1 /* 2 * libjingle 3 * Copyright 2014 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 #ifdef HAVE_WEBRTC_VIDEO 29 #include "talk/media/webrtc/webrtcvideoengine2.h" 30 31 #include <set> 32 #include <string> 33 34 #include "libyuv/convert_from.h" 35 #include "talk/media/base/videocapturer.h" 36 #include "talk/media/base/videorenderer.h" 37 #include "talk/media/webrtc/constants.h" 38 #include "talk/media/webrtc/webrtcvideocapturer.h" 39 #include "talk/media/webrtc/webrtcvideoframe.h" 40 #include "talk/media/webrtc/webrtcvoiceengine.h" 41 #include "webrtc/base/buffer.h" 42 #include "webrtc/base/logging.h" 43 #include "webrtc/base/stringutils.h" 44 #include "webrtc/call.h" 45 #include "webrtc/video_encoder.h" 46 47 #define UNIMPLEMENTED \ 48 LOG(LS_ERROR) << "Call to unimplemented function " << __FUNCTION__; \ 49 ASSERT(false) 50 51 namespace cricket { 52 53 // This constant is really an on/off, lower-level configurable NACK history 54 // duration hasn't been implemented. 55 static const int kNackHistoryMs = 1000; 56 57 static const int kDefaultQpMax = 56; 58 59 static const int kDefaultRtcpReceiverReportSsrc = 1; 60 61 struct VideoCodecPref { 62 int payload_type; 63 int width; 64 int height; 65 const char* name; 66 int rtx_payload_type; 67 } kDefaultVideoCodecPref = {100, 640, 400, kVp8CodecName, 96}; 68 69 VideoCodecPref kRedPref = {116, -1, -1, kRedCodecName, -1}; 70 VideoCodecPref kUlpfecPref = {117, -1, -1, kUlpfecCodecName, -1}; 71 72 static bool FindFirstMatchingCodec(const std::vector<VideoCodec>& codecs, 73 const VideoCodec& requested_codec, 74 VideoCodec* matching_codec) { 75 for (size_t i = 0; i < codecs.size(); ++i) { 76 if (requested_codec.Matches(codecs[i])) { 77 *matching_codec = codecs[i]; 78 return true; 79 } 80 } 81 return false; 82 } 83 84 static void AddDefaultFeedbackParams(VideoCodec* codec) { 85 const FeedbackParam kFir(kRtcpFbParamCcm, kRtcpFbCcmParamFir); 86 codec->AddFeedbackParam(kFir); 87 const FeedbackParam kNack(kRtcpFbParamNack, kParamValueEmpty); 88 codec->AddFeedbackParam(kNack); 89 const FeedbackParam kPli(kRtcpFbParamNack, kRtcpFbNackParamPli); 90 codec->AddFeedbackParam(kPli); 91 const FeedbackParam kRemb(kRtcpFbParamRemb, kParamValueEmpty); 92 codec->AddFeedbackParam(kRemb); 93 } 94 95 static bool IsNackEnabled(const VideoCodec& codec) { 96 return codec.HasFeedbackParam( 97 FeedbackParam(kRtcpFbParamNack, kParamValueEmpty)); 98 } 99 100 static bool IsRembEnabled(const VideoCodec& codec) { 101 return codec.HasFeedbackParam( 102 FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty)); 103 } 104 105 static VideoCodec DefaultVideoCodec() { 106 VideoCodec default_codec(kDefaultVideoCodecPref.payload_type, 107 kDefaultVideoCodecPref.name, 108 kDefaultVideoCodecPref.width, 109 kDefaultVideoCodecPref.height, 110 kDefaultFramerate, 111 0); 112 AddDefaultFeedbackParams(&default_codec); 113 return default_codec; 114 } 115 116 static VideoCodec DefaultRedCodec() { 117 return VideoCodec(kRedPref.payload_type, kRedPref.name, 0, 0, 0, 0); 118 } 119 120 static VideoCodec DefaultUlpfecCodec() { 121 return VideoCodec(kUlpfecPref.payload_type, kUlpfecPref.name, 0, 0, 0, 0); 122 } 123 124 static std::vector<VideoCodec> DefaultVideoCodecs() { 125 std::vector<VideoCodec> codecs; 126 codecs.push_back(DefaultVideoCodec()); 127 codecs.push_back(DefaultRedCodec()); 128 codecs.push_back(DefaultUlpfecCodec()); 129 if (kDefaultVideoCodecPref.rtx_payload_type != -1) { 130 codecs.push_back( 131 VideoCodec::CreateRtxCodec(kDefaultVideoCodecPref.rtx_payload_type, 132 kDefaultVideoCodecPref.payload_type)); 133 } 134 return codecs; 135 } 136 137 static bool ValidateRtpHeaderExtensionIds( 138 const std::vector<RtpHeaderExtension>& extensions) { 139 std::set<int> extensions_used; 140 for (size_t i = 0; i < extensions.size(); ++i) { 141 if (extensions[i].id < 0 || extensions[i].id >= 15 || 142 !extensions_used.insert(extensions[i].id).second) { 143 LOG(LS_ERROR) << "RTP extensions are with incorrect or duplicate ids."; 144 return false; 145 } 146 } 147 return true; 148 } 149 150 static std::vector<webrtc::RtpExtension> FilterRtpExtensions( 151 const std::vector<RtpHeaderExtension>& extensions) { 152 std::vector<webrtc::RtpExtension> webrtc_extensions; 153 for (size_t i = 0; i < extensions.size(); ++i) { 154 // Unsupported extensions will be ignored. 155 if (webrtc::RtpExtension::IsSupported(extensions[i].uri)) { 156 webrtc_extensions.push_back(webrtc::RtpExtension( 157 extensions[i].uri, extensions[i].id)); 158 } else { 159 LOG(LS_WARNING) << "Unsupported RTP extension: " << extensions[i].uri; 160 } 161 } 162 return webrtc_extensions; 163 } 164 165 WebRtcVideoEncoderFactory2::~WebRtcVideoEncoderFactory2() { 166 } 167 168 std::vector<webrtc::VideoStream> WebRtcVideoEncoderFactory2::CreateVideoStreams( 169 const VideoCodec& codec, 170 const VideoOptions& options, 171 size_t num_streams) { 172 assert(SupportsCodec(codec)); 173 if (num_streams != 1) { 174 LOG(LS_ERROR) << "Unsupported number of streams: " << num_streams; 175 return std::vector<webrtc::VideoStream>(); 176 } 177 178 webrtc::VideoStream stream; 179 stream.width = codec.width; 180 stream.height = codec.height; 181 stream.max_framerate = 182 codec.framerate != 0 ? codec.framerate : kDefaultFramerate; 183 184 int min_bitrate = kMinVideoBitrate; 185 codec.GetParam(kCodecParamMinBitrate, &min_bitrate); 186 int max_bitrate = kMaxVideoBitrate; 187 codec.GetParam(kCodecParamMaxBitrate, &max_bitrate); 188 stream.min_bitrate_bps = min_bitrate * 1000; 189 stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate * 1000; 190 191 int max_qp = kDefaultQpMax; 192 codec.GetParam(kCodecParamMaxQuantization, &max_qp); 193 stream.max_qp = max_qp; 194 std::vector<webrtc::VideoStream> streams; 195 streams.push_back(stream); 196 return streams; 197 } 198 199 webrtc::VideoEncoder* WebRtcVideoEncoderFactory2::CreateVideoEncoder( 200 const VideoCodec& codec, 201 const VideoOptions& options) { 202 assert(SupportsCodec(codec)); 203 if (_stricmp(codec.name.c_str(), kVp8CodecName) == 0) { 204 return webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp8); 205 } 206 // This shouldn't happen, we should be able to create encoders for all codecs 207 // we support. 208 assert(false); 209 return NULL; 210 } 211 212 void* WebRtcVideoEncoderFactory2::CreateVideoEncoderSettings( 213 const VideoCodec& codec, 214 const VideoOptions& options) { 215 assert(SupportsCodec(codec)); 216 if (_stricmp(codec.name.c_str(), kVp8CodecName) == 0) { 217 webrtc::VideoCodecVP8* settings = new webrtc::VideoCodecVP8( 218 webrtc::VideoEncoder::GetDefaultVp8Settings()); 219 options.video_noise_reduction.Get(&settings->denoisingOn); 220 return settings; 221 } 222 return NULL; 223 } 224 225 void WebRtcVideoEncoderFactory2::DestroyVideoEncoderSettings( 226 const VideoCodec& codec, 227 void* encoder_settings) { 228 assert(SupportsCodec(codec)); 229 if (encoder_settings == NULL) { 230 return; 231 } 232 if (_stricmp(codec.name.c_str(), kVp8CodecName) == 0) { 233 delete reinterpret_cast<webrtc::VideoCodecVP8*>(encoder_settings); 234 } 235 } 236 237 bool WebRtcVideoEncoderFactory2::SupportsCodec(const VideoCodec& codec) { 238 return _stricmp(codec.name.c_str(), kVp8CodecName) == 0; 239 } 240 241 DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler() 242 : default_recv_ssrc_(0), default_renderer_(NULL) {} 243 244 UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc( 245 VideoMediaChannel* channel, 246 uint32_t ssrc) { 247 if (default_recv_ssrc_ != 0) { // Already one default stream. 