| /* |
| * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "webrtc/media/engine/webrtcvideoengine2.h" |
| |
| #include <stdio.h> |
| #include <algorithm> |
| #include <set> |
| #include <string> |
| #include <utility> |
| |
| #include "webrtc/base/copyonwritebuffer.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/stringutils.h" |
| #include "webrtc/base/timeutils.h" |
| #include "webrtc/base/trace_event.h" |
| #include "webrtc/call.h" |
| #include "webrtc/media/engine/constants.h" |
| #include "webrtc/media/engine/simulcast.h" |
| #include "webrtc/media/engine/webrtcmediaengine.h" |
| #include "webrtc/media/engine/webrtcvideoencoderfactory.h" |
| #include "webrtc/media/engine/webrtcvideoframe.h" |
| #include "webrtc/media/engine/webrtcvoiceengine.h" |
| #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" |
| #include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h" |
| #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h" |
| #include "webrtc/system_wrappers/include/field_trial.h" |
| #include "webrtc/system_wrappers/include/metrics.h" |
| #include "webrtc/video_decoder.h" |
| #include "webrtc/video_encoder.h" |
| |
| namespace cricket { |
| namespace { |
| |
| // Wrap cricket::WebRtcVideoEncoderFactory as a webrtc::VideoEncoderFactory. |
| class EncoderFactoryAdapter : public webrtc::VideoEncoderFactory { |
| public: |
| // EncoderFactoryAdapter doesn't take ownership of |factory|, which is owned |
| // by e.g. PeerConnectionFactory. |
| explicit EncoderFactoryAdapter(cricket::WebRtcVideoEncoderFactory* factory) |
| : factory_(factory) {} |
| virtual ~EncoderFactoryAdapter() {} |
| |
| // Implement webrtc::VideoEncoderFactory. |
| webrtc::VideoEncoder* Create() override { |
| return factory_->CreateVideoEncoder(webrtc::kVideoCodecVP8); |
| } |
| |
| void Destroy(webrtc::VideoEncoder* encoder) override { |
| return factory_->DestroyVideoEncoder(encoder); |
| } |
| |
| private: |
| cricket::WebRtcVideoEncoderFactory* const factory_; |
| }; |
| |
| webrtc::Call::Config::BitrateConfig GetBitrateConfigForCodec( |
| const VideoCodec& codec) { |
| webrtc::Call::Config::BitrateConfig config; |
| int bitrate_kbps; |
| if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) && |
| bitrate_kbps > 0) { |
| config.min_bitrate_bps = bitrate_kbps * 1000; |
| } else { |
| config.min_bitrate_bps = 0; |
| } |
| if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) && |
| bitrate_kbps > 0) { |
| config.start_bitrate_bps = bitrate_kbps * 1000; |
| } else { |
| // Do not reconfigure start bitrate unless it's specified and positive. |
| config.start_bitrate_bps = -1; |
| } |
| if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) && |
| bitrate_kbps > 0) { |
| config.max_bitrate_bps = bitrate_kbps * 1000; |
| } else { |
| config.max_bitrate_bps = -1; |
| } |
| return config; |
| } |
| |
| // An encoder factory that wraps Create requests for simulcastable codec types |
| // with a webrtc::SimulcastEncoderAdapter. Non simulcastable codec type |
| // requests are just passed through to the contained encoder factory. |
| class WebRtcSimulcastEncoderFactory |
| : public cricket::WebRtcVideoEncoderFactory { |
| public: |
| // WebRtcSimulcastEncoderFactory doesn't take ownership of |factory|, which is |
| // owned by e.g. PeerConnectionFactory. |
| explicit WebRtcSimulcastEncoderFactory( |
| cricket::WebRtcVideoEncoderFactory* factory) |
| : factory_(factory) {} |
| |
| static bool UseSimulcastEncoderFactory( |
| const std::vector<VideoCodec>& codecs) { |
| // If any codec is VP8, use the simulcast factory. If asked to create a |
| // non-VP8 codec, we'll just return a contained factory encoder directly. |
| for (const auto& codec : codecs) { |
| if (codec.type == webrtc::kVideoCodecVP8) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| webrtc::VideoEncoder* CreateVideoEncoder( |
| webrtc::VideoCodecType type) override { |
| RTC_DCHECK(factory_ != NULL); |
| // If it's a codec type we can simulcast, create a wrapped encoder. |
| if (type == webrtc::kVideoCodecVP8) { |
| return new webrtc::SimulcastEncoderAdapter( |
| new EncoderFactoryAdapter(factory_)); |
| } |
| webrtc::VideoEncoder* encoder = factory_->CreateVideoEncoder(type); |
| if (encoder) { |
| non_simulcast_encoders_.push_back(encoder); |
| } |
| return encoder; |
| } |
| |
| const std::vector<VideoCodec>& codecs() const override { |
| return factory_->codecs(); |
| } |
| |
| bool EncoderTypeHasInternalSource( |
| webrtc::VideoCodecType type) const override { |
| return factory_->EncoderTypeHasInternalSource(type); |
| } |
| |
| void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) override { |
| // Check first to see if the encoder wasn't wrapped in a |
| // SimulcastEncoderAdapter. In that case, ask the factory to destroy it. |
| if (std::remove(non_simulcast_encoders_.begin(), |
| non_simulcast_encoders_.end(), |
| encoder) != non_simulcast_encoders_.end()) { |
| factory_->DestroyVideoEncoder(encoder); |
| return; |
| } |
| |
| // Otherwise, SimulcastEncoderAdapter can be deleted directly, and will call |
| // DestroyVideoEncoder on the factory for individual encoder instances. |
| delete encoder; |
| } |
| |
| private: |
| cricket::WebRtcVideoEncoderFactory* factory_; |
| // A list of encoders that were created without being wrapped in a |
| // SimulcastEncoderAdapter. |
| std::vector<webrtc::VideoEncoder*> non_simulcast_encoders_; |
| }; |
| |
| bool CodecIsInternallySupported(const std::string& codec_name) { |
| if (CodecNamesEq(codec_name, kVp8CodecName)) { |
| return true; |
| } |
| if (CodecNamesEq(codec_name, kVp9CodecName)) { |
| return webrtc::VP9Encoder::IsSupported() && |
| webrtc::VP9Decoder::IsSupported(); |
| } |
| if (CodecNamesEq(codec_name, kH264CodecName)) { |
| return webrtc::H264Encoder::IsSupported() && |
| webrtc::H264Decoder::IsSupported(); |
| } |
| return false; |
| } |
| |
| void AddDefaultFeedbackParams(VideoCodec* codec) { |
| codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir)); |
| codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kParamValueEmpty)); |
| codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli)); |
| codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty)); |
| codec->AddFeedbackParam( |
| FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty)); |
| } |
| |
| static VideoCodec MakeVideoCodecWithDefaultFeedbackParams(int payload_type, |
| const char* name) { |
| VideoCodec codec(payload_type, name, kDefaultVideoMaxWidth, |
| kDefaultVideoMaxHeight, kDefaultVideoMaxFramerate); |
| AddDefaultFeedbackParams(&codec); |
| return codec; |
| } |
| |
| static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) { |
| std::stringstream out; |
| out << '{'; |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| out << codecs[i].ToString(); |
| if (i != codecs.size() - 1) { |
| out << ", "; |
| } |
| } |
| out << '}'; |
| return out.str(); |
| } |
| |
| static bool ValidateCodecFormats(const std::vector<VideoCodec>& codecs) { |
| bool has_video = false; |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| if (!codecs[i].ValidateCodecFormat()) { |
| return false; |
| } |
| if (codecs[i].GetCodecType() == VideoCodec::CODEC_VIDEO) { |
| has_video = true; |
| } |
| } |
| if (!has_video) { |
| LOG(LS_ERROR) << "Setting codecs without a video codec is invalid: " |
| << CodecVectorToString(codecs); |
| return false; |
| } |
| return true; |
| } |
| |
| static bool ValidateStreamParams(const StreamParams& sp) { |
| if (sp.ssrcs.empty()) { |
| LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString(); |
| return false; |
| } |
| |
| std::vector<uint32_t> primary_ssrcs; |
| sp.GetPrimarySsrcs(&primary_ssrcs); |
| std::vector<uint32_t> rtx_ssrcs; |
| sp.GetFidSsrcs(primary_ssrcs, &rtx_ssrcs); |
| for (uint32_t rtx_ssrc : rtx_ssrcs) { |
| bool rtx_ssrc_present = false; |
| for (uint32_t sp_ssrc : sp.ssrcs) { |
| if (sp_ssrc == rtx_ssrc) { |
| rtx_ssrc_present = true; |
| break; |
| } |
| } |
| if (!rtx_ssrc_present) { |
| LOG(LS_ERROR) << "RTX SSRC '" << rtx_ssrc |
| << "' missing from StreamParams ssrcs: " << sp.ToString(); |
| return false; |
| } |
| } |
| if (!rtx_ssrcs.empty() && primary_ssrcs.size() != rtx_ssrcs.size()) { |
| LOG(LS_ERROR) |
| << "RTX SSRCs exist, but don't cover all SSRCs (unsupported): " |
| << sp.ToString(); |
| return false; |
| } |
| |
| return true; |
| } |
| |
| inline bool ContainsHeaderExtension( |
| const std::vector<webrtc::RtpExtension>& extensions, |
| const std::string& uri) { |
| for (const auto& kv : extensions) { |
| if (kv.uri == uri) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| // Returns true if the given codec is disallowed from doing simulcast. |
| bool IsCodecBlacklistedForSimulcast(const std::string& codec_name) { |
| return CodecNamesEq(codec_name, kH264CodecName) || |
| CodecNamesEq(codec_name, kVp9CodecName); |
| } |
| |
| // The selected thresholds for QVGA and VGA corresponded to a QP around 10. |
| // The change in QP declined above the selected bitrates. |
| static int GetMaxDefaultVideoBitrateKbps(int width, int height) { |
| if (width * height <= 320 * 240) { |
| return 600; |
| } else if (width * height <= 640 * 480) { |
| return 1700; |
| } else if (width * height <= 960 * 540) { |
| return 2000; |
| } else { |
| return 2500; |
| } |
| } |
| |
| bool GetVp9LayersFromFieldTrialGroup(int* num_spatial_layers, |
| int* num_temporal_layers) { |
| std::string group = webrtc::field_trial::FindFullName("WebRTC-SupportVP9SVC"); |
| if (group.empty()) |
| return false; |
| |
| if (sscanf(group.c_str(), "EnabledByFlag_%dSL%dTL", num_spatial_layers, |
| num_temporal_layers) != 2) { |
| return false; |
| } |
| const int kMaxSpatialLayers = 2; |
| if (*num_spatial_layers > kMaxSpatialLayers || *num_spatial_layers < 1) |
| return false; |
| |
| const int kMaxTemporalLayers = 3; |
| if (*num_temporal_layers > kMaxTemporalLayers || *num_temporal_layers < 1) |
| return false; |
| |
| return true; |
| } |
| |
| int GetDefaultVp9SpatialLayers() { |
| int num_sl; |
| int num_tl; |
| if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) { |
| return num_sl; |
| } |
| return 1; |
| } |
| |
| int GetDefaultVp9TemporalLayers() { |
| int num_sl; |
| int num_tl; |
| if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) { |
| return num_tl; |
| } |
| return 1; |
| } |
| |
| class EncoderStreamFactory |
| : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface { |
| public: |
| EncoderStreamFactory(std::string codec_name, |
| int max_qp, |
| int max_framerate, |
| bool is_screencast, |
| bool conference_mode) |
| : codec_name_(codec_name), |
| max_qp_(max_qp), |
| max_framerate_(max_framerate), |
| is_screencast_(is_screencast), |
| conference_mode_(conference_mode) {} |
