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| 1 /* | |
| 2 * libjingle | |
| 3 * Copyright 2004 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/filemediaengine.h" | |
| 29 | |
| 30 #include <algorithm> | |
| 31 #include <limits.h> | |
| 32 | |
| 33 #include "talk/media/base/rtpdump.h" | |
| 34 #include "talk/media/base/rtputils.h" | |
| 35 #include "talk/media/base/streamparams.h" | |
| 36 #include "webrtc/base/buffer.h" | |
| 37 #include "webrtc/base/event.h" | |
| 38 #include "webrtc/base/logging.h" | |
| 39 #include "webrtc/base/pathutils.h" | |
| 40 #include "webrtc/base/stream.h" | |
| 41 | |
| 42 namespace cricket { | |
| 43 | |
| 44 /////////////////////////////////////////////////////////////////////////// | |
| 45 // Implementation of FileMediaEngine. | |
| 46 /////////////////////////////////////////////////////////////////////////// | |
| 47 int FileMediaEngine::GetCapabilities() { | |
| 48 int capabilities = 0; | |
| 49 if (!voice_input_filename_.empty()) { | |
| 50 capabilities |= AUDIO_SEND; | |
| 51 } | |
| 52 if (!voice_output_filename_.empty()) { | |
| 53 capabilities |= AUDIO_RECV; | |
| 54 } | |
| 55 if (!video_input_filename_.empty()) { | |
| 56 capabilities |= VIDEO_SEND; | |
| 57 } | |
| 58 if (!video_output_filename_.empty()) { | |
| 59 capabilities |= VIDEO_RECV; | |
| 60 } | |
| 61 return capabilities; | |
| 62 } | |
| 63 | |
| 64 VoiceMediaChannel* FileMediaEngine::CreateChannel(const AudioOptions& options) { | |
| 65 rtc::FileStream* input_file_stream = nullptr; | |
| 66 rtc::FileStream* output_file_stream = nullptr; | |
| 67 | |
| 68 if (voice_input_filename_.empty() && voice_output_filename_.empty()) | |
| 69 return nullptr; | |
| 70 if (!voice_input_filename_.empty()) { | |
| 71 input_file_stream = rtc::Filesystem::OpenFile( | |
| 72 rtc::Pathname(voice_input_filename_), "rb"); | |
| 73 if (!input_file_stream) { | |
| 74 LOG(LS_ERROR) << "Not able to open the input audio stream file."; | |
| 75 return nullptr; | |
| 76 } | |
| 77 } | |
| 78 | |
| 79 if (!voice_output_filename_.empty()) { | |
| 80 output_file_stream = rtc::Filesystem::OpenFile( | |
| 81 rtc::Pathname(voice_output_filename_), "wb"); | |
| 82 if (!output_file_stream) { | |
| 83 delete input_file_stream; | |
| 84 LOG(LS_ERROR) << "Not able to open the output audio stream file."; | |
| 85 return nullptr; | |
| 86 } | |
| 87 } | |
| 88 | |
| 89 FileVoiceChannel* channel = new FileVoiceChannel( | |
| 90 input_file_stream, output_file_stream, rtp_sender_thread_); | |
| 91 channel->SetOptions(options); | |
| 92 return channel; | |
| 93 } | |
| 94 | |
| 95 VideoMediaChannel* FileMediaEngine::CreateVideoChannel( | |
| 96 const VideoOptions& options, | |
| 97 VoiceMediaChannel* voice_ch) { | |
| 98 rtc::FileStream* input_file_stream = NULL; | |
| 99 rtc::FileStream* output_file_stream = NULL; | |
| 100 | |
| 101 if (video_input_filename_.empty() && video_output_filename_.empty()) | |
| 102 return NULL; | |
| 103 | |
| 104 if (!video_input_filename_.empty()) { | |
| 105 input_file_stream = rtc::Filesystem::OpenFile( | |
| 106 rtc::Pathname(video_input_filename_), "rb"); | |
| 107 if (!input_file_stream) { | |
| 108 LOG(LS_ERROR) << "Not able to open the input video stream file."; | |
| 109 return NULL; | |
| 110 } | |
| 111 } | |
| 112 | |
| 113 if (!video_output_filename_.empty()) { | |
| 114 output_file_stream = rtc::Filesystem::OpenFile( | |
| 115 rtc::Pathname(video_output_filename_), "wb"); | |