248 LOG(LS_WARNING) << "Unknown SSRC, but default receive stream already set."; 249 return kDropPacket; 250 } 251 252 StreamParams sp; 253 sp.ssrcs.push_back(ssrc); 254 LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << "."; 255 if (!channel->AddRecvStream(sp)) { 256 LOG(LS_WARNING) << "Could not create default receive stream."; 257 } 258 259 channel->SetRenderer(ssrc, default_renderer_); 260 default_recv_ssrc_ = ssrc; 261 return kDeliverPacket; 262 } 263 264 VideoRenderer* DefaultUnsignalledSsrcHandler::GetDefaultRenderer() const { 265 return default_renderer_; 266 } 267 268 void DefaultUnsignalledSsrcHandler::SetDefaultRenderer( 269 VideoMediaChannel* channel, 270 VideoRenderer* renderer) { 271 default_renderer_ = renderer; 272 if (default_recv_ssrc_ != 0) { 273 channel->SetRenderer(default_recv_ssrc_, default_renderer_); 274 } 275 } 276 277 WebRtcVideoEngine2::WebRtcVideoEngine2() 278 : worker_thread_(NULL), 279 voice_engine_(NULL), 280 video_codecs_(DefaultVideoCodecs()), 281 default_codec_format_(kDefaultVideoCodecPref.width, 282 kDefaultVideoCodecPref.height, 283 FPS_TO_INTERVAL(kDefaultFramerate), 284 FOURCC_ANY), 285 initialized_(false), 286 cpu_monitor_(new rtc::CpuMonitor(NULL)), 287 channel_factory_(NULL), 288 external_decoder_factory_(NULL), 289 external_encoder_factory_(NULL) { 290 LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()"; 291 rtp_header_extensions_.push_back( 292 RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension, 293 kRtpTimestampOffsetHeaderExtensionDefaultId)); 294 rtp_header_extensions_.push_back( 295 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, 296 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); 297 } 298 299 void WebRtcVideoEngine2::SetChannelFactory( 300 WebRtcVideoChannelFactory* channel_factory) { 301 channel_factory_ = channel_factory; 302 } 303 304 WebRtcVideoEngine2::~WebRtcVideoEngine2() { 305 LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2"; 306 307 if (initialized_) { 308 Terminate(); 309 } 310 } 311 312 bool WebRtcVideoEngine2::Init(rtc::Thread* worker_thread) { 313 LOG(LS_INFO) << "WebRtcVideoEngine2::Init"; 314 worker_thread_ = worker_thread; 315 ASSERT(worker_thread_ != NULL); 316 317 cpu_monitor_->set_thread(worker_thread_); 318 if (!cpu_monitor_->Start(kCpuMonitorPeriodMs)) { 319 LOG(LS_ERROR) << "Failed to start CPU monitor."; 320 cpu_monitor_.reset(); 321 } 322 323 initialized_ = true; 324 return true; 325 } 326 327 void WebRtcVideoEngine2::Terminate() { 328 LOG(LS_INFO) << "WebRtcVideoEngine2::Terminate"; 329 330 cpu_monitor_->Stop(); 331 332 initialized_ = false; 333 } 334 335 int WebRtcVideoEngine2::GetCapabilities() { return VIDEO_RECV | VIDEO_SEND; } 336 337 bool WebRtcVideoEngine2::SetDefaultEncoderConfig( 338 const VideoEncoderConfig& config) { 339 const VideoCodec& codec = config.max_codec; 340 // TODO(pbos): Make use of external encoder factory. 341 if (!GetVideoEncoderFactory()->SupportsCodec(codec)) { 342 LOG(LS_ERROR) << "SetDefaultEncoderConfig, codec not supported:" 343 << codec.ToString(); 344 return false; 345 } 346 347 default_codec_format_ = 348 VideoFormat(codec.width, 349 codec.height, 350 VideoFormat::FpsToInterval(codec.framerate), 351 FOURCC_ANY); 352 video_codecs_.clear(); 353 video_codecs_.push_back(codec); 354 return true; 355 } 356 357 VideoEncoderConfig WebRtcVideoEngine2::GetDefaultEncoderConfig() const { 358 return VideoEncoderConfig(DefaultVideoCodec()); 359 } 360 361 WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel( 362 VoiceMediaChannel* voice_channel) { 363 LOG(LS_INFO) << "CreateChannel: " 364 << (voice_channel != NULL ? "With" : "Without") 365 << " voice channel."; 366 WebRtcVideoChannel2* channel = 367 channel_factory_ != NULL 368 ? channel_factory_->Create(this, voice_channel) 369 : new WebRtcVideoChannel2( 370 this, voice_channel, GetVideoEncoderFactory()); 371 if (!channel->Init()) { 372 delete channel; 373 return NULL; 374 } 375 channel->SetRecvCodecs(video_codecs_); 376 return channel; 377 } 378 379 const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const { 380 return video_codecs_; 381 } 382 383 const std::vector<RtpHeaderExtension>& 384 WebRtcVideoEngine2::rtp_header_extensions() const { 385 return rtp_header_extensions_; 386 } 387 388 void WebRtcVideoEngine2::SetLogging(int min_sev, const char* filter) { 389 // TODO(pbos): Set up logging. 390 LOG(LS_VERBOSE) << "SetLogging: " << min_sev << '"' << filter << '"'; 391 // if min_sev == -1, we keep the current log level. 392 if (min_sev < 0) { 393 assert(min_sev == -1); 394 return; 395 } 396 } 397 398 void WebRtcVideoEngine2::SetExternalDecoderFactory( 399 WebRtcVideoDecoderFactory* decoder_factory) { 400 external_decoder_factory_ = decoder_factory; 401 } 402 403 void WebRtcVideoEngine2::SetExternalEncoderFactory( 404 WebRtcVideoEncoderFactory* encoder_factory) { 405 if (external_encoder_factory_ == encoder_factory) { 406 return; 407 } 408 if (external_encoder_factory_) { 409 external_encoder_factory_->RemoveObserver(this); 410 } 411 external_encoder_factory_ = encoder_factory; 412 if (external_encoder_factory_) { 413 external_encoder_factory_->AddObserver(this); 414 } 415 416 // Invoke OnCodecAvailable() here in case the list of codecs is already 417 // available when the encoder factory is installed. If not the encoder 418 // factory will invoke the callback later when the codecs become available. 419 OnCodecsAvailable(); 420 } 421 422 bool WebRtcVideoEngine2::EnableTimedRender() { 423 // TODO(pbos): Figure out whether this can be removed. 424 return true; 425 } 426 427 // Checks to see whether we comprehend and could receive a particular codec 428 bool WebRtcVideoEngine2::FindCodec(const VideoCodec& in) { 429 // TODO(pbos): Probe encoder factory to figure out that the codec is supported 430 // if supported by the encoder factory. Add a corresponding test that fails 431 // with this code (that doesn't ask the factory). 432 for (size_t j = 0; j < video_codecs_.size(); ++j) { 433 VideoCodec codec(video_codecs_[j].id, video_codecs_[j].name, 0, 0, 0, 0); 434 if (codec.Matches(in)) { 435 return true; 436 } 437 } 438 return false; 439 } 440 441 // Tells whether the |requested| codec can be transmitted or not. If it can be 442 // transmitted |out| is set with the best settings supported. Aspect ratio will 443 // be set as close to |current|'s as possible. If not set |requested|'s 444 // dimensions will be used for aspect ratio matching. 445 bool WebRtcVideoEngine2::CanSendCodec(const VideoCodec& requested, 446 const VideoCodec& current, 447 VideoCodec* out) { 448 assert(out != NULL); 449 450 if (requested.width != requested.height && 451 (requested.height == 0 || requested.width == 0)) { 452 // 0xn and nx0 are invalid resolutions. 453 return false; 454 } 455 456 VideoCodec matching_codec; 457 if (!FindFirstMatchingCodec(video_codecs_, requested, &matching_codec)) { 458 // Codec not supported. 459 return false; 460 } 461 462 out->id = requested.id; 463 out->name = requested.name; 464 out->preference = requested.preference; 465 out->params = requested.params; 466 out->framerate = 467 rtc::_min(requested.framerate, matching_codec.framerate); 468 out->params = requested.params; 469 out->feedback_params = requested.feedback_params; 470 out->width = requested.width; 471 out->height = requested.height; 472 if (requested.width == 0 && requested.height == 0) { 473 return true; 474 } 475 476 while (out->width > matching_codec.width) { 477 out->width /= 2; 478 out->height /= 2; 479 } 480 481 return out->width > 0 && out->height > 0; 482 } 483 484 bool WebRtcVideoEngine2::SetVoiceEngine(WebRtcVoiceEngine* voice_engine) { 485 if (initialized_) { 486 LOG(LS_WARNING) << "SetVoiceEngine can not be called after Init"; 487 return false; 488 } 489 voice_engine_ = voice_engine; 490 return true; 491 } 492 493 // Ignore spammy trace messages, mostly from the stats API when we haven't 494 // gotten RTCP info yet from the remote side. 495 bool WebRtcVideoEngine2::ShouldIgnoreTrace(const std::string& trace) { 496 static const char* const kTracesToIgnore[] = {NULL}; 497 for (const char* const* p = kTracesToIgnore; *p; ++p) { 498 if (trace.find(*p) == 0) { 499 return true; 500 } 501 } 502 return false; 503 } 504 505 WebRtcVideoEncoderFactory2* WebRtcVideoEngine2::GetVideoEncoderFactory() { 506 return &default_video_encoder_factory_; 507 } 508 509 void WebRtcVideoEngine2::OnCodecsAvailable() { 510 // TODO(pbos): Implement. 