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| 1 /* | |
| 2 * libjingle | |
| 3 * Copyright 2012 Google Inc. | |
| 4 * | |
| 5 * Redistribution and use in source and binary forms, with or without | |
| 6 * modification, are permitted provided that the following conditions are met: | |
| 7 * | |
| 8 * 1. Redistributions of source code must retain the above copyright notice, | |
| 9 * this list of conditions and the following disclaimer. | |
| 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | |
| 11 * this list of conditions and the following disclaimer in the documentation | |
| 12 * and/or other materials provided with the distribution. | |
| 13 * 3. The name of the author may not be used to endorse or promote products | |
| 14 * derived from this software without specific prior written permission. | |
| 15 * | |
| 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED | |
| 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF | |
| 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO | |
| 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, | |
| 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, | |
| 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | |
| 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | |
| 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | |
| 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | |
| 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | |
| 26 */ | |
| 27 | |
| 28 #include "talk/media/base/rtpdataengine.h" | |
| 29 | |
| 30 #include "talk/media/base/codec.h" | |
| 31 #include "talk/media/base/constants.h" | |
| 32 #include "talk/media/base/rtputils.h" | |
| 33 #include "talk/media/base/streamparams.h" | |
| 34 #include "webrtc/base/buffer.h" | |
| 35 #include "webrtc/base/helpers.h" | |
| 36 #include "webrtc/base/logging.h" | |
| 37 #include "webrtc/base/ratelimiter.h" | |
| 38 #include "webrtc/base/timing.h" | |
| 39 | |
| 40 namespace cricket { | |
| 41 | |
| 42 // We want to avoid IP fragmentation. | |
| 43 static const size_t kDataMaxRtpPacketLen = 1200U; | |
| 44 // We reserve space after the RTP header for future wiggle room. | |
| 45 static const unsigned char kReservedSpace[] = { | |
| 46 0x00, 0x00, 0x00, 0x00 | |
| 47 }; | |
| 48 | |
| 49 // Amount of overhead SRTP may take. We need to leave room in the | |
| 50 // buffer for it, otherwise SRTP will fail later. If SRTP ever uses | |
| 51 // more than this, we need to increase this number. | |
| 52 static const size_t kMaxSrtpHmacOverhead = 16; | |
| 53 | |
| 54 RtpDataEngine::RtpDataEngine() { | |
| 55 data_codecs_.push_back( | |
| 56 DataCodec(kGoogleRtpDataCodecId, | |
| 57 kGoogleRtpDataCodecName, 0)); | |
| 58 SetTiming(new rtc::Timing()); | |
| 59 } | |
| 60 | |
| 61 DataMediaChannel* RtpDataEngine::CreateChannel( | |
| 62 DataChannelType data_channel_type) { | |
| 63 if (data_channel_type != DCT_RTP) { | |
| 64 return NULL; | |
| 65 } | |
| 66 return new RtpDataMediaChannel(timing_.get()); | |
| 67 } | |
| 68 | |
| 69 bool FindCodecByName(const std::vector<DataCodec>& codecs, | |
| 70 const std::string& name, DataCodec* codec_out) { | |
| 71 std::vector<DataCodec>::const_iterator iter; | |
| 72 for (iter = codecs.begin(); iter != codecs.end(); ++iter) { | |
| 73 if (iter->name == name) { | |
| 74 *codec_out = *iter; | |
| 75 return true; | |
| 76 } | |
| 77 } | |
| 78 return false; | |
| 79 } | |
| 80 | |
| 81 RtpDataMediaChannel::RtpDataMediaChannel(rtc::Timing* timing) { | |
| 82 Construct(timing); | |
| 83 } | |
| 84 | |
| 85 RtpDataMediaChannel::RtpDataMediaChannel() { | |
| 86 Construct(NULL); | |
| 87 } | |
