalid max payload - 1"); |
| 2042 | return -1; |
| 2043 | } |
| 2044 | } |
| 2045 | else if (32000 == sendCodec.plfreq) |
| 2046 | { |
| 2047 | if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesSwb) || |
| 2048 | (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb)) |
| 2049 | { |
| 2050 | _engineStatisticsPtr->SetLastError( |
| 2051 | VE_INVALID_ARGUMENT, kTraceError, |
| 2052 | "SetISACMaxPayloadSize() invalid max payload - 2"); |
| 2053 | return -1; |
| 2054 | } |
| 2055 | } |
| 2056 | if (_sending) |
| 2057 | { |
| 2058 | _engineStatisticsPtr->SetLastError( |
| 2059 | VE_SENDING, kTraceError, |
| 2060 | "SetISACMaxPayloadSize() unable to set max rate while sending"); |
| 2061 | return -1; |
| 2062 | } |
| 2063 | |
| 2064 | if (_audioCodingModule.SetISACMaxPayloadSize(sizeBytes) == -1) |
| 2065 | { |
| 2066 | _engineStatisticsPtr->SetLastError( |
| 2067 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 2068 | "SetISACMaxPayloadSize() failed to set max payload size"); |
| 2069 | return -1; |
| 2070 | } |
| 2071 | return 0; |
| 2072 | } |
| 2073 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2074 | int32_t Channel::RegisterExternalTransport(Transport& transport) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2075 | { |
| 2076 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 2077 | "Channel::RegisterExternalTransport()"); |
| 2078 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2079 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2080 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2081 | if (_externalTransport) |
| 2082 | { |
| 2083 | _engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION, |
| 2084 | kTraceError, |
| 2085 | "RegisterExternalTransport() external transport already enabled"); |
| 2086 | return -1; |
| 2087 | } |
| 2088 | _externalTransport = true; |
| 2089 | _transportPtr = &transport; |
| 2090 | return 0; |
| 2091 | } |
| 2092 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2093 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2094 | Channel::DeRegisterExternalTransport() |
| 2095 | { |
| 2096 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2097 | "Channel::DeRegisterExternalTransport()"); |
| 2098 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2099 | CriticalSectionScoped cs(&_callbackCritSect); |
xians@webrtc.org | 83661f5 | 2011-11-25 10:58:15 +0000 | [diff] [blame] | 2100 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2101 | if (!_transportPtr) |
| 2102 | { |
| 2103 | _engineStatisticsPtr->SetLastError( |
| 2104 | VE_INVALID_OPERATION, kTraceWarning, |
| 2105 | "DeRegisterExternalTransport() external transport already " |
| 2106 | "disabled"); |
| 2107 | return 0; |
| 2108 | } |
| 2109 | _externalTransport = false; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2110 | _transportPtr = NULL; |
| 2111 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2112 | "DeRegisterExternalTransport() all transport is disabled"); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2113 | return 0; |
| 2114 | } |
| 2115 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2116 | int32_t Channel::ReceivedRTPPacket(const int8_t* data, int32_t length) { |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 2117 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2118 | "Channel::ReceivedRTPPacket()"); |
| 2119 | |
| 2120 | // Store playout timestamp for the received RTP packet |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 2121 | UpdatePlayoutTimestamp(false); |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 2122 | |
| 2123 | // Dump the RTP packet to a file (if RTP dump is enabled). |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2124 | if (_rtpDumpIn.DumpPacket((const uint8_t*)data, |
| 2125 | (uint16_t)length) == -1) { |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 2126 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 2127 | VoEId(_instanceId,_channelId), |
| 2128 | "Channel::SendPacket() RTP dump to input file failed"); |
| 2129 | } |
| 2130 | |
| 2131 | // Deliver RTP packet to RTP/RTCP module for parsing |
| 2132 | // The packet will be pushed back to the channel thru the |
| 2133 | // OnReceivedPayloadData callback so we don't push it to the ACM here |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2134 | if (_rtpRtcpModule->IncomingPacket((const uint8_t*)data, |
| 2135 | (uint16_t)length) == -1) { |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 2136 | _engineStatisticsPtr->SetLastError( |
| 2137 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning, |
| 2138 | "Channel::IncomingRTPPacket() RTP packet is invalid"); |
| 2139 | } |
| 2140 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2141 | } |
| 2142 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2143 | int32_t Channel::ReceivedRTCPPacket(const int8_t* data, int32_t length) { |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 2144 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2145 | "Channel::ReceivedRTCPPacket()"); |
| 2146 | // Store playout timestamp for the received RTCP packet |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 2147 | UpdatePlayoutTimestamp(true); |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 2148 | |
| 2149 | // Dump the RTCP packet to a file (if RTP dump is enabled). |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2150 | if (_rtpDumpIn.DumpPacket((const uint8_t*)data, |
| 2151 | (uint16_t)length) == -1) { |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 2152 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 2153 | VoEId(_instanceId,_channelId), |
| 2154 | "Channel::SendPacket() RTCP dump to input file failed"); |
| 2155 | } |
| 2156 | |
| 2157 | // Deliver RTCP packet to RTP/RTCP module for parsing |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2158 | if (_rtpRtcpModule->IncomingPacket((const uint8_t*)data, |
| 2159 | (uint16_t)length) == -1) { |
pwestin@webrtc.org | 0c45957 | 2013-04-03 15:43:57 +0000 | [diff] [blame] | 2160 | _engineStatisticsPtr->SetLastError( |
| 2161 | VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning, |
| 2162 | "Channel::IncomingRTPPacket() RTCP packet is invalid"); |
| 2163 | } |
| 2164 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2165 | } |
| 2166 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2167 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2168 | Channel::SetPacketTimeoutNotification(bool enable, int timeoutSeconds) |
| 2169 | { |
| 2170 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2171 | "Channel::SetPacketTimeoutNotification()"); |
| 2172 | if (enable) |
| 2173 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2174 | const uint32_t RTPtimeoutMS = 1000*timeoutSeconds; |
| 2175 | const uint32_t RTCPtimeoutMS = 0; |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 2176 | _rtpRtcpModule->SetPacketTimeout(RTPtimeoutMS, RTCPtimeoutMS); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2177 | _rtpPacketTimeOutIsEnabled = true; |
| 2178 | _rtpTimeOutSeconds = timeoutSeconds; |
| 2179 | } |
| 2180 | else |
| 2181 | { |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 2182 | _rtpRtcpModule->SetPacketTimeout(0, 0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2183 | _rtpPacketTimeOutIsEnabled = false; |
| 2184 | _rtpTimeOutSeconds = 0; |
| 2185 | } |
| 2186 | return 0; |
| 2187 | } |
| 2188 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2189 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2190 | Channel::GetPacketTimeoutNotification(bool& enabled, int& timeoutSeconds) |
| 2191 | { |
| 2192 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2193 | "Channel::GetPacketTimeoutNotification()"); |
| 2194 | enabled = _rtpPacketTimeOutIsEnabled; |
| 2195 | if (enabled) |
| 2196 | { |
| 2197 | timeoutSeconds = _rtpTimeOutSeconds; |
| 2198 | } |
| 2199 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1), |
| 2200 | "GetPacketTimeoutNotification() => enabled=%d," |
| 2201 | " timeoutSeconds=%d", |
| 2202 | enabled, timeoutSeconds); |
| 2203 | return 0; |
| 2204 | } |
| 2205 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2206 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2207 | Channel::RegisterDeadOrAliveObserver(VoEConnectionObserver& observer) |
| 2208 | { |
| 2209 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2210 | "Channel::RegisterDeadOrAliveObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2211 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2212 | |
| 2213 | if (_connectionObserverPtr) |
| 2214 | { |
| 2215 | _engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION, kTraceError, |
| 2216 | "RegisterDeadOrAliveObserver() observer already enabled"); |
| 2217 | return -1; |
| 2218 | } |
| 2219 | |
| 2220 | _connectionObserverPtr = &observer; |
| 2221 | _connectionObserver = true; |
| 2222 | |
| 2223 | return 0; |
| 2224 | } |
| 2225 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2226 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2227 | Channel::DeRegisterDeadOrAliveObserver() |
| 2228 | { |
| 2229 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2230 | "Channel::DeRegisterDeadOrAliveObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2231 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2232 | |
| 2233 | if (!_connectionObserverPtr) |
| 2234 | { |
| 2235 | _engineStatisticsPtr->SetLastError( |
| 2236 | VE_INVALID_OPERATION, kTraceWarning, |
| 2237 | "DeRegisterDeadOrAliveObserver() observer already disabled"); |
| 2238 | return 0; |
| 2239 | } |
| 2240 | |
| 2241 | _connectionObserver = false; |
| 2242 | _connectionObserverPtr = NULL; |
| 2243 | |
| 2244 | return 0; |
| 2245 | } |
| 2246 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2247 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2248 | Channel::SetPeriodicDeadOrAliveStatus(bool enable, int sampleTimeSeconds) |
| 2249 | { |
| 2250 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2251 | "Channel::SetPeriodicDeadOrAliveStatus()"); |
| 2252 | if (!_connectionObserverPtr) |
| 2253 | { |
| 2254 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2255 | "SetPeriodicDeadOrAliveStatus() connection observer has" |
| 2256 | " not been registered"); |
| 2257 | } |
| 2258 | if (enable) |
| 2259 | { |
| 2260 | ResetDeadOrAliveCounters(); |
| 2261 | } |
| 2262 | bool enabled(false); |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2263 | uint8_t currentSampleTimeSec(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2264 | // Store last state (will be used later if dead-or-alive is disabled). |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 2265 | _rtpRtcpModule->PeriodicDeadOrAliveStatus(enabled, currentSampleTimeSec); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2266 | // Update the dead-or-alive state. |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 2267 | if (_rtpRtcpModule->SetPeriodicDeadOrAliveStatus( |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2268 | enable, (uint8_t)sampleTimeSeconds) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2269 | { |
| 2270 | _engineStatisticsPtr->SetLastError( |
| 2271 | VE_RTP_RTCP_MODULE_ERROR, |
| 2272 | kTraceError, |
| 2273 | "SetPeriodicDeadOrAliveStatus() failed to set dead-or-alive " |
| 2274 | "status"); |
| 2275 | return -1; |
| 2276 | } |
| 2277 | if (!enable) |
| 2278 | { |
| 2279 | // Restore last utilized sample time. |
| 2280 | // Without this, the sample time would always be reset to default |
| 2281 | // (2 sec), each time dead-or-alived was disabled without sample-time |
| 2282 | // parameter. |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 2283 | _rtpRtcpModule->SetPeriodicDeadOrAliveStatus(enable, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2284 | currentSampleTimeSec); |
| 2285 | } |
| 2286 | return 0; |
| 2287 | } |
| 2288 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2289 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2290 | Channel::GetPeriodicDeadOrAliveStatus(bool& enabled, int& sampleTimeSeconds) |
| 2291 | { |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 2292 | _rtpRtcpModule->PeriodicDeadOrAliveStatus( |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2293 | enabled, |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2294 | (uint8_t&)sampleTimeSeconds); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2295 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1), |
| 2296 | "GetPeriodicDeadOrAliveStatus() => enabled=%d," |
| 2297 | " sampleTimeSeconds=%d", |
| 2298 | enabled, sampleTimeSeconds); |
| 2299 | return 0; |
| 2300 | } |
| 2301 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2302 | int Channel::StartPlayingFileLocally(const char* fileName, |
| 2303 | const bool loop, |
| 2304 | const FileFormats format, |
| 2305 | const int startPosition, |
| 2306 | const float volumeScaling, |
| 2307 | const int stopPosition, |
| 2308 | const CodecInst* codecInst) |
| 2309 | { |
| 2310 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2311 | "Channel::StartPlayingFileLocally(fileNameUTF8[]=%s, loop=%d," |
| 2312 | " format=%d, volumeScaling=%5.3f, startPosition=%d, " |
| 2313 | "stopPosition=%d)", fileName, loop, format, volumeScaling, |
| 2314 | startPosition, stopPosition); |
| 2315 | |
| 2316 | if (_outputFilePlaying) |
| 2317 | { |
| 2318 | _engineStatisticsPtr->SetLastError( |
| 2319 | VE_ALREADY_PLAYING, kTraceError, |
| 2320 | "StartPlayingFileLocally() is already playing"); |
| 2321 | return -1; |
| 2322 | } |
| 2323 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2324 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2325 | CriticalSectionScoped cs(&_fileCritSect); |
henrike@webrtc.org | b37c628 | 2011-10-31 23:53:04 +0000 | [diff] [blame] | 2326 | |
| 2327 | if (_outputFilePlayerPtr) |
| 2328 | { |
| 2329 | _outputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 2330 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 2331 | _outputFilePlayerPtr = NULL; |
| 2332 | } |
| 2333 | |
| 2334 | _outputFilePlayerPtr = FilePlayer::CreateFilePlayer( |
| 2335 | _outputFilePlayerId, (const FileFormats)format); |
| 2336 | |
| 2337 | if (_outputFilePlayerPtr == NULL) |
| 2338 | { |
| 2339 | _engineStatisticsPtr->SetLastError( |
| 2340 | VE_INVALID_ARGUMENT, kTraceError, |
henrike@webrtc.org | 31d3070 | 2011-11-18 19:59:32 +0000 | [diff] [blame] | 2341 | "StartPlayingFileLocally() filePlayer format is not correct"); |
henrike@webrtc.org | b37c628 | 2011-10-31 23:53:04 +0000 | [diff] [blame] | 2342 | return -1; |
| 2343 | } |
| 2344 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2345 | const uint32_t notificationTime(0); |
henrike@webrtc.org | b37c628 | 2011-10-31 23:53:04 +0000 | [diff] [blame] | 2346 | |
| 2347 | if (_outputFilePlayerPtr->StartPlayingFile( |
| 2348 | fileName, |
| 2349 | loop, |
| 2350 | startPosition, |
| 2351 | volumeScaling, |
| 2352 | notificationTime, |
| 2353 | stopPosition, |
| 2354 | (const CodecInst*)codecInst) != 0) |
| 2355 | { |
| 2356 | _engineStatisticsPtr->SetLastError( |
| 2357 | VE_BAD_FILE, kTraceError, |
| 2358 | "StartPlayingFile() failed to start file playout"); |
| 2359 | _outputFilePlayerPtr->StopPlayingFile(); |
| 2360 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 2361 | _outputFilePlayerPtr = NULL; |
| 2362 | return -1; |
| 2363 | } |
| 2364 | _outputFilePlayerPtr->RegisterModuleFileCallback(this); |
| 2365 | _outputFilePlaying = true; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2366 | } |
braveyao@webrtc.org | ab12990 | 2012-06-04 03:26:39 +0000 | [diff] [blame] | 2367 | |
| 2368 | if (RegisterFilePlayingToMixer() != 0) |
henrike@webrtc.org | 066f9e5 | 2011-10-28 23:15:47 +0000 | [diff] [blame] | 2369 | return -1; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2370 | |
| 2371 | return 0; |
| 2372 | } |
| 2373 | |
| 2374 | int Channel::StartPlayingFileLocally(InStream* stream, |
| 2375 | const FileFormats format, |
| 2376 | const int startPosition, |
| 2377 | const float volumeScaling, |
| 2378 | const int stopPosition, |
| 2379 | const CodecInst* codecInst) |
| 2380 | { |
| 2381 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2382 | "Channel::StartPlayingFileLocally(format=%d," |
| 2383 | " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)", |
| 2384 | format, volumeScaling, startPosition, stopPosition); |
| 2385 | |
| 2386 | if(stream == NULL) |
| 2387 | { |
| 2388 | _engineStatisticsPtr->SetLastError( |
| 2389 | VE_BAD_FILE, kTraceError, |
| 2390 | "StartPlayingFileLocally() NULL as input stream"); |
| 2391 | return -1; |
| 2392 | } |
| 2393 | |
| 2394 | |
| 2395 | if (_outputFilePlaying) |
| 2396 | { |
| 2397 | _engineStatisticsPtr->SetLastError( |
| 2398 | VE_ALREADY_PLAYING, kTraceError, |
| 2399 | "StartPlayingFileLocally() is already playing"); |
| 2400 | return -1; |
| 2401 | } |
| 2402 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2403 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2404 | CriticalSectionScoped cs(&_fileCritSect); |
henrike@webrtc.org | b37c628 | 2011-10-31 23:53:04 +0000 | [diff] [blame] | 2405 | |
| 2406 | // Destroy the old instance |
| 2407 | if (_outputFilePlayerPtr) |
| 2408 | { |
| 2409 | _outputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 2410 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 2411 | _outputFilePlayerPtr = NULL; |
| 2412 | } |
| 2413 | |
| 2414 | // Create the instance |
| 2415 | _outputFilePlayerPtr = FilePlayer::CreateFilePlayer( |
| 2416 | _outputFilePlayerId, |
| 2417 | (const FileFormats)format); |
| 2418 | |