| |
| private: |
| std::vector<webrtc::VideoStream> CreateEncoderStreams( |
| int width, |
| int height, |
| const webrtc::VideoEncoderConfig& encoder_config) override { |
| RTC_DCHECK(encoder_config.number_of_streams > 1 ? !is_screencast_ : true); |
| if (encoder_config.number_of_streams > 1) { |
| return GetSimulcastConfig(encoder_config.number_of_streams, width, height, |
| encoder_config.max_bitrate_bps, max_qp_, |
| max_framerate_); |
| } |
| |
| // For unset max bitrates set default bitrate for non-simulcast. |
| int max_bitrate_bps = |
| (encoder_config.max_bitrate_bps > 0) |
| ? encoder_config.max_bitrate_bps |
| : GetMaxDefaultVideoBitrateKbps(width, height) * 1000; |
| |
| webrtc::VideoStream stream; |
| stream.width = width; |
| stream.height = height; |
| stream.max_framerate = max_framerate_; |
| stream.min_bitrate_bps = kMinVideoBitrateKbps * 1000; |
| stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate_bps; |
| stream.max_qp = max_qp_; |
| |
| // Conference mode screencast uses 2 temporal layers split at 100kbit. |
| if (conference_mode_ && is_screencast_) { |
| ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault(); |
| // For screenshare in conference mode, tl0 and tl1 bitrates are |
| // piggybacked |
| // on the VideoCodec struct as target and max bitrates, respectively. |
| // See eg. webrtc::VP8EncoderImpl::SetRates(). |
| stream.target_bitrate_bps = config.tl0_bitrate_kbps * 1000; |
| stream.max_bitrate_bps = config.tl1_bitrate_kbps * 1000; |
| stream.temporal_layer_thresholds_bps.clear(); |
| stream.temporal_layer_thresholds_bps.push_back(config.tl0_bitrate_kbps * |
| 1000); |
| } |
| |
| if (CodecNamesEq(codec_name_, kVp9CodecName) && !is_screencast_) { |
| stream.temporal_layer_thresholds_bps.resize( |
| GetDefaultVp9TemporalLayers() - 1); |
| } |
| |
| std::vector<webrtc::VideoStream> streams; |
| streams.push_back(stream); |
| return streams; |
| } |
| |
| const std::string codec_name_; |
| const int max_qp_; |
| const int max_framerate_; |
| const bool is_screencast_; |
| const bool conference_mode_; |
| }; |
| |
| } // namespace |
| |
| // Constants defined in webrtc/media/engine/constants.h |
| // TODO(pbos): Move these to a separate constants.cc file. |
| const int kMinVideoBitrateKbps = 30; |
| |
| const int kVideoMtu = 1200; |
| const int kVideoRtpBufferSize = 65536; |
| |
| // This constant is really an on/off, lower-level configurable NACK history |
| // duration hasn't been implemented. |
| static const int kNackHistoryMs = 1000; |
| |
| static const int kDefaultQpMax = 56; |
| |
| static const int kDefaultRtcpReceiverReportSsrc = 1; |
| |
| // Down grade resolution at most 2 times for CPU reasons. |
| static const int kMaxCpuDowngrades = 2; |
| |
| // Minimum time interval for logging stats. |
| static const int64_t kStatsLogIntervalMs = 10000; |
| |
| // Adds |codec| to |list|, and also adds an RTX codec if |codec|'s name is |
| // recognized. |
| // TODO(deadbeef): Should we add RTX codecs for external codecs whose names we |
| // don't recognize? |
| void AddCodecAndMaybeRtxCodec(const VideoCodec& codec, |
| std::vector<VideoCodec>* codecs) { |
| codecs->push_back(codec); |
| int rtx_payload_type = 0; |
| if (CodecNamesEq(codec.name, kVp8CodecName)) { |
| rtx_payload_type = kDefaultRtxVp8PlType; |
| } else if (CodecNamesEq(codec.name, kVp9CodecName)) { |
| rtx_payload_type = kDefaultRtxVp9PlType; |
| } else if (CodecNamesEq(codec.name, kH264CodecName)) { |
| rtx_payload_type = kDefaultRtxH264PlType; |
| } else if (CodecNamesEq(codec.name, kRedCodecName)) { |
| rtx_payload_type = kDefaultRtxRedPlType; |
| } else { |
| return; |
| } |
| codecs->push_back(VideoCodec::CreateRtxCodec(rtx_payload_type, codec.id)); |
| } |
| |
| std::vector<VideoCodec> DefaultVideoCodecList() { |
| std::vector<VideoCodec> codecs; |
| AddCodecAndMaybeRtxCodec( |
| MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType, kVp8CodecName), |
| &codecs); |
| if (CodecIsInternallySupported(kVp9CodecName)) { |
| AddCodecAndMaybeRtxCodec(MakeVideoCodecWithDefaultFeedbackParams( |
| kDefaultVp9PlType, kVp9CodecName), |
| &codecs); |
| } |
| if (CodecIsInternallySupported(kH264CodecName)) { |
| VideoCodec codec = MakeVideoCodecWithDefaultFeedbackParams( |
| kDefaultH264PlType, kH264CodecName); |
| // TODO(hta): Move all parameter generation for SDP into the codec |
| // implementation, for all codecs and parameters. |
| // TODO(hta): Move selection of profile-level-id to H.264 codec |
| // implementation. |
| // TODO(hta): Set FMTP parameters for all codecs of type H264. |
| codec.SetParam(kH264FmtpProfileLevelId, |
| kH264ProfileLevelConstrainedBaseline); |
| codec.SetParam(kH264FmtpLevelAsymmetryAllowed, "1"); |
| codec.SetParam(kH264FmtpPacketizationMode, "1"); |
| AddCodecAndMaybeRtxCodec(codec, &codecs); |
| } |
| AddCodecAndMaybeRtxCodec(VideoCodec(kDefaultRedPlType, kRedCodecName), |
| &codecs); |
| codecs.push_back(VideoCodec(kDefaultUlpfecType, kUlpfecCodecName)); |
| return codecs; |
| } |
| |
| rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings> |
| WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings( |
| const VideoCodec& codec) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| bool is_screencast = parameters_.options.is_screencast.value_or(false); |
| // No automatic resizing when using simulcast or screencast. |
| bool automatic_resize = |
| !is_screencast && parameters_.config.rtp.ssrcs.size() == 1; |
| bool frame_dropping = !is_screencast; |
| bool denoising; |
| bool codec_default_denoising = false; |
| if (is_screencast) { |
| denoising = false; |
| } else { |
| // Use codec default if video_noise_reduction is unset. |
| codec_default_denoising = !parameters_.options.video_noise_reduction; |
| denoising = parameters_.options.video_noise_reduction.value_or(false); |
| } |
| |
| if (CodecNamesEq(codec.name, kH264CodecName)) { |
| webrtc::VideoCodecH264 h264_settings = |
| webrtc::VideoEncoder::GetDefaultH264Settings(); |
| h264_settings.frameDroppingOn = frame_dropping; |
| return new rtc::RefCountedObject< |
| webrtc::VideoEncoderConfig::H264EncoderSpecificSettings>(h264_settings); |
| } |
| if (CodecNamesEq(codec.name, kVp8CodecName)) { |
| webrtc::VideoCodecVP8 vp8_settings = |
| webrtc::VideoEncoder::GetDefaultVp8Settings(); |
| vp8_settings.automaticResizeOn = automatic_resize; |
| // VP8 denoising is enabled by default. |
| vp8_settings.denoisingOn = codec_default_denoising ? true : denoising; |
| vp8_settings.frameDroppingOn = frame_dropping; |
| return new rtc::RefCountedObject< |
| webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings); |
| } |
| if (CodecNamesEq(codec.name, kVp9CodecName)) { |
| webrtc::VideoCodecVP9 vp9_settings = |
| webrtc::VideoEncoder::GetDefaultVp9Settings(); |
| if (is_screencast) { |
| // TODO(asapersson): Set to 2 for now since there is a DCHECK in |
| // VideoSendStream::ReconfigureVideoEncoder. |
| vp9_settings.numberOfSpatialLayers = 2; |
| } else { |
| vp9_settings.numberOfSpatialLayers = GetDefaultVp9SpatialLayers(); |
| } |
| // VP9 denoising is disabled by default. |
| vp9_settings.denoisingOn = codec_default_denoising ? false : denoising; |
| vp9_settings.frameDroppingOn = frame_dropping; |
| return new rtc::RefCountedObject< |
| webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings); |
| } |
| return nullptr; |
| } |
| |
| DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler() |
| : default_recv_ssrc_(0), default_sink_(NULL) {} |
| |
| UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc( |
| WebRtcVideoChannel2* channel, |
| uint32_t ssrc) { |
| if (default_recv_ssrc_ != 0) { // Already one default stream. |
| LOG(LS_WARNING) << "Unknown SSRC, but default receive stream already set."; |
| return kDropPacket; |
| } |
| |
| StreamParams sp; |
| sp.ssrcs.push_back(ssrc); |
| LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << "."; |
| if (!channel->AddRecvStream(sp, true)) { |
| LOG(LS_WARNING) << "Could not create default receive stream."; |
| } |
| |
| channel->SetSink(ssrc, default_sink_); |
| default_recv_ssrc_ = ssrc; |
| return kDeliverPacket; |
| } |
| |
| rtc::VideoSinkInterface<VideoFrame>* |
| DefaultUnsignalledSsrcHandler::GetDefaultSink() const { |
| return default_sink_; |
| } |
| |
| void DefaultUnsignalledSsrcHandler::SetDefaultSink( |
| VideoMediaChannel* channel, |
| rtc::VideoSinkInterface<VideoFrame>* sink) { |
| default_sink_ = sink; |
| if (default_recv_ssrc_ != 0) { |
| channel->SetSink(default_recv_ssrc_, default_sink_); |
| } |
| } |
| |
| WebRtcVideoEngine2::WebRtcVideoEngine2() |
| : initialized_(false), |
| external_decoder_factory_(NULL), |
| external_encoder_factory_(NULL) { |
| LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()"; |
| video_codecs_ = GetSupportedCodecs(); |
| } |
| |
| WebRtcVideoEngine2::~WebRtcVideoEngine2() { |
| LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2"; |
| } |
| |
| void WebRtcVideoEngine2::Init() { |
| LOG(LS_INFO) << "WebRtcVideoEngine2::Init"; |
| initialized_ = true; |
| } |
| |
| WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options) { |
| RTC_DCHECK(initialized_); |
| LOG(LS_INFO) << "CreateChannel. Options: " << options.ToString(); |
| return new WebRtcVideoChannel2(call, config, options, video_codecs_, |
| external_encoder_factory_, |
| external_decoder_factory_); |
| } |
| |
| const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const { |
| return video_codecs_; |
| } |
| |
| RtpCapabilities WebRtcVideoEngine2::GetCapabilities() const { |
| RtpCapabilities capabilities; |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kTimestampOffsetUri, |
| webrtc::RtpExtension::kTimestampOffsetDefaultId)); |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri, |
| webrtc::RtpExtension::kAbsSendTimeDefaultId)); |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kVideoRotationUri, |
| webrtc::RtpExtension::kVideoRotationDefaultId)); |
| capabilities.header_extensions.push_back(webrtc::RtpExtension( |
| webrtc::RtpExtension::kTransportSequenceNumberUri, |
| webrtc::RtpExtension::kTransportSequenceNumberDefaultId)); |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kPlayoutDelayUri, |
| webrtc::RtpExtension::kPlayoutDelayDefaultId)); |
| return capabilities; |
| } |
| |
| void WebRtcVideoEngine2::SetExternalDecoderFactory( |
| WebRtcVideoDecoderFactory* decoder_factory) { |
| RTC_DCHECK(!initialized_); |
| external_decoder_factory_ = decoder_factory; |
| } |
| |
| void WebRtcVideoEngine2::SetExternalEncoderFactory( |
| WebRtcVideoEncoderFactory* encoder_factory) { |
| RTC_DCHECK(!initialized_); |
| if (external_encoder_factory_ == encoder_factory) |
| return; |
| |
| // No matter what happens we shouldn't hold on to a stale |