| 88 | |
| 89 void RtpDataMediaChannel::Construct(rtc::Timing* timing) { | |
| 90 sending_ = false; | |
| 91 receiving_ = false; | |
| 92 timing_ = timing; | |
| 93 send_limiter_.reset(new rtc::RateLimiter(kDataMaxBandwidth / 8, 1.0)); | |
| 94 } | |
| 95 | |
| 96 | |
| 97 RtpDataMediaChannel::~RtpDataMediaChannel() { | |
| 98 std::map<uint32_t, RtpClock*>::const_iterator iter; | |
| 99 for (iter = rtp_clock_by_send_ssrc_.begin(); | |
| 100 iter != rtp_clock_by_send_ssrc_.end(); | |
| 101 ++iter) { | |
| 102 delete iter->second; | |
| 103 } | |
| 104 } | |
| 105 | |
| 106 void RtpClock::Tick(double now, int* seq_num, uint32_t* timestamp) { | |
| 107 *seq_num = ++last_seq_num_; | |
| 108 *timestamp = timestamp_offset_ + static_cast<uint32_t>(now * clockrate_); | |
| 109 } | |
| 110 | |
| 111 const DataCodec* FindUnknownCodec(const std::vector<DataCodec>& codecs) { | |
| 112 DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0); | |
| 113 std::vector<DataCodec>::const_iterator iter; | |
| 114 for (iter = codecs.begin(); iter != codecs.end(); ++iter) { | |
| 115 if (!iter->Matches(data_codec)) { | |
| 116 return &(*iter); | |
| 117 } | |
| 118 } | |
| 119 return NULL; | |
| 120 } | |
| 121 | |
| 122 const DataCodec* FindKnownCodec(const std::vector<DataCodec>& codecs) { | |
| 123 DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0); | |
| 124 std::vector<DataCodec>::const_iterator iter; | |
| 125 for (iter = codecs.begin(); iter != codecs.end(); ++iter) { | |
| 126 if (iter->Matches(data_codec)) { | |
| 127 return &(*iter); | |
| 128 } | |
| 129 } | |
| 130 return NULL; | |
| 131 } | |
| 132 | |
| 133 bool RtpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) { | |
| 134 const DataCodec* unknown_codec = FindUnknownCodec(codecs); | |
| 135 if (unknown_codec) { | |
| 136 LOG(LS_WARNING) << "Failed to SetRecvCodecs because of unknown codec: " | |
| 137 << unknown_codec->ToString(); | |
| 138 return false; | |
| 139 } | |
| 140 | |
| 141 recv_codecs_ = codecs; | |
| 142 return true; | |
| 143 } | |
| 144 | |
| 145 bool RtpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) { | |
| 146 const DataCodec* known_codec = FindKnownCodec(codecs); | |
| 147 if (!known_codec) { | |
| 148 LOG(LS_WARNING) << | |
| 149 "Failed to SetSendCodecs because there is no known codec."; | |
| 150 return false; | |
| 151 } | |
| 152 | |
| 153 send_codecs_ = codecs; | |
| 154 return true; | |
| 155 } | |
| 156 | |
| 157 bool RtpDataMediaChannel::SetSendParameters(const DataSendParameters& params) { | |
| 158 return (SetSendCodecs(params.codecs) && | |
| 159 SetMaxSendBandwidth(params.max_bandwidth_bps)); | |
| 160 } | |
| 161 | |
| 162 bool RtpDataMediaChannel::SetRecvParameters(const DataRecvParameters& params) { | |
| 163 return SetRecvCodecs(params.codecs); | |
| 164 } | |
| 165 | |
| 166 bool RtpDataMediaChannel::AddSendStream(const StreamParams& stream) { | |
| 167 if (!stream.has_ssrcs()) { | |
| 168 return false; | |
| 169 } | |
| 170 | |
| 171 if (GetStreamBySsrc(send_streams_, stream.first_ssrc())) { | |
| 172 LOG(LS_WARNING) << "Not adding data send stream '" << stream.id | |
| 173 << "' with ssrc=" << stream.first_ssrc() | |
| 174 << " because stream already exists."; | |
| 175 return false; | |
| 176 } | |
| 177 | |
| 178 send_streams_.push_back(stream); | |
| 179 // TODO(pthatcher): This should be per-stream, not per-ssrc. | |
| 180 // And we should probably allow more than one per stream. | |
| 181 rtp_clock_by_send_ssrc_[stream.first_ssrc()] = new RtpClock( | |
| 182 kDataCodecClockrate, | |
| 183 rtc::CreateRandomNonZeroId(), rtc::CreateRandomNonZeroId()); | |