511 } 512 // Thin map between VideoFrame and an existing webrtc::I420VideoFrame 513 // to avoid having to copy the rendered VideoFrame prematurely. 514 // This implementation is only safe to use in a const context and should never 515 // be written to. 516 class WebRtcVideoRenderFrame : public VideoFrame { 517 public: 518 explicit WebRtcVideoRenderFrame(const webrtc::I420VideoFrame* frame) 519 : frame_(frame) {} 520 521 virtual bool InitToBlack(int w, 522 int h, 523 size_t pixel_width, 524 size_t pixel_height, 525 int64 elapsed_time, 526 int64 time_stamp) OVERRIDE { 527 UNIMPLEMENTED; 528 return false; 529 } 530 531 virtual bool Reset(uint32 fourcc, 532 int w, 533 int h, 534 int dw, 535 int dh, 536 uint8* sample, 537 size_t sample_size, 538 size_t pixel_width, 539 size_t pixel_height, 540 int64 elapsed_time, 541 int64 time_stamp, 542 int rotation) OVERRIDE { 543 UNIMPLEMENTED; 544 return false; 545 } 546 547 virtual size_t GetWidth() const OVERRIDE { 548 return static_cast<size_t>(frame_->width()); 549 } 550 virtual size_t GetHeight() const OVERRIDE { 551 return static_cast<size_t>(frame_->height()); 552 } 553 554 virtual const uint8* GetYPlane() const OVERRIDE { 555 return frame_->buffer(webrtc::kYPlane); 556 } 557 virtual const uint8* GetUPlane() const OVERRIDE { 558 return frame_->buffer(webrtc::kUPlane); 559 } 560 virtual const uint8* GetVPlane() const OVERRIDE { 561 return frame_->buffer(webrtc::kVPlane); 562 } 563 564 virtual uint8* GetYPlane() OVERRIDE { 565 UNIMPLEMENTED; 566 return NULL; 567 } 568 virtual uint8* GetUPlane() OVERRIDE { 569 UNIMPLEMENTED; 570 return NULL; 571 } 572 virtual uint8* GetVPlane() OVERRIDE { 573 UNIMPLEMENTED; 574 return NULL; 575 } 576 577 virtual int32 GetYPitch() const OVERRIDE { 578 return frame_->stride(webrtc::kYPlane); 579 } 580 virtual int32 GetUPitch() const OVERRIDE { 581 return frame_->stride(webrtc::kUPlane); 582 } 583 virtual int32 GetVPitch() const OVERRIDE { 584 return frame_->stride(webrtc::kVPlane); 585 } 586 587 virtual void* GetNativeHandle() const OVERRIDE { return NULL; } 588 589 virtual size_t GetPixelWidth() const OVERRIDE { return 1; } 590 virtual size_t GetPixelHeight() const OVERRIDE { return 1; } 591 592 virtual int64 GetElapsedTime() const OVERRIDE { 593 // Convert millisecond render time to ns timestamp. 594 return frame_->render_time_ms() * rtc::kNumNanosecsPerMillisec; 595 } 596 virtual int64 GetTimeStamp() const OVERRIDE { 597 // Convert 90K rtp timestamp to ns timestamp. 598 return (frame_->timestamp() / 90) * rtc::kNumNanosecsPerMillisec; 599 } 600 virtual void SetElapsedTime(int64 elapsed_time) OVERRIDE { UNIMPLEMENTED; } 601 virtual void SetTimeStamp(int64 time_stamp) OVERRIDE { UNIMPLEMENTED; } 602 603 virtual int GetRotation() const OVERRIDE { 604 UNIMPLEMENTED; 605 return ROTATION_0; 606 } 607 608 virtual VideoFrame* Copy() const OVERRIDE { 609 UNIMPLEMENTED; 610 return NULL; 611 } 612 613 virtual bool MakeExclusive() OVERRIDE { 614 UNIMPLEMENTED; 615 return false; 616 } 617 618 virtual size_t CopyToBuffer(uint8* buffer, size_t size) const { 619 UNIMPLEMENTED; 620 return 0; 621 } 622 623 // TODO(fbarchard): Refactor into base class and share with LMI 624 virtual size_t ConvertToRgbBuffer(uint32 to_fourcc, 625 uint8* buffer, 626 size_t size, 627 int stride_rgb) const OVERRIDE { 628 size_t width = GetWidth(); 629 size_t height = GetHeight(); 630 size_t needed = (stride_rgb >= 0 ? stride_rgb : -stride_rgb) * height; 631 if (size < needed) { 632 LOG(LS_WARNING) << "RGB buffer is not large enough"; 633 return needed; 634 } 635 636 if (libyuv::ConvertFromI420(GetYPlane(), 637 GetYPitch(), 638 GetUPlane(), 639 GetUPitch(), 640 GetVPlane(), 641 GetVPitch(), 642 buffer, 643 stride_rgb, 644 static_cast<int>(width), 645 static_cast<int>(height), 646 to_fourcc)) { 647 LOG(LS_ERROR) << "RGB type not supported: " << to_fourcc; 648 return 0; // 0 indicates error 649 } 650 return needed; 651 } 652 653 protected: 654 virtual VideoFrame* CreateEmptyFrame(int w, 655 int h, 656 size_t pixel_width, 657 size_t pixel_height, 658 int64 elapsed_time, 659 int64 time_stamp) const OVERRIDE { 660 WebRtcVideoFrame* frame = new WebRtcVideoFrame(); 661 frame->InitToBlack( 662 w, h, pixel_width, pixel_height, elapsed_time, time_stamp); 663 return frame; 664 } 665 666 private: 667 const webrtc::I420VideoFrame* const frame_; 668 }; 669 670 WebRtcVideoChannel2::WebRtcVideoChannel2( 671 WebRtcVideoEngine2* engine, 672 VoiceMediaChannel* voice_channel, 673 WebRtcVideoEncoderFactory2* encoder_factory) 674 : unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_), 675 encoder_factory_(encoder_factory) { 676 // TODO(pbos): Connect the video and audio with |voice_channel|. 677 webrtc::Call::Config config(this); 678 Construct(webrtc::Call::Create(config), engine); 679 } 680 681 WebRtcVideoChannel2::WebRtcVideoChannel2( 682 webrtc::Call* call, 683 WebRtcVideoEngine2* engine, 684 WebRtcVideoEncoderFactory2* encoder_factory) 685 : unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_), 686 encoder_factory_(encoder_factory) { 687 Construct(call, engine); 688 } 689 690 void WebRtcVideoChannel2::Construct(webrtc::Call* call, 691 WebRtcVideoEngine2* engine) { 692 rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc; 693 sending_ = false; 694 call_.reset(call); 695 default_send_ssrc_ = 0; 696 697 SetDefaultOptions(); 698 } 699 700 void WebRtcVideoChannel2::SetDefaultOptions() { 701 options_.video_noise_reduction.Set(true); 702 options_.use_payload_padding.Set(false); 703 options_.suspend_below_min_bitrate.Set(false); 704 } 705 706 WebRtcVideoChannel2::~WebRtcVideoChannel2() { 707 for (std::map<uint32, WebRtcVideoSendStream*>::iterator it = 708 send_streams_.begin(); 709 it != send_streams_.end(); 710 ++it) { 711 delete it->second; 712 } 713 714 for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it = 715 receive_streams_.begin(); 716 it != receive_streams_.end(); 717 ++it) { 718 delete it->second; 719 } 720 } 721 722 bool WebRtcVideoChannel2::Init() { return true; } 723 724 namespace { 725 726 static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) { 727 std::stringstream out; 728 out << '{'; 729 for (size_t i = 0; i < codecs.size(); ++i) { 730 out << codecs[i].ToString(); 731 if (i != codecs.size() - 1) { 732 out << ", "; 733 } 734 } 735 out << '}'; 736 return out.str(); 737 } 738 739 static bool ValidateCodecFormats(const std::vector<VideoCodec>& codecs) { 740 bool has_video = false; 741 for (size_t i = 0; i < codecs.size(); ++i) { 742 if (!codecs[i].ValidateCodecFormat()) { 743 return false; 744 } 745 if (codecs[i].GetCodecType() == VideoCodec::CODEC_VIDEO) { 746 has_video = true; 747 } 748 } 749 if (!has_video) { 750 LOG(LS_ERROR) << "Setting codecs without a video codec is invalid: " 751 << CodecVectorToString(codecs); 752 return false; 753 } 754 return true; 755 } 756 757 static std::string RtpExtensionsToString( 758 const std::vector<RtpHeaderExtension>& extensions) { 759 std::stringstream out; 760 out << '{'; 761 for (size_t i = 0; i < extensions.size(); ++i) { 762 out << "{" << extensions[i].uri << ": " << extensions[i].id << "}"; 763 if (i != extensions.size() - 1) { 764 out << ", "; 765 } 766 } 767 out << '}'; 768 return out.str(); 769 } 770 771 } // namespace 772 773 bool WebRtcVideoChannel2::SetRecvCodecs(const std::vector<VideoCodec>& codecs) { 774 LOG(LS_INFO) << "SetRecvCodecs: " << CodecVectorToString(codecs); 775 if (!ValidateCodecFormats(codecs)) { 776 return false; 777 } 778 779 const std::vector<VideoCodecSettings> mapped_codecs = MapCodecs(codecs); 780 if (mapped_codecs.empty()) { 781 LOG(LS_ERROR) << "SetRecvCodecs called without video codec payloads."; 782 return false; 783 } 784 785 // TODO(pbos): Add a decoder factory which controls supported codecs. 