| 2419 | if (_outputFilePlayerPtr == NULL) |
| 2420 | { |
| 2421 | _engineStatisticsPtr->SetLastError( |
| 2422 | VE_INVALID_ARGUMENT, kTraceError, |
| 2423 | "StartPlayingFileLocally() filePlayer format isnot correct"); |
| 2424 | return -1; |
| 2425 | } |
| 2426 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2427 | const uint32_t notificationTime(0); |
henrike@webrtc.org | b37c628 | 2011-10-31 23:53:04 +0000 | [diff] [blame] | 2428 | |
| 2429 | if (_outputFilePlayerPtr->StartPlayingFile(*stream, startPosition, |
| 2430 | volumeScaling, |
| 2431 | notificationTime, |
| 2432 | stopPosition, codecInst) != 0) |
| 2433 | { |
| 2434 | _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| 2435 | "StartPlayingFile() failed to " |
| 2436 | "start file playout"); |
| 2437 | _outputFilePlayerPtr->StopPlayingFile(); |
| 2438 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 2439 | _outputFilePlayerPtr = NULL; |
| 2440 | return -1; |
| 2441 | } |
| 2442 | _outputFilePlayerPtr->RegisterModuleFileCallback(this); |
| 2443 | _outputFilePlaying = true; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2444 | } |
braveyao@webrtc.org | ab12990 | 2012-06-04 03:26:39 +0000 | [diff] [blame] | 2445 | |
| 2446 | if (RegisterFilePlayingToMixer() != 0) |
henrike@webrtc.org | 066f9e5 | 2011-10-28 23:15:47 +0000 | [diff] [blame] | 2447 | return -1; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2448 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2449 | return 0; |
| 2450 | } |
| 2451 | |
| 2452 | int Channel::StopPlayingFileLocally() |
| 2453 | { |
| 2454 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2455 | "Channel::StopPlayingFileLocally()"); |
| 2456 | |
| 2457 | if (!_outputFilePlaying) |
| 2458 | { |
| 2459 | _engineStatisticsPtr->SetLastError( |
| 2460 | VE_INVALID_OPERATION, kTraceWarning, |
| 2461 | "StopPlayingFileLocally() isnot playing"); |
| 2462 | return 0; |
| 2463 | } |
| 2464 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2465 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2466 | CriticalSectionScoped cs(&_fileCritSect); |
henrike@webrtc.org | b37c628 | 2011-10-31 23:53:04 +0000 | [diff] [blame] | 2467 | |
| 2468 | if (_outputFilePlayerPtr->StopPlayingFile() != 0) |
| 2469 | { |
| 2470 | _engineStatisticsPtr->SetLastError( |
| 2471 | VE_STOP_RECORDING_FAILED, kTraceError, |
| 2472 | "StopPlayingFile() could not stop playing"); |
| 2473 | return -1; |
| 2474 | } |
| 2475 | _outputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 2476 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 2477 | _outputFilePlayerPtr = NULL; |
| 2478 | _outputFilePlaying = false; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2479 | } |
henrike@webrtc.org | b37c628 | 2011-10-31 23:53:04 +0000 | [diff] [blame] | 2480 | // _fileCritSect cannot be taken while calling |
| 2481 | // SetAnonymousMixibilityStatus. Refer to comments in |
| 2482 | // StartPlayingFileLocally(const char* ...) for more details. |
henrike@webrtc.org | 066f9e5 | 2011-10-28 23:15:47 +0000 | [diff] [blame] | 2483 | if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, false) != 0) |
| 2484 | { |
| 2485 | _engineStatisticsPtr->SetLastError( |
| 2486 | VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError, |
henrike@webrtc.org | b37c628 | 2011-10-31 23:53:04 +0000 | [diff] [blame] | 2487 | "StopPlayingFile() failed to stop participant from playing as" |
| 2488 | "file in the mixer"); |
henrike@webrtc.org | 066f9e5 | 2011-10-28 23:15:47 +0000 | [diff] [blame] | 2489 | return -1; |
| 2490 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2491 | |
| 2492 | return 0; |
| 2493 | } |
| 2494 | |
| 2495 | int Channel::IsPlayingFileLocally() const |
| 2496 | { |
| 2497 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2498 | "Channel::IsPlayingFileLocally()"); |
| 2499 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2500 | return (int32_t)_outputFilePlaying; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2501 | } |
| 2502 | |
braveyao@webrtc.org | ab12990 | 2012-06-04 03:26:39 +0000 | [diff] [blame] | 2503 | int Channel::RegisterFilePlayingToMixer() |
| 2504 | { |
| 2505 | // Return success for not registering for file playing to mixer if: |
| 2506 | // 1. playing file before playout is started on that channel. |
| 2507 | // 2. starting playout without file playing on that channel. |
| 2508 | if (!_playing || !_outputFilePlaying) |
| 2509 | { |
| 2510 | return 0; |
| 2511 | } |
| 2512 | |
| 2513 | // |_fileCritSect| cannot be taken while calling |
| 2514 | // SetAnonymousMixabilityStatus() since as soon as the participant is added |
| 2515 | // frames can be pulled by the mixer. Since the frames are generated from |
| 2516 | // the file, _fileCritSect will be taken. This would result in a deadlock. |
| 2517 | if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0) |
| 2518 | { |
| 2519 | CriticalSectionScoped cs(&_fileCritSect); |
| 2520 | _outputFilePlaying = false; |
| 2521 | _engineStatisticsPtr->SetLastError( |
| 2522 | VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError, |
| 2523 | "StartPlayingFile() failed to add participant as file to mixer"); |
| 2524 | _outputFilePlayerPtr->StopPlayingFile(); |
| 2525 | FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr); |
| 2526 | _outputFilePlayerPtr = NULL; |
| 2527 | return -1; |
| 2528 | } |
| 2529 | |
| 2530 | return 0; |
| 2531 | } |
| 2532 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2533 | int Channel::ScaleLocalFilePlayout(const float scale) |
| 2534 | { |
| 2535 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2536 | "Channel::ScaleLocalFilePlayout(scale=%5.3f)", scale); |
| 2537 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2538 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2539 | |
| 2540 | if (!_outputFilePlaying) |
| 2541 | { |
| 2542 | _engineStatisticsPtr->SetLastError( |
| 2543 | VE_INVALID_OPERATION, kTraceError, |
| 2544 | "ScaleLocalFilePlayout() isnot playing"); |
| 2545 | return -1; |
| 2546 | } |
| 2547 | if ((_outputFilePlayerPtr == NULL) || |
| 2548 | (_outputFilePlayerPtr->SetAudioScaling(scale) != 0)) |
| 2549 | { |
| 2550 | _engineStatisticsPtr->SetLastError( |
| 2551 | VE_BAD_ARGUMENT, kTraceError, |
| 2552 | "SetAudioScaling() failed to scale the playout"); |
| 2553 | return -1; |
| 2554 | } |
| 2555 | |
| 2556 | return 0; |
| 2557 | } |
| 2558 | |
| 2559 | int Channel::GetLocalPlayoutPosition(int& positionMs) |
| 2560 | { |
| 2561 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2562 | "Channel::GetLocalPlayoutPosition(position=?)"); |
| 2563 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2564 | uint32_t position; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2565 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2566 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2567 | |
| 2568 | if (_outputFilePlayerPtr == NULL) |
| 2569 | { |
| 2570 | _engineStatisticsPtr->SetLastError( |
| 2571 | VE_INVALID_OPERATION, kTraceError, |
| 2572 | "GetLocalPlayoutPosition() filePlayer instance doesnot exist"); |
| 2573 | return -1; |
| 2574 | } |
| 2575 | |
| 2576 | if (_outputFilePlayerPtr->GetPlayoutPosition(position) != 0) |
| 2577 | { |
| 2578 | _engineStatisticsPtr->SetLastError( |
| 2579 | VE_BAD_FILE, kTraceError, |
| 2580 | "GetLocalPlayoutPosition() failed"); |
| 2581 | return -1; |
| 2582 | } |
| 2583 | positionMs = position; |
| 2584 | |
| 2585 | return 0; |
| 2586 | } |
| 2587 | |
| 2588 | int Channel::StartPlayingFileAsMicrophone(const char* fileName, |
| 2589 | const bool loop, |
| 2590 | const FileFormats format, |
| 2591 | const int startPosition, |
| 2592 | const float volumeScaling, |
| 2593 | const int stopPosition, |
| 2594 | const CodecInst* codecInst) |
| 2595 | { |
| 2596 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2597 | "Channel::StartPlayingFileAsMicrophone(fileNameUTF8[]=%s, " |
| 2598 | "loop=%d, format=%d, volumeScaling=%5.3f, startPosition=%d, " |
| 2599 | "stopPosition=%d)", fileName, loop, format, volumeScaling, |
| 2600 | startPosition, stopPosition); |
| 2601 | |
| 2602 | if (_inputFilePlaying) |
| 2603 | { |
| 2604 | _engineStatisticsPtr->SetLastError( |
| 2605 | VE_ALREADY_PLAYING, kTraceWarning, |
| 2606 | "StartPlayingFileAsMicrophone() filePlayer is playing"); |
| 2607 | return 0; |
| 2608 | } |
| 2609 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2610 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2611 | |
| 2612 | // Destroy the old instance |
| 2613 | if (_inputFilePlayerPtr) |
| 2614 | { |
| 2615 | _inputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 2616 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 2617 | _inputFilePlayerPtr = NULL; |
| 2618 | } |
| 2619 | |
| 2620 | // Create the instance |
| 2621 | _inputFilePlayerPtr = FilePlayer::CreateFilePlayer( |
| 2622 | _inputFilePlayerId, (const FileFormats)format); |
| 2623 | |
| 2624 | if (_inputFilePlayerPtr == NULL) |
| 2625 | { |
| 2626 | _engineStatisticsPtr->SetLastError( |
| 2627 | VE_INVALID_ARGUMENT, kTraceError, |
| 2628 | "StartPlayingFileAsMicrophone() filePlayer format isnot correct"); |
| 2629 | return -1; |
| 2630 | } |
| 2631 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2632 | const uint32_t notificationTime(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2633 | |
| 2634 | if (_inputFilePlayerPtr->StartPlayingFile( |
| 2635 | fileName, |
| 2636 | loop, |
| 2637 | startPosition, |
| 2638 | volumeScaling, |
| 2639 | notificationTime, |
| 2640 | stopPosition, |
| 2641 | (const CodecInst*)codecInst) != 0) |
| 2642 | { |
| 2643 | _engineStatisticsPtr->SetLastError( |
| 2644 | VE_BAD_FILE, kTraceError, |
| 2645 | "StartPlayingFile() failed to start file playout"); |
| 2646 | _inputFilePlayerPtr->StopPlayingFile(); |
| 2647 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 2648 | _inputFilePlayerPtr = NULL; |
| 2649 | return -1; |
| 2650 | } |
| 2651 | _inputFilePlayerPtr->RegisterModuleFileCallback(this); |
| 2652 | _inputFilePlaying = true; |
| 2653 | |
| 2654 | return 0; |
| 2655 | } |
| 2656 | |
| 2657 | int Channel::StartPlayingFileAsMicrophone(InStream* stream, |
| 2658 | const FileFormats format, |
| 2659 | const int startPosition, |
| 2660 | const float volumeScaling, |
| 2661 | const int stopPosition, |
| 2662 | const CodecInst* codecInst) |
| 2663 | { |
| 2664 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2665 | "Channel::StartPlayingFileAsMicrophone(format=%d, " |
| 2666 | "volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)", |
| 2667 | format, volumeScaling, startPosition, stopPosition); |
| 2668 | |
| 2669 | if(stream == NULL) |
| 2670 | { |
| 2671 | _engineStatisticsPtr->SetLastError( |
| 2672 | VE_BAD_FILE, kTraceError, |
| 2673 | "StartPlayingFileAsMicrophone NULL as input stream"); |
| 2674 | return -1; |
| 2675 | } |
| 2676 | |
| 2677 | if (_inputFilePlaying) |
| 2678 | { |
| 2679 | _engineStatisticsPtr->SetLastError( |
| 2680 | VE_ALREADY_PLAYING, kTraceWarning, |
| 2681 | "StartPlayingFileAsMicrophone() is playing"); |
| 2682 | return 0; |
| 2683 | } |
| 2684 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2685 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2686 | |
| 2687 | // Destroy the old instance |
| 2688 | if (_inputFilePlayerPtr) |
| 2689 | { |
| 2690 | _inputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 2691 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 2692 | _inputFilePlayerPtr = NULL; |
| 2693 | } |
| 2694 | |
| 2695 | // Create the instance |
| 2696 | _inputFilePlayerPtr = FilePlayer::CreateFilePlayer( |
| 2697 | _inputFilePlayerId, (const FileFormats)format); |
| 2698 | |
| 2699 | if (_inputFilePlayerPtr == NULL) |
| 2700 | { |
| 2701 | _engineStatisticsPtr->SetLastError( |
| 2702 | VE_INVALID_ARGUMENT, kTraceError, |
| 2703 | "StartPlayingInputFile() filePlayer format isnot correct"); |
| 2704 | return -1; |
| 2705 | } |
| 2706 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2707 | const uint32_t notificationTime(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2708 | |
| 2709 | if (_inputFilePlayerPtr->StartPlayingFile(*stream, startPosition, |
| 2710 | volumeScaling, notificationTime, |
| 2711 | stopPosition, codecInst) != 0) |
| 2712 | { |
| 2713 | _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| 2714 | "StartPlayingFile() failed to start " |
| 2715 | "file playout"); |
| 2716 | _inputFilePlayerPtr->StopPlayingFile(); |
| 2717 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 2718 | _inputFilePlayerPtr = NULL; |
| 2719 | return -1; |
| 2720 | } |
andrew@webrtc.org | ae1a58b | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 2721 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2722 | _inputFilePlayerPtr->RegisterModuleFileCallback(this); |
| 2723 | _inputFilePlaying = true; |
| 2724 | |
| 2725 | return 0; |
| 2726 | } |
| 2727 | |
| 2728 | int Channel::StopPlayingFileAsMicrophone() |
| 2729 | { |
| 2730 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2731 | "Channel::StopPlayingFileAsMicrophone()"); |
| 2732 | |
| 2733 | if (!_inputFilePlaying) |
| 2734 | { |
| 2735 | _engineStatisticsPtr->SetLastError( |
| 2736 | VE_INVALID_OPERATION, kTraceWarning, |
| 2737 | "StopPlayingFileAsMicrophone() isnot playing"); |
| 2738 | return 0; |
| 2739 | } |
| 2740 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2741 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2742 | if (_inputFilePlayerPtr->StopPlayingFile() != 0) |
| 2743 | { |
| 2744 | _engineStatisticsPtr->SetLastError( |
| 2745 | VE_STOP_RECORDING_FAILED, kTraceError, |
| 2746 | "StopPlayingFile() could not stop playing"); |
| 2747 | return -1; |
| 2748 | } |
| 2749 | _inputFilePlayerPtr->RegisterModuleFileCallback(NULL); |
| 2750 | FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr); |
| 2751 | _inputFilePlayerPtr = NULL; |
| 2752 | _inputFilePlaying = false; |
| 2753 | |
| 2754 | return 0; |
| 2755 | } |
| 2756 | |
| 2757 | int Channel::IsPlayingFileAsMicrophone() const |
| 2758 | { |
| 2759 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2760 | "Channel::IsPlayingFileAsMicrophone()"); |
| 2761 | |
| 2762 | return _inputFilePlaying; |
| 2763 | } |
| 2764 | |
| 2765 | int Channel::ScaleFileAsMicrophonePlayout(const float scale) |
| 2766 | { |
| 2767 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2768 | "Channel::ScaleFileAsMicrophonePlayout(scale=%5.3f)", scale); |
| 2769 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2770 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2771 | |
| 2772 | if (!_inputFilePlaying) |
| 2773 | { |
| 2774 | _engineStatisticsPtr->SetLastError( |
| 2775 | VE_INVALID_OPERATION, kTraceError, |
| 2776 | "ScaleFileAsMicrophonePlayout() isnot playing"); |
| 2777 | return -1; |
| 2778 | } |
| 2779 | |
| 2780 | if ((_inputFilePlayerPtr == NULL) || |
| 2781 | (_inputFilePlayerPtr->SetAudioScaling(scale) != 0)) |
| 2782 | { |
| 2783 | _engineStatisticsPtr->SetLastError( |
| 2784 | VE_BAD_ARGUMENT, kTraceError, |
| 2785 | "SetAudioScaling() failed to scale playout"); |
| 2786 | return -1; |
| 2787 | } |
| 2788 | |
| 2789 | return 0; |
| 2790 | } |
| 2791 | |
leozwang@webrtc.org | 813e4b0 | 2012-03-01 18:34:25 +0000 | [diff] [blame] | 2792 | int Channel::StartRecordingPlayout(const char* fileName, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2793 | const CodecInst* codecInst) |
| 2794 | { |
| 2795 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2796 | "Channel::StartRecordingPlayout(fileName=%s)", fileName); |
| 2797 | |
| 2798 | if (_outputFileRecording) |
| 2799 | { |
| 2800 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1), |
| 2801 | "StartRecordingPlayout() is already recording"); |
| 2802 | return 0; |
| 2803 | } |
| 2804 | |
| 2805 | FileFormats format; |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2806 | const uint32_t notificationTime(0); // Not supported in VoE |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2807 | CodecInst dummyCodec={100,"L16",16000,320,1,320000}; |
| 2808 | |
niklas.enbom@webrtc.org | 40197d7 | 2012-03-26 08:45:47 +0000 | [diff] [blame] | 2809 | if ((codecInst != NULL) && |
| 2810 | ((codecInst->channels < 1) || (codecInst->channels > 2))) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2811 | { |
| 2812 | _engineStatisticsPtr->SetLastError( |
| 2813 | VE_BAD_ARGUMENT, kTraceError, |
| 2814 | "StartRecordingPlayout() invalid compression"); |
| 2815 | return(-1); |
| 2816 | } |
| 2817 | if(codecInst == NULL) |
| 2818 | { |
| 2819 | format = kFileFormatPcm16kHzFile; |
| 2820 | codecInst=&dummyCodec; |
| 2821 | } |
| 2822 | else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| 2823 | (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| 2824 | (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| 2825 | { |
| 2826 | format = kFileFormatWavFile; |
| 2827 | } |
| 2828 | else |
| 2829 | { |
| 2830 | format = kFileFormatCompressedFile; |
| 2831 | } |
| 2832 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2833 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2834 | |
| 2835 | // Destroy the old instance |
| 2836 | if (_outputFileRecorderPtr) |
| 2837 | { |
| 2838 | _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 2839 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 2840 | _outputFileRecorderPtr = NULL; |
| 2841 | } |
| 2842 | |
| 2843 | _outputFileRecorderPtr = FileRecorder::CreateFileRecorder( |
| 2844 | _outputFileRecorderId, (const FileFormats)format); |
| 2845 | if (_outputFileRecorderPtr == NULL) |
| 2846 | { |
| 2847 | _engineStatisticsPtr->SetLastError( |