| 116 if (!output_file_stream) { | |
| 117 delete input_file_stream; | |
| 118 LOG(LS_ERROR) << "Not able to open the output video stream file."; | |
| 119 return NULL; | |
| 120 } | |
| 121 } | |
| 122 | |
| 123 FileVideoChannel* channel = new FileVideoChannel( | |
| 124 input_file_stream, output_file_stream, rtp_sender_thread_); | |
| 125 channel->SetOptions(options); | |
| 126 return channel; | |
| 127 } | |
| 128 | |
| 129 /////////////////////////////////////////////////////////////////////////// | |
| 130 // Definition of RtpSenderReceiver. | |
| 131 /////////////////////////////////////////////////////////////////////////// | |
| 132 class RtpSenderReceiver : public rtc::MessageHandler { | |
| 133 public: | |
| 134 RtpSenderReceiver(MediaChannel* channel, | |
| 135 rtc::StreamInterface* input_file_stream, | |
| 136 rtc::StreamInterface* output_file_stream, | |
| 137 rtc::Thread* sender_thread); | |
| 138 virtual ~RtpSenderReceiver(); | |
| 139 | |
| 140 // Called by media channel. Context: media channel thread. | |
| 141 bool SetSend(bool send); | |
| 142 void SetSendSsrc(uint32 ssrc); | |
| 143 void OnPacketReceived(rtc::Buffer* packet); | |
| 144 | |
| 145 // Override virtual method of parent MessageHandler. Context: Worker Thread. | |
| 146 virtual void OnMessage(rtc::Message* pmsg); | |
| 147 | |
| 148 private: | |
| 149 // Read the next RTP dump packet, whose RTP SSRC is the same as first_ssrc_. | |
| 150 // Return true if successful. | |
| 151 bool ReadNextPacket(RtpDumpPacket* packet); | |
| 152 // Send a RTP packet to the network. The input parameter data points to the | |
| 153 // start of the RTP packet and len is the packet size. Return true if the sent | |
| 154 // size is equal to len. | |
| 155 bool SendRtpPacket(const void* data, size_t len); | |
| 156 | |
| 157 MediaChannel* media_channel_; | |
| 158 rtc::scoped_ptr<rtc::StreamInterface> input_stream_; | |
| 159 rtc::scoped_ptr<rtc::StreamInterface> output_stream_; | |
| 160 rtc::scoped_ptr<RtpDumpLoopReader> rtp_dump_reader_; | |
| 161 rtc::scoped_ptr<RtpDumpWriter> rtp_dump_writer_; | |
| 162 rtc::Thread* sender_thread_; | |
| 163 bool own_sender_thread_; | |
| 164 // RTP dump packet read from the input stream. | |
| 165 RtpDumpPacket rtp_dump_packet_; | |
| 166 uint32 start_send_time_; | |
| 167 bool sending_; | |
| 168 bool first_packet_; | |
| 169 uint32 first_ssrc_; | |
| 170 | |
| 171 DISALLOW_COPY_AND_ASSIGN(RtpSenderReceiver); | |
| 172 }; | |
| 173 | |
| 174 /////////////////////////////////////////////////////////////////////////// | |
| 175 // Implementation of RtpSenderReceiver. | |
| 176 /////////////////////////////////////////////////////////////////////////// | |
| 177 RtpSenderReceiver::RtpSenderReceiver( | |
| 178 MediaChannel* channel, | |
| 179 rtc::StreamInterface* input_file_stream, | |
| 180 rtc::StreamInterface* output_file_stream, | |
| 181 rtc::Thread* sender_thread) | |
| 182 : media_channel_(channel), | |
| 183 input_stream_(input_file_stream), | |
| 184 output_stream_(output_file_stream), | |
| 185 sending_(false), | |
| 186 first_packet_(true) { | |
| 187 if (sender_thread == NULL) { | |
| 188 sender_thread_ = new rtc::Thread(); | |
| 189 own_sender_thread_ = true; | |
| 190 } else { | |
| 191 sender_thread_ = sender_thread; | |
| 192 own_sender_thread_ = false; | |
| 193 } | |
| 194 | |
| 195 if (input_stream_) { | |