| // WebRtcSimulcastEncoderFactory. |
| simulcast_encoder_factory_.reset(); |
| |
| if (encoder_factory && |
| WebRtcSimulcastEncoderFactory::UseSimulcastEncoderFactory( |
| encoder_factory->codecs())) { |
| simulcast_encoder_factory_.reset( |
| new WebRtcSimulcastEncoderFactory(encoder_factory)); |
| encoder_factory = simulcast_encoder_factory_.get(); |
| } |
| external_encoder_factory_ = encoder_factory; |
| |
| video_codecs_ = GetSupportedCodecs(); |
| } |
| |
| std::vector<VideoCodec> WebRtcVideoEngine2::GetSupportedCodecs() const { |
| std::vector<VideoCodec> supported_codecs = DefaultVideoCodecList(); |
| |
| if (external_encoder_factory_ == NULL) { |
| LOG(LS_INFO) << "Supported codecs: " |
| << CodecVectorToString(supported_codecs); |
| return supported_codecs; |
| } |
| |
| std::stringstream out; |
| const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs = |
| external_encoder_factory_->codecs(); |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| out << codecs[i].name; |
| if (i != codecs.size() - 1) { |
| out << ", "; |
| } |
| // Don't add internally-supported codecs twice. |
| if (CodecIsInternallySupported(codecs[i].name)) { |
| continue; |
| } |
| |
| // External video encoders are given payloads 120-127. This also means that |
| // we only support up to 8 external payload types. |
| // TODO(deadbeef): mediasession.cc already has code to dynamically |
| // determine a payload type. We should be able to just leave the payload |
| // type empty and let mediasession determine it. However, currently RTX |
| // codecs are associated to codecs by payload type, meaning we DO need |
| // to allocate unique payload types here. So to make this change we would |
| // need to make RTX codecs associated by name instead. |
| const int kExternalVideoPayloadTypeBase = 120; |
| size_t payload_type = kExternalVideoPayloadTypeBase + i; |
| RTC_DCHECK(payload_type < 128); |
| VideoCodec codec(static_cast<int>(payload_type), codecs[i].name, |
| codecs[i].max_width, codecs[i].max_height, |
| codecs[i].max_fps); |
| |
| AddDefaultFeedbackParams(&codec); |
| AddCodecAndMaybeRtxCodec(codec, &supported_codecs); |
| } |
| LOG(LS_INFO) << "Supported codecs (incl. external codecs): " |
| << CodecVectorToString(supported_codecs); |
| LOG(LS_INFO) << "Codecs supported by the external encoder factory: " |
| << out.str(); |
| return supported_codecs; |
| } |
| |
| WebRtcVideoChannel2::WebRtcVideoChannel2( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options, |
| const std::vector<VideoCodec>& recv_codecs, |
| WebRtcVideoEncoderFactory* external_encoder_factory, |
| WebRtcVideoDecoderFactory* external_decoder_factory) |
| : VideoMediaChannel(config), |
| call_(call), |
| unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_), |
| video_config_(config.video), |
| external_encoder_factory_(external_encoder_factory), |
| external_decoder_factory_(external_decoder_factory), |
| default_send_options_(options), |
| red_disabled_by_remote_side_(false), |
| last_stats_log_ms_(-1) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| |
| rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc; |
| sending_ = false; |
| RTC_DCHECK(ValidateCodecFormats(recv_codecs)); |
| recv_codecs_ = FilterSupportedCodecs(MapCodecs(recv_codecs)); |
| } |
| |
| WebRtcVideoChannel2::~WebRtcVideoChannel2() { |
| for (auto& kv : send_streams_) |
| delete kv.second; |
| for (auto& kv : receive_streams_) |
| delete kv.second; |
| } |
| |
| bool WebRtcVideoChannel2::CodecIsExternallySupported( |
| const std::string& name) const { |
| if (external_encoder_factory_ == NULL) { |
| return false; |
| } |
| |
| const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs = |
| external_encoder_factory_->codecs(); |
| for (size_t c = 0; c < external_codecs.size(); ++c) { |
| if (CodecNamesEq(name, external_codecs[c].name)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| std::vector<WebRtcVideoChannel2::VideoCodecSettings> |
| WebRtcVideoChannel2::FilterSupportedCodecs( |
| const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs) |
| const { |
| std::vector<VideoCodecSettings> supported_codecs; |
| for (size_t i = 0; i < mapped_codecs.size(); ++i) { |
| const VideoCodecSettings& codec = mapped_codecs[i]; |
| if (CodecIsInternallySupported(codec.codec.name) || |
| CodecIsExternallySupported(codec.codec.name)) { |
| supported_codecs.push_back(codec); |
| } |
| } |
| return supported_codecs; |
| } |
| |
| bool WebRtcVideoChannel2::ReceiveCodecsHaveChanged( |
| std::vector<VideoCodecSettings> before, |
| std::vector<VideoCodecSettings> after) { |
| if (before.size() != after.size()) { |
| return true; |
| } |
| // The receive codec order doesn't matter, so we sort the codecs before |
| // comparing. This is necessary because currently the |
| // only way to change the send codec is to munge SDP, which causes |
| // the receive codec list to change order, which causes the streams |
| // to be recreates which causes a "blink" of black video. In order |
| // to support munging the SDP in this way without recreating receive |
| // streams, we ignore the order of the received codecs so that |
| // changing the order doesn't cause this "blink". |
| auto comparison = |
| [](const VideoCodecSettings& codec1, const VideoCodecSettings& codec2) { |
| return codec1.codec.id > codec2.codec.id; |
| }; |
| std::sort(before.begin(), before.end(), comparison); |
| std::sort(after.begin(), after.end(), comparison); |
| return before != after; |
| } |
| |
| bool WebRtcVideoChannel2::GetChangedSendParameters( |
| const VideoSendParameters& params, |
| ChangedSendParameters* changed_params) const { |
| if (!ValidateCodecFormats(params.codecs) || |
| !ValidateRtpExtensions(params.extensions)) { |
| return false; |
| } |
| |
| // Handle send codec. |
| const std::vector<VideoCodecSettings> supported_codecs = |
| FilterSupportedCodecs(MapCodecs(params.codecs)); |
| |
| if (supported_codecs.empty()) { |
| LOG(LS_ERROR) << "No video codecs supported."; |
| return false; |
| } |
| |
| if (!send_codec_ || supported_codecs.front() != *send_codec_) { |
| changed_params->codec = |
| rtc::Optional<VideoCodecSettings>(supported_codecs.front()); |
| } |
| |
| // Handle RTP header extensions. |
| std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( |
| params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true); |
| if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) { |
| changed_params->rtp_header_extensions = |
| rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions); |
| } |
| |
| // Handle max bitrate. |
| if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps && |
| params.max_bandwidth_bps >= 0) { |
| // 0 uncaps max bitrate (-1). |
| changed_params->max_bandwidth_bps = rtc::Optional<int>( |
| params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps); |
| } |
| |
| // Handle conference mode. |
| if (params.conference_mode != send_params_.conference_mode) { |
| changed_params->conference_mode = |
| rtc::Optional<bool>(params.conference_mode); |
| } |
| |
| // Handle RTCP mode. |
| if (params.rtcp.reduced_size != send_params_.rtcp.reduced_size) { |
| changed_params->rtcp_mode = rtc::Optional<webrtc::RtcpMode>( |
| params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound); |
| } |
| |
| return true; |
| } |
| |
| rtc::DiffServCodePoint WebRtcVideoChannel2::PreferredDscp() const { |
| return rtc::DSCP_AF41; |
| } |
| |
| bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendParameters"); |
| LOG(LS_INFO) << "SetSendParameters: " << params.ToString(); |
| ChangedSendParameters changed_params; |
| if (!GetChangedSendParameters(params, &changed_params)) { |
| return false; |
| } |
| |
| if (changed_params.codec) { |
| const VideoCodecSettings& codec_settings = *changed_params.codec; |
| send_codec_ = rtc::Optional<VideoCodecSettings>(codec_settings); |
| LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString(); |
| } |
| |
| if (changed_params.rtp_header_extensions) { |
| send_rtp_extensions_ = changed_params.rtp_header_extensions; |
| } |
| |
| if (changed_params.codec || changed_params.max_bandwidth_bps) { |
| if (send_codec_) { |
| // TODO(holmer): Changing the codec parameters shouldn't necessarily mean |
| // that we change the min/max of bandwidth estimation. Reevaluate this. |
| bitrate_config_ = GetBitrateConfigForCodec(send_codec_->codec); |
| if (!changed_params.codec) { |
| // If the codec isn't changing, set the start bitrate to -1 which means |
| // "unchanged" so that BWE isn't affected. |
| bitrate_config_.start_bitrate_bps = -1; |
| } |
| } |
| if (params.max_bandwidth_bps >= 0) { |
| // Note that max_bandwidth_bps intentionally takes priority over the |
| // bitrate config for the codec. This allows FEC to be applied above the |
| // codec target bitrate. |
| // TODO(pbos): Figure out whether b=AS means max bitrate for this |
| // WebRtcVideoChannel2 (in which case we're good), or per sender (SSRC), |
| // in which case this should not set a Call::BitrateConfig but rather |
| // reconfigure all senders. |
| bitrate_config_.max_bitrate_bps = |
| params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps; |
| } |
| call_->SetBitrateConfig(bitrate_config_); |
| } |
| |
| { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (auto& kv : send_streams_) { |
| kv.second->SetSendParameters(changed_params); |
| } |
| if (changed_params.codec || changed_params.rtcp_mode) { |
| // Update receive feedback parameters from new codec or RTCP mode. |
| LOG(LS_INFO) |
| << "SetFeedbackOptions on all the receive streams because the send " |
| "codec or RTCP mode has changed."; |
| for (auto& kv : receive_streams_) { |
| RTC_DCHECK(kv.second != nullptr); |
| kv.second->SetFeedbackParameters( |
| HasNack(send_codec_->codec), HasRemb(send_codec_->codec), |
| HasTransportCc(send_codec_->codec), |
| params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound); |
| } |
| } |
| if (changed_params.codec) { |
| bool red_was_disabled = red_disabled_by_remote_side_; |
| red_disabled_by_remote_side_ = |
| changed_params.codec->fec.red_payload_type == -1; |
| if (red_was_disabled != red_disabled_by_remote_side_) { |
| for (auto& kv : receive_streams_) { |
| // In practice VideoChannel::SetRemoteContent appears to most of the |
| // time also call UpdateRemoteStreams, which recreates the receive |
| // streams. If that's always true this call isn't needed. |
| kv.second->SetFecDisabledRemotely(red_disabled_by_remote_side_); |
| } |
| } |
| } |
| } |
| send_params_ = params; |
| return true; |
| } |
| |
| webrtc::RtpParameters WebRtcVideoChannel2::GetRtpSendParameters( |
| uint32_t ssrc) const { |
| rtc::CritScope stream_lock(&stream_crit_); |
| auto it = send_streams_.find(ssrc); |