| 2848 | VE_INVALID_ARGUMENT, kTraceError, |
| 2849 | "StartRecordingPlayout() fileRecorder format isnot correct"); |
| 2850 | return -1; |
| 2851 | } |
| 2852 | |
| 2853 | if (_outputFileRecorderPtr->StartRecordingAudioFile( |
| 2854 | fileName, (const CodecInst&)*codecInst, notificationTime) != 0) |
| 2855 | { |
| 2856 | _engineStatisticsPtr->SetLastError( |
| 2857 | VE_BAD_FILE, kTraceError, |
| 2858 | "StartRecordingAudioFile() failed to start file recording"); |
| 2859 | _outputFileRecorderPtr->StopRecording(); |
| 2860 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 2861 | _outputFileRecorderPtr = NULL; |
| 2862 | return -1; |
| 2863 | } |
| 2864 | _outputFileRecorderPtr->RegisterModuleFileCallback(this); |
| 2865 | _outputFileRecording = true; |
| 2866 | |
| 2867 | return 0; |
| 2868 | } |
| 2869 | |
| 2870 | int Channel::StartRecordingPlayout(OutStream* stream, |
| 2871 | const CodecInst* codecInst) |
| 2872 | { |
| 2873 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 2874 | "Channel::StartRecordingPlayout()"); |
| 2875 | |
| 2876 | if (_outputFileRecording) |
| 2877 | { |
| 2878 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1), |
| 2879 | "StartRecordingPlayout() is already recording"); |
| 2880 | return 0; |
| 2881 | } |
| 2882 | |
| 2883 | FileFormats format; |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2884 | const uint32_t notificationTime(0); // Not supported in VoE |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2885 | CodecInst dummyCodec={100,"L16",16000,320,1,320000}; |
| 2886 | |
| 2887 | if (codecInst != NULL && codecInst->channels != 1) |
| 2888 | { |
| 2889 | _engineStatisticsPtr->SetLastError( |
| 2890 | VE_BAD_ARGUMENT, kTraceError, |
| 2891 | "StartRecordingPlayout() invalid compression"); |
| 2892 | return(-1); |
| 2893 | } |
| 2894 | if(codecInst == NULL) |
| 2895 | { |
| 2896 | format = kFileFormatPcm16kHzFile; |
| 2897 | codecInst=&dummyCodec; |
| 2898 | } |
| 2899 | else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| 2900 | (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| 2901 | (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| 2902 | { |
| 2903 | format = kFileFormatWavFile; |
| 2904 | } |
| 2905 | else |
| 2906 | { |
| 2907 | format = kFileFormatCompressedFile; |
| 2908 | } |
| 2909 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2910 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2911 | |
| 2912 | // Destroy the old instance |
| 2913 | if (_outputFileRecorderPtr) |
| 2914 | { |
| 2915 | _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 2916 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 2917 | _outputFileRecorderPtr = NULL; |
| 2918 | } |
| 2919 | |
| 2920 | _outputFileRecorderPtr = FileRecorder::CreateFileRecorder( |
| 2921 | _outputFileRecorderId, (const FileFormats)format); |
| 2922 | if (_outputFileRecorderPtr == NULL) |
| 2923 | { |
| 2924 | _engineStatisticsPtr->SetLastError( |
| 2925 | VE_INVALID_ARGUMENT, kTraceError, |
| 2926 | "StartRecordingPlayout() fileRecorder format isnot correct"); |
| 2927 | return -1; |
| 2928 | } |
| 2929 | |
| 2930 | if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, *codecInst, |
| 2931 | notificationTime) != 0) |
| 2932 | { |
| 2933 | _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| 2934 | "StartRecordingPlayout() failed to " |
| 2935 | "start file recording"); |
| 2936 | _outputFileRecorderPtr->StopRecording(); |
| 2937 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 2938 | _outputFileRecorderPtr = NULL; |
| 2939 | return -1; |
| 2940 | } |
andrew@webrtc.org | ae1a58b | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 2941 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2942 | _outputFileRecorderPtr->RegisterModuleFileCallback(this); |
| 2943 | _outputFileRecording = true; |
| 2944 | |
| 2945 | return 0; |
| 2946 | } |
| 2947 | |
| 2948 | int Channel::StopRecordingPlayout() |
| 2949 | { |
| 2950 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| 2951 | "Channel::StopRecordingPlayout()"); |
| 2952 | |
| 2953 | if (!_outputFileRecording) |
| 2954 | { |
| 2955 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1), |
| 2956 | "StopRecordingPlayout() isnot recording"); |
| 2957 | return -1; |
| 2958 | } |
| 2959 | |
| 2960 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 2961 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2962 | |
| 2963 | if (_outputFileRecorderPtr->StopRecording() != 0) |
| 2964 | { |
| 2965 | _engineStatisticsPtr->SetLastError( |
| 2966 | VE_STOP_RECORDING_FAILED, kTraceError, |
| 2967 | "StopRecording() could not stop recording"); |
| 2968 | return(-1); |
| 2969 | } |
| 2970 | _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 2971 | FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| 2972 | _outputFileRecorderPtr = NULL; |
| 2973 | _outputFileRecording = false; |
| 2974 | |
| 2975 | return 0; |
| 2976 | } |
| 2977 | |
| 2978 | void |
| 2979 | Channel::SetMixWithMicStatus(bool mix) |
| 2980 | { |
| 2981 | _mixFileWithMicrophone=mix; |
| 2982 | } |
| 2983 | |
| 2984 | int |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2985 | Channel::GetSpeechOutputLevel(uint32_t& level) const |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2986 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2987 | int8_t currentLevel = _outputAudioLevel.Level(); |
| 2988 | level = static_cast<int32_t> (currentLevel); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2989 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 2990 | VoEId(_instanceId,_channelId), |
| 2991 | "GetSpeechOutputLevel() => level=%u", level); |
| 2992 | return 0; |
| 2993 | } |
| 2994 | |
| 2995 | int |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2996 | Channel::GetSpeechOutputLevelFullRange(uint32_t& level) const |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 2997 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 2998 | int16_t currentLevel = _outputAudioLevel.LevelFullRange(); |
| 2999 | level = static_cast<int32_t> (currentLevel); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3000 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3001 | VoEId(_instanceId,_channelId), |
| 3002 | "GetSpeechOutputLevelFullRange() => level=%u", level); |
| 3003 | return 0; |
| 3004 | } |
| 3005 | |
| 3006 | int |
| 3007 | Channel::SetMute(bool enable) |
| 3008 | { |
| 3009 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3010 | "Channel::SetMute(enable=%d)", enable); |
| 3011 | _mute = enable; |
| 3012 | return 0; |
| 3013 | } |
| 3014 | |
| 3015 | bool |
| 3016 | Channel::Mute() const |
| 3017 | { |
| 3018 | return _mute; |
| 3019 | } |
| 3020 | |
| 3021 | int |
| 3022 | Channel::SetOutputVolumePan(float left, float right) |
| 3023 | { |
| 3024 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3025 | "Channel::SetOutputVolumePan()"); |
| 3026 | _panLeft = left; |
| 3027 | _panRight = right; |
| 3028 | return 0; |
| 3029 | } |
| 3030 | |
| 3031 | int |
| 3032 | Channel::GetOutputVolumePan(float& left, float& right) const |
| 3033 | { |
| 3034 | left = _panLeft; |
| 3035 | right = _panRight; |
| 3036 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3037 | VoEId(_instanceId,_channelId), |
| 3038 | "GetOutputVolumePan() => left=%3.2f, right=%3.2f", left, right); |
| 3039 | return 0; |
| 3040 | } |
| 3041 | |
| 3042 | int |
| 3043 | Channel::SetChannelOutputVolumeScaling(float scaling) |
| 3044 | { |
| 3045 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3046 | "Channel::SetChannelOutputVolumeScaling()"); |
| 3047 | _outputGain = scaling; |
| 3048 | return 0; |
| 3049 | } |
| 3050 | |
| 3051 | int |
| 3052 | Channel::GetChannelOutputVolumeScaling(float& scaling) const |
| 3053 | { |
| 3054 | scaling = _outputGain; |
| 3055 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3056 | VoEId(_instanceId,_channelId), |
| 3057 | "GetChannelOutputVolumeScaling() => scaling=%3.2f", scaling); |
| 3058 | return 0; |
| 3059 | } |
| 3060 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3061 | int |
| 3062 | Channel::RegisterExternalEncryption(Encryption& encryption) |
| 3063 | { |
| 3064 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3065 | "Channel::RegisterExternalEncryption()"); |
| 3066 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 3067 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3068 | |
| 3069 | if (_encryptionPtr) |
| 3070 | { |
| 3071 | _engineStatisticsPtr->SetLastError( |
| 3072 | VE_INVALID_OPERATION, kTraceError, |
| 3073 | "RegisterExternalEncryption() encryption already enabled"); |
| 3074 | return -1; |
| 3075 | } |
| 3076 | |
| 3077 | _encryptionPtr = &encryption; |
| 3078 | |
| 3079 | _decrypting = true; |
| 3080 | _encrypting = true; |
| 3081 | |
| 3082 | return 0; |
| 3083 | } |
| 3084 | |
| 3085 | int |
| 3086 | Channel::DeRegisterExternalEncryption() |
| 3087 | { |
| 3088 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3089 | "Channel::DeRegisterExternalEncryption()"); |
| 3090 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 3091 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3092 | |
| 3093 | if (!_encryptionPtr) |
| 3094 | { |
| 3095 | _engineStatisticsPtr->SetLastError( |
| 3096 | VE_INVALID_OPERATION, kTraceWarning, |
| 3097 | "DeRegisterExternalEncryption() encryption already disabled"); |
| 3098 | return 0; |
| 3099 | } |
| 3100 | |
| 3101 | _decrypting = false; |
| 3102 | _encrypting = false; |
| 3103 | |
| 3104 | _encryptionPtr = NULL; |
| 3105 | |
| 3106 | return 0; |
| 3107 | } |
| 3108 | |
| 3109 | int Channel::SendTelephoneEventOutband(unsigned char eventCode, |
| 3110 | int lengthMs, int attenuationDb, |
| 3111 | bool playDtmfEvent) |
| 3112 | { |
| 3113 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 3114 | "Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)", |
| 3115 | playDtmfEvent); |
| 3116 | |
| 3117 | _playOutbandDtmfEvent = playDtmfEvent; |
| 3118 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3119 | if (_rtpRtcpModule->SendTelephoneEventOutband(eventCode, lengthMs, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3120 | attenuationDb) != 0) |
| 3121 | { |
| 3122 | _engineStatisticsPtr->SetLastError( |
| 3123 | VE_SEND_DTMF_FAILED, |
| 3124 | kTraceWarning, |
| 3125 | "SendTelephoneEventOutband() failed to send event"); |
| 3126 | return -1; |
| 3127 | } |
| 3128 | return 0; |
| 3129 | } |
| 3130 | |
| 3131 | int Channel::SendTelephoneEventInband(unsigned char eventCode, |
| 3132 | int lengthMs, |
| 3133 | int attenuationDb, |
| 3134 | bool playDtmfEvent) |
| 3135 | { |
| 3136 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 3137 | "Channel::SendTelephoneEventInband(..., playDtmfEvent=%d)", |
| 3138 | playDtmfEvent); |
| 3139 | |
| 3140 | _playInbandDtmfEvent = playDtmfEvent; |
| 3141 | _inbandDtmfQueue.AddDtmf(eventCode, lengthMs, attenuationDb); |
| 3142 | |
| 3143 | return 0; |
| 3144 | } |
| 3145 | |
| 3146 | int |
| 3147 | Channel::SetDtmfPlayoutStatus(bool enable) |
| 3148 | { |
| 3149 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3150 | "Channel::SetDtmfPlayoutStatus()"); |
| 3151 | if (_audioCodingModule.SetDtmfPlayoutStatus(enable) != 0) |
| 3152 | { |
| 3153 | _engineStatisticsPtr->SetLastError( |
| 3154 | VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning, |
| 3155 | "SetDtmfPlayoutStatus() failed to set Dtmf playout"); |
| 3156 | return -1; |
| 3157 | } |
| 3158 | return 0; |
| 3159 | } |
| 3160 | |
| 3161 | bool |
| 3162 | Channel::DtmfPlayoutStatus() const |
| 3163 | { |
| 3164 | return _audioCodingModule.DtmfPlayoutStatus(); |
| 3165 | } |
| 3166 | |
| 3167 | int |
| 3168 | Channel::SetSendTelephoneEventPayloadType(unsigned char type) |
| 3169 | { |
| 3170 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3171 | "Channel::SetSendTelephoneEventPayloadType()"); |
andrew@webrtc.org | f81f9f8 | 2011-08-19 22:56:22 +0000 | [diff] [blame] | 3172 | if (type > 127) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3173 | { |
| 3174 | _engineStatisticsPtr->SetLastError( |
| 3175 | VE_INVALID_ARGUMENT, kTraceError, |
| 3176 | "SetSendTelephoneEventPayloadType() invalid type"); |
| 3177 | return -1; |
| 3178 | } |
pwestin@webrtc.org | 1da1ce0 | 2011-10-13 15:19:55 +0000 | [diff] [blame] | 3179 | CodecInst codec; |
| 3180 | codec.plfreq = 8000; |
| 3181 | codec.pltype = type; |
| 3182 | memcpy(codec.plname, "telephone-event", 16); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3183 | if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3184 | { |
| 3185 | _engineStatisticsPtr->SetLastError( |
| 3186 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 3187 | "SetSendTelephoneEventPayloadType() failed to register send" |
| 3188 | "payload type"); |
| 3189 | return -1; |
| 3190 | } |
| 3191 | _sendTelephoneEventPayloadType = type; |
| 3192 | return 0; |
| 3193 | } |
| 3194 | |
| 3195 | int |
| 3196 | Channel::GetSendTelephoneEventPayloadType(unsigned char& type) |
| 3197 | { |
| 3198 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3199 | "Channel::GetSendTelephoneEventPayloadType()"); |
| 3200 | type = _sendTelephoneEventPayloadType; |
| 3201 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3202 | VoEId(_instanceId,_channelId), |
| 3203 | "GetSendTelephoneEventPayloadType() => type=%u", type); |
| 3204 | return 0; |
| 3205 | } |
| 3206 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3207 | int |
| 3208 | Channel::UpdateRxVadDetection(AudioFrame& audioFrame) |
| 3209 | { |
| 3210 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3211 | "Channel::UpdateRxVadDetection()"); |
| 3212 | |
| 3213 | int vadDecision = 1; |
| 3214 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 3215 | vadDecision = (audioFrame.vad_activity_ == AudioFrame::kVadActive)? 1 : 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3216 | |
| 3217 | if ((vadDecision != _oldVadDecision) && _rxVadObserverPtr) |
| 3218 | { |
| 3219 | OnRxVadDetected(vadDecision); |
| 3220 | _oldVadDecision = vadDecision; |
| 3221 | } |
| 3222 | |
| 3223 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3224 | "Channel::UpdateRxVadDetection() => vadDecision=%d", |
| 3225 | vadDecision); |
| 3226 | return 0; |
| 3227 | } |
| 3228 | |
| 3229 | int |
| 3230 | Channel::RegisterRxVadObserver(VoERxVadCallback &observer) |
| 3231 | { |
| 3232 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3233 | "Channel::RegisterRxVadObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 3234 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3235 | |
| 3236 | if (_rxVadObserverPtr) |
| 3237 | { |
| 3238 | _engineStatisticsPtr->SetLastError( |
| 3239 | VE_INVALID_OPERATION, kTraceError, |
| 3240 | "RegisterRxVadObserver() observer already enabled"); |
| 3241 | return -1; |
| 3242 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3243 | _rxVadObserverPtr = &observer; |
| 3244 | _RxVadDetection = true; |
| 3245 | return 0; |
| 3246 | } |
| 3247 | |
| 3248 | int |
| 3249 | Channel::DeRegisterRxVadObserver() |
| 3250 | { |
| 3251 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3252 | "Channel::DeRegisterRxVadObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 3253 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3254 | |
| 3255 | if (!_rxVadObserverPtr) |
| 3256 | { |
| 3257 | _engineStatisticsPtr->SetLastError( |
| 3258 | VE_INVALID_OPERATION, kTraceWarning, |
| 3259 | "DeRegisterRxVadObserver() observer already disabled"); |
| 3260 | return 0; |
| 3261 | } |
| 3262 | _rxVadObserverPtr = NULL; |
| 3263 | _RxVadDetection = false; |
| 3264 | return 0; |
| 3265 | } |
| 3266 | |
| 3267 | int |
| 3268 | Channel::VoiceActivityIndicator(int &activity) |
| 3269 | { |
| 3270 | activity = _sendFrameType; |
| 3271 | |
| 3272 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3273 | "Channel::VoiceActivityIndicator(indicator=%d)", activity); |
| 3274 | return 0; |
| 3275 | } |
| 3276 | |
| 3277 | #ifdef WEBRTC_VOICE_ENGINE_AGC |
| 3278 | |
| 3279 | int |
| 3280 | Channel::SetRxAgcStatus(const bool enable, const AgcModes mode) |
| 3281 | { |
| 3282 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3283 | "Channel::SetRxAgcStatus(enable=%d, mode=%d)", |
| 3284 | (int)enable, (int)mode); |
| 3285 | |
| 3286 | GainControl::Mode agcMode(GainControl::kFixedDigital); |
| 3287 | switch (mode) |
| 3288 | { |
| 3289 | case kAgcDefault: |
| 3290 | agcMode = GainControl::kAdaptiveDigital; |
| 3291 | break; |
| 3292 | case kAgcUnchanged: |
| 3293 | agcMode = _rxAudioProcessingModulePtr->gain_control()->mode(); |
| 3294 | break; |
| 3295 | case kAgcFixedDigital: |
| 3296 | agcMode = GainControl::kFixedDigital; |
| 3297 | break; |
| 3298 | case kAgcAdaptiveDigital: |
| 3299 | agcMode =GainControl::kAdaptiveDigital; |
| 3300 | break; |
| 3301 | default: |
| 3302 | _engineStatisticsPtr->SetLastError( |
| 3303 | VE_INVALID_ARGUMENT, kTraceError, |
| 3304 | "SetRxAgcStatus() invalid Agc mode"); |
| 3305 | return -1; |
| 3306 | } |
| 3307 | |
| 3308 | if (_rxAudioProcessingModulePtr->gain_control()->set_mode(agcMode) != 0) |
| 3309 | { |
| 3310 | _engineStatisticsPtr->SetLastError( |
| 3311 | VE_APM_ERROR, kTraceError, |
| 3312 | "SetRxAgcStatus() failed to set Agc mode"); |
| 3313 | return -1; |
| 3314 | } |
| 3315 | if (_rxAudioProcessingModulePtr->gain_control()->Enable(enable) != 0) |
| 3316 | { |
| 3317 | _engineStatisticsPtr->SetLastError( |
| 3318 | VE_APM_ERROR, kTraceError, |
| 3319 | "SetRxAgcStatus() failed to set Agc state"); |
| 3320 | return -1; |
| 3321 | } |
| 3322 | |