| 184 | |
| 185 LOG(LS_INFO) << "Added data send stream '" << stream.id | |
| 186 << "' with ssrc=" << stream.first_ssrc(); | |
| 187 return true; | |
| 188 } | |
| 189 | |
| 190 bool RtpDataMediaChannel::RemoveSendStream(uint32_t ssrc) { | |
| 191 if (!GetStreamBySsrc(send_streams_, ssrc)) { | |
| 192 return false; | |
| 193 } | |
| 194 | |
| 195 RemoveStreamBySsrc(&send_streams_, ssrc); | |
| 196 delete rtp_clock_by_send_ssrc_[ssrc]; | |
| 197 rtp_clock_by_send_ssrc_.erase(ssrc); | |
| 198 return true; | |
| 199 } | |
| 200 | |
| 201 bool RtpDataMediaChannel::AddRecvStream(const StreamParams& stream) { | |
| 202 if (!stream.has_ssrcs()) { | |
| 203 return false; | |
| 204 } | |
| 205 | |
| 206 if (GetStreamBySsrc(recv_streams_, stream.first_ssrc())) { | |
| 207 LOG(LS_WARNING) << "Not adding data recv stream '" << stream.id | |
| 208 << "' with ssrc=" << stream.first_ssrc() | |
| 209 << " because stream already exists."; | |
| 210 return false; | |
| 211 } | |
| 212 | |
| 213 recv_streams_.push_back(stream); | |
| 214 LOG(LS_INFO) << "Added data recv stream '" << stream.id | |
| 215 << "' with ssrc=" << stream.first_ssrc(); | |
| 216 return true; | |
| 217 } | |
| 218 | |
| 219 bool RtpDataMediaChannel::RemoveRecvStream(uint32_t ssrc) { | |
| 220 RemoveStreamBySsrc(&recv_streams_, ssrc); | |
| 221 return true; | |
| 222 } | |
| 223 | |
| 224 void RtpDataMediaChannel::OnPacketReceived( | |
| 225 rtc::Buffer* packet, const rtc::PacketTime& packet_time) { | |
| 226 RtpHeader header; | |
| 227 if (!GetRtpHeader(packet->data(), packet->size(), &header)) { | |
| 228 // Don't want to log for every corrupt packet. | |
| 229 // LOG(LS_WARNING) << "Could not read rtp header from packet of length " | |
| 230 // << packet->length() << "."; | |
| 231 return; | |
| 232 } | |
| 233 | |
| 234 size_t header_length; | |
| 235 if (!GetRtpHeaderLen(packet->data(), packet->size(), &header_length)) { | |
| 236 // Don't want to log for every corrupt packet. | |
| 237 // LOG(LS_WARNING) << "Could not read rtp header" | |
| 238 // << length from packet of length " | |
| 239 // << packet->length() << "."; | |
| 240 return; | |
| 241 } | |
| 242 const char* data = | |
| 243 packet->data<char>() + header_length + sizeof(kReservedSpace); | |
| 244 size_t data_len = packet->size() - header_length - sizeof(kReservedSpace); | |
| 245 | |
| 246 if (!receiving_) { | |
| 247 LOG(LS_WARNING) << "Not receiving packet " | |
| 248 << header.ssrc << ":" << header.seq_num | |
| 249 << " before SetReceive(true) called."; | |
| 250 return; | |
| 251 } | |
| 252 | |
| 253 DataCodec codec; | |
| 254 if (!FindCodecById(recv_codecs_, header.payload_type, &codec)) { | |
| 255 // For bundling, this will be logged for every message. | |
| 256 // So disable this logging. | |
| 257 // LOG(LS_WARNING) << "Not receiving packet " | |
| 258 // << header.ssrc << ":" << header.seq_num | |
| 259 // << " (" << data_len << ")" | |
| 260 // << " because unknown payload id: " << header.payload_type; | |
| 261 return; | |
| 262 } | |
| 263 | |
| 264 if (!GetStreamBySsrc(recv_streams_, header.ssrc)) { | |
| 265 LOG(LS_WARNING) << "Received packet for unknown ssrc: " << header.ssrc; | |
| 266 return; | |
| 267 } | |
| 268 | |
| 269 // Uncomment this for easy debugging. | |
| 270 // const auto* found_stream = GetStreamBySsrc(recv_streams_, header.ssrc); | |
| 271 // LOG(LS_INFO) << "Received packet" | |
| 272 // << " groupid=" << found_stream.groupid | |
| 273 // << ", ssrc=" << header.ssrc | |
| 274 // << ", seqnum=" << header.seq_num | |
| 275 // << ", timestamp=" << header.timestamp | |
| 276 // << ", len=" << data_len; | |