| if (it == send_streams_.end()) { |
| LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream " |
| << "with ssrc " << ssrc << " which doesn't exist."; |
| return webrtc::RtpParameters(); |
| } |
| |
| webrtc::RtpParameters rtp_params = it->second->GetRtpParameters(); |
| // Need to add the common list of codecs to the send stream-specific |
| // RTP parameters. |
| for (const VideoCodec& codec : send_params_.codecs) { |
| rtp_params.codecs.push_back(codec.ToCodecParameters()); |
| } |
| return rtp_params; |
| } |
| |
| bool WebRtcVideoChannel2::SetRtpSendParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpSendParameters"); |
| rtc::CritScope stream_lock(&stream_crit_); |
| auto it = send_streams_.find(ssrc); |
| if (it == send_streams_.end()) { |
| LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream " |
| << "with ssrc " << ssrc << " which doesn't exist."; |
| return false; |
| } |
| |
| // TODO(deadbeef): Handle setting parameters with a list of codecs in a |
| // different order (which should change the send codec). |
| webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc); |
| if (current_parameters.codecs != parameters.codecs) { |
| LOG(LS_ERROR) << "Using SetParameters to change the set of codecs " |
| << "is not currently supported."; |
| return false; |
| } |
| |
| return it->second->SetRtpParameters(parameters); |
| } |
| |
| webrtc::RtpParameters WebRtcVideoChannel2::GetRtpReceiveParameters( |
| uint32_t ssrc) const { |
| rtc::CritScope stream_lock(&stream_crit_); |
| auto it = receive_streams_.find(ssrc); |
| if (it == receive_streams_.end()) { |
| LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream " |
| << "with ssrc " << ssrc << " which doesn't exist."; |
| return webrtc::RtpParameters(); |
| } |
| |
| // TODO(deadbeef): Return stream-specific parameters. |
| webrtc::RtpParameters rtp_params = CreateRtpParametersWithOneEncoding(); |
| for (const VideoCodec& codec : recv_params_.codecs) { |
| rtp_params.codecs.push_back(codec.ToCodecParameters()); |
| } |
| rtp_params.encodings[0].ssrc = it->second->GetFirstPrimarySsrc(); |
| return rtp_params; |
| } |
| |
| bool WebRtcVideoChannel2::SetRtpReceiveParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpReceiveParameters"); |
| rtc::CritScope stream_lock(&stream_crit_); |
| auto it = receive_streams_.find(ssrc); |
| if (it == receive_streams_.end()) { |
| LOG(LS_ERROR) << "Attempting to set RTP receive parameters for stream " |
| << "with ssrc " << ssrc << " which doesn't exist."; |
| return false; |
| } |
| |
| webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc); |
| if (current_parameters != parameters) { |
| LOG(LS_ERROR) << "Changing the RTP receive parameters is currently " |
| << "unsupported."; |
| return false; |
| } |
| return true; |
| } |
| |
| bool WebRtcVideoChannel2::GetChangedRecvParameters( |
| const VideoRecvParameters& params, |
| ChangedRecvParameters* changed_params) const { |
| if (!ValidateCodecFormats(params.codecs) || |
| !ValidateRtpExtensions(params.extensions)) { |
| return false; |
| } |
| |
| // Handle receive codecs. |
| const std::vector<VideoCodecSettings> mapped_codecs = |
| MapCodecs(params.codecs); |
| if (mapped_codecs.empty()) { |
| LOG(LS_ERROR) << "SetRecvParameters called without any video codecs."; |
| return false; |
| } |
| |
| std::vector<VideoCodecSettings> supported_codecs = |
| FilterSupportedCodecs(mapped_codecs); |
| |
| if (mapped_codecs.size() != supported_codecs.size()) { |
| LOG(LS_ERROR) << "SetRecvParameters called with unsupported video codecs."; |
| return false; |
| } |
| |
| if (ReceiveCodecsHaveChanged(recv_codecs_, supported_codecs)) { |
| changed_params->codec_settings = |
| rtc::Optional<std::vector<VideoCodecSettings>>(supported_codecs); |
| } |
| |
| // Handle RTP header extensions. |
| std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( |
| params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false); |
| if (filtered_extensions != recv_rtp_extensions_) { |
| changed_params->rtp_header_extensions = |
| rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions); |
| } |
| |
| return true; |
| } |
| |
| bool WebRtcVideoChannel2::SetRecvParameters(const VideoRecvParameters& params) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvParameters"); |
| LOG(LS_INFO) << "SetRecvParameters: " << params.ToString(); |
| ChangedRecvParameters changed_params; |
| if (!GetChangedRecvParameters(params, &changed_params)) { |
| return false; |
| } |
| if (changed_params.rtp_header_extensions) { |
| recv_rtp_extensions_ = *changed_params.rtp_header_extensions; |
| } |
| if (changed_params.codec_settings) { |
| LOG(LS_INFO) << "Changing recv codecs from " |
| << CodecSettingsVectorToString(recv_codecs_) << " to " |
| << CodecSettingsVectorToString(*changed_params.codec_settings); |
| recv_codecs_ = *changed_params.codec_settings; |
| } |
| |
| { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (auto& kv : receive_streams_) { |
| kv.second->SetRecvParameters(changed_params); |
| } |
| } |
| recv_params_ = params; |
| return true; |
| } |
| |
| std::string WebRtcVideoChannel2::CodecSettingsVectorToString( |
| const std::vector<VideoCodecSettings>& codecs) { |
| std::stringstream out; |
| out << '{'; |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| out << codecs[i].codec.ToString(); |
| if (i != codecs.size() - 1) { |
| out << ", "; |
| } |
| } |
| out << '}'; |
| return out.str(); |
| } |
| |
| bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) { |
| if (!send_codec_) { |
| LOG(LS_VERBOSE) << "GetSendCodec: No send codec set."; |
| return false; |
| } |
| *codec = send_codec_->codec; |
| return true; |
| } |
| |
| bool WebRtcVideoChannel2::SetSend(bool send) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSend"); |
| LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false"); |
| if (send && !send_codec_) { |
| LOG(LS_ERROR) << "SetSend(true) called before setting codec."; |
| return false; |
| } |
| { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (const auto& kv : send_streams_) { |
| kv.second->SetSend(send); |
| } |
| } |
| sending_ = send; |
| return true; |
| } |
| |
| // TODO(nisse): The enable argument was used for mute logic which has |
| // been moved to VideoBroadcaster. So remove the argument from this |
| // method. |
| bool WebRtcVideoChannel2::SetVideoSend( |
| uint32_t ssrc, |
| bool enable, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<cricket::VideoFrame>* source) { |
| TRACE_EVENT0("webrtc", "SetVideoSend"); |
| RTC_DCHECK(ssrc != 0); |
| LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", enable = " << enable |
| << ", options: " << (options ? options->ToString() : "nullptr") |
| << ", source = " << (source ? "(source)" : "nullptr") << ")"; |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| const auto& kv = send_streams_.find(ssrc); |
| if (kv == send_streams_.end()) { |
| // Allow unknown ssrc only if source is null. |
| RTC_CHECK(source == nullptr); |
| LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; |
| return false; |
| } |
| |
| return kv->second->SetVideoSend(enable, options, source); |
| } |
| |
| bool WebRtcVideoChannel2::ValidateSendSsrcAvailability( |
| const StreamParams& sp) const { |
| for (uint32_t ssrc : sp.ssrcs) { |
| if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) { |
| LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists."; |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| bool WebRtcVideoChannel2::ValidateReceiveSsrcAvailability( |
| const StreamParams& sp) const { |
| for (uint32_t ssrc : sp.ssrcs) { |
| if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) { |
| LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc |
| << "' already exists."; |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) { |
| LOG(LS_INFO) << "AddSendStream: " << sp.ToString(); |
| if (!ValidateStreamParams(sp)) |
| return false; |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| |
| if (!ValidateSendSsrcAvailability(sp)) |
| return false; |
| |
| for (uint32_t used_ssrc : sp.ssrcs) |
| send_ssrcs_.insert(used_ssrc); |
| |
| webrtc::VideoSendStream::Config config(this); |
| config.suspend_below_min_bitrate = video_config_.suspend_below_min_bitrate; |
| WebRtcVideoSendStream* stream = new WebRtcVideoSendStream( |
| call_, sp, std::move(config), default_send_options_, |
| external_encoder_factory_, video_config_.enable_cpu_overuse_detection, |
| bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_, |
| send_params_); |
| |
| uint32_t ssrc = sp.first_ssrc(); |
| RTC_DCHECK(ssrc != 0); |
| send_streams_[ssrc] = stream; |
| |
| if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) { |
| rtcp_receiver_report_ssrc_ = ssrc; |
| LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added " |
| "a send stream."; |
| for (auto& kv : receive_streams_) |
| kv.second->SetLocalSsrc(ssrc); |
| } |
| if (sending_) { |
| stream->SetSend(true); |
| } |
| |
| return true; |
| } |
| |
| bool WebRtcVideoChannel2::RemoveSendStream(uint32_t ssrc) { |
| LOG(LS_INFO) << "RemoveSendStream: " << ssrc; |
| |
| WebRtcVideoSendStream* removed_stream; |
| { |
| rtc::CritScope stream_lock(&stream_crit_); |
| std::map<uint32_t, WebRtcVideoSendStream*>::iterator it = |
| send_streams_.find(ssrc); |
| if (it == send_streams_.end()) { |
| return false; |
| } |
| |
| for (uint32_t old_ssrc : it->second->GetSsrcs()) |
| send_ssrcs_.erase(old_ssrc); |
| |
| removed_stream = it->second; |
| send_streams_.erase(it); |
| |
| // Switch receiver report SSRCs, the one in use is no longer valid. |
| if (rtcp_receiver_report_ssrc_ == ssrc) { |
| rtcp_receiver_report_ssrc_ = send_streams_.empty() |
| ? kDefaultRtcpReceiverReportSsrc |
| : send_streams_.begin()->first; |
| LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because the " |
| "previous local SSRC was removed."; |
| |
| for (auto& kv : receive_streams_) { |
| kv.second->SetLocalSsrc(rtcp_receiver_report_ssrc_); |
| } |
| } |
| } |
| |
| delete removed_stream; |
| |
| return true; |
| } |
| |
| void WebRtcVideoChannel2::DeleteReceiveStream( |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream* stream) { |
| for (uint32_t old_ssrc : stream->GetSsrcs()) |
| receive_ssrcs_.erase(old_ssrc); |
| delete stream; |
| } |
| |
| bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp) { |
| return AddRecvStream(sp, false); |
| } |
| |
| bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp, |
| bool default_stream) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| |
| LOG(LS_INFO) << "AddRecvStream" << (default_stream ? " (default stream)" : "") |
| << ": " << sp.ToString(); |
| if (!ValidateStreamParams(sp)) |
| return false; |
| |
| uint32_t ssrc = sp.first_ssrc(); |