| 3323 | _rxAgcIsEnabled = enable; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3324 | _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true)); |
| 3325 | |
| 3326 | return 0; |
| 3327 | } |
| 3328 | |
| 3329 | int |
| 3330 | Channel::GetRxAgcStatus(bool& enabled, AgcModes& mode) |
| 3331 | { |
| 3332 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3333 | "Channel::GetRxAgcStatus(enable=?, mode=?)"); |
| 3334 | |
| 3335 | bool enable = _rxAudioProcessingModulePtr->gain_control()->is_enabled(); |
| 3336 | GainControl::Mode agcMode = |
| 3337 | _rxAudioProcessingModulePtr->gain_control()->mode(); |
| 3338 | |
| 3339 | enabled = enable; |
| 3340 | |
| 3341 | switch (agcMode) |
| 3342 | { |
| 3343 | case GainControl::kFixedDigital: |
| 3344 | mode = kAgcFixedDigital; |
| 3345 | break; |
| 3346 | case GainControl::kAdaptiveDigital: |
| 3347 | mode = kAgcAdaptiveDigital; |
| 3348 | break; |
| 3349 | default: |
| 3350 | _engineStatisticsPtr->SetLastError( |
| 3351 | VE_APM_ERROR, kTraceError, |
| 3352 | "GetRxAgcStatus() invalid Agc mode"); |
| 3353 | return -1; |
| 3354 | } |
| 3355 | |
| 3356 | return 0; |
| 3357 | } |
| 3358 | |
| 3359 | int |
| 3360 | Channel::SetRxAgcConfig(const AgcConfig config) |
| 3361 | { |
| 3362 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3363 | "Channel::SetRxAgcConfig()"); |
| 3364 | |
| 3365 | if (_rxAudioProcessingModulePtr->gain_control()->set_target_level_dbfs( |
| 3366 | config.targetLeveldBOv) != 0) |
| 3367 | { |
| 3368 | _engineStatisticsPtr->SetLastError( |
| 3369 | VE_APM_ERROR, kTraceError, |
| 3370 | "SetRxAgcConfig() failed to set target peak |level|" |
| 3371 | "(or envelope) of the Agc"); |
| 3372 | return -1; |
| 3373 | } |
| 3374 | if (_rxAudioProcessingModulePtr->gain_control()->set_compression_gain_db( |
| 3375 | config.digitalCompressionGaindB) != 0) |
| 3376 | { |
| 3377 | _engineStatisticsPtr->SetLastError( |
| 3378 | VE_APM_ERROR, kTraceError, |
| 3379 | "SetRxAgcConfig() failed to set the range in |gain| the" |
| 3380 | " digital compression stage may apply"); |
| 3381 | return -1; |
| 3382 | } |
| 3383 | if (_rxAudioProcessingModulePtr->gain_control()->enable_limiter( |
| 3384 | config.limiterEnable) != 0) |
| 3385 | { |
| 3386 | _engineStatisticsPtr->SetLastError( |
| 3387 | VE_APM_ERROR, kTraceError, |
| 3388 | "SetRxAgcConfig() failed to set hard limiter to the signal"); |
| 3389 | return -1; |
| 3390 | } |
| 3391 | |
| 3392 | return 0; |
| 3393 | } |
| 3394 | |
| 3395 | int |
| 3396 | Channel::GetRxAgcConfig(AgcConfig& config) |
| 3397 | { |
| 3398 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3399 | "Channel::GetRxAgcConfig(config=%?)"); |
| 3400 | |
| 3401 | config.targetLeveldBOv = |
| 3402 | _rxAudioProcessingModulePtr->gain_control()->target_level_dbfs(); |
| 3403 | config.digitalCompressionGaindB = |
| 3404 | _rxAudioProcessingModulePtr->gain_control()->compression_gain_db(); |
| 3405 | config.limiterEnable = |
| 3406 | _rxAudioProcessingModulePtr->gain_control()->is_limiter_enabled(); |
| 3407 | |
| 3408 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3409 | VoEId(_instanceId,_channelId), "GetRxAgcConfig() => " |
| 3410 | "targetLeveldBOv=%u, digitalCompressionGaindB=%u," |
| 3411 | " limiterEnable=%d", |
| 3412 | config.targetLeveldBOv, |
| 3413 | config.digitalCompressionGaindB, |
| 3414 | config.limiterEnable); |
| 3415 | |
| 3416 | return 0; |
| 3417 | } |
| 3418 | |
| 3419 | #endif // #ifdef WEBRTC_VOICE_ENGINE_AGC |
| 3420 | |
| 3421 | #ifdef WEBRTC_VOICE_ENGINE_NR |
| 3422 | |
| 3423 | int |
| 3424 | Channel::SetRxNsStatus(const bool enable, const NsModes mode) |
| 3425 | { |
| 3426 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3427 | "Channel::SetRxNsStatus(enable=%d, mode=%d)", |
| 3428 | (int)enable, (int)mode); |
| 3429 | |
| 3430 | NoiseSuppression::Level nsLevel( |
| 3431 | (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE); |
| 3432 | switch (mode) |
| 3433 | { |
| 3434 | |
| 3435 | case kNsDefault: |
| 3436 | nsLevel = (NoiseSuppression::Level) |
| 3437 | WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE; |
| 3438 | break; |
| 3439 | case kNsUnchanged: |
| 3440 | nsLevel = _rxAudioProcessingModulePtr->noise_suppression()->level(); |
| 3441 | break; |
| 3442 | case kNsConference: |
| 3443 | nsLevel = NoiseSuppression::kHigh; |
| 3444 | break; |
| 3445 | case kNsLowSuppression: |
| 3446 | nsLevel = NoiseSuppression::kLow; |
| 3447 | break; |
| 3448 | case kNsModerateSuppression: |
| 3449 | nsLevel = NoiseSuppression::kModerate; |
| 3450 | break; |
| 3451 | case kNsHighSuppression: |
| 3452 | nsLevel = NoiseSuppression::kHigh; |
| 3453 | break; |
| 3454 | case kNsVeryHighSuppression: |
| 3455 | nsLevel = NoiseSuppression::kVeryHigh; |
| 3456 | break; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3457 | } |
| 3458 | |
| 3459 | if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(nsLevel) |
| 3460 | != 0) |
| 3461 | { |
| 3462 | _engineStatisticsPtr->SetLastError( |
| 3463 | VE_APM_ERROR, kTraceError, |
| 3464 | "SetRxAgcStatus() failed to set Ns level"); |
| 3465 | return -1; |
| 3466 | } |
| 3467 | if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(enable) != 0) |
| 3468 | { |
| 3469 | _engineStatisticsPtr->SetLastError( |
| 3470 | VE_APM_ERROR, kTraceError, |
| 3471 | "SetRxAgcStatus() failed to set Agc state"); |
| 3472 | return -1; |
| 3473 | } |
| 3474 | |
| 3475 | _rxNsIsEnabled = enable; |
| 3476 | _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true)); |
| 3477 | |
| 3478 | return 0; |
| 3479 | } |
| 3480 | |
| 3481 | int |
| 3482 | Channel::GetRxNsStatus(bool& enabled, NsModes& mode) |
| 3483 | { |
| 3484 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3485 | "Channel::GetRxNsStatus(enable=?, mode=?)"); |
| 3486 | |
| 3487 | bool enable = |
| 3488 | _rxAudioProcessingModulePtr->noise_suppression()->is_enabled(); |
| 3489 | NoiseSuppression::Level ncLevel = |
| 3490 | _rxAudioProcessingModulePtr->noise_suppression()->level(); |
| 3491 | |
| 3492 | enabled = enable; |
| 3493 | |
| 3494 | switch (ncLevel) |
| 3495 | { |
| 3496 | case NoiseSuppression::kLow: |
| 3497 | mode = kNsLowSuppression; |
| 3498 | break; |
| 3499 | case NoiseSuppression::kModerate: |
| 3500 | mode = kNsModerateSuppression; |
| 3501 | break; |
| 3502 | case NoiseSuppression::kHigh: |
| 3503 | mode = kNsHighSuppression; |
| 3504 | break; |
| 3505 | case NoiseSuppression::kVeryHigh: |
| 3506 | mode = kNsVeryHighSuppression; |
| 3507 | break; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3508 | } |
| 3509 | |
| 3510 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3511 | VoEId(_instanceId,_channelId), |
| 3512 | "GetRxNsStatus() => enabled=%d, mode=%d", enabled, mode); |
| 3513 | return 0; |
| 3514 | } |
| 3515 | |
| 3516 | #endif // #ifdef WEBRTC_VOICE_ENGINE_NR |
| 3517 | |
| 3518 | int |
| 3519 | Channel::RegisterRTPObserver(VoERTPObserver& observer) |
| 3520 | { |
| 3521 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 3522 | "Channel::RegisterRTPObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 3523 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3524 | |
| 3525 | if (_rtpObserverPtr) |
| 3526 | { |
| 3527 | _engineStatisticsPtr->SetLastError( |
| 3528 | VE_INVALID_OPERATION, kTraceError, |
| 3529 | "RegisterRTPObserver() observer already enabled"); |
| 3530 | return -1; |
| 3531 | } |
| 3532 | |
| 3533 | _rtpObserverPtr = &observer; |
| 3534 | _rtpObserver = true; |
| 3535 | |
| 3536 | return 0; |
| 3537 | } |
| 3538 | |
| 3539 | int |
| 3540 | Channel::DeRegisterRTPObserver() |
| 3541 | { |
| 3542 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3543 | "Channel::DeRegisterRTPObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 3544 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3545 | |
| 3546 | if (!_rtpObserverPtr) |
| 3547 | { |
| 3548 | _engineStatisticsPtr->SetLastError( |
| 3549 | VE_INVALID_OPERATION, kTraceWarning, |
| 3550 | "DeRegisterRTPObserver() observer already disabled"); |
| 3551 | return 0; |
| 3552 | } |
| 3553 | |
| 3554 | _rtpObserver = false; |
| 3555 | _rtpObserverPtr = NULL; |
| 3556 | |
| 3557 | return 0; |
| 3558 | } |
| 3559 | |
| 3560 | int |
| 3561 | Channel::RegisterRTCPObserver(VoERTCPObserver& observer) |
| 3562 | { |
| 3563 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3564 | "Channel::RegisterRTCPObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 3565 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3566 | |
| 3567 | if (_rtcpObserverPtr) |
| 3568 | { |
| 3569 | _engineStatisticsPtr->SetLastError( |
| 3570 | VE_INVALID_OPERATION, kTraceError, |
| 3571 | "RegisterRTCPObserver() observer already enabled"); |
| 3572 | return -1; |
| 3573 | } |
| 3574 | |
| 3575 | _rtcpObserverPtr = &observer; |
| 3576 | _rtcpObserver = true; |
| 3577 | |
| 3578 | return 0; |
| 3579 | } |
| 3580 | |
| 3581 | int |
| 3582 | Channel::DeRegisterRTCPObserver() |
| 3583 | { |
| 3584 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 3585 | "Channel::DeRegisterRTCPObserver()"); |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 3586 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3587 | |
| 3588 | if (!_rtcpObserverPtr) |
| 3589 | { |
| 3590 | _engineStatisticsPtr->SetLastError( |
| 3591 | VE_INVALID_OPERATION, kTraceWarning, |
| 3592 | "DeRegisterRTCPObserver() observer already disabled"); |
| 3593 | return 0; |
| 3594 | } |
| 3595 | |
| 3596 | _rtcpObserver = false; |
| 3597 | _rtcpObserverPtr = NULL; |
| 3598 | |
| 3599 | return 0; |
| 3600 | } |
| 3601 | |
| 3602 | int |
| 3603 | Channel::SetLocalSSRC(unsigned int ssrc) |
| 3604 | { |
| 3605 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 3606 | "Channel::SetLocalSSRC()"); |
| 3607 | if (_sending) |
| 3608 | { |
| 3609 | _engineStatisticsPtr->SetLastError( |
| 3610 | VE_ALREADY_SENDING, kTraceError, |
| 3611 | "SetLocalSSRC() already sending"); |
| 3612 | return -1; |
| 3613 | } |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3614 | if (_rtpRtcpModule->SetSSRC(ssrc) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3615 | { |
| 3616 | _engineStatisticsPtr->SetLastError( |
| 3617 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 3618 | "SetLocalSSRC() failed to set SSRC"); |
| 3619 | return -1; |
| 3620 | } |
| 3621 | return 0; |
| 3622 | } |
| 3623 | |
| 3624 | int |
| 3625 | Channel::GetLocalSSRC(unsigned int& ssrc) |
| 3626 | { |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3627 | ssrc = _rtpRtcpModule->SSRC(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3628 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3629 | VoEId(_instanceId,_channelId), |
| 3630 | "GetLocalSSRC() => ssrc=%lu", ssrc); |
| 3631 | return 0; |
| 3632 | } |
| 3633 | |
| 3634 | int |
| 3635 | Channel::GetRemoteSSRC(unsigned int& ssrc) |
| 3636 | { |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3637 | ssrc = _rtpRtcpModule->RemoteSSRC(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3638 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3639 | VoEId(_instanceId,_channelId), |
| 3640 | "GetRemoteSSRC() => ssrc=%lu", ssrc); |
| 3641 | return 0; |
| 3642 | } |
| 3643 | |
| 3644 | int |
| 3645 | Channel::GetRemoteCSRCs(unsigned int arrCSRC[15]) |
| 3646 | { |
| 3647 | if (arrCSRC == NULL) |
| 3648 | { |
| 3649 | _engineStatisticsPtr->SetLastError( |
| 3650 | VE_INVALID_ARGUMENT, kTraceError, |
| 3651 | "GetRemoteCSRCs() invalid array argument"); |
| 3652 | return -1; |
| 3653 | } |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 3654 | uint32_t arrOfCSRC[kRtpCsrcSize]; |
| 3655 | int32_t CSRCs(0); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3656 | CSRCs = _rtpRtcpModule->CSRCs(arrOfCSRC); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3657 | if (CSRCs > 0) |
| 3658 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 3659 | memcpy(arrCSRC, arrOfCSRC, CSRCs * sizeof(uint32_t)); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3660 | for (int i = 0; i < (int) CSRCs; i++) |
| 3661 | { |
| 3662 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3663 | VoEId(_instanceId, _channelId), |
| 3664 | "GetRemoteCSRCs() => arrCSRC[%d]=%lu", i, arrCSRC[i]); |
| 3665 | } |
| 3666 | } else |
| 3667 | { |
| 3668 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3669 | VoEId(_instanceId, _channelId), |
| 3670 | "GetRemoteCSRCs() => list is empty!"); |
| 3671 | } |
| 3672 | return CSRCs; |
| 3673 | } |
| 3674 | |
| 3675 | int |
| 3676 | Channel::SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID) |
| 3677 | { |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 3678 | if (_rtpAudioProc.get() == NULL) |
| 3679 | { |
| 3680 | _rtpAudioProc.reset(AudioProcessing::Create(VoEModuleId(_instanceId, |
| 3681 | _channelId))); |
| 3682 | if (_rtpAudioProc.get() == NULL) |
| 3683 | { |
| 3684 | _engineStatisticsPtr->SetLastError(VE_NO_MEMORY, kTraceCritical, |
| 3685 | "Failed to create AudioProcessing"); |
| 3686 | return -1; |
| 3687 | } |
| 3688 | } |
| 3689 | |
| 3690 | if (_rtpAudioProc->level_estimator()->Enable(enable) != |
| 3691 | AudioProcessing::kNoError) |
| 3692 | { |
| 3693 | _engineStatisticsPtr->SetLastError(VE_APM_ERROR, kTraceWarning, |
| 3694 | "Failed to enable AudioProcessing::level_estimator()"); |
| 3695 | } |
| 3696 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3697 | _includeAudioLevelIndication = enable; |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3698 | return _rtpRtcpModule->SetRTPAudioLevelIndicationStatus(enable, ID); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3699 | } |
| 3700 | int |
| 3701 | Channel::GetRTPAudioLevelIndicationStatus(bool& enabled, unsigned char& ID) |
| 3702 | { |
| 3703 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3704 | VoEId(_instanceId,_channelId), |
| 3705 | "GetRTPAudioLevelIndicationStatus() => enabled=%d, ID=%u", |
| 3706 | enabled, ID); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3707 | return _rtpRtcpModule->GetRTPAudioLevelIndicationStatus(enabled, ID); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3708 | } |
| 3709 | |
| 3710 | int |
| 3711 | Channel::SetRTCPStatus(bool enable) |
| 3712 | { |
| 3713 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 3714 | "Channel::SetRTCPStatus()"); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3715 | if (_rtpRtcpModule->SetRTCPStatus(enable ? |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3716 | kRtcpCompound : kRtcpOff) != 0) |
| 3717 | { |
| 3718 | _engineStatisticsPtr->SetLastError( |
| 3719 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 3720 | "SetRTCPStatus() failed to set RTCP status"); |
| 3721 | return -1; |
| 3722 | } |
| 3723 | return 0; |
| 3724 | } |
| 3725 | |
| 3726 | int |
| 3727 | Channel::GetRTCPStatus(bool& enabled) |
| 3728 | { |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3729 | RTCPMethod method = _rtpRtcpModule->RTCP(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3730 | enabled = (method != kRtcpOff); |
| 3731 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3732 | VoEId(_instanceId,_channelId), |
| 3733 | "GetRTCPStatus() => enabled=%d", enabled); |
| 3734 | return 0; |
| 3735 | } |
| 3736 | |
| 3737 | int |
| 3738 | Channel::SetRTCP_CNAME(const char cName[256]) |
| 3739 | { |
| 3740 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 3741 | "Channel::SetRTCP_CNAME()"); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3742 | if (_rtpRtcpModule->SetCNAME(cName) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3743 | { |
| 3744 | _engineStatisticsPtr->SetLastError( |
| 3745 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 3746 | "SetRTCP_CNAME() failed to set RTCP CNAME"); |
| 3747 | return -1; |
| 3748 | } |
| 3749 | return 0; |
| 3750 | } |
| 3751 | |
| 3752 | int |
| 3753 | Channel::GetRTCP_CNAME(char cName[256]) |
| 3754 | { |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3755 | if (_rtpRtcpModule->CNAME(cName) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3756 | { |
| 3757 | _engineStatisticsPtr->SetLastError( |
| 3758 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 3759 | "GetRTCP_CNAME() failed to retrieve RTCP CNAME"); |
| 3760 | return -1; |
| 3761 | } |
| 3762 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3763 | VoEId(_instanceId, _channelId), |
| 3764 | "GetRTCP_CNAME() => cName=%s", cName); |
| 3765 | return 0; |
| 3766 | } |
| 3767 | |
| 3768 | int |
| 3769 | Channel::GetRemoteRTCP_CNAME(char cName[256]) |
| 3770 | { |
| 3771 | if (cName == NULL) |
| 3772 | { |
| 3773 | _engineStatisticsPtr->SetLastError( |
| 3774 | VE_INVALID_ARGUMENT, kTraceError, |
| 3775 | "GetRemoteRTCP_CNAME() invalid CNAME input buffer"); |
| 3776 | return -1; |
| 3777 | } |
leozwang@webrtc.org | 813e4b0 | 2012-03-01 18:34:25 +0000 | [diff] [blame] | 3778 | char cname[RTCP_CNAME_SIZE]; |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 3779 | const uint32_t remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3780 | if (_rtpRtcpModule->RemoteCNAME(remoteSSRC, cname) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3781 | { |
| 3782 | _engineStatisticsPtr->SetLastError( |
| 3783 | VE_CANNOT_RETRIEVE_CNAME, kTraceError, |
| 3784 | "GetRemoteRTCP_CNAME() failed to retrieve remote RTCP CNAME"); |
| 3785 | return -1; |
| 3786 | } |
| 3787 | strcpy(cName, cname); |
| 3788 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3789 | VoEId(_instanceId, _channelId), |
| 3790 | "GetRemoteRTCP_CNAME() => cName=%s", cName); |
| 3791 | return 0; |
| 3792 | } |
| 3793 | |
| 3794 | int |
| 3795 | Channel::GetRemoteRTCPData( |