| 196 rtp_dump_reader_.reset(new RtpDumpLoopReader(input_stream_.get())); | |
| 197 // Start the sender thread, which reads rtp dump records, waits based on | |
| 198 // the record timestamps, and sends the RTP packets to the network. | |
| 199 if (own_sender_thread_) { | |
| 200 sender_thread_->Start(); | |
| 201 } | |
| 202 } | |
| 203 | |
| 204 // Create a rtp dump writer for the output RTP dump stream. | |
| 205 if (output_stream_) { | |
| 206 rtp_dump_writer_.reset(new RtpDumpWriter(output_stream_.get())); | |
| 207 } | |
| 208 } | |
| 209 | |
| 210 RtpSenderReceiver::~RtpSenderReceiver() { | |
| 211 if (own_sender_thread_) { | |
| 212 sender_thread_->Stop(); | |
| 213 delete sender_thread_; | |
| 214 } | |
| 215 } | |
| 216 | |
| 217 bool RtpSenderReceiver::SetSend(bool send) { | |
| 218 bool was_sending = sending_; | |
| 219 sending_ = send; | |
| 220 if (!was_sending && sending_) { | |
| 221 sender_thread_->PostDelayed(0, this); // Wake up the send thread. | |
| 222 start_send_time_ = rtc::Time(); | |
| 223 } | |
| 224 return true; | |
| 225 } | |
| 226 | |
| 227 void RtpSenderReceiver::SetSendSsrc(uint32 ssrc) { | |
| 228 if (rtp_dump_reader_) { | |
| 229 rtp_dump_reader_->SetSsrc(ssrc); | |
| 230 } | |
| 231 } | |
| 232 | |
| 233 void RtpSenderReceiver::OnPacketReceived(rtc::Buffer* packet) { | |
| 234 if (rtp_dump_writer_) { | |
| 235 rtp_dump_writer_->WriteRtpPacket(packet->data(), packet->size()); | |
| 236 } | |
| 237 } | |
| 238 | |
| 239 void RtpSenderReceiver::OnMessage(rtc::Message* pmsg) { | |
| 240 if (!sending_) { | |
| 241 // If the sender thread is not sending, ignore this message. The thread goes | |
| 242 // to sleep until SetSend(true) wakes it up. | |
| 243 return; | |
| 244 } | |
| 245 if (!first_packet_) { | |
| 246 // Send the previously read packet. | |
| 247 SendRtpPacket(&rtp_dump_packet_.data[0], rtp_dump_packet_.data.size()); | |
| 248 } | |
| 249 | |
| 250 if (ReadNextPacket(&rtp_dump_packet_)) { | |
| 251 int wait = rtc::TimeUntil( | |
| 252 start_send_time_ + rtp_dump_packet_.elapsed_time); | |
| 253 wait = std::max(0, wait); | |
| 254 sender_thread_->PostDelayed(wait, this); | |
| 255 } else { | |
| 256 sender_thread_->Quit(); | |
| 257 } | |
| 258 } | |
| 259 | |
| 260 bool RtpSenderReceiver::ReadNextPacket(RtpDumpPacket* packet) { | |
| 261 while (rtc::SR_SUCCESS == rtp_dump_reader_->ReadPacket(packet)) { | |
| 262 uint32 ssrc; | |
| 263 if (!packet->GetRtpSsrc(&ssrc)) { | |
| 264 return false; | |
| 265 } | |
| 266 if (first_packet_) { | |
| 267 first_packet_ = false; | |
| 268 first_ssrc_ = ssrc; | |
| 269 } | |
| 270 if (ssrc == first_ssrc_) { | |
| 271 return true; | |
| 272 } | |
| 273 } | |
| 274 return false; | |
| 275 } | |
| 276 | |
| 277 bool RtpSenderReceiver::SendRtpPacket(const void* data, size_t len) { | |
| 278 if (!media_channel_) | |
| 279 return false; | |
| 280 | |
| 281 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len, | |
| 282 kMaxRtpPacketLen); | |
| 283 return media_channel_->SendPacket(&packet); | |
| 284 } | |
| 285 | |
| 286 /////////////////////////////////////////////////////////////////////////// | |
| 287 // Implementation of FileVoiceChannel. | |
| 288 /////////////////////////////////////////////////////////////////////////// | |
| 289 FileVoiceChannel::FileVoiceChannel( | |
| 290 rtc::StreamInterface* input_file_stream, | |
| 291 rtc::StreamInterface* output_file_stream, | |
| 292 rtc::Thread* rtp_sender_thread) | |