786 // Blocked on webrtc:2854. 787 for (size_t i = 0; i < mapped_codecs.size(); ++i) { 788 if (_stricmp(mapped_codecs[i].codec.name.c_str(), kVp8CodecName) != 0) { 789 LOG(LS_ERROR) << "SetRecvCodecs called with unsupported codec: '" 790 << mapped_codecs[i].codec.name << "'"; 791 return false; 792 } 793 } 794 795 recv_codecs_ = mapped_codecs; 796 797 for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it = 798 receive_streams_.begin(); 799 it != receive_streams_.end(); 800 ++it) { 801 it->second->SetRecvCodecs(recv_codecs_); 802 } 803 804 return true; 805 } 806 807 bool WebRtcVideoChannel2::SetSendCodecs(const std::vector<VideoCodec>& codecs) { 808 LOG(LS_INFO) << "SetSendCodecs: " << CodecVectorToString(codecs); 809 if (!ValidateCodecFormats(codecs)) { 810 return false; 811 } 812 813 const std::vector<VideoCodecSettings> supported_codecs = 814 FilterSupportedCodecs(MapCodecs(codecs)); 815 816 if (supported_codecs.empty()) { 817 LOG(LS_ERROR) << "No video codecs supported by encoder factory."; 818 return false; 819 } 820 821 send_codec_.Set(supported_codecs.front()); 822 LOG(LS_INFO) << "Using codec: " << supported_codecs.front().codec.ToString(); 823 824 for (std::map<uint32, WebRtcVideoSendStream*>::iterator it = 825 send_streams_.begin(); 826 it != send_streams_.end(); 827 ++it) { 828 assert(it->second != NULL); 829 it->second->SetCodec(supported_codecs.front()); 830 } 831 832 return true; 833 } 834 835 bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) { 836 VideoCodecSettings codec_settings; 837 if (!send_codec_.Get(&codec_settings)) { 838 LOG(LS_VERBOSE) << "GetSendCodec: No send codec set."; 839 return false; 840 } 841 *codec = codec_settings.codec; 842 return true; 843 } 844 845 bool WebRtcVideoChannel2::SetSendStreamFormat(uint32 ssrc, 846 const VideoFormat& format) { 847 LOG(LS_VERBOSE) << "SetSendStreamFormat:" << ssrc << " -> " 848 << format.ToString(); 849 if (send_streams_.find(ssrc) == send_streams_.end()) { 850 return false; 851 } 852 return send_streams_[ssrc]->SetVideoFormat(format); 853 } 854 855 bool WebRtcVideoChannel2::SetRender(bool render) { 856 // TODO(pbos): Implement. Or refactor away as it shouldn't be needed. 857 LOG(LS_VERBOSE) << "SetRender: " << (render ? "true" : "false"); 858 return true; 859 } 860 861 bool WebRtcVideoChannel2::SetSend(bool send) { 862 LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false"); 863 if (send && !send_codec_.IsSet()) { 864 LOG(LS_ERROR) << "SetSend(true) called before setting codec."; 865 return false; 866 } 867 if (send) { 868 StartAllSendStreams(); 869 } else { 870 StopAllSendStreams(); 871 } 872 sending_ = send; 873 return true; 874 } 875 876 bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) { 877 LOG(LS_INFO) << "AddSendStream: " << sp.ToString(); 878 if (sp.ssrcs.empty()) { 879 LOG(LS_ERROR) << "No SSRCs in stream parameters."; 880 return false; 881 } 882 883 uint32 ssrc = sp.first_ssrc(); 884 assert(ssrc != 0); 885 // TODO(pbos): Make sure none of sp.ssrcs are used, not just the identifying 886 // ssrc. 887 if (send_streams_.find(ssrc) != send_streams_.end()) { 888 LOG(LS_ERROR) << "Send stream with ssrc '" << ssrc << "' already exists."; 889 return false; 890 } 891 892 std::vector<uint32> primary_ssrcs; 893 sp.GetPrimarySsrcs(&primary_ssrcs); 894 std::vector<uint32> rtx_ssrcs; 895 sp.GetFidSsrcs(primary_ssrcs, &rtx_ssrcs); 896 if (!rtx_ssrcs.empty() && primary_ssrcs.size() != rtx_ssrcs.size()) { 897 LOG(LS_ERROR) 898 << "RTX SSRCs exist, but don't cover all SSRCs (unsupported): " 899 << sp.ToString(); 900 return false; 901 } 902 903 WebRtcVideoSendStream* stream = 904 new WebRtcVideoSendStream(call_.get(), 905 encoder_factory_, 906 options_, 907 send_codec_, 908 sp, 909 send_rtp_extensions_); 910 911 send_streams_[ssrc] = stream; 912 913 if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) { 914 rtcp_receiver_report_ssrc_ = ssrc; 915 } 916 if (default_send_ssrc_ == 0) { 917 default_send_ssrc_ = ssrc; 918 } 919 if (sending_) { 920 stream->Start(); 921 } 922 923 return true; 924 } 925 926 bool WebRtcVideoChannel2::RemoveSendStream(uint32 ssrc) { 927 LOG(LS_INFO) << "RemoveSendStream: " << ssrc; 928 929 if (ssrc == 0) { 930 if (default_send_ssrc_ == 0) { 931 LOG(LS_ERROR) << "No default send stream active."; 932 return false; 933 } 934 935 LOG(LS_VERBOSE) << "Removing default stream: " << default_send_ssrc_; 936 ssrc = default_send_ssrc_; 937 } 938 939 std::map<uint32, WebRtcVideoSendStream*>::iterator it = 940 send_streams_.find(ssrc); 941 if (it == send_streams_.end()) { 942 return false; 943 } 944 945 delete it->second; 946 send_streams_.erase(it); 947 948 if (ssrc == default_send_ssrc_) { 949 default_send_ssrc_ = 0; 950 } 951 952 return true; 953 } 954 955 bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp) { 956 LOG(LS_INFO) << "AddRecvStream: " << sp.ToString(); 957 assert(sp.ssrcs.size() > 0); 958 959 uint32 ssrc = sp.first_ssrc(); 960 assert(ssrc != 0); // TODO(pbos): Is this ever valid? 961 962 // TODO(pbos): Check if any of the SSRCs overlap. 963 if (receive_streams_.find(ssrc) != receive_streams_.end()) { 964 LOG(LS_ERROR) << "Receive stream for SSRC " << ssrc << "already exists."; 965 return false; 966 } 967 968 webrtc::VideoReceiveStream::Config config; 969 ConfigureReceiverRtp(&config, sp); 970 receive_streams_[ssrc] = 971 new WebRtcVideoReceiveStream(call_.get(), config, recv_codecs_); 972 973 return true; 974 } 975 976 void WebRtcVideoChannel2::ConfigureReceiverRtp( 977 webrtc::VideoReceiveStream::Config* config, 978 const StreamParams& sp) const { 979 uint32 ssrc = sp.first_ssrc(); 980 981 config->rtp.remote_ssrc = ssrc; 982 config->rtp.local_ssrc = rtcp_receiver_report_ssrc_; 983 984 config->rtp.extensions = recv_rtp_extensions_; 985 986 // TODO(pbos): This protection is against setting the same local ssrc as 987 // remote which is not permitted by the lower-level API. RTCP requires a 988 // corresponding sender SSRC. Figure out what to do when we don't have 989 // (receive-only) or know a good local SSRC. 990 if (config->rtp.remote_ssrc == config->rtp.local_ssrc) { 991 if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) { 992 config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc; 993 } else { 994 config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc + 1; 995 } 996 } 997 998 for (size_t i = 0; i < recv_codecs_.size(); ++i) { 999 if (recv_codecs_[i].codec.id == kDefaultVideoCodecPref.payload_type) { 1000 config->rtp.fec = recv_codecs_[i].fec; 1001 uint32 rtx_ssrc; 1002 if (recv_codecs_[i].rtx_payload_type != -1 && 1003 sp.GetFidSsrc(ssrc, &rtx_ssrc)) { 1004 config->rtp.rtx[kDefaultVideoCodecPref.payload_type].ssrc = rtx_ssrc; 1005 config->rtp.rtx[kDefaultVideoCodecPref.payload_type].payload_type = 1006 recv_codecs_[i].rtx_payload_type; 1007 } 1008 break; 1009 } 1010 } 1011 1012 } 1013 1014 bool WebRtcVideoChannel2::RemoveRecvStream(uint32 ssrc) { 1015 LOG(LS_INFO) << "RemoveRecvStream: " << ssrc; 1016 if (ssrc == 0) { 1017 LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported."; 1018 return false; 1019 } 1020 1021 std::map<uint32, WebRtcVideoReceiveStream*>::iterator stream = 1022 receive_streams_.find(ssrc); 1023 if (stream == receive_streams_.end()) { 1024 LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc; 1025 return false; 1026 } 1027 delete stream->second; 1028 receive_streams_.erase(stream); 1029 1030 return true; 1031 } 1032 1033 bool WebRtcVideoChannel2::SetRenderer(uint32 ssrc, VideoRenderer* renderer) { 1034 LOG(LS_INFO) << "SetRenderer: ssrc:" << ssrc << " " 1035 << (renderer ? "(ptr)" : "NULL"); 1036 if (ssrc == 0) { 1037 default_unsignalled_ssrc_handler_.SetDefaultRenderer(this, renderer); 1038 return true; 1039 } 1040 1041 std::map<uint32, WebRtcVideoReceiveStream*>::iterator it = 1042 receive_streams_.find(ssrc); 1043 if (it == receive_streams_.end()) { 1044 return false; 1045 } 1046 1047 it->second->SetRenderer(renderer); 1048 return true; 1049 } 1050 1051 bool WebRtcVideoChannel2::GetRenderer(uint32 ssrc, VideoRenderer** renderer) { 1052 if (ssrc == 0) { 1053 *renderer = default_unsignalled_ssrc_handler_.GetDefaultRenderer(); 1054 return *renderer != NULL; 1055 } 1056 1057 std::map<uint32, WebRtcVideoReceiveStream*>::iterator it = 1058 receive_streams_.find(ssrc); 1059 if (it == receive_streams_.end()) { 1060 return false; 1061 } 1062 *renderer = it->second->GetRenderer(); 1063 return true; 1064 } 1065 1066 bool WebRtcVideoChannel2::GetStats(const StatsOptions& options, 1067 VideoMediaInfo* info) { 1068 info->Clear(); 1069 FillSenderStats(info); 1070 FillReceiverStats(info); 1071 FillBandwidthEstimationStats(info); 1072 return true; 1073 } 1074 1075 void WebRtcVideoChannel2::FillSenderStats(VideoMediaInfo* video_media_info) { 1076 for (std::map<uint32, WebRtcVideoSendStream*>::iterator it = 1077 send_streams_.begin(); 1078 it != send_streams_.end(); 1079 ++it) { 1080 video_media_info->senders.push_back(it->second->GetVideoSenderInfo()); 1081 } 1082 } 1083 1084 void WebRtcVideoChannel2::FillReceiverStats(VideoMediaInfo* video_media_info) { 1085 for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it = 1086 receive_streams_.begin(); 1087 it != receive_streams_.end(); 1088 ++it) { 1089 video_media_info->receivers.push_back(it->second->GetVideoReceiverInfo()); 1090 } 1091 } 1092 1093 void WebRtcVideoChannel2::FillBandwidthEstimationStats( 1094 VideoMediaInfo* video_media_info) { 1095 // TODO(pbos): Implement. 1096 } 1097 1098 bool WebRtcVideoChannel2::SetCapturer(uint32 ssrc, VideoCapturer* capturer) { 1099 LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> " 1100 << (capturer != NULL ? "(capturer)" : "NULL"); 1101 assert(ssrc != 0); 1102 if (send_streams_.find(ssrc) == send_streams_.end()) { 1103 LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; 1104 return false; 1105 } 1106 return send_streams_[ssrc]->SetCapturer(capturer); 1107 } 1108 1109 bool WebRtcVideoChannel2::SendIntraFrame() { 1110 // TODO(pbos): Implement. 1111 LOG(LS_VERBOSE) << "SendIntraFrame()."; 1112 return true; 1113 } 1114 1115 bool WebRtcVideoChannel2::RequestIntraFrame() { 1116 // TODO(pbos): Implement. 1117 LOG(LS_VERBOSE) << "SendIntraFrame()."; 1118 return true; 1119 } 1120 1121 void WebRtcVideoChannel2::OnPacketReceived( 1122 rtc::Buffer* packet, 1123 const rtc::PacketTime& packet_time) { 1124 const webrtc::PacketReceiver::DeliveryStatus delivery_result = 1125 call_->Receiver()->DeliverPacket( 1126 reinterpret_cast<const uint8_t*>(packet->data()), packet->length()); 1127 switch (delivery_result) { 1128 case webrtc::PacketReceiver::DELIVERY_OK: 1129 return; 1130 case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR: 1131 return; 1132 case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC: 1133 break; 1134 } 1135 1136 uint32 ssrc = 0; 1137 if (!GetRtpSsrc(packet->data(), packet->length(), &ssrc)) { 1138 return; 1139 } 1140 1141 // TODO(pbos): Make sure that the unsignalled SSRC uses the video payload. 1142 // Also figure out whether RTX needs to be handled. 1143 switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) { 1144 case UnsignalledSsrcHandler::kDropPacket: 1145 return; 1146 case UnsignalledSsrcHandler::kDeliverPacket: 1147 break; 1148 } 1149 1150 if (call_->Receiver()->DeliverPacket( 1151 reinterpret_cast<const uint8_t*>(packet->data()), packet->length()) != 1152 webrtc::PacketReceiver::DELIVERY_OK) { 1153 LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery."; 1154 return; 1155 } 1156 } 1157 1158 void WebRtcVideoChannel2::OnRtcpReceived( 1159 rtc::Buffer* packet, 1160 const rtc::PacketTime& packet_time) { 1161 if (call_->Receiver()->DeliverPacket( 1162 reinterpret_cast<const uint8_t*>(packet->data()), packet->length()) != 1163 webrtc::PacketReceiver::DELIVERY_OK) { 1164 LOG(LS_WARNING) << "Failed to deliver RTCP packet."; 1165 } 1166 } 1167 1168 void WebRtcVideoChannel2::OnReadyToSend(bool ready) { 1169 LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready."); 1170 call_->SignalNetworkState(ready ? webrtc::Call::kNetworkUp 1171 : webrtc::Call::kNetworkDown); 1172 } 1173 1174 bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) { 1175 LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> " 1176 << (mute ? "mute" : "unmute"); 1177 assert(ssrc != 0); 1178 if (send_streams_.find(ssrc) == send_streams_.end()) { 1179 LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; 1180 return false; 1181 } 1182 1183 send_streams_[ssrc]->MuteStream(mute); 1184 return true; 1185 } 1186 1187 bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions( 1188 const std::vector<RtpHeaderExtension>& extensions) { 1189 LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: " 1190 << RtpExtensionsToString(extensions); 1191 if (!ValidateRtpHeaderExtensionIds(extensions)) 1192 return false; 1193 1194 recv_rtp_extensions_ = FilterRtpExtensions(extensions); 1195 for (std::map<uint32, WebRtcVideoReceiveStream*>::iterator it = 1196 receive_streams_.begin(); 1197 it != receive_streams_.end(); 1198 ++it) { 1199 it->second->SetRtpExtensions(recv_rtp_extensions_); 1200 } 1201 return true; 1202 } 1203 1204 bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions( 1205 const std::vector<RtpHeaderExtension>& extensions) { 1206 LOG(LS_INFO) << "SetSendRtpHeaderExtensions: " 1207 << RtpExtensionsToString(extensions); 1208 if (!ValidateRtpHeaderExtensionIds(extensions)) 1209 return false; 1210 1211 send_rtp_extensions_ = FilterRtpExtensions(extensions); 1212 for (std::map<uint32, WebRtcVideoSendStream*>::iterator it = 1213 send_streams_.begin(); 1214 it != send_streams_.end(); 1215 ++it) { 1216 it->second->SetRtpExtensions(send_rtp_extensions_); 1217 } 1218 return true; 1219 } 1220 1221 bool WebRtcVideoChannel2::SetStartSendBandwidth(int bps) { 1222 // TODO(pbos): Implement. 1223 LOG(LS_VERBOSE) << "SetStartSendBandwidth: " << bps; 1224 return true; 1225 } 1226 1227 bool WebRtcVideoChannel2::SetMaxSendBandwidth(int bps) { 1228 // TODO(pbos): Implement. 1229 LOG(LS_VERBOSE) << "SetMaxSendBandwidth: " << bps; 1230 return true; 1231 } 1232 1233 bool WebRtcVideoChannel2::SetOptions(const VideoOptions& options) { 1234 LOG(LS_VERBOSE) << "SetOptions: " << options.ToString(); 1235 options_.SetAll(options); 1236 for (std::map<uint32, WebRtcVideoSendStream*>::iterator it = 1237 send_streams_.begin(); 1238 it != send_streams_.end(); 1239 ++it) { 1240 it->second->SetOptions(options_); 1241 } 1242 return true; 1243 } 1244 1245 void WebRtcVideoChannel2::SetInterface(NetworkInterface* iface) { 1246 MediaChannel::SetInterface(iface); 1247 // Set the RTP recv/send buffer to a bigger size 1248 MediaChannel::SetOption(NetworkInterface::ST_RTP, 1249 rtc::Socket::OPT_RCVBUF, 1250 kVideoRtpBufferSize); 1251 1252 // TODO(sriniv): Remove or re-enable this. 1253 // As part of b/8030474, send-buffer is size now controlled through 1254 // portallocator flags. 1255 // network_interface_->SetOption(NetworkInterface::ST_RTP, 1256 // rtc::Socket::OPT_SNDBUF, 1257 // kVideoRtpBufferSize); 1258 } 1259 1260 void WebRtcVideoChannel2::UpdateAspectRatio(int ratio_w, int ratio_h) { 1261 // TODO(pbos): Implement. 1262 } 1263 1264 void WebRtcVideoChannel2::OnMessage(rtc::Message* msg) { 1265 // Ignored. 