| 3796 | unsigned int& NTPHigh, |
| 3797 | unsigned int& NTPLow, |
| 3798 | unsigned int& timestamp, |
| 3799 | unsigned int& playoutTimestamp, |
| 3800 | unsigned int* jitter, |
| 3801 | unsigned short* fractionLost) |
| 3802 | { |
| 3803 | // --- Information from sender info in received Sender Reports |
| 3804 | |
| 3805 | RTCPSenderInfo senderInfo; |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3806 | if (_rtpRtcpModule->RemoteRTCPStat(&senderInfo) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3807 | { |
| 3808 | _engineStatisticsPtr->SetLastError( |
| 3809 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 3810 | "GetRemoteRTCPData() failed to retrieve sender info for remote " |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3811 | "side"); |
| 3812 | return -1; |
| 3813 | } |
| 3814 | |
| 3815 | // We only utilize 12 out of 20 bytes in the sender info (ignores packet |
| 3816 | // and octet count) |
| 3817 | NTPHigh = senderInfo.NTPseconds; |
| 3818 | NTPLow = senderInfo.NTPfraction; |
| 3819 | timestamp = senderInfo.RTPtimeStamp; |
| 3820 | |
| 3821 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3822 | VoEId(_instanceId, _channelId), |
| 3823 | "GetRemoteRTCPData() => NTPHigh=%lu, NTPLow=%lu, " |
| 3824 | "timestamp=%lu", |
| 3825 | NTPHigh, NTPLow, timestamp); |
| 3826 | |
| 3827 | // --- Locally derived information |
| 3828 | |
| 3829 | // This value is updated on each incoming RTCP packet (0 when no packet |
| 3830 | // has been received) |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 3831 | playoutTimestamp = playout_timestamp_rtcp_; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3832 | |
| 3833 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3834 | VoEId(_instanceId, _channelId), |
| 3835 | "GetRemoteRTCPData() => playoutTimestamp=%lu", |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 3836 | playout_timestamp_rtcp_); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3837 | |
| 3838 | if (NULL != jitter || NULL != fractionLost) |
| 3839 | { |
perkj@webrtc.org | ce5990c | 2012-01-11 13:00:08 +0000 | [diff] [blame] | 3840 | // Get all RTCP receiver report blocks that have been received on this |
| 3841 | // channel. If we receive RTP packets from a remote source we know the |
| 3842 | // remote SSRC and use the report block from him. |
| 3843 | // Otherwise use the first report block. |
| 3844 | std::vector<RTCPReportBlock> remote_stats; |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3845 | if (_rtpRtcpModule->RemoteRTCPStat(&remote_stats) != 0 || |
perkj@webrtc.org | ce5990c | 2012-01-11 13:00:08 +0000 | [diff] [blame] | 3846 | remote_stats.empty()) { |
| 3847 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 3848 | VoEId(_instanceId, _channelId), |
| 3849 | "GetRemoteRTCPData() failed to measure statistics due" |
| 3850 | " to lack of received RTP and/or RTCP packets"); |
| 3851 | return -1; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3852 | } |
perkj@webrtc.org | ce5990c | 2012-01-11 13:00:08 +0000 | [diff] [blame] | 3853 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 3854 | uint32_t remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
perkj@webrtc.org | ce5990c | 2012-01-11 13:00:08 +0000 | [diff] [blame] | 3855 | std::vector<RTCPReportBlock>::const_iterator it = remote_stats.begin(); |
| 3856 | for (; it != remote_stats.end(); ++it) { |
| 3857 | if (it->remoteSSRC == remoteSSRC) |
| 3858 | break; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3859 | } |
perkj@webrtc.org | ce5990c | 2012-01-11 13:00:08 +0000 | [diff] [blame] | 3860 | |
| 3861 | if (it == remote_stats.end()) { |
| 3862 | // If we have not received any RTCP packets from this SSRC it probably |
| 3863 | // means that we have not received any RTP packets. |
| 3864 | // Use the first received report block instead. |
| 3865 | it = remote_stats.begin(); |
| 3866 | remoteSSRC = it->remoteSSRC; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3867 | } |
perkj@webrtc.org | ce5990c | 2012-01-11 13:00:08 +0000 | [diff] [blame] | 3868 | |
xians@webrtc.org | 79af734 | 2012-01-31 12:22:14 +0000 | [diff] [blame] | 3869 | if (jitter) { |
| 3870 | *jitter = it->jitter; |
| 3871 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3872 | VoEId(_instanceId, _channelId), |
| 3873 | "GetRemoteRTCPData() => jitter = %lu", *jitter); |
| 3874 | } |
perkj@webrtc.org | ce5990c | 2012-01-11 13:00:08 +0000 | [diff] [blame] | 3875 | |
xians@webrtc.org | 79af734 | 2012-01-31 12:22:14 +0000 | [diff] [blame] | 3876 | if (fractionLost) { |
| 3877 | *fractionLost = it->fractionLost; |
| 3878 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3879 | VoEId(_instanceId, _channelId), |
| 3880 | "GetRemoteRTCPData() => fractionLost = %lu", |
| 3881 | *fractionLost); |
| 3882 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3883 | } |
| 3884 | return 0; |
| 3885 | } |
| 3886 | |
| 3887 | int |
| 3888 | Channel::SendApplicationDefinedRTCPPacket(const unsigned char subType, |
| 3889 | unsigned int name, |
| 3890 | const char* data, |
| 3891 | unsigned short dataLengthInBytes) |
| 3892 | { |
| 3893 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 3894 | "Channel::SendApplicationDefinedRTCPPacket()"); |
| 3895 | if (!_sending) |
| 3896 | { |
| 3897 | _engineStatisticsPtr->SetLastError( |
| 3898 | VE_NOT_SENDING, kTraceError, |
| 3899 | "SendApplicationDefinedRTCPPacket() not sending"); |
| 3900 | return -1; |
| 3901 | } |
| 3902 | if (NULL == data) |
| 3903 | { |
| 3904 | _engineStatisticsPtr->SetLastError( |
| 3905 | VE_INVALID_ARGUMENT, kTraceError, |
| 3906 | "SendApplicationDefinedRTCPPacket() invalid data value"); |
| 3907 | return -1; |
| 3908 | } |
| 3909 | if (dataLengthInBytes % 4 != 0) |
| 3910 | { |
| 3911 | _engineStatisticsPtr->SetLastError( |
| 3912 | VE_INVALID_ARGUMENT, kTraceError, |
| 3913 | "SendApplicationDefinedRTCPPacket() invalid length value"); |
| 3914 | return -1; |
| 3915 | } |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3916 | RTCPMethod status = _rtpRtcpModule->RTCP(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3917 | if (status == kRtcpOff) |
| 3918 | { |
| 3919 | _engineStatisticsPtr->SetLastError( |
| 3920 | VE_RTCP_ERROR, kTraceError, |
| 3921 | "SendApplicationDefinedRTCPPacket() RTCP is disabled"); |
| 3922 | return -1; |
| 3923 | } |
| 3924 | |
| 3925 | // Create and schedule the RTCP APP packet for transmission |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3926 | if (_rtpRtcpModule->SetRTCPApplicationSpecificData( |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3927 | subType, |
| 3928 | name, |
| 3929 | (const unsigned char*) data, |
| 3930 | dataLengthInBytes) != 0) |
| 3931 | { |
| 3932 | _engineStatisticsPtr->SetLastError( |
| 3933 | VE_SEND_ERROR, kTraceError, |
| 3934 | "SendApplicationDefinedRTCPPacket() failed to send RTCP packet"); |
| 3935 | return -1; |
| 3936 | } |
| 3937 | return 0; |
| 3938 | } |
| 3939 | |
| 3940 | int |
| 3941 | Channel::GetRTPStatistics( |
| 3942 | unsigned int& averageJitterMs, |
| 3943 | unsigned int& maxJitterMs, |
| 3944 | unsigned int& discardedPackets) |
| 3945 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 3946 | uint8_t fraction_lost(0); |
| 3947 | uint32_t cum_lost(0); |
| 3948 | uint32_t ext_max(0); |
| 3949 | uint32_t jitter(0); |
| 3950 | uint32_t max_jitter(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3951 | |
| 3952 | // The jitter statistics is updated for each received RTP packet and is |
| 3953 | // based on received packets. |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 3954 | if (_rtpRtcpModule->StatisticsRTP(&fraction_lost, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3955 | &cum_lost, |
| 3956 | &ext_max, |
| 3957 | &jitter, |
| 3958 | &max_jitter) != 0) |
| 3959 | { |
| 3960 | _engineStatisticsPtr->SetLastError( |
| 3961 | VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning, |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 3962 | "GetRTPStatistics() failed to read RTP statistics from the " |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3963 | "RTP/RTCP module"); |
| 3964 | } |
| 3965 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 3966 | const int32_t playoutFrequency = |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3967 | _audioCodingModule.PlayoutFrequency(); |
| 3968 | if (playoutFrequency > 0) |
| 3969 | { |
| 3970 | // Scale RTP statistics given the current playout frequency |
| 3971 | maxJitterMs = max_jitter / (playoutFrequency / 1000); |
| 3972 | averageJitterMs = jitter / (playoutFrequency / 1000); |
| 3973 | } |
| 3974 | |
| 3975 | discardedPackets = _numberOfDiscardedPackets; |
| 3976 | |
| 3977 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 3978 | VoEId(_instanceId, _channelId), |
| 3979 | "GetRTPStatistics() => averageJitterMs = %lu, maxJitterMs = %lu," |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 3980 | " discardedPackets = %lu)", |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 3981 | averageJitterMs, maxJitterMs, discardedPackets); |
| 3982 | return 0; |
| 3983 | } |
| 3984 | |
henrika@webrtc.org | 8a2fc88 | 2012-08-22 08:53:55 +0000 | [diff] [blame] | 3985 | int Channel::GetRemoteRTCPSenderInfo(SenderInfo* sender_info) { |
| 3986 | if (sender_info == NULL) { |
| 3987 | _engineStatisticsPtr->SetLastError(VE_INVALID_ARGUMENT, kTraceError, |
| 3988 | "GetRemoteRTCPSenderInfo() invalid sender_info."); |
| 3989 | return -1; |
| 3990 | } |
| 3991 | |
| 3992 | // Get the sender info from the latest received RTCP Sender Report. |
| 3993 | RTCPSenderInfo rtcp_sender_info; |
| 3994 | if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_sender_info) != 0) { |
| 3995 | _engineStatisticsPtr->SetLastError(VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 3996 | "GetRemoteRTCPSenderInfo() failed to read RTCP SR sender info."); |
| 3997 | return -1; |
| 3998 | } |
| 3999 | |
| 4000 | sender_info->NTP_timestamp_high = rtcp_sender_info.NTPseconds; |
| 4001 | sender_info->NTP_timestamp_low = rtcp_sender_info.NTPfraction; |
| 4002 | sender_info->RTP_timestamp = rtcp_sender_info.RTPtimeStamp; |
| 4003 | sender_info->sender_packet_count = rtcp_sender_info.sendPacketCount; |
| 4004 | sender_info->sender_octet_count = rtcp_sender_info.sendOctetCount; |
| 4005 | return 0; |
| 4006 | } |
| 4007 | |
| 4008 | int Channel::GetRemoteRTCPReportBlocks( |
| 4009 | std::vector<ReportBlock>* report_blocks) { |
| 4010 | if (report_blocks == NULL) { |
| 4011 | _engineStatisticsPtr->SetLastError(VE_INVALID_ARGUMENT, kTraceError, |
| 4012 | "GetRemoteRTCPReportBlock()s invalid report_blocks."); |
| 4013 | return -1; |
| 4014 | } |
| 4015 | |
| 4016 | // Get the report blocks from the latest received RTCP Sender or Receiver |
| 4017 | // Report. Each element in the vector contains the sender's SSRC and a |
| 4018 | // report block according to RFC 3550. |
| 4019 | std::vector<RTCPReportBlock> rtcp_report_blocks; |
| 4020 | if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks) != 0) { |
| 4021 | _engineStatisticsPtr->SetLastError(VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 4022 | "GetRemoteRTCPReportBlocks() failed to read RTCP SR/RR report block."); |
| 4023 | return -1; |
| 4024 | } |
| 4025 | |
| 4026 | if (rtcp_report_blocks.empty()) |
| 4027 | return 0; |
| 4028 | |
| 4029 | std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin(); |
| 4030 | for (; it != rtcp_report_blocks.end(); ++it) { |
| 4031 | ReportBlock report_block; |
| 4032 | report_block.sender_SSRC = it->remoteSSRC; |
| 4033 | report_block.source_SSRC = it->sourceSSRC; |
| 4034 | report_block.fraction_lost = it->fractionLost; |
| 4035 | report_block.cumulative_num_packets_lost = it->cumulativeLost; |
| 4036 | report_block.extended_highest_sequence_number = it->extendedHighSeqNum; |
| 4037 | report_block.interarrival_jitter = it->jitter; |
| 4038 | report_block.last_SR_timestamp = it->lastSR; |
| 4039 | report_block.delay_since_last_SR = it->delaySinceLastSR; |
| 4040 | report_blocks->push_back(report_block); |
| 4041 | } |
| 4042 | return 0; |
| 4043 | } |
| 4044 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4045 | int |
| 4046 | Channel::GetRTPStatistics(CallStatistics& stats) |
| 4047 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4048 | uint8_t fraction_lost(0); |
| 4049 | uint32_t cum_lost(0); |
| 4050 | uint32_t ext_max(0); |
| 4051 | uint32_t jitter(0); |
| 4052 | uint32_t max_jitter(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4053 | |
| 4054 | // --- Part one of the final structure (four values) |
| 4055 | |
| 4056 | // The jitter statistics is updated for each received RTP packet and is |
| 4057 | // based on received packets. |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4058 | if (_rtpRtcpModule->StatisticsRTP(&fraction_lost, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4059 | &cum_lost, |
| 4060 | &ext_max, |
| 4061 | &jitter, |
| 4062 | &max_jitter) != 0) |
| 4063 | { |
| 4064 | _engineStatisticsPtr->SetLastError( |
| 4065 | VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning, |
| 4066 | "GetRTPStatistics() failed to read RTP statistics from the " |
| 4067 | "RTP/RTCP module"); |
| 4068 | } |
| 4069 | |
| 4070 | stats.fractionLost = fraction_lost; |
| 4071 | stats.cumulativeLost = cum_lost; |
| 4072 | stats.extendedMax = ext_max; |
| 4073 | stats.jitterSamples = jitter; |
| 4074 | |
| 4075 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4076 | VoEId(_instanceId, _channelId), |
| 4077 | "GetRTPStatistics() => fractionLost=%lu, cumulativeLost=%lu," |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 4078 | " extendedMax=%lu, jitterSamples=%li)", |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4079 | stats.fractionLost, stats.cumulativeLost, stats.extendedMax, |
| 4080 | stats.jitterSamples); |
| 4081 | |
| 4082 | // --- Part two of the final structure (one value) |
| 4083 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4084 | uint16_t RTT(0); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4085 | RTCPMethod method = _rtpRtcpModule->RTCP(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4086 | if (method == kRtcpOff) |
| 4087 | { |
| 4088 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4089 | VoEId(_instanceId, _channelId), |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 4090 | "GetRTPStatistics() RTCP is disabled => valid RTT " |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4091 | "measurements cannot be retrieved"); |
| 4092 | } else |
| 4093 | { |
| 4094 | // The remote SSRC will be zero if no RTP packet has been received. |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4095 | uint32_t remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4096 | if (remoteSSRC > 0) |
| 4097 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4098 | uint16_t avgRTT(0); |
| 4099 | uint16_t maxRTT(0); |
| 4100 | uint16_t minRTT(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4101 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4102 | if (_rtpRtcpModule->RTT(remoteSSRC, &RTT, &avgRTT, &minRTT, &maxRTT) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4103 | != 0) |
| 4104 | { |
| 4105 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4106 | VoEId(_instanceId, _channelId), |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 4107 | "GetRTPStatistics() failed to retrieve RTT from " |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4108 | "the RTP/RTCP module"); |
| 4109 | } |
| 4110 | } else |
| 4111 | { |
| 4112 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4113 | VoEId(_instanceId, _channelId), |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 4114 | "GetRTPStatistics() failed to measure RTT since no " |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4115 | "RTP packets have been received yet"); |
| 4116 | } |
| 4117 | } |
| 4118 | |
| 4119 | stats.rttMs = static_cast<int> (RTT); |
| 4120 | |
| 4121 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4122 | VoEId(_instanceId, _channelId), |
| 4123 | "GetRTPStatistics() => rttMs=%d", stats.rttMs); |
| 4124 | |
| 4125 | // --- Part three of the final structure (four values) |
| 4126 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4127 | uint32_t bytesSent(0); |
| 4128 | uint32_t packetsSent(0); |
| 4129 | uint32_t bytesReceived(0); |
| 4130 | uint32_t packetsReceived(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4131 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4132 | if (_rtpRtcpModule->DataCountersRTP(&bytesSent, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4133 | &packetsSent, |
| 4134 | &bytesReceived, |
| 4135 | &packetsReceived) != 0) |
| 4136 | { |
| 4137 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4138 | VoEId(_instanceId, _channelId), |
| 4139 | "GetRTPStatistics() failed to retrieve RTP datacounters =>" |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 4140 | " output will not be complete"); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4141 | } |
| 4142 | |
| 4143 | stats.bytesSent = bytesSent; |
| 4144 | stats.packetsSent = packetsSent; |
| 4145 | stats.bytesReceived = bytesReceived; |
| 4146 | stats.packetsReceived = packetsReceived; |
| 4147 | |
| 4148 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4149 | VoEId(_instanceId, _channelId), |
| 4150 | "GetRTPStatistics() => bytesSent=%d, packetsSent=%d," |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 4151 | " bytesReceived=%d, packetsReceived=%d)", |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4152 | stats.bytesSent, stats.packetsSent, stats.bytesReceived, |