| RTC_DCHECK(ssrc != 0); // TODO(pbos): Is this ever valid? |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| // Remove running stream if this was a default stream. |
| const auto& prev_stream = receive_streams_.find(ssrc); |
| if (prev_stream != receive_streams_.end()) { |
| if (default_stream || !prev_stream->second->IsDefaultStream()) { |
| LOG(LS_ERROR) << "Receive stream for SSRC '" << ssrc |
| << "' already exists."; |
| return false; |
| } |
| DeleteReceiveStream(prev_stream->second); |
| receive_streams_.erase(prev_stream); |
| } |
| |
| if (!ValidateReceiveSsrcAvailability(sp)) |
| return false; |
| |
| for (uint32_t used_ssrc : sp.ssrcs) |
| receive_ssrcs_.insert(used_ssrc); |
| |
| webrtc::VideoReceiveStream::Config config(this); |
| ConfigureReceiverRtp(&config, sp); |
| |
| // Set up A/V sync group based on sync label. |
| config.sync_group = sp.sync_label; |
| |
| config.rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false; |
| config.rtp.transport_cc = |
| send_codec_ ? HasTransportCc(send_codec_->codec) : false; |
| config.disable_prerenderer_smoothing = |
| video_config_.disable_prerenderer_smoothing; |
| |
| receive_streams_[ssrc] = new WebRtcVideoReceiveStream( |
| call_, sp, std::move(config), external_decoder_factory_, default_stream, |
| recv_codecs_, red_disabled_by_remote_side_); |
| |
| return true; |
| } |
| |
| void WebRtcVideoChannel2::ConfigureReceiverRtp( |
| webrtc::VideoReceiveStream::Config* config, |
| const StreamParams& sp) const { |
| uint32_t ssrc = sp.first_ssrc(); |
| |
| config->rtp.remote_ssrc = ssrc; |
| config->rtp.local_ssrc = rtcp_receiver_report_ssrc_; |
| |
| config->rtp.extensions = recv_rtp_extensions_; |
| // Whether or not the receive stream sends reduced size RTCP is determined |
| // by the send params. |
| // TODO(deadbeef): Once we change "send_params" to "sender_params" and |
| // "recv_params" to "receiver_params", we should get this out of |
| // receiver_params_. |
| config->rtp.rtcp_mode = send_params_.rtcp.reduced_size |
| ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound; |
| |
| // TODO(pbos): This protection is against setting the same local ssrc as |
| // remote which is not permitted by the lower-level API. RTCP requires a |
| // corresponding sender SSRC. Figure out what to do when we don't have |
| // (receive-only) or know a good local SSRC. |
| if (config->rtp.remote_ssrc == config->rtp.local_ssrc) { |
| if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) { |
| config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc; |
| } else { |
| config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc + 1; |
| } |
| } |
| |
| for (size_t i = 0; i < recv_codecs_.size(); ++i) { |
| uint32_t rtx_ssrc; |
| if (recv_codecs_[i].rtx_payload_type != -1 && |
| sp.GetFidSsrc(ssrc, &rtx_ssrc)) { |
| webrtc::VideoReceiveStream::Config::Rtp::Rtx& rtx = |
| config->rtp.rtx[recv_codecs_[i].codec.id]; |
| rtx.ssrc = rtx_ssrc; |
| rtx.payload_type = recv_codecs_[i].rtx_payload_type; |
| } |
| } |
| } |
| |
| bool WebRtcVideoChannel2::RemoveRecvStream(uint32_t ssrc) { |
| LOG(LS_INFO) << "RemoveRecvStream: " << ssrc; |
| if (ssrc == 0) { |
| LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported."; |
| return false; |
| } |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator stream = |
| receive_streams_.find(ssrc); |
| if (stream == receive_streams_.end()) { |
| LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc; |
| return false; |
| } |
| DeleteReceiveStream(stream->second); |
| receive_streams_.erase(stream); |
| |
| return true; |
| } |
| |
| bool WebRtcVideoChannel2::SetSink(uint32_t ssrc, |
| rtc::VideoSinkInterface<VideoFrame>* sink) { |
| LOG(LS_INFO) << "SetSink: ssrc:" << ssrc << " " |
| << (sink ? "(ptr)" : "nullptr"); |
| if (ssrc == 0) { |
| default_unsignalled_ssrc_handler_.SetDefaultSink(this, sink); |
| return true; |
| } |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it = |
| receive_streams_.find(ssrc); |
| if (it == receive_streams_.end()) { |
| return false; |
| } |
| |
| it->second->SetSink(sink); |
| return true; |
| } |
| |
| bool WebRtcVideoChannel2::GetStats(VideoMediaInfo* info) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::GetStats"); |
| |
| // Log stats periodically. |
| bool log_stats = false; |
| int64_t now_ms = rtc::TimeMillis(); |
| if (last_stats_log_ms_ == -1 || |
| now_ms - last_stats_log_ms_ > kStatsLogIntervalMs) { |
| last_stats_log_ms_ = now_ms; |
| log_stats = true; |
| } |
| |
| info->Clear(); |
| FillSenderStats(info, log_stats); |
| FillReceiverStats(info, log_stats); |
| webrtc::Call::Stats stats = call_->GetStats(); |
| FillBandwidthEstimationStats(stats, info); |
| if (stats.rtt_ms != -1) { |
| for (size_t i = 0; i < info->senders.size(); ++i) { |
| info->senders[i].rtt_ms = stats.rtt_ms; |
| } |
| } |
| |
| if (log_stats) |
| LOG(LS_INFO) << stats.ToString(now_ms); |
| |
| return true; |
| } |
| |
| void WebRtcVideoChannel2::FillSenderStats(VideoMediaInfo* video_media_info, |
| bool log_stats) { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it = |
| send_streams_.begin(); |
| it != send_streams_.end(); ++it) { |
| video_media_info->senders.push_back( |
| it->second->GetVideoSenderInfo(log_stats)); |
| } |
| } |
| |
| void WebRtcVideoChannel2::FillReceiverStats(VideoMediaInfo* video_media_info, |
| bool log_stats) { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it = |
| receive_streams_.begin(); |
| it != receive_streams_.end(); ++it) { |
| video_media_info->receivers.push_back( |
| it->second->GetVideoReceiverInfo(log_stats)); |
| } |
| } |
| |
| void WebRtcVideoChannel2::FillBandwidthEstimationStats( |
| const webrtc::Call::Stats& stats, |
| VideoMediaInfo* video_media_info) { |
| BandwidthEstimationInfo bwe_info; |
| bwe_info.available_send_bandwidth = stats.send_bandwidth_bps; |
| bwe_info.available_recv_bandwidth = stats.recv_bandwidth_bps; |
| bwe_info.bucket_delay = stats.pacer_delay_ms; |
| |
| // Get send stream bitrate stats. |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream = |
| send_streams_.begin(); |
| stream != send_streams_.end(); ++stream) { |
| stream->second->FillBandwidthEstimationInfo(&bwe_info); |
| } |
| video_media_info->bw_estimations.push_back(bwe_info); |
| } |
| |
| void WebRtcVideoChannel2::OnPacketReceived( |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time) { |
| const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, |
| packet_time.not_before); |
| const webrtc::PacketReceiver::DeliveryStatus delivery_result = |
| call_->Receiver()->DeliverPacket( |
| webrtc::MediaType::VIDEO, |
| packet->cdata(), packet->size(), |
| webrtc_packet_time); |
| switch (delivery_result) { |
| case webrtc::PacketReceiver::DELIVERY_OK: |
| return; |
| case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR: |
| return; |
| case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC: |
| break; |
| } |
| |
| uint32_t ssrc = 0; |
| if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) { |
| return; |
| } |
| |
| int payload_type = 0; |
| if (!GetRtpPayloadType(packet->cdata(), packet->size(), &payload_type)) { |
| return; |
| } |
| |
| // See if this payload_type is registered as one that usually gets its own |
| // SSRC (RTX) or at least is safe to drop either way (ULPFEC). If it is, and |
| // it wasn't handled above by DeliverPacket, that means we don't know what |
| // stream it associates with, and we shouldn't ever create an implicit channel |
| // for these. |
| for (auto& codec : recv_codecs_) { |
| if (payload_type == codec.rtx_payload_type || |
| payload_type == codec.fec.red_rtx_payload_type || |
| payload_type == codec.fec.ulpfec_payload_type) { |
| return; |
| } |
| } |
| |
| switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) { |
| case UnsignalledSsrcHandler::kDropPacket: |
| return; |
| case UnsignalledSsrcHandler::kDeliverPacket: |
| break; |
| } |
| |
| if (call_->Receiver()->DeliverPacket( |
| webrtc::MediaType::VIDEO, |
| packet->cdata(), packet->size(), |
| webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) { |
| LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery."; |
| return; |
| } |
| } |
| |
| void WebRtcVideoChannel2::OnRtcpReceived( |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time) { |
| const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, |
| packet_time.not_before); |
| // TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver |
| // for both audio and video on the same path. Since BundleFilter doesn't |
| // filter RTCP anymore incoming RTCP packets could've been going to audio (so |
| // logging failures spam the log). |
| call_->Receiver()->DeliverPacket( |
| webrtc::MediaType::VIDEO, |
| packet->cdata(), packet->size(), |
| webrtc_packet_time); |
| } |
| |
| void WebRtcVideoChannel2::OnReadyToSend(bool ready) { |
| LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready."); |
| call_->SignalChannelNetworkState( |
| webrtc::MediaType::VIDEO, |
| ready ? webrtc::kNetworkUp : webrtc::kNetworkDown); |
| } |
| |
| void WebRtcVideoChannel2::OnNetworkRouteChanged( |
| const std::string& transport_name, |
| const rtc::NetworkRoute& network_route) { |
| call_->OnNetworkRouteChanged(transport_name, network_route); |
| } |
| |
| void WebRtcVideoChannel2::SetInterface(NetworkInterface* iface) { |
| MediaChannel::SetInterface(iface); |
| // Set the RTP recv/send buffer to a bigger size |
| MediaChannel::SetOption(NetworkInterface::ST_RTP, |
| rtc::Socket::OPT_RCVBUF, |
| kVideoRtpBufferSize); |
| |
| // Speculative change to increase the outbound socket buffer size. |
| // In b/15152257, we are seeing a significant number of packets discarded |
| // due to lack of socket buffer space, although it's not yet clear what the |
| // ideal value should be. |
| MediaChannel::SetOption(NetworkInterface::ST_RTP, |
| rtc::Socket::OPT_SNDBUF, |
| kVideoRtpBufferSize); |
| } |
| |
| bool WebRtcVideoChannel2::SendRtp(const uint8_t* data, |
| size_t len, |
| const webrtc::PacketOptions& options) { |
| rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen); |
| rtc::PacketOptions rtc_options; |
| rtc_options.packet_id = options.packet_id; |
| return MediaChannel::SendPacket(&packet, rtc_options); |
| } |
| |
| bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) { |
| rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen); |
| return MediaChannel::SendRtcp(&packet, rtc::PacketOptions()); |
| } |
| |
| WebRtcVideoChannel2::WebRtcVideoSendStream::VideoSendStreamParameters:: |
| VideoSendStreamParameters( |
| webrtc::VideoSendStream::Config config, |
| const VideoOptions& options, |
| int max_bitrate_bps, |
| const rtc::Optional<VideoCodecSettings>& codec_settings) |
| : config(std::move(config)), |