1266 } 1267 1268 bool WebRtcVideoChannel2::SendRtp(const uint8_t* data, size_t len) { 1269 rtc::Buffer packet(data, len, kMaxRtpPacketLen); 1270 return MediaChannel::SendPacket(&packet); 1271 } 1272 1273 bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) { 1274 rtc::Buffer packet(data, len, kMaxRtpPacketLen); 1275 return MediaChannel::SendRtcp(&packet); 1276 } 1277 1278 void WebRtcVideoChannel2::StartAllSendStreams() { 1279 for (std::map<uint32, WebRtcVideoSendStream*>::iterator it = 1280 send_streams_.begin(); 1281 it != send_streams_.end(); 1282 ++it) { 1283 it->second->Start(); 1284 } 1285 } 1286 1287 void WebRtcVideoChannel2::StopAllSendStreams() { 1288 for (std::map<uint32, WebRtcVideoSendStream*>::iterator it = 1289 send_streams_.begin(); 1290 it != send_streams_.end(); 1291 ++it) { 1292 it->second->Stop(); 1293 } 1294 } 1295 1296 WebRtcVideoChannel2::WebRtcVideoSendStream::VideoSendStreamParameters:: 1297 VideoSendStreamParameters( 1298 const webrtc::VideoSendStream::Config& config, 1299 const VideoOptions& options, 1300 const Settable<VideoCodecSettings>& codec_settings) 1301 : config(config), options(options), codec_settings(codec_settings) { 1302 } 1303 1304 WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream( 1305 webrtc::Call* call, 1306 WebRtcVideoEncoderFactory2* encoder_factory, 1307 const VideoOptions& options, 1308 const Settable<VideoCodecSettings>& codec_settings, 1309 const StreamParams& sp, 1310 const std::vector<webrtc::RtpExtension>& rtp_extensions) 1311 : call_(call), 1312 encoder_factory_(encoder_factory), 1313 stream_(NULL), 1314 parameters_(webrtc::VideoSendStream::Config(), options, codec_settings), 1315 capturer_(NULL), 1316 sending_(false), 1317 muted_(false) { 1318 parameters_.config.rtp.max_packet_size = kVideoMtu; 1319 1320 sp.GetPrimarySsrcs(¶meters_.config.rtp.ssrcs); 1321 sp.GetFidSsrcs(parameters_.config.rtp.ssrcs, 1322 ¶meters_.config.rtp.rtx.ssrcs); 1323 parameters_.config.rtp.c_name = sp.cname; 1324 parameters_.config.rtp.extensions = rtp_extensions; 1325 1326 VideoCodecSettings params; 1327 if (codec_settings.Get(¶ms)) { 1328 SetCodec(params); 1329 } 1330 } 1331 1332 WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() { 1333 DisconnectCapturer(); 1334 if (stream_ != NULL) { 1335 call_->DestroyVideoSendStream(stream_); 1336 } 1337 delete parameters_.config.encoder_settings.encoder; 1338 } 1339 1340 static void SetWebRtcFrameToBlack(webrtc::I420VideoFrame* video_frame) { 1341 assert(video_frame != NULL); 1342 memset(video_frame->buffer(webrtc::kYPlane), 1343 16, 1344 video_frame->allocated_size(webrtc::kYPlane)); 1345 memset(video_frame->buffer(webrtc::kUPlane), 1346 128, 1347 video_frame->allocated_size(webrtc::kUPlane)); 1348 memset(video_frame->buffer(webrtc::kVPlane), 1349 128, 1350 video_frame->allocated_size(webrtc::kVPlane)); 1351 } 1352 1353 static void CreateBlackFrame(webrtc::I420VideoFrame* video_frame, 1354 int width, 1355 int height) { 1356 video_frame->CreateEmptyFrame( 1357 width, height, width, (width + 1) / 2, (width + 1) / 2); 1358 SetWebRtcFrameToBlack(video_frame); 1359 } 1360 1361 static void ConvertToI420VideoFrame(const VideoFrame& frame, 1362 webrtc::I420VideoFrame* i420_frame) { 1363 i420_frame->CreateFrame( 1364 static_cast<int>(frame.GetYPitch() * frame.GetHeight()), 1365 frame.GetYPlane(), 1366 static_cast<int>(frame.GetUPitch() * ((frame.GetHeight() + 1) / 2)), 1367 frame.GetUPlane(), 1368 static_cast<int>(frame.GetVPitch() * ((frame.GetHeight() + 1) / 2)), 1369 frame.GetVPlane(), 1370 static_cast<int>(frame.GetWidth()), 1371 static_cast<int>(frame.GetHeight()), 1372 static_cast<int>(frame.GetYPitch()), 1373 static_cast<int>(frame.GetUPitch()), 1374 static_cast<int>(frame.GetVPitch())); 1375 } 1376 1377 void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame( 1378 VideoCapturer* capturer, 1379 const VideoFrame* frame) { 1380 LOG(LS_VERBOSE) << "InputFrame: " << frame->GetWidth() << "x" 1381 << frame->GetHeight(); 1382 // Lock before copying, can be called concurrently when swapping input source. 1383 rtc::CritScope frame_cs(&frame_lock_); 1384 ConvertToI420VideoFrame(*frame, &video_frame_); 1385 1386 rtc::CritScope cs(&lock_); 1387 if (stream_ == NULL) { 1388 LOG(LS_WARNING) << "Capturer inputting frames before send codecs are " 1389 "configured, dropping."; 1390 return; 1391 } 1392 if (format_.width == 0) { // Dropping frames. 1393 assert(format_.height == 0); 1394 LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame."; 1395 return; 1396 } 1397 if (muted_) { 1398 // Create a black frame to transmit instead. 1399 CreateBlackFrame(&video_frame_, 1400 static_cast<int>(frame->GetWidth()), 1401 static_cast<int>(frame->GetHeight())); 1402 } 1403 // Reconfigure codec if necessary. 1404 SetDimensions( 1405 video_frame_.width(), video_frame_.height(), capturer->IsScreencast()); 1406 1407 LOG(LS_VERBOSE) << "SwapFrame: " << video_frame_.width() << "x" 1408 << video_frame_.height() << " -> (codec) " 1409 << parameters_.encoder_config.streams.back().width << "x" 1410 << parameters_.encoder_config.streams.back().height; 1411 stream_->Input()->SwapFrame(&video_frame_); 1412 } 1413 1414 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer( 1415 VideoCapturer* capturer) { 1416 if (!DisconnectCapturer() && capturer == NULL) { 1417 return false; 1418 } 1419 1420 { 1421 rtc::CritScope cs(&lock_); 1422 1423 if (capturer == NULL) { 1424 if (stream_ != NULL) { 1425 LOG(LS_VERBOSE) << "Disabling capturer, sending black frame."; 1426 webrtc::I420VideoFrame black_frame; 1427 1428 int width = format_.width; 1429 int height = format_.height; 1430 int half_width = (width + 1) / 2; 1431 black_frame.CreateEmptyFrame( 1432 width, height, width, half_width, half_width); 1433 SetWebRtcFrameToBlack(&black_frame); 1434 SetDimensions(width, height, false); 1435 stream_->Input()->SwapFrame(&black_frame); 1436 } 1437 1438 capturer_ = NULL; 1439 return true; 1440 } 1441 1442 capturer_ = capturer; 1443 } 1444 // Lock cannot be held while connecting the capturer to prevent lock-order 1445 // violations. 1446 capturer->SignalVideoFrame.connect(this, &WebRtcVideoSendStream::InputFrame); 1447 return true; 1448 } 1449 1450 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoFormat( 1451 const VideoFormat& format) { 1452 if ((format.width == 0 || format.height == 0) && 1453 format.width != format.height) { 1454 LOG(LS_ERROR) << "Can't set VideoFormat, width or height is zero (but not " 1455 "both, 0x0 drops frames)."; 1456 return false; 1457 } 1458 1459 rtc::CritScope cs(&lock_); 1460 if (format.width == 0 && format.height == 0) { 1461 LOG(LS_INFO) 1462 << "0x0 resolution selected. Captured frames will be dropped for ssrc: " 1463 << parameters_.config.rtp.ssrcs[0] << "."; 1464 } else { 1465 // TODO(pbos): Fix me, this only affects the last stream! 1466 parameters_.encoder_config.streams.back().max_framerate = 1467 VideoFormat::IntervalToFps(format.interval); 1468 SetDimensions(format.width, format.height, false); 1469 } 1470 1471 format_ = format; 1472 return true; 1473 } 1474 1475 void WebRtcVideoChannel2::WebRtcVideoSendStream::MuteStream(bool mute) { 1476 rtc::CritScope cs(&lock_); 1477 muted_ = mute; 1478 } 1479 1480 bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() { 1481 rtc::CritScope cs(&lock_); 1482 if (capturer_ == NULL) { 1483 return false; 1484 } 1485 capturer_->SignalVideoFrame.disconnect(this); 1486 capturer_ = NULL; 1487 return true; 1488 } 1489 1490 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions( 1491 const VideoOptions& options) { 1492 rtc::CritScope cs(&lock_); 1493 VideoCodecSettings codec_settings; 1494 if (parameters_.codec_settings.Get(&codec_settings)) { 1495 SetCodecAndOptions(codec_settings, options); 1496 } else { 1497 parameters_.options = options; 1498 } 1499 } 1500 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec( 1501 const VideoCodecSettings& codec_settings) { 1502 rtc::CritScope cs(&lock_); 1503 SetCodecAndOptions(codec_settings, parameters_.options); 1504 } 1505 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodecAndOptions( 1506 const VideoCodecSettings& codec_settings, 1507 const VideoOptions& options) { 1508 std::vector<webrtc::VideoStream> video_streams = 1509 encoder_factory_->CreateVideoStreams( 1510 codec_settings.codec, options, parameters_.config.rtp.ssrcs.size()); 1511 if (video_streams.empty()) { 1512 return; 1513 } 1514 parameters_.encoder_config.streams = video_streams; 1515 format_ = VideoFormat(codec_settings.codec.width, 1516 codec_settings.codec.height, 1517 VideoFormat::FpsToInterval(30), 1518 FOURCC_I420); 1519 1520 webrtc::VideoEncoder* old_encoder = 1521 parameters_.config.encoder_settings.encoder; 1522 parameters_.config.encoder_settings.encoder = 1523 encoder_factory_->CreateVideoEncoder(codec_settings.codec, options); 1524 parameters_.config.encoder_settings.payload_name = codec_settings.codec.name; 1525 parameters_.config.encoder_settings.payload_type = codec_settings.codec.id; 1526 parameters_.config.rtp.fec = codec_settings.fec; 1527 1528 // Set RTX payload type if RTX is enabled. 