| 4153 | stats.packetsReceived); |
| 4154 | |
| 4155 | return 0; |
| 4156 | } |
| 4157 | |
turaj@webrtc.org | 42259e7 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 4158 | int Channel::SetFECStatus(bool enable, int redPayloadtype) { |
| 4159 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 4160 | "Channel::SetFECStatus()"); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4161 | |
turaj@webrtc.org | 8c8ad85 | 2013-01-31 18:20:17 +0000 | [diff] [blame] | 4162 | if (enable) { |
| 4163 | if (redPayloadtype < 0 || redPayloadtype > 127) { |
| 4164 | _engineStatisticsPtr->SetLastError( |
| 4165 | VE_PLTYPE_ERROR, kTraceError, |
| 4166 | "SetFECStatus() invalid RED payload type"); |
| 4167 | return -1; |
| 4168 | } |
| 4169 | |
| 4170 | if (SetRedPayloadType(redPayloadtype) < 0) { |
| 4171 | _engineStatisticsPtr->SetLastError( |
| 4172 | VE_CODEC_ERROR, kTraceError, |
| 4173 | "SetSecondarySendCodec() Failed to register RED ACM"); |
| 4174 | return -1; |
| 4175 | } |
turaj@webrtc.org | 42259e7 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 4176 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4177 | |
turaj@webrtc.org | 42259e7 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 4178 | if (_audioCodingModule.SetFECStatus(enable) != 0) { |
| 4179 | _engineStatisticsPtr->SetLastError( |
| 4180 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 4181 | "SetFECStatus() failed to set FEC state in the ACM"); |
| 4182 | return -1; |
| 4183 | } |
| 4184 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4185 | } |
| 4186 | |
| 4187 | int |
| 4188 | Channel::GetFECStatus(bool& enabled, int& redPayloadtype) |
| 4189 | { |
| 4190 | enabled = _audioCodingModule.FECStatus(); |
| 4191 | if (enabled) |
| 4192 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4193 | int8_t payloadType(0); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4194 | if (_rtpRtcpModule->SendREDPayloadType(payloadType) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4195 | { |
| 4196 | _engineStatisticsPtr->SetLastError( |
| 4197 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 4198 | "GetFECStatus() failed to retrieve RED PT from RTP/RTCP " |
| 4199 | "module"); |
| 4200 | return -1; |
| 4201 | } |
| 4202 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4203 | VoEId(_instanceId, _channelId), |
| 4204 | "GetFECStatus() => enabled=%d, redPayloadtype=%d", |
| 4205 | enabled, redPayloadtype); |
| 4206 | return 0; |
| 4207 | } |
| 4208 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4209 | VoEId(_instanceId, _channelId), |
| 4210 | "GetFECStatus() => enabled=%d", enabled); |
| 4211 | return 0; |
| 4212 | } |
| 4213 | |
| 4214 | int |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4215 | Channel::StartRTPDump(const char fileNameUTF8[1024], |
| 4216 | RTPDirections direction) |
| 4217 | { |
| 4218 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 4219 | "Channel::StartRTPDump()"); |
| 4220 | if ((direction != kRtpIncoming) && (direction != kRtpOutgoing)) |
| 4221 | { |
| 4222 | _engineStatisticsPtr->SetLastError( |
| 4223 | VE_INVALID_ARGUMENT, kTraceError, |
| 4224 | "StartRTPDump() invalid RTP direction"); |
| 4225 | return -1; |
| 4226 | } |
| 4227 | RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ? |
| 4228 | &_rtpDumpIn : &_rtpDumpOut; |
| 4229 | if (rtpDumpPtr == NULL) |
| 4230 | { |
| 4231 | assert(false); |
| 4232 | return -1; |
| 4233 | } |
| 4234 | if (rtpDumpPtr->IsActive()) |
| 4235 | { |
| 4236 | rtpDumpPtr->Stop(); |
| 4237 | } |
| 4238 | if (rtpDumpPtr->Start(fileNameUTF8) != 0) |
| 4239 | { |
| 4240 | _engineStatisticsPtr->SetLastError( |
| 4241 | VE_BAD_FILE, kTraceError, |
| 4242 | "StartRTPDump() failed to create file"); |
| 4243 | return -1; |
| 4244 | } |
| 4245 | return 0; |
| 4246 | } |
| 4247 | |
| 4248 | int |
| 4249 | Channel::StopRTPDump(RTPDirections direction) |
| 4250 | { |
| 4251 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 4252 | "Channel::StopRTPDump()"); |
| 4253 | if ((direction != kRtpIncoming) && (direction != kRtpOutgoing)) |
| 4254 | { |
| 4255 | _engineStatisticsPtr->SetLastError( |
| 4256 | VE_INVALID_ARGUMENT, kTraceError, |
| 4257 | "StopRTPDump() invalid RTP direction"); |
| 4258 | return -1; |
| 4259 | } |
| 4260 | RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ? |
| 4261 | &_rtpDumpIn : &_rtpDumpOut; |
| 4262 | if (rtpDumpPtr == NULL) |
| 4263 | { |
| 4264 | assert(false); |
| 4265 | return -1; |
| 4266 | } |
| 4267 | if (!rtpDumpPtr->IsActive()) |
| 4268 | { |
| 4269 | return 0; |
| 4270 | } |
| 4271 | return rtpDumpPtr->Stop(); |
| 4272 | } |
| 4273 | |
| 4274 | bool |
| 4275 | Channel::RTPDumpIsActive(RTPDirections direction) |
| 4276 | { |
| 4277 | if ((direction != kRtpIncoming) && |
| 4278 | (direction != kRtpOutgoing)) |
| 4279 | { |
| 4280 | _engineStatisticsPtr->SetLastError( |
| 4281 | VE_INVALID_ARGUMENT, kTraceError, |
| 4282 | "RTPDumpIsActive() invalid RTP direction"); |
| 4283 | return false; |
| 4284 | } |
| 4285 | RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ? |
| 4286 | &_rtpDumpIn : &_rtpDumpOut; |
| 4287 | return rtpDumpPtr->IsActive(); |
| 4288 | } |
| 4289 | |
| 4290 | int |
| 4291 | Channel::InsertExtraRTPPacket(unsigned char payloadType, |
| 4292 | bool markerBit, |
| 4293 | const char* payloadData, |
| 4294 | unsigned short payloadSize) |
| 4295 | { |
| 4296 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
| 4297 | "Channel::InsertExtraRTPPacket()"); |
| 4298 | if (payloadType > 127) |
| 4299 | { |
| 4300 | _engineStatisticsPtr->SetLastError( |
| 4301 | VE_INVALID_PLTYPE, kTraceError, |
| 4302 | "InsertExtraRTPPacket() invalid payload type"); |
| 4303 | return -1; |
| 4304 | } |
| 4305 | if (payloadData == NULL) |
| 4306 | { |
| 4307 | _engineStatisticsPtr->SetLastError( |
| 4308 | VE_INVALID_ARGUMENT, kTraceError, |
| 4309 | "InsertExtraRTPPacket() invalid payload data"); |
| 4310 | return -1; |
| 4311 | } |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4312 | if (payloadSize > _rtpRtcpModule->MaxDataPayloadLength()) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4313 | { |
| 4314 | _engineStatisticsPtr->SetLastError( |
| 4315 | VE_INVALID_ARGUMENT, kTraceError, |
| 4316 | "InsertExtraRTPPacket() invalid payload size"); |
| 4317 | return -1; |
| 4318 | } |
| 4319 | if (!_sending) |
| 4320 | { |
| 4321 | _engineStatisticsPtr->SetLastError( |
| 4322 | VE_NOT_SENDING, kTraceError, |
| 4323 | "InsertExtraRTPPacket() not sending"); |
| 4324 | return -1; |
| 4325 | } |
| 4326 | |
| 4327 | // Create extra RTP packet by calling RtpRtcp::SendOutgoingData(). |
| 4328 | // Transport::SendPacket() will be called by the module when the RTP packet |
| 4329 | // is created. |
| 4330 | // The call to SendOutgoingData() does *not* modify the timestamp and |
| 4331 | // payloadtype to ensure that the RTP module generates a valid RTP packet |
| 4332 | // (user might utilize a non-registered payload type). |
| 4333 | // The marker bit and payload type will be replaced just before the actual |
| 4334 | // transmission, i.e., the actual modification is done *after* the RTP |
| 4335 | // module has delivered its RTP packet back to the VoE. |
| 4336 | // We will use the stored values above when the packet is modified |
| 4337 | // (see Channel::SendPacket()). |
| 4338 | |
| 4339 | _extraPayloadType = payloadType; |
| 4340 | _extraMarkerBit = markerBit; |
| 4341 | _insertExtraRTPPacket = true; |
| 4342 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4343 | if (_rtpRtcpModule->SendOutgoingData(kAudioFrameSpeech, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4344 | _lastPayloadType, |
| 4345 | _lastLocalTimeStamp, |
stefan@webrtc.org | ddfdfed | 2012-07-03 13:21:22 +0000 | [diff] [blame] | 4346 | // Leaving the time when this frame was |
| 4347 | // received from the capture device as |
| 4348 | // undefined for voice for now. |
| 4349 | -1, |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4350 | (const uint8_t*) payloadData, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4351 | payloadSize) != 0) |
| 4352 | { |
| 4353 | _engineStatisticsPtr->SetLastError( |
| 4354 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 4355 | "InsertExtraRTPPacket() failed to send extra RTP packet"); |
| 4356 | return -1; |
| 4357 | } |
| 4358 | |
| 4359 | return 0; |
| 4360 | } |
| 4361 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4362 | uint32_t |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 4363 | Channel::Demultiplex(const AudioFrame& audioFrame) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4364 | { |
| 4365 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 4366 | "Channel::Demultiplex()"); |
andrew@webrtc.org | ae1a58b | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 4367 | _audioFrame.CopyFrom(audioFrame); |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4368 | _audioFrame.id_ = _channelId; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4369 | return 0; |
| 4370 | } |
| 4371 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4372 | uint32_t |
xians@google.com | 0b0665a | 2011-08-08 08:18:44 +0000 | [diff] [blame] | 4373 | Channel::PrepareEncodeAndSend(int mixingFrequency) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4374 | { |
| 4375 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4376 | "Channel::PrepareEncodeAndSend()"); |
| 4377 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4378 | if (_audioFrame.samples_per_channel_ == 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4379 | { |
| 4380 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4381 | "Channel::PrepareEncodeAndSend() invalid audio frame"); |
| 4382 | return -1; |
| 4383 | } |
| 4384 | |
| 4385 | if (_inputFilePlaying) |
| 4386 | { |
| 4387 | MixOrReplaceAudioWithFile(mixingFrequency); |
| 4388 | } |
| 4389 | |
| 4390 | if (_mute) |
| 4391 | { |
| 4392 | AudioFrameOperations::Mute(_audioFrame); |
| 4393 | } |
| 4394 | |
| 4395 | if (_inputExternalMedia) |
| 4396 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 4397 | CriticalSectionScoped cs(&_callbackCritSect); |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4398 | const bool isStereo = (_audioFrame.num_channels_ == 2); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4399 | if (_inputExternalMediaCallbackPtr) |
| 4400 | { |
| 4401 | _inputExternalMediaCallbackPtr->Process( |
| 4402 | _channelId, |
| 4403 | kRecordingPerChannel, |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4404 | (int16_t*)_audioFrame.data_, |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4405 | _audioFrame.samples_per_channel_, |
| 4406 | _audioFrame.sample_rate_hz_, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4407 | isStereo); |
| 4408 | } |
| 4409 | } |
| 4410 | |
| 4411 | InsertInbandDtmfTone(); |
| 4412 | |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 4413 | if (_includeAudioLevelIndication) |
| 4414 | { |
| 4415 | assert(_rtpAudioProc.get() != NULL); |
| 4416 | |
| 4417 | // Check if settings need to be updated. |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4418 | if (_rtpAudioProc->sample_rate_hz() != _audioFrame.sample_rate_hz_) |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 4419 | { |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4420 | if (_rtpAudioProc->set_sample_rate_hz(_audioFrame.sample_rate_hz_) != |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 4421 | AudioProcessing::kNoError) |
| 4422 | { |
| 4423 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4424 | VoEId(_instanceId, _channelId), |
| 4425 | "Error setting AudioProcessing sample rate"); |
| 4426 | return -1; |
| 4427 | } |
| 4428 | } |
| 4429 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4430 | if (_rtpAudioProc->num_input_channels() != _audioFrame.num_channels_) |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 4431 | { |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4432 | if (_rtpAudioProc->set_num_channels(_audioFrame.num_channels_, |
| 4433 | _audioFrame.num_channels_) |
andrew@webrtc.org | 755b04a | 2011-11-15 16:57:56 +0000 | [diff] [blame] | 4434 | != AudioProcessing::kNoError) |
| 4435 | { |
| 4436 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4437 | VoEId(_instanceId, _channelId), |
| 4438 | "Error setting AudioProcessing channels"); |
| 4439 | return -1; |
| 4440 | } |
| 4441 | } |
| 4442 | |
| 4443 | // Performs level analysis only; does not affect the signal. |
| 4444 | _rtpAudioProc->ProcessStream(&_audioFrame); |
| 4445 | } |
| 4446 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4447 | return 0; |
| 4448 | } |
| 4449 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4450 | uint32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4451 | Channel::EncodeAndSend() |
| 4452 | { |
| 4453 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4454 | "Channel::EncodeAndSend()"); |
| 4455 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4456 | assert(_audioFrame.num_channels_ <= 2); |
| 4457 | if (_audioFrame.samples_per_channel_ == 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4458 | { |
| 4459 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4460 | "Channel::EncodeAndSend() invalid audio frame"); |
| 4461 | return -1; |
| 4462 | } |
| 4463 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4464 | _audioFrame.id_ = _channelId; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4465 | |
| 4466 | // --- Add 10ms of raw (PCM) audio data to the encoder @ 32kHz. |
| 4467 | |
| 4468 | // The ACM resamples internally. |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4469 | _audioFrame.timestamp_ = _timeStamp; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4470 | if (_audioCodingModule.Add10MsData((AudioFrame&)_audioFrame) != 0) |
| 4471 | { |
| 4472 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4473 | "Channel::EncodeAndSend() ACM encoding failed"); |
| 4474 | return -1; |
| 4475 | } |
| 4476 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4477 | _timeStamp += _audioFrame.samples_per_channel_; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4478 | |
| 4479 | // --- Encode if complete frame is ready |
| 4480 | |
| 4481 | // This call will trigger AudioPacketizationCallback::SendData if encoding |
| 4482 | // is done and payload is ready for packetization and transmission. |
| 4483 | return _audioCodingModule.Process(); |
| 4484 | } |
| 4485 | |
| 4486 | int Channel::RegisterExternalMediaProcessing( |
| 4487 | ProcessingTypes type, |
| 4488 | VoEMediaProcess& processObject) |
| 4489 | { |
| 4490 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4491 | "Channel::RegisterExternalMediaProcessing()"); |
| 4492 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 4493 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4494 | |
| 4495 | if (kPlaybackPerChannel == type) |
| 4496 | { |
| 4497 | if (_outputExternalMediaCallbackPtr) |
| 4498 | { |
| 4499 | _engineStatisticsPtr->SetLastError( |
| 4500 | VE_INVALID_OPERATION, kTraceError, |
| 4501 | "Channel::RegisterExternalMediaProcessing() " |
| 4502 | "output external media already enabled"); |
| 4503 | return -1; |
| 4504 | } |
| 4505 | _outputExternalMediaCallbackPtr = &processObject; |
| 4506 | _outputExternalMedia = true; |
| 4507 | } |
| 4508 | else if (kRecordingPerChannel == type) |
| 4509 | { |
| 4510 | if (_inputExternalMediaCallbackPtr) |
| 4511 | { |
| 4512 | _engineStatisticsPtr->SetLastError( |
| 4513 | VE_INVALID_OPERATION, kTraceError, |
| 4514 | "Channel::RegisterExternalMediaProcessing() " |
| 4515 | "output external media already enabled"); |
| 4516 | return -1; |
| 4517 | } |
| 4518 | _inputExternalMediaCallbackPtr = &processObject; |
| 4519 | _inputExternalMedia = true; |
| 4520 | } |
| 4521 | return 0; |
| 4522 | } |
| 4523 | |
| 4524 | int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type) |
| 4525 | { |
| 4526 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4527 | "Channel::DeRegisterExternalMediaProcessing()"); |
| 4528 | |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 4529 | CriticalSectionScoped cs(&_callbackCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4530 | |
| 4531 | if (kPlaybackPerChannel == type) |
| 4532 | { |
| 4533 | if (!_outputExternalMediaCallbackPtr) |
| 4534 | { |
| 4535 | _engineStatisticsPtr->SetLastError( |
| 4536 | VE_INVALID_OPERATION, kTraceWarning, |
| 4537 | "Channel::DeRegisterExternalMediaProcessing() " |
| 4538 | "output external media already disabled"); |
| 4539 | return 0; |
| 4540 | } |
| 4541 | _outputExternalMedia = false; |
| 4542 | _outputExternalMediaCallbackPtr = NULL; |
| 4543 | } |
| 4544 | else if (kRecordingPerChannel == type) |
| 4545 | { |
| 4546 | if (!_inputExternalMediaCallbackPtr) |
| 4547 | { |
| 4548 | _engineStatisticsPtr->SetLastError( |
| 4549 | VE_INVALID_OPERATION, kTraceWarning, |
| 4550 | "Channel::DeRegisterExternalMediaProcessing() " |
| 4551 | "input external media already disabled"); |
| 4552 | return 0; |
| 4553 | } |
| 4554 | _inputExternalMedia = false; |
| 4555 | _inputExternalMediaCallbackPtr = NULL; |
| 4556 | } |
| 4557 | |
| 4558 | return 0; |
| 4559 | } |
| 4560 | |
roosa@google.com | 1b60ceb | 2012-12-12 23:00:29 +0000 | [diff] [blame] | 4561 | int Channel::SetExternalMixing(bool enabled) { |
| 4562 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4563 | "Channel::SetExternalMixing(enabled=%d)", enabled); |
| 4564 | |
| 4565 | if (_playing) |
| 4566 | { |
| 4567 | _engineStatisticsPtr->SetLastError( |