| options(options), |
| max_bitrate_bps(max_bitrate_bps), |
| conference_mode(false), |
| codec_settings(codec_settings) {} |
| |
| WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder::AllocatedEncoder( |
| webrtc::VideoEncoder* encoder, |
| webrtc::VideoCodecType type, |
| bool external) |
| : encoder(encoder), |
| external_encoder(nullptr), |
| type(type), |
| external(external) { |
| if (external) { |
| external_encoder = encoder; |
| this->encoder = |
| new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder); |
| } |
| } |
| |
| WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream( |
| webrtc::Call* call, |
| const StreamParams& sp, |
| webrtc::VideoSendStream::Config config, |
| const VideoOptions& options, |
| WebRtcVideoEncoderFactory* external_encoder_factory, |
| bool enable_cpu_overuse_detection, |
| int max_bitrate_bps, |
| const rtc::Optional<VideoCodecSettings>& codec_settings, |
| const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions, |
| // TODO(deadbeef): Don't duplicate information between send_params, |
| // rtp_extensions, options, etc. |
| const VideoSendParameters& send_params) |
| : worker_thread_(rtc::Thread::Current()), |
| ssrcs_(sp.ssrcs), |
| ssrc_groups_(sp.ssrc_groups), |
| call_(call), |
| cpu_restricted_counter_(0), |
| number_of_cpu_adapt_changes_(0), |
| frame_count_(0), |
| cpu_restricted_frame_count_(0), |
| source_(nullptr), |
| external_encoder_factory_(external_encoder_factory), |
| stream_(nullptr), |
| encoder_sink_(nullptr), |
| parameters_(std::move(config), options, max_bitrate_bps, codec_settings), |
| rtp_parameters_(CreateRtpParametersWithOneEncoding()), |
| allocated_encoder_(nullptr, webrtc::kVideoCodecUnknown, false), |
| sending_(false), |
| last_frame_timestamp_us_(0) { |
| parameters_.config.rtp.max_packet_size = kVideoMtu; |
| parameters_.conference_mode = send_params.conference_mode; |
| |
| sp.GetPrimarySsrcs(¶meters_.config.rtp.ssrcs); |
| sp.GetFidSsrcs(parameters_.config.rtp.ssrcs, |
| ¶meters_.config.rtp.rtx.ssrcs); |
| parameters_.config.rtp.c_name = sp.cname; |
| if (rtp_extensions) { |
| parameters_.config.rtp.extensions = *rtp_extensions; |
| } |
| parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size |
| ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound; |
| parameters_.config.overuse_callback = |
| enable_cpu_overuse_detection ? this : nullptr; |
| |
| // Only request rotation at the source when we positively know that the remote |
| // side doesn't support the rotation extension. This allows us to prepare the |
| // encoder in the expectation that rotation is supported - which is the common |
| // case. |
| sink_wants_.rotation_applied = |
| rtp_extensions && |
| !ContainsHeaderExtension(*rtp_extensions, |
| webrtc::RtpExtension::kVideoRotationUri); |
| |
| if (codec_settings) { |
| SetCodec(*codec_settings); |
| } |
| } |
| |
| WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() { |
| DisconnectSource(); |
| if (stream_ != NULL) { |
| call_->DestroyVideoSendStream(stream_); |
| } |
| DestroyVideoEncoder(&allocated_encoder_); |
| UpdateHistograms(); |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::UpdateHistograms() const { |
| const int kMinRequiredFrames = 200; |
| if (frame_count_ > kMinRequiredFrames) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.CpuLimitedResolutionInPercent", |
| cpu_restricted_frame_count_ * 100 / frame_count_); |
| } |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame( |
| const VideoFrame& frame) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::OnFrame"); |
| webrtc::VideoFrame video_frame(frame.video_frame_buffer(), |
| frame.rotation(), |
| frame.timestamp_us()); |
| |
| rtc::CritScope cs(&lock_); |
| |
| if (video_frame.width() != last_frame_info_.width || |
| video_frame.height() != last_frame_info_.height || |
| video_frame.rotation() != last_frame_info_.rotation || |
| video_frame.is_texture() != last_frame_info_.is_texture) { |
| last_frame_info_.width = video_frame.width(); |
| last_frame_info_.height = video_frame.height(); |
| last_frame_info_.rotation = video_frame.rotation(); |
| last_frame_info_.is_texture = video_frame.is_texture(); |
| |
| LOG(LS_INFO) << "Video frame parameters changed: dimensions=" |
| << last_frame_info_.width << "x" << last_frame_info_.height |
| << ", rotation=" << last_frame_info_.rotation |
| << ", texture=" << last_frame_info_.is_texture; |
| } |
| |
| if (encoder_sink_ == NULL) { |
| // Frame input before send codecs are configured, dropping frame. |
| return; |
| } |
| |
| last_frame_timestamp_us_ = video_frame.timestamp_us(); |
| |
| ++frame_count_; |
| if (cpu_restricted_counter_ > 0) |
| ++cpu_restricted_frame_count_; |
| |
| // Forward frame to the encoder regardless if we are sending or not. This is |
| // to ensure that the encoder can be reconfigured with the correct frame size |
| // as quickly as possible. |
| encoder_sink_->OnFrame(video_frame); |
| } |
| |
| bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoSend( |
| bool enable, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<cricket::VideoFrame>* source) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetVideoSend"); |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| // Ignore |options| pointer if |enable| is false. |
| bool options_present = enable && options; |
| bool source_changing = source_ != source; |
| if (source_changing) { |
| DisconnectSource(); |
| } |
| |
| if (options_present) { |
| VideoOptions old_options = parameters_.options; |
| parameters_.options.SetAll(*options); |
| // If options has changed and SetCodec has been called. |
| if (parameters_.options != old_options && stream_) { |
| ReconfigureEncoder(); |
| } |
| } |
| |
| if (source_changing) { |
| rtc::CritScope cs(&lock_); |
| if (source == nullptr && encoder_sink_ != nullptr && |
| last_frame_info_.width > 0) { |
| LOG(LS_VERBOSE) << "Disabling capturer, sending black frame."; |
| // Force this black frame not to be dropped due to timestamp order |
| // check. As IncomingCapturedFrame will drop the frame if this frame's |
| // timestamp is less than or equal to last frame's timestamp, it is |
| // necessary to give this black frame a larger timestamp than the |
| // previous one. |
| last_frame_timestamp_us_ += rtc::kNumMicrosecsPerMillisec; |
| rtc::scoped_refptr<webrtc::I420Buffer> black_buffer( |
| webrtc::I420Buffer::Create(last_frame_info_.width, |
| last_frame_info_.height)); |
| black_buffer->SetToBlack(); |
| |
| encoder_sink_->OnFrame(webrtc::VideoFrame( |
| black_buffer, last_frame_info_.rotation, last_frame_timestamp_us_)); |
| } |
| source_ = source; |
| } |
| |
| if (source_changing && source_) { |
| // |source_->AddOrUpdateSink| may not be called while holding |lock_| since |
| // that might cause a lock order inversion. |
| source_->AddOrUpdateSink(this, sink_wants_); |
| } |
| return true; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectSource() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (source_ == nullptr) { |
| return; |
| } |
| |
| // |source_->RemoveSink| may not be called while holding |lock_| since |
| // that might cause a lock order inversion. |
| source_->RemoveSink(this); |
| source_ = nullptr; |
| // Reset |cpu_restricted_counter_| if the source is changed. It is not |
| // possible to know if the video resolution is restricted by CPU usage after |
| // the source is changed since the next source might be screen capture |
| // with another resolution and frame rate. |
| cpu_restricted_counter_ = 0; |
| } |
| |
| const std::vector<uint32_t>& |
| WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const { |
| return ssrcs_; |
| } |
| |
| webrtc::VideoCodecType CodecTypeFromName(const std::string& name) { |
| if (CodecNamesEq(name, kVp8CodecName)) { |
| return webrtc::kVideoCodecVP8; |
| } else if (CodecNamesEq(name, kVp9CodecName)) { |
| return webrtc::kVideoCodecVP9; |
| } else if (CodecNamesEq(name, kH264CodecName)) { |
| return webrtc::kVideoCodecH264; |
| } |
| return webrtc::kVideoCodecUnknown; |
| } |
| |
| WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder |
| WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder( |
| const VideoCodec& codec) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| webrtc::VideoCodecType type = CodecTypeFromName(codec.name); |
| |
| // Do not re-create encoders of the same type. |
| if (type == allocated_encoder_.type && allocated_encoder_.encoder != NULL) { |
| return allocated_encoder_; |
| } |
| |
| if (external_encoder_factory_ != NULL) { |
| webrtc::VideoEncoder* encoder = |
| external_encoder_factory_->CreateVideoEncoder(type); |
| if (encoder != NULL) { |
| return AllocatedEncoder(encoder, type, true); |
| } |
| } |
| |
| if (type == webrtc::kVideoCodecVP8) { |
| return AllocatedEncoder( |
| webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp8), type, false); |
| } else if (type == webrtc::kVideoCodecVP9) { |
| return AllocatedEncoder( |
| webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp9), type, false); |
| } else if (type == webrtc::kVideoCodecH264) { |
| return AllocatedEncoder( |
| webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kH264), type, false); |
| } |
| |
| // This shouldn't happen, we should not be trying to create something we don't |
| // support. |
| RTC_DCHECK(false); |
| return AllocatedEncoder(NULL, webrtc::kVideoCodecUnknown, false); |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder( |
| AllocatedEncoder* encoder) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (encoder->external) { |
| external_encoder_factory_->DestroyVideoEncoder(encoder->external_encoder); |
| } |
| delete encoder->encoder; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec( |
| const VideoCodecSettings& codec_settings) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| parameters_.encoder_config = CreateVideoEncoderConfig(codec_settings.codec); |
| RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0u); |
| |
| AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec); |
| parameters_.config.encoder_settings.encoder = new_encoder.encoder; |
| parameters_.config.encoder_settings.full_overuse_time = new_encoder.external; |
| parameters_.config.encoder_settings.payload_name = codec_settings.codec.name; |
| parameters_.config.encoder_settings.payload_type = codec_settings.codec.id; |
| if (new_encoder.external) { |
| webrtc::VideoCodecType type = CodecTypeFromName(codec_settings.codec.name); |
| parameters_.config.encoder_settings.internal_source = |
| external_encoder_factory_->EncoderTypeHasInternalSource(type); |
| } |
| parameters_.config.rtp.fec = codec_settings.fec; |
| |
| // Set RTX payload type if RTX is enabled. |
| if (!parameters_.config.rtp.rtx.ssrcs.empty()) { |