| 293 : send_ssrc_(0), | |
| 294 rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream, | |
| 295 output_file_stream, | |
| 296 rtp_sender_thread)) {} | |
| 297 | |
| 298 FileVoiceChannel::~FileVoiceChannel() {} | |
| 299 | |
| 300 bool FileVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) { | |
| 301 // TODO(whyuan): Check the format of RTP dump input. | |
| 302 return true; | |
| 303 } | |
| 304 | |
| 305 bool FileVoiceChannel::SetSend(SendFlags flag) { | |
| 306 return rtp_sender_receiver_->SetSend(flag != SEND_NOTHING); | |
| 307 } | |
| 308 | |
| 309 bool FileVoiceChannel::AddSendStream(const StreamParams& sp) { | |
| 310 if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) { | |
| 311 LOG(LS_ERROR) << "FileVoiceChannel only supports one send stream."; | |
| 312 return false; | |
| 313 } | |
| 314 send_ssrc_ = sp.ssrcs[0]; | |
| 315 rtp_sender_receiver_->SetSendSsrc(send_ssrc_); | |
| 316 return true; | |
| 317 } | |
| 318 | |
| 319 bool FileVoiceChannel::RemoveSendStream(uint32 ssrc) { | |
| 320 if (ssrc != send_ssrc_) | |
| 321 return false; | |
| 322 send_ssrc_ = 0; | |
| 323 rtp_sender_receiver_->SetSendSsrc(send_ssrc_); | |
| 324 return true; | |
| 325 } | |
| 326 | |
| 327 void FileVoiceChannel::OnPacketReceived( | |
| 328 rtc::Buffer* packet, const rtc::PacketTime& packet_time) { | |
| 329 rtp_sender_receiver_->OnPacketReceived(packet); | |
| 330 } | |
| 331 | |
| 332 /////////////////////////////////////////////////////////////////////////// | |
| 333 // Implementation of FileVideoChannel. | |
| 334 /////////////////////////////////////////////////////////////////////////// | |
| 335 FileVideoChannel::FileVideoChannel( | |
| 336 rtc::StreamInterface* input_file_stream, | |
| 337 rtc::StreamInterface* output_file_stream, | |
| 338 rtc::Thread* rtp_sender_thread) | |
| 339 : send_ssrc_(0), | |
| 340 rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream, | |
| 341 output_file_stream, | |
| 342 rtp_sender_thread)) {} | |
| 343 | |
| 344 FileVideoChannel::~FileVideoChannel() {} | |
| 345 | |
| 346 bool FileVideoChannel::SetSendCodecs(const std::vector<VideoCodec>& codecs) { | |
| 347 // TODO(whyuan): Check the format of RTP dump input. | |
| 348 return true; | |
| 349 } | |
| 350 | |
| 351 bool FileVideoChannel::SetSend(bool send) { | |
| 352 return rtp_sender_receiver_->SetSend(send); | |
| 353 } | |
| 354 | |
| 355 bool FileVideoChannel::AddSendStream(const StreamParams& sp) { | |
| 356 if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) { | |
| 357 LOG(LS_ERROR) << "FileVideoChannel only support one send stream."; | |
| 358 return false; | |
| 359 } | |
| 360 send_ssrc_ = sp.ssrcs[0]; | |
| 361 rtp_sender_receiver_->SetSendSsrc(send_ssrc_); | |
| 362 return true; | |
| 363 } | |
| 364 | |
| 365 bool FileVideoChannel::RemoveSendStream(uint32 ssrc) { | |
| 366 if (ssrc != send_ssrc_) | |
| 367 return false; | |
| 368 send_ssrc_ = 0; | |
| 369 rtp_sender_receiver_->SetSendSsrc(send_ssrc_); | |
| 370 return true; | |
| 371 } | |
| 372 | |
| 373 void FileVideoChannel::OnPacketReceived( | |
| 374 rtc::Buffer* packet, const rtc::PacketTime& packet_time) { | |
| 375 rtp_sender_receiver_->OnPacketReceived(packet); | |
| 376 } | |
| 377 | |
| 378 } // namespace cricket | |
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Chromium Code Reviews
Issue 1169833004:
Remove FileMediaEngine. (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master