| 4568 | VE_INVALID_OPERATION, kTraceError, |
| 4569 | "Channel::SetExternalMixing() " |
| 4570 | "external mixing cannot be changed while playing."); |
| 4571 | return -1; |
| 4572 | } |
| 4573 | |
| 4574 | _externalMixing = enabled; |
| 4575 | |
| 4576 | return 0; |
| 4577 | } |
| 4578 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4579 | int |
| 4580 | Channel::ResetRTCPStatistics() |
| 4581 | { |
| 4582 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4583 | "Channel::ResetRTCPStatistics()"); |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4584 | uint32_t remoteSSRC(0); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4585 | remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
| 4586 | return _rtpRtcpModule->ResetRTT(remoteSSRC); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4587 | } |
| 4588 | |
| 4589 | int |
| 4590 | Channel::GetRoundTripTimeSummary(StatVal& delaysMs) const |
| 4591 | { |
| 4592 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4593 | "Channel::GetRoundTripTimeSummary()"); |
| 4594 | // Override default module outputs for the case when RTCP is disabled. |
| 4595 | // This is done to ensure that we are backward compatible with the |
| 4596 | // VoiceEngine where we did not use RTP/RTCP module. |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4597 | if (!_rtpRtcpModule->RTCP()) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4598 | { |
| 4599 | delaysMs.min = -1; |
| 4600 | delaysMs.max = -1; |
| 4601 | delaysMs.average = -1; |
| 4602 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4603 | "Channel::GetRoundTripTimeSummary() RTCP is disabled =>" |
| 4604 | " valid RTT measurements cannot be retrieved"); |
| 4605 | return 0; |
| 4606 | } |
| 4607 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4608 | uint32_t remoteSSRC; |
| 4609 | uint16_t RTT; |
| 4610 | uint16_t avgRTT; |
| 4611 | uint16_t maxRTT; |
| 4612 | uint16_t minRTT; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4613 | // The remote SSRC will be zero if no RTP packet has been received. |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4614 | remoteSSRC = _rtpRtcpModule->RemoteSSRC(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4615 | if (remoteSSRC == 0) |
| 4616 | { |
| 4617 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4618 | "Channel::GetRoundTripTimeSummary() unable to measure RTT" |
| 4619 | " since no RTP packet has been received yet"); |
| 4620 | } |
| 4621 | |
| 4622 | // Retrieve RTT statistics from the RTP/RTCP module for the specified |
| 4623 | // channel and SSRC. The SSRC is required to parse out the correct source |
| 4624 | // in conference scenarios. |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4625 | if (_rtpRtcpModule->RTT(remoteSSRC, &RTT, &avgRTT, &minRTT,&maxRTT) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4626 | { |
| 4627 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4628 | "GetRoundTripTimeSummary unable to retrieve RTT values" |
| 4629 | " from the RTCP layer"); |
| 4630 | delaysMs.min = -1; delaysMs.max = -1; delaysMs.average = -1; |
| 4631 | } |
| 4632 | else |
| 4633 | { |
| 4634 | delaysMs.min = minRTT; |
| 4635 | delaysMs.max = maxRTT; |
| 4636 | delaysMs.average = avgRTT; |
| 4637 | } |
| 4638 | return 0; |
| 4639 | } |
| 4640 | |
| 4641 | int |
| 4642 | Channel::GetNetworkStatistics(NetworkStatistics& stats) |
| 4643 | { |
| 4644 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4645 | "Channel::GetNetworkStatistics()"); |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 4646 | ACMNetworkStatistics acm_stats; |
| 4647 | int return_value = _audioCodingModule.NetworkStatistics(&acm_stats); |
| 4648 | if (return_value >= 0) { |
| 4649 | memcpy(&stats, &acm_stats, sizeof(NetworkStatistics)); |
| 4650 | } |
| 4651 | return return_value; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4652 | } |
| 4653 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 4654 | bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms, |
| 4655 | int* playout_buffer_delay_ms) const { |
| 4656 | if (_average_jitter_buffer_delay_us == 0) { |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4657 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 4658 | "Channel::GetDelayEstimate() no valid estimate."); |
| 4659 | return false; |
| 4660 | } |
| 4661 | *jitter_buffer_delay_ms = (_average_jitter_buffer_delay_us + 500) / 1000 + |
| 4662 | _recPacketDelayMs; |
| 4663 | *playout_buffer_delay_ms = playout_delay_ms_; |
| 4664 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4665 | "Channel::GetDelayEstimate()"); |
| 4666 | return true; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4667 | } |
| 4668 | |
turaj@webrtc.org | 6388c3e | 2013-02-12 21:42:18 +0000 | [diff] [blame] | 4669 | int Channel::SetInitialPlayoutDelay(int delay_ms) |
| 4670 | { |
| 4671 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4672 | "Channel::SetInitialPlayoutDelay()"); |
| 4673 | if ((delay_ms < kVoiceEngineMinMinPlayoutDelayMs) || |
| 4674 | (delay_ms > kVoiceEngineMaxMinPlayoutDelayMs)) |
| 4675 | { |
| 4676 | _engineStatisticsPtr->SetLastError( |
| 4677 | VE_INVALID_ARGUMENT, kTraceError, |
| 4678 | "SetInitialPlayoutDelay() invalid min delay"); |
| 4679 | return -1; |
| 4680 | } |
| 4681 | if (_audioCodingModule.SetInitialPlayoutDelay(delay_ms) != 0) |
| 4682 | { |
| 4683 | _engineStatisticsPtr->SetLastError( |
| 4684 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 4685 | "SetInitialPlayoutDelay() failed to set min playout delay"); |
| 4686 | return -1; |
| 4687 | } |
| 4688 | return 0; |
| 4689 | } |
| 4690 | |
| 4691 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4692 | int |
| 4693 | Channel::SetMinimumPlayoutDelay(int delayMs) |
| 4694 | { |
| 4695 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4696 | "Channel::SetMinimumPlayoutDelay()"); |
| 4697 | if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) || |
| 4698 | (delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) |
| 4699 | { |
| 4700 | _engineStatisticsPtr->SetLastError( |
| 4701 | VE_INVALID_ARGUMENT, kTraceError, |
| 4702 | "SetMinimumPlayoutDelay() invalid min delay"); |
| 4703 | return -1; |
| 4704 | } |
| 4705 | if (_audioCodingModule.SetMinimumPlayoutDelay(delayMs) != 0) |
| 4706 | { |
| 4707 | _engineStatisticsPtr->SetLastError( |
| 4708 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 4709 | "SetMinimumPlayoutDelay() failed to set min playout delay"); |
| 4710 | return -1; |
| 4711 | } |
| 4712 | return 0; |
| 4713 | } |
| 4714 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 4715 | void Channel::UpdatePlayoutTimestamp(bool rtcp) { |
| 4716 | uint32_t playout_timestamp = 0; |
| 4717 | |
| 4718 | if (_audioCodingModule.PlayoutTimestamp(&playout_timestamp) == -1) { |
| 4719 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4720 | "Channel::UpdatePlayoutTimestamp() failed to read playout" |
| 4721 | " timestamp from the ACM"); |
| 4722 | _engineStatisticsPtr->SetLastError( |
| 4723 | VE_CANNOT_RETRIEVE_VALUE, kTraceError, |
| 4724 | "UpdatePlayoutTimestamp() failed to retrieve timestamp"); |
| 4725 | return; |
| 4726 | } |
| 4727 | |
| 4728 | uint16_t delay_ms = 0; |
| 4729 | if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) { |
| 4730 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4731 | "Channel::UpdatePlayoutTimestamp() failed to read playout" |
| 4732 | " delay from the ADM"); |
| 4733 | _engineStatisticsPtr->SetLastError( |
| 4734 | VE_CANNOT_RETRIEVE_VALUE, kTraceError, |
| 4735 | "UpdatePlayoutTimestamp() failed to retrieve playout delay"); |
| 4736 | return; |
| 4737 | } |
| 4738 | |
| 4739 | int32_t playout_frequency = _audioCodingModule.PlayoutFrequency(); |
| 4740 | CodecInst current_recive_codec; |
| 4741 | if (_audioCodingModule.ReceiveCodec(¤t_recive_codec) == 0) { |
| 4742 | if (STR_CASE_CMP("G722", current_recive_codec.plname) == 0) { |
| 4743 | playout_frequency = 8000; |
| 4744 | } else if (STR_CASE_CMP("opus", current_recive_codec.plname) == 0) { |
| 4745 | playout_frequency = 48000; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4746 | } |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 4747 | } |
| 4748 | |
| 4749 | // Remove the playout delay. |
| 4750 | playout_timestamp -= (delay_ms * (playout_frequency / 1000)); |
| 4751 | |
| 4752 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4753 | "Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu", |
| 4754 | playout_timestamp); |
| 4755 | |
| 4756 | if (rtcp) { |
| 4757 | playout_timestamp_rtcp_ = playout_timestamp; |
| 4758 | } else { |
| 4759 | playout_timestamp_rtp_ = playout_timestamp; |
| 4760 | } |
| 4761 | playout_delay_ms_ = delay_ms; |
| 4762 | } |
| 4763 | |
| 4764 | int Channel::GetPlayoutTimestamp(unsigned int& timestamp) { |
| 4765 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4766 | "Channel::GetPlayoutTimestamp()"); |
| 4767 | if (playout_timestamp_rtp_ == 0) { |
| 4768 | _engineStatisticsPtr->SetLastError( |
| 4769 | VE_CANNOT_RETRIEVE_VALUE, kTraceError, |
| 4770 | "GetPlayoutTimestamp() failed to retrieve timestamp"); |
| 4771 | return -1; |
| 4772 | } |
| 4773 | timestamp = playout_timestamp_rtp_; |
| 4774 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| 4775 | VoEId(_instanceId,_channelId), |
| 4776 | "GetPlayoutTimestamp() => timestamp=%u", timestamp); |
| 4777 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4778 | } |
| 4779 | |
| 4780 | int |
| 4781 | Channel::SetInitTimestamp(unsigned int timestamp) |
| 4782 | { |
| 4783 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4784 | "Channel::SetInitTimestamp()"); |
| 4785 | if (_sending) |
| 4786 | { |
| 4787 | _engineStatisticsPtr->SetLastError( |
| 4788 | VE_SENDING, kTraceError, "SetInitTimestamp() already sending"); |
| 4789 | return -1; |
| 4790 | } |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4791 | if (_rtpRtcpModule->SetStartTimestamp(timestamp) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4792 | { |
| 4793 | _engineStatisticsPtr->SetLastError( |
| 4794 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 4795 | "SetInitTimestamp() failed to set timestamp"); |
| 4796 | return -1; |
| 4797 | } |
| 4798 | return 0; |
| 4799 | } |
| 4800 | |
| 4801 | int |
| 4802 | Channel::SetInitSequenceNumber(short sequenceNumber) |
| 4803 | { |
| 4804 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4805 | "Channel::SetInitSequenceNumber()"); |
| 4806 | if (_sending) |
| 4807 | { |
| 4808 | _engineStatisticsPtr->SetLastError( |
| 4809 | VE_SENDING, kTraceError, |
| 4810 | "SetInitSequenceNumber() already sending"); |
| 4811 | return -1; |
| 4812 | } |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4813 | if (_rtpRtcpModule->SetSequenceNumber(sequenceNumber) != 0) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4814 | { |
| 4815 | _engineStatisticsPtr->SetLastError( |
| 4816 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 4817 | "SetInitSequenceNumber() failed to set sequence number"); |
| 4818 | return -1; |
| 4819 | } |
| 4820 | return 0; |
| 4821 | } |
| 4822 | |
| 4823 | int |
| 4824 | Channel::GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const |
| 4825 | { |
| 4826 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 4827 | "Channel::GetRtpRtcp()"); |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 4828 | rtpRtcpModule = _rtpRtcpModule.get(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4829 | return 0; |
| 4830 | } |
| 4831 | |
andrew@webrtc.org | e59a0ac | 2012-05-08 17:12:40 +0000 | [diff] [blame] | 4832 | // TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use |
| 4833 | // a shared helper. |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4834 | int32_t |
xians@google.com | 0b0665a | 2011-08-08 08:18:44 +0000 | [diff] [blame] | 4835 | Channel::MixOrReplaceAudioWithFile(const int mixingFrequency) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4836 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4837 | scoped_array<int16_t> fileBuffer(new int16_t[640]); |
andrew@webrtc.org | e59a0ac | 2012-05-08 17:12:40 +0000 | [diff] [blame] | 4838 | int fileSamples(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4839 | |
| 4840 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 4841 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4842 | |
| 4843 | if (_inputFilePlayerPtr == NULL) |
| 4844 | { |
| 4845 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4846 | VoEId(_instanceId, _channelId), |
| 4847 | "Channel::MixOrReplaceAudioWithFile() fileplayer" |
| 4848 | " doesnt exist"); |
| 4849 | return -1; |
| 4850 | } |
| 4851 | |
braveyao@webrtc.org | d713143 | 2012-03-29 10:39:44 +0000 | [diff] [blame] | 4852 | if (_inputFilePlayerPtr->Get10msAudioFromFile(fileBuffer.get(), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4853 | fileSamples, |
| 4854 | mixingFrequency) == -1) |
| 4855 | { |
| 4856 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4857 | VoEId(_instanceId, _channelId), |
| 4858 | "Channel::MixOrReplaceAudioWithFile() file mixing " |
| 4859 | "failed"); |
| 4860 | return -1; |
| 4861 | } |
| 4862 | if (fileSamples == 0) |
| 4863 | { |
| 4864 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4865 | VoEId(_instanceId, _channelId), |
| 4866 | "Channel::MixOrReplaceAudioWithFile() file is ended"); |
| 4867 | return 0; |
| 4868 | } |
| 4869 | } |
| 4870 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4871 | assert(_audioFrame.samples_per_channel_ == fileSamples); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4872 | |
| 4873 | if (_mixFileWithMicrophone) |
| 4874 | { |
braveyao@webrtc.org | d713143 | 2012-03-29 10:39:44 +0000 | [diff] [blame] | 4875 | // Currently file stream is always mono. |
| 4876 | // TODO(xians): Change the code when FilePlayer supports real stereo. |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4877 | Utility::MixWithSat(_audioFrame.data_, |
andrew@webrtc.org | e59a0ac | 2012-05-08 17:12:40 +0000 | [diff] [blame] | 4878 | _audioFrame.num_channels_, |
braveyao@webrtc.org | d713143 | 2012-03-29 10:39:44 +0000 | [diff] [blame] | 4879 | fileBuffer.get(), |
| 4880 | 1, |
andrew@webrtc.org | e59a0ac | 2012-05-08 17:12:40 +0000 | [diff] [blame] | 4881 | fileSamples); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4882 | } |
| 4883 | else |
| 4884 | { |
braveyao@webrtc.org | d713143 | 2012-03-29 10:39:44 +0000 | [diff] [blame] | 4885 | // Replace ACM audio with file. |
| 4886 | // Currently file stream is always mono. |
| 4887 | // TODO(xians): Change the code when FilePlayer supports real stereo. |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4888 | _audioFrame.UpdateFrame(_channelId, |
| 4889 | -1, |
braveyao@webrtc.org | d713143 | 2012-03-29 10:39:44 +0000 | [diff] [blame] | 4890 | fileBuffer.get(), |
andrew@webrtc.org | e59a0ac | 2012-05-08 17:12:40 +0000 | [diff] [blame] | 4891 | fileSamples, |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4892 | mixingFrequency, |
| 4893 | AudioFrame::kNormalSpeech, |
| 4894 | AudioFrame::kVadUnknown, |
| 4895 | 1); |
| 4896 | |
| 4897 | } |
| 4898 | return 0; |
| 4899 | } |
| 4900 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4901 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4902 | Channel::MixAudioWithFile(AudioFrame& audioFrame, |
xians@google.com | 0b0665a | 2011-08-08 08:18:44 +0000 | [diff] [blame] | 4903 | const int mixingFrequency) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4904 | { |
| 4905 | assert(mixingFrequency <= 32000); |
| 4906 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4907 | scoped_array<int16_t> fileBuffer(new int16_t[640]); |
andrew@webrtc.org | e59a0ac | 2012-05-08 17:12:40 +0000 | [diff] [blame] | 4908 | int fileSamples(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4909 | |
| 4910 | { |
mflodman@webrtc.org | 9a065d1 | 2012-03-07 08:12:21 +0000 | [diff] [blame] | 4911 | CriticalSectionScoped cs(&_fileCritSect); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4912 | |
| 4913 | if (_outputFilePlayerPtr == NULL) |
| 4914 | { |
| 4915 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4916 | VoEId(_instanceId, _channelId), |
| 4917 | "Channel::MixAudioWithFile() file mixing failed"); |
| 4918 | return -1; |
| 4919 | } |
| 4920 | |
| 4921 | // We should get the frequency we ask for. |
braveyao@webrtc.org | d713143 | 2012-03-29 10:39:44 +0000 | [diff] [blame] | 4922 | if (_outputFilePlayerPtr->Get10msAudioFromFile(fileBuffer.get(), |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4923 | fileSamples, |
| 4924 | mixingFrequency) == -1) |
| 4925 | { |
| 4926 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 4927 | VoEId(_instanceId, _channelId), |
| 4928 | "Channel::MixAudioWithFile() file mixing failed"); |
| 4929 | return -1; |
| 4930 | } |
| 4931 | } |
| 4932 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4933 | if (audioFrame.samples_per_channel_ == fileSamples) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4934 | { |
braveyao@webrtc.org | d713143 | 2012-03-29 10:39:44 +0000 | [diff] [blame] | 4935 | // Currently file stream is always mono. |
| 4936 | // TODO(xians): Change the code when FilePlayer supports real stereo. |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4937 | Utility::MixWithSat(audioFrame.data_, |
andrew@webrtc.org | e59a0ac | 2012-05-08 17:12:40 +0000 | [diff] [blame] | 4938 | audioFrame.num_channels_, |
braveyao@webrtc.org | d713143 | 2012-03-29 10:39:44 +0000 | [diff] [blame] | 4939 | fileBuffer.get(), |
| 4940 | 1, |