| if (codec_settings.rtx_payload_type == -1) { |
| LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX " |
| "payload type. Ignoring."; |
| parameters_.config.rtp.rtx.ssrcs.clear(); |
| } else { |
| parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type; |
| } |
| } |
| |
| parameters_.config.rtp.nack.rtp_history_ms = |
| HasNack(codec_settings.codec) ? kNackHistoryMs : 0; |
| |
| parameters_.codec_settings = |
| rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>(codec_settings); |
| |
| LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetCodec."; |
| RecreateWebRtcStream(); |
| if (allocated_encoder_.encoder != new_encoder.encoder) { |
| DestroyVideoEncoder(&allocated_encoder_); |
| allocated_encoder_ = new_encoder; |
| } |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters( |
| const ChangedSendParameters& params) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| // |recreate_stream| means construction-time parameters have changed and the |
| // sending stream needs to be reset with the new config. |
| bool recreate_stream = false; |
| if (params.rtcp_mode) { |
| parameters_.config.rtp.rtcp_mode = *params.rtcp_mode; |
| recreate_stream = true; |
| } |
| if (params.rtp_header_extensions) { |
| parameters_.config.rtp.extensions = *params.rtp_header_extensions; |
| recreate_stream = true; |
| } |
| if (params.max_bandwidth_bps) { |
| parameters_.max_bitrate_bps = *params.max_bandwidth_bps; |
| ReconfigureEncoder(); |
| } |
| if (params.conference_mode) { |
| parameters_.conference_mode = *params.conference_mode; |
| } |
| |
| // Set codecs and options. |
| if (params.codec) { |
| SetCodec(*params.codec); |
| recreate_stream = false; // SetCodec has already recreated the stream. |
| } else if (params.conference_mode && parameters_.codec_settings) { |
| SetCodec(*parameters_.codec_settings); |
| recreate_stream = false; // SetCodec has already recreated the stream. |
| } |
| if (recreate_stream) { |
| LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters"; |
| RecreateWebRtcStream(); |
| } |
| |
| // |source_->AddOrUpdateSink| may not be called while holding |lock_| since |
| // that might cause a lock order inversion. |
| if (params.rtp_header_extensions) { |
| sink_wants_.rotation_applied = !ContainsHeaderExtension( |
| *params.rtp_header_extensions, webrtc::RtpExtension::kVideoRotationUri); |
| if (source_) { |
| source_->AddOrUpdateSink(this, sink_wants_); |
| } |
| } |
| } |
| |
| bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpParameters( |
| const webrtc::RtpParameters& new_parameters) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (!ValidateRtpParameters(new_parameters)) { |
| return false; |
| } |
| |
| bool reconfigure_encoder = new_parameters.encodings[0].max_bitrate_bps != |
| rtp_parameters_.encodings[0].max_bitrate_bps; |
| rtp_parameters_ = new_parameters; |
| // Codecs are currently handled at the WebRtcVideoChannel2 level. |
| rtp_parameters_.codecs.clear(); |
| if (reconfigure_encoder) { |
| ReconfigureEncoder(); |
| } |
| // Encoding may have been activated/deactivated. |
| UpdateSendState(); |
| return true; |
| } |
| |
| webrtc::RtpParameters |
| WebRtcVideoChannel2::WebRtcVideoSendStream::GetRtpParameters() const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| return rtp_parameters_; |
| } |
| |
| bool WebRtcVideoChannel2::WebRtcVideoSendStream::ValidateRtpParameters( |
| const webrtc::RtpParameters& rtp_parameters) { |
| if (rtp_parameters.encodings.size() != 1) { |
| LOG(LS_ERROR) |
| << "Attempted to set RtpParameters without exactly one encoding"; |
| return false; |
| } |
| return true; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::UpdateSendState() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| // TODO(deadbeef): Need to handle more than one encoding in the future. |
| RTC_DCHECK(rtp_parameters_.encodings.size() == 1u); |
| if (sending_ && rtp_parameters_.encodings[0].active) { |
| RTC_DCHECK(stream_ != nullptr); |
| stream_->Start(); |
| } else { |
| if (stream_ != nullptr) { |
| stream_->Stop(); |
| } |
| } |
| } |
| |
| webrtc::VideoEncoderConfig |
| WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig( |
| const VideoCodec& codec) const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| webrtc::VideoEncoderConfig encoder_config; |
| bool is_screencast = parameters_.options.is_screencast.value_or(false); |
| if (is_screencast) { |
| encoder_config.min_transmit_bitrate_bps = |
| 1000 * parameters_.options.screencast_min_bitrate_kbps.value_or(0); |
| encoder_config.content_type = |
| webrtc::VideoEncoderConfig::ContentType::kScreen; |
| } else { |
| encoder_config.min_transmit_bitrate_bps = 0; |
| encoder_config.content_type = |
| webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo; |
| } |
| |
| // By default, the stream count for the codec configuration should match the |
| // number of negotiated ssrcs. But if the codec is blacklisted for simulcast |
| // or a screencast, only configure a single stream. |
| encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size(); |
| if (IsCodecBlacklistedForSimulcast(codec.name) || is_screencast) { |
| encoder_config.number_of_streams = 1; |
| } |
| |
| int stream_max_bitrate = |
| MinPositive(rtp_parameters_.encodings[0].max_bitrate_bps, |
| parameters_.max_bitrate_bps); |
| |
| int codec_max_bitrate_kbps; |
| if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps)) { |
| stream_max_bitrate = codec_max_bitrate_kbps * 1000; |
| } |
| encoder_config.max_bitrate_bps = stream_max_bitrate; |
| |
| int max_qp = kDefaultQpMax; |
| codec.GetParam(kCodecParamMaxQuantization, &max_qp); |
| int max_framerate = |
| codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate; |
| |
| encoder_config.video_stream_factory = |
| new rtc::RefCountedObject<EncoderStreamFactory>( |
| codec.name, max_qp, max_framerate, is_screencast, |
| parameters_.conference_mode); |
| return encoder_config; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::ReconfigureEncoder() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0u); |
| |
| RTC_CHECK(parameters_.codec_settings); |
| VideoCodecSettings codec_settings = *parameters_.codec_settings; |
| |
| webrtc::VideoEncoderConfig encoder_config = |
| CreateVideoEncoderConfig(codec_settings.codec); |
| |
| encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings( |
| codec_settings.codec); |
| |
| stream_->ReconfigureVideoEncoder(encoder_config.Copy()); |
| |
| encoder_config.encoder_specific_settings = NULL; |
| |
| parameters_.encoder_config = std::move(encoder_config); |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSend(bool send) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| sending_ = send; |
| UpdateSendState(); |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::AddOrUpdateSink( |
| VideoSinkInterface<webrtc::VideoFrame>* sink, |
| const rtc::VideoSinkWants& wants) { |
| // TODO(perkj): Actually consider the encoder |wants| and remove |
| // WebRtcVideoSendStream::OnLoadUpdate(Load load). |
| rtc::CritScope cs(&lock_); |
| RTC_DCHECK(!encoder_sink_ || encoder_sink_ == sink); |
| encoder_sink_ = sink; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::RemoveSink( |
| VideoSinkInterface<webrtc::VideoFrame>* sink) { |
| rtc::CritScope cs(&lock_); |
| RTC_DCHECK_EQ(encoder_sink_, sink); |
| encoder_sink_ = nullptr; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate(Load load) { |
| if (worker_thread_ != rtc::Thread::Current()) { |
| invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, worker_thread_, |
| rtc::Bind(&WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate, |
| this, load)); |
| return; |
| } |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (!source_) { |
| return; |
| } |
| |
| LOG(LS_INFO) << "OnLoadUpdate " << load << ", is_screencast: " |
| << (parameters_.options.is_screencast |
| ? (*parameters_.options.is_screencast ? "true" : "false") |
| : "unset"); |
| // Do not adapt resolution for screen content as this will likely result in |
| // blurry and unreadable text. |
| if (parameters_.options.is_screencast.value_or(false)) |
| return; |
| |
| rtc::Optional<int> max_pixel_count; |
| rtc::Optional<int> max_pixel_count_step_up; |
| if (load == kOveruse) { |
| rtc::CritScope cs(&lock_); |
| if (cpu_restricted_counter_ >= kMaxCpuDowngrades) { |
| return; |
| } |
| // The input video frame size will have a resolution with less than or |
| // equal to |max_pixel_count| depending on how the source can scale the |
| // input frame size. |
| max_pixel_count = rtc::Optional<int>( |
| (last_frame_info_.height * last_frame_info_.width * 3) / 5); |
| // Increase |number_of_cpu_adapt_changes_| if |
| // sink_wants_.max_pixel_count will be changed since |
| // last time |source_->AddOrUpdateSink| was called. That is, this will |
| // result in a new request for the source to change resolution. |
| if (!sink_wants_.max_pixel_count || |
| *sink_wants_.max_pixel_count > *max_pixel_count) { |
| ++number_of_cpu_adapt_changes_; |
| ++cpu_restricted_counter_; |
| } |
| } else { |
| RTC_DCHECK(load == kUnderuse); |
| rtc::CritScope cs(&lock_); |
| // The input video frame size will have a resolution with "one step up" |
| // pixels than |max_pixel_count_step_up| where "one step up" depends on |
| // how the source can scale the input frame size. |
| max_pixel_count_step_up = |
| rtc::Optional<int>(last_frame_info_.height * last_frame_info_.width); |
| // Increase |number_of_cpu_adapt_changes_| if |
| // sink_wants_.max_pixel_count_step_up will be changed since |
| // last time |source_->AddOrUpdateSink| was called. That is, this will |
| // result in a new request for the source to change resolution. |
| if (sink_wants_.max_pixel_count || |
| (sink_wants_.max_pixel_count_step_up && |
| *sink_wants_.max_pixel_count_step_up < *max_pixel_count_step_up)) { |
| ++number_of_cpu_adapt_changes_; |
| --cpu_restricted_counter_; |
| } |
| } |
| sink_wants_.max_pixel_count = max_pixel_count; |
| sink_wants_.max_pixel_count_step_up = max_pixel_count_step_up; |
| // |source_->AddOrUpdateSink| may not be called while holding |lock_| since |
| // that might cause a lock order inversion. |
| source_->AddOrUpdateSink(this, sink_wants_); |
| } |
| |
| VideoSenderInfo WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo( |
| bool log_stats) { |
| VideoSenderInfo info; |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| for (uint32_t ssrc : parameters_.config.rtp.ssrcs) |
| info.add_ssrc(ssrc); |
| |
| if (parameters_.codec_settings) |
| info.codec_name = parameters_.codec_settings->codec.name; |
| |
| if (stream_ == NULL) |
| return info; |
| |
| webrtc::VideoSendStream::Stats stats = stream_->GetStats(); |
| |
| if (log_stats) |