1529 if (!parameters_.config.rtp.rtx.ssrcs.empty()) { 1530 parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type; 1531 1532 options.use_payload_padding.Get( 1533 ¶meters_.config.rtp.rtx.pad_with_redundant_payloads); 1534 } 1535 1536 if (IsNackEnabled(codec_settings.codec)) { 1537 parameters_.config.rtp.nack.rtp_history_ms = kNackHistoryMs; 1538 } 1539 1540 options.suspend_below_min_bitrate.Get( 1541 ¶meters_.config.suspend_below_min_bitrate); 1542 1543 parameters_.codec_settings.Set(codec_settings); 1544 parameters_.options = options; 1545 1546 RecreateWebRtcStream(); 1547 delete old_encoder; 1548 } 1549 1550 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpExtensions( 1551 const std::vector<webrtc::RtpExtension>& rtp_extensions) { 1552 rtc::CritScope cs(&lock_); 1553 parameters_.config.rtp.extensions = rtp_extensions; 1554 RecreateWebRtcStream(); 1555 } 1556 1557 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions( 1558 int width, 1559 int height, 1560 bool override_max) { 1561 assert(!parameters_.encoder_config.streams.empty()); 1562 LOG(LS_VERBOSE) << "SetDimensions: " << width << "x" << height; 1563 1564 VideoCodecSettings codec_settings; 1565 parameters_.codec_settings.Get(&codec_settings); 1566 // Restrict dimensions according to codec max. 1567 if (!override_max) { 1568 if (codec_settings.codec.width < width) 1569 width = codec_settings.codec.width; 1570 if (codec_settings.codec.height < height) 1571 height = codec_settings.codec.height; 1572 } 1573 1574 if (parameters_.encoder_config.streams.back().width == width && 1575 parameters_.encoder_config.streams.back().height == height) { 1576 return; 1577 } 1578 1579 webrtc::VideoEncoderConfig encoder_config = parameters_.encoder_config; 1580 encoder_config.encoder_specific_settings = 1581 encoder_factory_->CreateVideoEncoderSettings(codec_settings.codec, 1582 parameters_.options); 1583 1584 VideoCodec codec = codec_settings.codec; 1585 codec.width = width; 1586 codec.height = height; 1587 1588 encoder_config.streams = encoder_factory_->CreateVideoStreams( 1589 codec, parameters_.options, parameters_.config.rtp.ssrcs.size()); 1590 1591 bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config); 1592 1593 encoder_factory_->DestroyVideoEncoderSettings( 1594 codec_settings.codec, 1595 encoder_config.encoder_specific_settings); 1596 1597 encoder_config.encoder_specific_settings = NULL; 1598 1599 if (!stream_reconfigured) { 1600 LOG(LS_WARNING) << "Failed to reconfigure video encoder for dimensions: " 1601 << width << "x" << height; 1602 return; 1603 } 1604 1605 parameters_.encoder_config = encoder_config; 1606 } 1607 1608 void WebRtcVideoChannel2::WebRtcVideoSendStream::Start() { 1609 rtc::CritScope cs(&lock_); 1610 assert(stream_ != NULL); 1611 stream_->Start(); 1612 sending_ = true; 1613 } 1614 1615 void WebRtcVideoChannel2::WebRtcVideoSendStream::Stop() { 1616 rtc::CritScope cs(&lock_); 1617 if (stream_ != NULL) { 1618 stream_->Stop(); 1619 } 1620 sending_ = false; 1621 } 1622 1623 VideoSenderInfo 1624 WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() { 1625 VideoSenderInfo info; 1626 rtc::CritScope cs(&lock_); 1627 for (size_t i = 0; i < parameters_.config.rtp.ssrcs.size(); ++i) { 1628 info.add_ssrc(parameters_.config.rtp.ssrcs[i]); 1629 } 1630 1631 if (stream_ == NULL) { 1632 return info; 1633 } 1634 1635 webrtc::VideoSendStream::Stats stats = stream_->GetStats(); 1636 info.framerate_input = stats.input_frame_rate; 1637 info.framerate_sent = stats.encode_frame_rate; 1638 1639 for (std::map<uint32_t, webrtc::StreamStats>::iterator it = 1640 stats.substreams.begin(); 1641 it != stats.substreams.end(); 1642 ++it) { 1643 // TODO(pbos): Wire up additional stats, such as padding bytes. 1644 webrtc::StreamStats stream_stats = it->second; 1645 info.bytes_sent += stream_stats.rtp_stats.bytes + 1646 stream_stats.rtp_stats.header_bytes + 1647 stream_stats.rtp_stats.padding_bytes; 1648 info.packets_sent += stream_stats.rtp_stats.packets; 1649 info.packets_lost += stream_stats.rtcp_stats.cumulative_lost; 1650 } 1651 1652 if (!stats.substreams.empty()) { 1653 // TODO(pbos): Report fraction lost per SSRC. 1654 webrtc::StreamStats first_stream_stats = stats.substreams.begin()->second; 1655 info.fraction_lost = 1656 static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) / 1657 (1 << 8); 1658 } 1659 1660 if (capturer_ != NULL && !capturer_->IsMuted()) { 1661 VideoFormat last_captured_frame_format; 1662 capturer_->GetStats(&info.adapt_frame_drops, 1663 &info.effects_frame_drops, 1664 &info.capturer_frame_time, 1665 &last_captured_frame_format); 1666 info.input_frame_width = last_captured_frame_format.width; 1667 info.input_frame_height = last_captured_frame_format.height; 1668 info.send_frame_width = 1669 static_cast<int>(parameters_.encoder_config.streams.front().width); 1670 info.send_frame_height = 1671 static_cast<int>(parameters_.encoder_config.streams.front().height); 1672 } 1673 1674 // TODO(pbos): Support or remove the following stats. 1675 info.packets_cached = -1; 1676 info.rtt_ms = -1; 1677 1678 return info; 1679 } 1680 1681 void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() { 1682 if (stream_ != NULL) { 1683 call_->DestroyVideoSendStream(stream_); 1684 } 1685 1686 VideoCodecSettings codec_settings; 1687 parameters_.codec_settings.Get(&codec_settings); 1688 parameters_.encoder_config.encoder_specific_settings = 1689 encoder_factory_->CreateVideoEncoderSettings(codec_settings.codec, 1690 parameters_.options); 1691 1692 stream_ = call_->CreateVideoSendStream(parameters_.config, 1693 parameters_.encoder_config); 1694 1695 encoder_factory_->DestroyVideoEncoderSettings( 1696 codec_settings.codec, 1697 parameters_.encoder_config.encoder_specific_settings); 1698 1699 parameters_.encoder_config.encoder_specific_settings = NULL; 1700 1701 if (sending_) { 1702 stream_->Start(); 1703 } 1704 } 1705 1706 WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream( 1707 webrtc::Call* call, 1708 const webrtc::VideoReceiveStream::Config& config, 1709 const std::vector<VideoCodecSettings>& recv_codecs) 1710 : call_(call), 1711 stream_(NULL), 1712 config_(config), 1713 renderer_(NULL), 1714 last_width_(-1), 1715 last_height_(-1) { 1716 config_.renderer = this; 1717 // SetRecvCodecs will also reset (start) the VideoReceiveStream. 1718 SetRecvCodecs(recv_codecs); 1719 } 1720 1721 WebRtcVideoChannel2::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() { 1722 call_->DestroyVideoReceiveStream(stream_); 1723 } 1724 1725 void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvCodecs( 1726 const std::vector<VideoCodecSettings>& recv_codecs) { 1727 // TODO(pbos): Reconfigure RTX based on incoming recv_codecs. 