andrew@webrtc.org | e59a0ac | 2012-05-08 17:12:40 +0000 | [diff] [blame] | 4941 | fileSamples); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4942 | } |
| 4943 | else |
| 4944 | { |
| 4945 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId), |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4946 | "Channel::MixAudioWithFile() samples_per_channel_(%d) != " |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4947 | "fileSamples(%d)", |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4948 | audioFrame.samples_per_channel_, fileSamples); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4949 | return -1; |
| 4950 | } |
| 4951 | |
| 4952 | return 0; |
| 4953 | } |
| 4954 | |
| 4955 | int |
| 4956 | Channel::InsertInbandDtmfTone() |
| 4957 | { |
niklas.enbom@webrtc.org | af26f64 | 2011-11-16 12:41:36 +0000 | [diff] [blame] | 4958 | // Check if we should start a new tone. |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4959 | if (_inbandDtmfQueue.PendingDtmf() && |
| 4960 | !_inbandDtmfGenerator.IsAddingTone() && |
| 4961 | _inbandDtmfGenerator.DelaySinceLastTone() > |
| 4962 | kMinTelephoneEventSeparationMs) |
| 4963 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4964 | int8_t eventCode(0); |
| 4965 | uint16_t lengthMs(0); |
| 4966 | uint8_t attenuationDb(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4967 | |
| 4968 | eventCode = _inbandDtmfQueue.NextDtmf(&lengthMs, &attenuationDb); |
| 4969 | _inbandDtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb); |
| 4970 | if (_playInbandDtmfEvent) |
| 4971 | { |
| 4972 | // Add tone to output mixer using a reduced length to minimize |
| 4973 | // risk of echo. |
| 4974 | _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs - 80, |
| 4975 | attenuationDb); |
| 4976 | } |
| 4977 | } |
| 4978 | |
| 4979 | if (_inbandDtmfGenerator.IsAddingTone()) |
| 4980 | { |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4981 | uint16_t frequency(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4982 | _inbandDtmfGenerator.GetSampleRate(frequency); |
| 4983 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 4984 | if (frequency != _audioFrame.sample_rate_hz_) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4985 | { |
| 4986 | // Update sample rate of Dtmf tone since the mixing frequency |
| 4987 | // has changed. |
| 4988 | _inbandDtmfGenerator.SetSampleRate( |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4989 | (uint16_t) (_audioFrame.sample_rate_hz_)); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4990 | // Reset the tone to be added taking the new sample rate into |
| 4991 | // account. |
| 4992 | _inbandDtmfGenerator.ResetTone(); |
| 4993 | } |
andrew@webrtc.org | ae1a58b | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 4994 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 4995 | int16_t toneBuffer[320]; |
| 4996 | uint16_t toneSamples(0); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 4997 | // Get 10ms tone segment and set time since last tone to zero |
| 4998 | if (_inbandDtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1) |
| 4999 | { |
| 5000 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 5001 | VoEId(_instanceId, _channelId), |
| 5002 | "Channel::EncodeAndSend() inserting Dtmf failed"); |
| 5003 | return -1; |
| 5004 | } |
| 5005 | |
niklas.enbom@webrtc.org | af26f64 | 2011-11-16 12:41:36 +0000 | [diff] [blame] | 5006 | // Replace mixed audio with DTMF tone. |
andrew@webrtc.org | ae1a58b | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 5007 | for (int sample = 0; |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 5008 | sample < _audioFrame.samples_per_channel_; |
niklas.enbom@webrtc.org | af26f64 | 2011-11-16 12:41:36 +0000 | [diff] [blame] | 5009 | sample++) |
| 5010 | { |
andrew@webrtc.org | ae1a58b | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 5011 | for (int channel = 0; |
| 5012 | channel < _audioFrame.num_channels_; |
niklas.enbom@webrtc.org | af26f64 | 2011-11-16 12:41:36 +0000 | [diff] [blame] | 5013 | channel++) |
| 5014 | { |
andrew@webrtc.org | ae1a58b | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 5015 | const int index = sample * _audioFrame.num_channels_ + channel; |
| 5016 | _audioFrame.data_[index] = toneBuffer[sample]; |
niklas.enbom@webrtc.org | af26f64 | 2011-11-16 12:41:36 +0000 | [diff] [blame] | 5017 | } |
| 5018 | } |
andrew@webrtc.org | ae1a58b | 2013-01-22 04:44:30 +0000 | [diff] [blame] | 5019 | |
andrew@webrtc.org | 63a5098 | 2012-05-02 23:56:37 +0000 | [diff] [blame] | 5020 | assert(_audioFrame.samples_per_channel_ == toneSamples); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5021 | } else |
| 5022 | { |
| 5023 | // Add 10ms to "delay-since-last-tone" counter |
| 5024 | _inbandDtmfGenerator.UpdateDelaySinceLastTone(); |
| 5025 | } |
| 5026 | return 0; |
| 5027 | } |
| 5028 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5029 | void |
| 5030 | Channel::ResetDeadOrAliveCounters() |
| 5031 | { |
| 5032 | _countDeadDetections = 0; |
| 5033 | _countAliveDetections = 0; |
| 5034 | } |
| 5035 | |
| 5036 | void |
| 5037 | Channel::UpdateDeadOrAliveCounters(bool alive) |
| 5038 | { |
| 5039 | if (alive) |
| 5040 | _countAliveDetections++; |
| 5041 | else |
| 5042 | _countDeadDetections++; |
| 5043 | } |
| 5044 | |
| 5045 | int |
| 5046 | Channel::GetDeadOrAliveCounters(int& countDead, int& countAlive) const |
| 5047 | { |
| 5048 | bool enabled; |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 5049 | uint8_t timeSec; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5050 | |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 5051 | _rtpRtcpModule->PeriodicDeadOrAliveStatus(enabled, timeSec); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5052 | if (!enabled) |
| 5053 | return (-1); |
| 5054 | |
| 5055 | countDead = static_cast<int> (_countDeadDetections); |
| 5056 | countAlive = static_cast<int> (_countAliveDetections); |
| 5057 | return 0; |
| 5058 | } |
| 5059 | |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 5060 | int32_t |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5061 | Channel::SendPacketRaw(const void *data, int len, bool RTCP) |
| 5062 | { |
| 5063 | if (_transportPtr == NULL) |
| 5064 | { |
| 5065 | return -1; |
| 5066 | } |
| 5067 | if (!RTCP) |
| 5068 | { |
| 5069 | return _transportPtr->SendPacket(_channelId, data, len); |
| 5070 | } |
| 5071 | else |
| 5072 | { |
| 5073 | return _transportPtr->SendRTCPPacket(_channelId, data, len); |
| 5074 | } |
| 5075 | } |
| 5076 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5077 | // Called for incoming RTP packets after successful RTP header parsing. |
| 5078 | void Channel::UpdatePacketDelay(uint32_t rtp_timestamp, |
| 5079 | uint16_t sequence_number) { |
| 5080 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5081 | "Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)", |
| 5082 | rtp_timestamp, sequence_number); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5083 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5084 | // Get frequency of last received payload |
| 5085 | int rtp_receive_frequency = _audioCodingModule.ReceiveFrequency(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5086 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5087 | CodecInst current_receive_codec; |
| 5088 | if (_audioCodingModule.ReceiveCodec(¤t_receive_codec) != 0) { |
| 5089 | return; |
| 5090 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5091 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5092 | if (STR_CASE_CMP("G722", current_receive_codec.plname) == 0) { |
| 5093 | // Even though the actual sampling rate for G.722 audio is |
| 5094 | // 16,000 Hz, the RTP clock rate for the G722 payload format is |
| 5095 | // 8,000 Hz because that value was erroneously assigned in |
| 5096 | // RFC 1890 and must remain unchanged for backward compatibility. |
| 5097 | rtp_receive_frequency = 8000; |
| 5098 | } else if (STR_CASE_CMP("opus", current_receive_codec.plname) == 0) { |
| 5099 | // We are resampling Opus internally to 32,000 Hz until all our |
| 5100 | // DSP routines can operate at 48,000 Hz, but the RTP clock |
| 5101 | // rate for the Opus payload format is standardized to 48,000 Hz, |
| 5102 | // because that is the maximum supported decoding sampling rate. |
| 5103 | rtp_receive_frequency = 48000; |
| 5104 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5105 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5106 | // playout_timestamp_rtp_ updated in UpdatePlayoutTimestamp for every incoming |
| 5107 | // packet. |
| 5108 | uint32_t timestamp_diff_ms = (rtp_timestamp - playout_timestamp_rtp_) / |
| 5109 | (rtp_receive_frequency / 1000); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5110 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5111 | uint16_t packet_delay_ms = (rtp_timestamp - _previousTimestamp) / |
| 5112 | (rtp_receive_frequency / 1000); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5113 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5114 | _previousTimestamp = rtp_timestamp; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5115 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5116 | if (timestamp_diff_ms > (2 * kVoiceEngineMaxMinPlayoutDelayMs)) { |
| 5117 | timestamp_diff_ms = 0; |
| 5118 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5119 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5120 | if (timestamp_diff_ms == 0) return; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5121 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5122 | if (packet_delay_ms >= 10 && packet_delay_ms <= 60) { |
| 5123 | _recPacketDelayMs = packet_delay_ms; |
| 5124 | } |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5125 | |
pwestin@webrtc.org | 1de0135 | 2013-04-11 20:23:35 +0000 | [diff] [blame^] | 5126 | if (_average_jitter_buffer_delay_us == 0) { |
| 5127 | _average_jitter_buffer_delay_us = timestamp_diff_ms * 1000; |
| 5128 | return; |
| 5129 | } |
| 5130 | |
| 5131 | // Filter average delay value using exponential filter (alpha is |
| 5132 | // 7/8). We derive 1000 *_average_jitter_buffer_delay_us here (reduces |
| 5133 | // risk of rounding error) and compensate for it in GetDelayEstimate() |
| 5134 | // later. |
| 5135 | _average_jitter_buffer_delay_us = (_average_jitter_buffer_delay_us * 7 + |
| 5136 | 1000 * timestamp_diff_ms + 500) / 8; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5137 | } |
| 5138 | |
| 5139 | void |
| 5140 | Channel::RegisterReceiveCodecsToRTPModule() |
| 5141 | { |
| 5142 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |
| 5143 | "Channel::RegisterReceiveCodecsToRTPModule()"); |
| 5144 | |
| 5145 | |
| 5146 | CodecInst codec; |
pbos@webrtc.org | 6141e13 | 2013-04-09 10:09:10 +0000 | [diff] [blame] | 5147 | const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs(); |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5148 | |
| 5149 | for (int idx = 0; idx < nSupportedCodecs; idx++) |
| 5150 | { |
| 5151 | // Open up the RTP/RTCP receiver for all supported codecs |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 5152 | if ((_audioCodingModule.Codec(idx, &codec) == -1) || |
pwestin@webrtc.org | 2853dde | 2012-05-11 11:08:54 +0000 | [diff] [blame] | 5153 | (_rtpRtcpModule->RegisterReceivePayload(codec) == -1)) |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5154 | { |
| 5155 | WEBRTC_TRACE( |
| 5156 | kTraceWarning, |
| 5157 | kTraceVoice, |
| 5158 | VoEId(_instanceId, _channelId), |
| 5159 | "Channel::RegisterReceiveCodecsToRTPModule() unable" |
| 5160 | " to register %s (%d/%d/%d/%d) to RTP/RTCP receiver", |
| 5161 | codec.plname, codec.pltype, codec.plfreq, |
| 5162 | codec.channels, codec.rate); |
| 5163 | } |
| 5164 | else |
| 5165 | { |
| 5166 | WEBRTC_TRACE( |
| 5167 | kTraceInfo, |
| 5168 | kTraceVoice, |
| 5169 | VoEId(_instanceId, _channelId), |
| 5170 | "Channel::RegisterReceiveCodecsToRTPModule() %s " |
wu@webrtc.org | fcd12b3 | 2011-09-15 20:49:50 +0000 | [diff] [blame] | 5171 | "(%d/%d/%d/%d) has been added to the RTP/RTCP " |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5172 | "receiver", |
| 5173 | codec.plname, codec.pltype, codec.plfreq, |
| 5174 | codec.channels, codec.rate); |
| 5175 | } |
| 5176 | } |
| 5177 | } |
| 5178 | |
andrew@webrtc.org | 50419b0 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 5179 | int Channel::ApmProcessRx(AudioFrame& frame) { |
| 5180 | AudioProcessing* audioproc = _rxAudioProcessingModulePtr; |
| 5181 | // Register the (possibly new) frame parameters. |
| 5182 | if (audioproc->set_sample_rate_hz(frame.sample_rate_hz_) != 0) { |
andrew@webrtc.org | 655d8f5 | 2012-11-20 07:34:45 +0000 | [diff] [blame] | 5183 | LOG_FERR1(LS_WARNING, set_sample_rate_hz, frame.sample_rate_hz_); |
andrew@webrtc.org | 50419b0 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 5184 | } |
| 5185 | if (audioproc->set_num_channels(frame.num_channels_, |
| 5186 | frame.num_channels_) != 0) { |
andrew@webrtc.org | 655d8f5 | 2012-11-20 07:34:45 +0000 | [diff] [blame] | 5187 | LOG_FERR1(LS_WARNING, set_num_channels, frame.num_channels_); |
andrew@webrtc.org | 50419b0 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 5188 | } |
| 5189 | if (audioproc->ProcessStream(&frame) != 0) { |
andrew@webrtc.org | 655d8f5 | 2012-11-20 07:34:45 +0000 | [diff] [blame] | 5190 | LOG_FERR0(LS_WARNING, ProcessStream); |
andrew@webrtc.org | 50419b0 | 2012-11-14 19:07:54 +0000 | [diff] [blame] | 5191 | } |
| 5192 | return 0; |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5193 | } |
| 5194 | |
turaj@webrtc.org | 42259e7 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 5195 | int Channel::SetSecondarySendCodec(const CodecInst& codec, |
| 5196 | int red_payload_type) { |
turaj@webrtc.org | 8c8ad85 | 2013-01-31 18:20:17 +0000 | [diff] [blame] | 5197 | // Sanity check for payload type. |
| 5198 | if (red_payload_type < 0 || red_payload_type > 127) { |
| 5199 | _engineStatisticsPtr->SetLastError( |
| 5200 | VE_PLTYPE_ERROR, kTraceError, |
| 5201 | "SetRedPayloadType() invalid RED payload type"); |
| 5202 | return -1; |
| 5203 | } |
| 5204 | |
turaj@webrtc.org | 42259e7 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 5205 | if (SetRedPayloadType(red_payload_type) < 0) { |
| 5206 | _engineStatisticsPtr->SetLastError( |
| 5207 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 5208 | "SetSecondarySendCodec() Failed to register RED ACM"); |
| 5209 | return -1; |
| 5210 | } |
| 5211 | if (_audioCodingModule.RegisterSecondarySendCodec(codec) < 0) { |
| 5212 | _engineStatisticsPtr->SetLastError( |
| 5213 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 5214 | "SetSecondarySendCodec() Failed to register secondary send codec in " |
| 5215 | "ACM"); |
| 5216 | return -1; |
| 5217 | } |
| 5218 | |
| 5219 | return 0; |
| 5220 | } |
| 5221 | |
| 5222 | void Channel::RemoveSecondarySendCodec() { |
| 5223 | _audioCodingModule.UnregisterSecondarySendCodec(); |
| 5224 | } |
| 5225 | |
| 5226 | int Channel::GetSecondarySendCodec(CodecInst* codec) { |
| 5227 | if (_audioCodingModule.SecondarySendCodec(codec) < 0) { |
| 5228 | _engineStatisticsPtr->SetLastError( |
| 5229 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 5230 | "GetSecondarySendCodec() Failed to get secondary sent codec from ACM"); |
| 5231 | return -1; |
| 5232 | } |
| 5233 | return 0; |
| 5234 | } |
| 5235 | |
turaj@webrtc.org | 8c8ad85 | 2013-01-31 18:20:17 +0000 | [diff] [blame] | 5236 | // Assuming this method is called with valid payload type. |
turaj@webrtc.org | 42259e7 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 5237 | int Channel::SetRedPayloadType(int red_payload_type) { |
turaj@webrtc.org | 42259e7 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 5238 | CodecInst codec; |
| 5239 | bool found_red = false; |
| 5240 | |
| 5241 | // Get default RED settings from the ACM database |
| 5242 | const int num_codecs = AudioCodingModule::NumberOfCodecs(); |
| 5243 | for (int idx = 0; idx < num_codecs; idx++) { |
tina.legrand@webrtc.org | 7a7a008 | 2013-02-21 10:27:48 +0000 | [diff] [blame] | 5244 | _audioCodingModule.Codec(idx, &codec); |
turaj@webrtc.org | 42259e7 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 5245 | if (!STR_CASE_CMP(codec.plname, "RED")) { |
| 5246 | found_red = true; |
| 5247 | break; |
| 5248 | } |
| 5249 | } |
| 5250 | |
| 5251 | if (!found_red) { |
| 5252 | _engineStatisticsPtr->SetLastError( |
| 5253 | VE_CODEC_ERROR, kTraceError, |
| 5254 | "SetRedPayloadType() RED is not supported"); |
| 5255 | return -1; |
| 5256 | } |
| 5257 | |
turaj@webrtc.org | 9d532fd | 2013-01-31 18:34:19 +0000 | [diff] [blame] | 5258 | codec.pltype = red_payload_type; |
turaj@webrtc.org | 42259e7 | 2012-12-11 02:15:12 +0000 | [diff] [blame] | 5259 | if (_audioCodingModule.RegisterSendCodec(codec) < 0) { |
| 5260 | _engineStatisticsPtr->SetLastError( |
| 5261 | VE_AUDIO_CODING_MODULE_ERROR, kTraceError, |
| 5262 | "SetRedPayloadType() RED registration in ACM module failed"); |
| 5263 | return -1; |
| 5264 | } |
| 5265 | |
| 5266 | if (_rtpRtcpModule->SetSendREDPayloadType(red_payload_type) != 0) { |
| 5267 | _engineStatisticsPtr->SetLastError( |
| 5268 | VE_RTP_RTCP_MODULE_ERROR, kTraceError, |
| 5269 | "SetRedPayloadType() RED registration in RTP/RTCP module failed"); |
| 5270 | return -1; |
| 5271 | } |
| 5272 | return 0; |
| 5273 | } |
| 5274 | |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5275 | } // namespace voe |
niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame] | 5276 | } // namespace webrtc |