| 277 | |
| 278 ReceiveDataParams params; | |
| 279 params.ssrc = header.ssrc; | |
| 280 params.seq_num = header.seq_num; | |
| 281 params.timestamp = header.timestamp; | |
| 282 SignalDataReceived(params, data, data_len); | |
| 283 } | |
| 284 | |
| 285 bool RtpDataMediaChannel::SetMaxSendBandwidth(int bps) { | |
| 286 if (bps <= 0) { | |
| 287 bps = kDataMaxBandwidth; | |
| 288 } | |
| 289 send_limiter_.reset(new rtc::RateLimiter(bps / 8, 1.0)); | |
| 290 LOG(LS_INFO) << "RtpDataMediaChannel::SetSendBandwidth to " << bps << "bps."; | |
| 291 return true; | |
| 292 } | |
| 293 | |
| 294 bool RtpDataMediaChannel::SendData( | |
| 295 const SendDataParams& params, | |
| 296 const rtc::Buffer& payload, | |
| 297 SendDataResult* result) { | |
| 298 if (result) { | |
| 299 // If we return true, we'll set this to SDR_SUCCESS. | |
| 300 *result = SDR_ERROR; | |
| 301 } | |
| 302 if (!sending_) { | |
| 303 LOG(LS_WARNING) << "Not sending packet with ssrc=" << params.ssrc | |
| 304 << " len=" << payload.size() << " before SetSend(true)."; | |
| 305 return false; | |
| 306 } | |
| 307 | |
| 308 if (params.type != cricket::DMT_TEXT) { | |
| 309 LOG(LS_WARNING) << "Not sending data because binary type is unsupported."; | |
| 310 return false; | |
| 311 } | |
| 312 | |
| 313 const StreamParams* found_stream = | |
| 314 GetStreamBySsrc(send_streams_, params.ssrc); | |
| 315 if (!found_stream) { | |
| 316 LOG(LS_WARNING) << "Not sending data because ssrc is unknown: " | |
| 317 << params.ssrc; | |
| 318 return false; | |
| 319 } | |
| 320 | |
| 321 DataCodec found_codec; | |
| 322 if (!FindCodecByName(send_codecs_, kGoogleRtpDataCodecName, &found_codec)) { | |
| 323 LOG(LS_WARNING) << "Not sending data because codec is unknown: " | |
| 324 << kGoogleRtpDataCodecName; | |
| 325 return false; | |
| 326 } | |
| 327 | |
| 328 size_t packet_len = (kMinRtpPacketLen + sizeof(kReservedSpace) + | |
| 329 payload.size() + kMaxSrtpHmacOverhead); | |
| 330 if (packet_len > kDataMaxRtpPacketLen) { | |
| 331 return false; | |
| 332 } | |
| 333 | |
| 334 double now = timing_->TimerNow(); | |
| 335 | |
| 336 if (!send_limiter_->CanUse(packet_len, now)) { | |
| 337 LOG(LS_VERBOSE) << "Dropped data packet of len=" << packet_len | |
| 338 << "; already sent " << send_limiter_->used_in_period() | |
| 339 << "/" << send_limiter_->max_per_period(); | |
| 340 return false; | |
| 341 } | |
| 342 | |
| 343 RtpHeader header; | |
| 344 header.payload_type = found_codec.id; | |
| 345 header.ssrc = params.ssrc; | |
| 346 rtp_clock_by_send_ssrc_[header.ssrc]->Tick( | |
| 347 now, &header.seq_num, &header.timestamp); | |
| 348 | |
| 349 rtc::Buffer packet(kMinRtpPacketLen, packet_len); | |
| 350 if (!SetRtpHeader(packet.data(), packet.size(), header)) { | |
| 351 return false; | |
| 352 } | |
| 353 packet.AppendData(kReservedSpace); | |
| 354 packet.AppendData(payload); | |
| 355 | |
| 356 LOG(LS_VERBOSE) << "Sent RTP data packet: " | |
| 357 << " stream=" << found_stream->id << " ssrc=" << header.ssrc | |
| 358 << ", seqnum=" << header.seq_num | |
| 359 << ", timestamp=" << header.timestamp | |
| 360 << ", len=" << payload.size(); | |
| 361 | |
| 362 MediaChannel::SendPacket(&packet, rtc::PacketOptions()); | |
| 363 send_limiter_->Use(packet_len, now); | |
| 364 if (result) { | |
| 365 *result = SDR_SUCCESS; | |
| 366 } | |
| 367 return true; | |
| 368 } | |
| 369 | |
| 370 } // namespace cricket | |
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Chromium Code Reviews
Issue 1587193006:
Move talk/media to webrtc/media (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master