1728 // TODO(pbos): Base receive codecs off recv_codecs_ and set up using a 1729 // DecoderFactory similar to send side. Pending webrtc:2854. 1730 // Also set up default codecs if there's nothing in recv_codecs_. 1731 webrtc::VideoCodec codec; 1732 memset(&codec, 0, sizeof(codec)); 1733 1734 codec.plType = kDefaultVideoCodecPref.payload_type; 1735 strcpy(codec.plName, kDefaultVideoCodecPref.name); 1736 codec.codecType = webrtc::kVideoCodecVP8; 1737 codec.codecSpecific.VP8.resilience = webrtc::kResilientStream; 1738 codec.codecSpecific.VP8.numberOfTemporalLayers = 1; 1739 codec.codecSpecific.VP8.denoisingOn = true; 1740 codec.codecSpecific.VP8.errorConcealmentOn = false; 1741 codec.codecSpecific.VP8.automaticResizeOn = false; 1742 codec.codecSpecific.VP8.frameDroppingOn = true; 1743 codec.codecSpecific.VP8.keyFrameInterval = 3000; 1744 // Bitrates don't matter and are ignored for the receiver. This is put in to 1745 // have the current underlying implementation accept the VideoCodec. 1746 codec.minBitrate = codec.startBitrate = codec.maxBitrate = 300; 1747 config_.codecs.clear(); 1748 config_.codecs.push_back(codec); 1749 1750 config_.rtp.fec = recv_codecs.front().fec; 1751 1752 config_.rtp.nack.rtp_history_ms = 1753 IsNackEnabled(recv_codecs.begin()->codec) ? kNackHistoryMs : 0; 1754 config_.rtp.remb = IsRembEnabled(recv_codecs.begin()->codec); 1755 1756 RecreateWebRtcStream(); 1757 } 1758 1759 void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRtpExtensions( 1760 const std::vector<webrtc::RtpExtension>& extensions) { 1761 config_.rtp.extensions = extensions; 1762 RecreateWebRtcStream(); 1763 } 1764 1765 void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() { 1766 if (stream_ != NULL) { 1767 call_->DestroyVideoReceiveStream(stream_); 1768 } 1769 stream_ = call_->CreateVideoReceiveStream(config_); 1770 stream_->Start(); 1771 } 1772 1773 void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RenderFrame( 1774 const webrtc::I420VideoFrame& frame, 1775 int time_to_render_ms) { 1776 rtc::CritScope crit(&renderer_lock_); 1777 if (renderer_ == NULL) { 1778 LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoRenderer."; 1779 return; 1780 } 1781 1782 if (frame.width() != last_width_ || frame.height() != last_height_) { 1783 SetSize(frame.width(), frame.height()); 1784 } 1785 1786 LOG(LS_VERBOSE) << "RenderFrame: (" << frame.width() << "x" << frame.height() 1787 << ")"; 1788 1789 const WebRtcVideoRenderFrame render_frame(&frame); 1790 renderer_->RenderFrame(&render_frame); 1791 } 1792 1793 void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRenderer( 1794 cricket::VideoRenderer* renderer) { 1795 rtc::CritScope crit(&renderer_lock_); 1796 renderer_ = renderer; 1797 if (renderer_ != NULL && last_width_ != -1) { 1798 SetSize(last_width_, last_height_); 1799 } 1800 } 1801 1802 VideoRenderer* WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetRenderer() { 1803 // TODO(pbos): Remove GetRenderer and all uses of it, it's thread-unsafe by 1804 // design. 1805 rtc::CritScope crit(&renderer_lock_); 1806 return renderer_; 1807 } 1808 1809 void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetSize(int width, 1810 int height) { 1811 rtc::CritScope crit(&renderer_lock_); 1812 if (!renderer_->SetSize(width, height, 0)) { 1813 LOG(LS_ERROR) << "Could not set renderer size."; 1814 } 1815 last_width_ = width; 1816 last_height_ = height; 1817 } 1818 1819 VideoReceiverInfo 1820 WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo() { 1821 VideoReceiverInfo info; 1822 info.add_ssrc(config_.rtp.remote_ssrc); 1823 webrtc::VideoReceiveStream::Stats stats = stream_->GetStats(); 1824 info.bytes_rcvd = stats.rtp_stats.bytes + stats.rtp_stats.header_bytes + 1825 stats.rtp_stats.padding_bytes; 1826 info.packets_rcvd = stats.rtp_stats.packets; 1827 1828 info.framerate_rcvd = stats.network_frame_rate; 1829 info.framerate_decoded = stats.decode_frame_rate; 1830 info.framerate_output = stats.render_frame_rate; 1831 1832 rtc::CritScope frame_cs(&renderer_lock_); 1833 info.frame_width = last_width_; 1834 info.frame_height = last_height_; 1835 1836 // TODO(pbos): Support or remove the following stats. 1837 info.packets_concealed = -1; 1838 1839 return info; 1840 } 1841 1842 WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings() 1843 : rtx_payload_type(-1) {} 1844 1845 std::vector<WebRtcVideoChannel2::VideoCodecSettings> 1846 WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) { 1847 assert(!codecs.empty()); 1848 1849 std::vector<VideoCodecSettings> video_codecs; 1850 std::map<int, bool> payload_used; 1851 std::map<int, VideoCodec::CodecType> payload_codec_type; 1852 std::map<int, int> rtx_mapping; // video payload type -> rtx payload type. 1853 1854 webrtc::FecConfig fec_settings; 1855 1856 for (size_t i = 0; i < codecs.size(); ++i) { 1857 const VideoCodec& in_codec = codecs[i]; 1858 int payload_type = in_codec.id; 1859 1860 if (payload_used[payload_type]) { 1861 LOG(LS_ERROR) << "Payload type already registered: " 1862 << in_codec.ToString(); 1863 return std::vector<VideoCodecSettings>(); 1864 } 1865 payload_used[payload_type] = true; 1866 payload_codec_type[payload_type] = in_codec.GetCodecType(); 1867 1868 switch (in_codec.GetCodecType()) { 1869 case VideoCodec::CODEC_RED: { 1870 // RED payload type, should not have duplicates. 1871 assert(fec_settings.red_payload_type == -1); 1872 fec_settings.red_payload_type = in_codec.id; 1873 continue; 1874 } 1875 1876 case VideoCodec::CODEC_ULPFEC: { 1877 // ULPFEC payload type, should not have duplicates. 1878 assert(fec_settings.ulpfec_payload_type == -1); 1879 fec_settings.ulpfec_payload_type = in_codec.id; 1880 continue; 1881 } 1882 1883 case VideoCodec::CODEC_RTX: { 1884 int associated_payload_type; 1885 if (!in_codec.GetParam(kCodecParamAssociatedPayloadType, 1886 &associated_payload_type)) { 1887 LOG(LS_ERROR) << "RTX codec without associated payload type: " 1888 << in_codec.ToString(); 1889 return std::vector<VideoCodecSettings>(); 1890 } 1891 rtx_mapping[associated_payload_type] = in_codec.id; 1892 continue; 1893 } 1894 1895 case VideoCodec::CODEC_VIDEO: 1896 break; 1897 } 1898 1899 video_codecs.push_back(VideoCodecSettings()); 1900 video_codecs.back().codec = in_codec; 1901 } 1902 1903 // One of these codecs should have been a video codec. Only having FEC 1904 // parameters into this code is a logic error. 1905 assert(!video_codecs.empty()); 1906 1907 for (std::map<int, int>::const_iterator it = rtx_mapping.begin(); 1908 it != rtx_mapping.end(); 1909 ++it) { 1910 if (!payload_used[it->first]) { 1911 LOG(LS_ERROR) << "RTX mapped to payload not in codec list."; 1912 return std::vector<VideoCodecSettings>(); 1913 } 1914 if (payload_codec_type[it->first] != VideoCodec::CODEC_VIDEO) { 1915 LOG(LS_ERROR) << "RTX not mapped to regular video codec."; 1916 return std::vector<VideoCodecSettings>(); 1917 } 1918 } 1919 1920 // TODO(pbos): Write tests that figure out that I have not verified that RTX 1921 // codecs aren't mapped to bogus payloads. 1922 for (size_t i = 0; i < video_codecs.size(); ++i) { 1923 video_codecs[i].fec = fec_settings; 1924 if (rtx_mapping[video_codecs[i].codec.id] != 0) { 1925 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; 1926 } 1927 } 1928 1929 return video_codecs; 1930 } 1931 1932 std::vector<WebRtcVideoChannel2::VideoCodecSettings> 1933 WebRtcVideoChannel2::FilterSupportedCodecs( 1934 const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs) { 1935 std::vector<VideoCodecSettings> supported_codecs; 1936 for (size_t i = 0; i < mapped_codecs.size(); ++i) { 1937 if (encoder_factory_->SupportsCodec(mapped_codecs[i].codec)) { 1938 supported_codecs.push_back(mapped_codecs[i]); 1939 } 1940 } 1941 return supported_codecs; 1942 } 1943 1944 } // namespace cricket 1945 1946 #endif // HAVE_WEBRTC_VIDEO 1947