| LOG(LS_INFO) << stats.ToString(rtc::TimeMillis()); |
| |
| info.adapt_changes = number_of_cpu_adapt_changes_; |
| info.adapt_reason = |
| cpu_restricted_counter_ <= 0 ? ADAPTREASON_NONE : ADAPTREASON_CPU; |
| |
| // Get bandwidth limitation info from stream_->GetStats(). |
| // Input resolution (output from video_adapter) can be further scaled down or |
| // higher video layer(s) can be dropped due to bitrate constraints. |
| // Note, adapt_changes only include changes from the video_adapter. |
| if (stats.bw_limited_resolution) |
| info.adapt_reason |= ADAPTREASON_BANDWIDTH; |
| |
| info.encoder_implementation_name = stats.encoder_implementation_name; |
| info.ssrc_groups = ssrc_groups_; |
| info.framerate_input = stats.input_frame_rate; |
| info.framerate_sent = stats.encode_frame_rate; |
| info.avg_encode_ms = stats.avg_encode_time_ms; |
| info.encode_usage_percent = stats.encode_usage_percent; |
| |
| info.nominal_bitrate = stats.media_bitrate_bps; |
| info.preferred_bitrate = stats.preferred_media_bitrate_bps; |
| |
| info.send_frame_width = 0; |
| info.send_frame_height = 0; |
| for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it = |
| stats.substreams.begin(); |
| it != stats.substreams.end(); ++it) { |
| // TODO(pbos): Wire up additional stats, such as padding bytes. |
| webrtc::VideoSendStream::StreamStats stream_stats = it->second; |
| info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes + |
| stream_stats.rtp_stats.transmitted.header_bytes + |
| stream_stats.rtp_stats.transmitted.padding_bytes; |
| info.packets_sent += stream_stats.rtp_stats.transmitted.packets; |
| info.packets_lost += stream_stats.rtcp_stats.cumulative_lost; |
| if (stream_stats.width > info.send_frame_width) |
| info.send_frame_width = stream_stats.width; |
| if (stream_stats.height > info.send_frame_height) |
| info.send_frame_height = stream_stats.height; |
| info.firs_rcvd += stream_stats.rtcp_packet_type_counts.fir_packets; |
| info.nacks_rcvd += stream_stats.rtcp_packet_type_counts.nack_packets; |
| info.plis_rcvd += stream_stats.rtcp_packet_type_counts.pli_packets; |
| } |
| |
| if (!stats.substreams.empty()) { |
| // TODO(pbos): Report fraction lost per SSRC. |
| webrtc::VideoSendStream::StreamStats first_stream_stats = |
| stats.substreams.begin()->second; |
| info.fraction_lost = |
| static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) / |
| (1 << 8); |
| } |
| |
| return info; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::FillBandwidthEstimationInfo( |
| BandwidthEstimationInfo* bwe_info) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (stream_ == NULL) { |
| return; |
| } |
| webrtc::VideoSendStream::Stats stats = stream_->GetStats(); |
| for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it = |
| stats.substreams.begin(); |
| it != stats.substreams.end(); ++it) { |
| bwe_info->transmit_bitrate += it->second.total_bitrate_bps; |
| bwe_info->retransmit_bitrate += it->second.retransmit_bitrate_bps; |
| } |
| bwe_info->target_enc_bitrate += stats.target_media_bitrate_bps; |
| bwe_info->actual_enc_bitrate += stats.media_bitrate_bps; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (stream_ != NULL) { |
| call_->DestroyVideoSendStream(stream_); |
| } |
| |
| RTC_CHECK(parameters_.codec_settings); |
| RTC_DCHECK_EQ((parameters_.encoder_config.content_type == |
| webrtc::VideoEncoderConfig::ContentType::kScreen), |
| parameters_.options.is_screencast.value_or(false)) |
| << "encoder content type inconsistent with screencast option"; |
| parameters_.encoder_config.encoder_specific_settings = |
| ConfigureVideoEncoderSettings(parameters_.codec_settings->codec); |
| |
| webrtc::VideoSendStream::Config config = parameters_.config.Copy(); |
| if (!config.rtp.rtx.ssrcs.empty() && config.rtp.rtx.payload_type == -1) { |
| LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX " |
| "payload type the set codec. Ignoring RTX."; |
| config.rtp.rtx.ssrcs.clear(); |
| } |
| stream_ = call_->CreateVideoSendStream(std::move(config), |
| parameters_.encoder_config.Copy()); |
| stream_->SetSource(this); |
| |
| parameters_.encoder_config.encoder_specific_settings = NULL; |
| |
| // Call stream_->Start() if necessary conditions are met. |
| UpdateSendState(); |
| } |
| |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream( |
| webrtc::Call* call, |
| const StreamParams& sp, |
| webrtc::VideoReceiveStream::Config config, |
| WebRtcVideoDecoderFactory* external_decoder_factory, |
| bool default_stream, |
| const std::vector<VideoCodecSettings>& recv_codecs, |
| bool red_disabled_by_remote_side) |
| : call_(call), |
| stream_params_(sp), |
| stream_(NULL), |
| default_stream_(default_stream), |
| config_(std::move(config)), |
| red_disabled_by_remote_side_(red_disabled_by_remote_side), |
| external_decoder_factory_(external_decoder_factory), |
| sink_(NULL), |
| first_frame_timestamp_(-1), |
| estimated_remote_start_ntp_time_ms_(0) { |
| config_.renderer = this; |
| std::vector<AllocatedDecoder> old_decoders; |
| ConfigureCodecs(recv_codecs, &old_decoders); |
| RecreateWebRtcStream(); |
| RTC_DCHECK(old_decoders.empty()); |
| } |
| |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder:: |
| AllocatedDecoder(webrtc::VideoDecoder* decoder, |
| webrtc::VideoCodecType type, |
| bool external) |
| : decoder(decoder), |
| external_decoder(nullptr), |
| type(type), |
| external(external) { |
| if (external) { |
| external_decoder = decoder; |
| this->decoder = |
| new webrtc::VideoDecoderSoftwareFallbackWrapper(type, external_decoder); |
| } |
| } |
| |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() { |
| call_->DestroyVideoReceiveStream(stream_); |
| ClearDecoders(&allocated_decoders_); |
| } |
| |
| const std::vector<uint32_t>& |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetSsrcs() const { |
| return stream_params_.ssrcs; |
| } |
| |
| rtc::Optional<uint32_t> |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetFirstPrimarySsrc() const { |
| std::vector<uint32_t> primary_ssrcs; |
| stream_params_.GetPrimarySsrcs(&primary_ssrcs); |
| |
| if (primary_ssrcs.empty()) { |
| LOG(LS_WARNING) << "Empty primary ssrcs vector, returning empty optional"; |
| return rtc::Optional<uint32_t>(); |
| } else { |
| return rtc::Optional<uint32_t>(primary_ssrcs[0]); |
| } |
| } |
| |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder( |
| std::vector<AllocatedDecoder>* old_decoders, |
| const VideoCodec& codec) { |
| webrtc::VideoCodecType type = CodecTypeFromName(codec.name); |
| |
| for (size_t i = 0; i < old_decoders->size(); ++i) { |
| if ((*old_decoders)[i].type == type) { |
| AllocatedDecoder decoder = (*old_decoders)[i]; |
| (*old_decoders)[i] = old_decoders->back(); |
| old_decoders->pop_back(); |
| return decoder; |
| } |
| } |
| |
| if (external_decoder_factory_ != NULL) { |
| webrtc::VideoDecoder* decoder = |
| external_decoder_factory_->CreateVideoDecoderWithParams( |
| type, {stream_params_.id}); |
| if (decoder != NULL) { |
| return AllocatedDecoder(decoder, type, true); |
| } |
| } |
| |
| if (type == webrtc::kVideoCodecVP8) { |
| return AllocatedDecoder( |
| webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp8), type, false); |
| } |
| |
| if (type == webrtc::kVideoCodecVP9) { |
| return AllocatedDecoder( |
| webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp9), type, false); |
| } |
| |
| if (type == webrtc::kVideoCodecH264) { |
| return AllocatedDecoder( |
| webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kH264), type, false); |
| } |
| |
| return AllocatedDecoder( |
| webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kUnsupportedCodec), |
| webrtc::kVideoCodecUnknown, false); |
| } |
| |
| void ConfigureDecoderSpecifics(webrtc::VideoReceiveStream::Decoder* decoder, |
| const cricket::VideoCodec& recv_video_codec) { |
| if (recv_video_codec.name.compare("H264") == 0) { |
| auto it = recv_video_codec.params.find("sprop-parameter-sets"); |
| if (it != recv_video_codec.params.end()) { |
| decoder->decoder_specific.h264_extra_settings = |
| rtc::Optional<webrtc::VideoDecoderH264Settings>( |
| webrtc::VideoDecoderH264Settings()); |
| decoder->decoder_specific.h264_extra_settings->sprop_parameter_sets = |
| it->second; |
| } |
| } |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ConfigureCodecs( |
| const std::vector<VideoCodecSettings>& recv_codecs, |
| std::vector<AllocatedDecoder>* old_decoders) { |
| *old_decoders = allocated_decoders_; |
| allocated_decoders_.clear(); |
| config_.decoders.clear(); |
| for (size_t i = 0; i < recv_codecs.size(); ++i) { |
| AllocatedDecoder allocated_decoder = |
| CreateOrReuseVideoDecoder(old_decoders, recv_codecs[i].codec); |
| allocated_decoders_.push_back(allocated_decoder); |
| |
| webrtc::VideoReceiveStream::Decoder decoder; |
| decoder.decoder = allocated_decoder.decoder; |
| decoder.payload_type = recv_codecs[i].codec.id; |
| decoder.payload_name = recv_codecs[i].codec.name; |
| ConfigureDecoderSpecifics(&decoder, recv_codecs[i].codec); |
| config_.decoders.push_back(decoder); |
| } |
| |
| // TODO(pbos): Reconfigure RTX based on incoming recv_codecs. |
| config_.rtp.fec = recv_codecs.front().fec; |
| config_.rtp.nack.rtp_history_ms = |
| HasNack(recv_codecs.begin()->codec) ? kNackHistoryMs : 0; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetLocalSsrc( |
| uint32_t local_ssrc) { |
| // TODO(pbos): Consider turning this sanity check into a RTC_DCHECK. You |
| // should not be able to create a sender with the same SSRC as a receiver, but |
| // right now this can't be done due to unittests depending on receiving what |
| // they are sending from the same MediaChannel. |
| if (local_ssrc == config_.rtp.remote_ssrc) { |
| LOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are " |
| "unchanged; local_ssrc=" << local_ssrc; |
| return; |
| } |
| |
| config_.rtp.local_ssrc = local_ssrc; |
| LOG(LS_INFO) |
| << "RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=" |
| << local_ssrc; |
| RecreateWebRtcStream(); |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetFeedbackParameters( |
| bool nack_enabled, |
| bool remb_enabled, |
| bool transport_cc_enabled, |
| webrtc::RtcpMode rtcp_mode) { |
| int nack_history_ms = nack_enabled ? kNackHistoryMs : 0; |
| if (config_.rtp.nack.rtp_history_ms == nack_history_ms && |
| config_.rtp.remb == remb_enabled && |
| config_.rtp.transport_cc == transport_cc_enabled && |
| config_.rtp.rtcp_mode == rtcp_mode) { |
| LOG(LS_INFO) |
| << "Ignoring call to SetFeedbackParameters because parameters are " |
| "unchanged; nack=" |
| << nack_enabled << ", remb=" << remb_enabled |
| << ", transport_cc=" << transport_cc_enabled; |
| return; |
| } |
| config_.rtp.remb = remb_enabled; |
| config_.rtp.nack.rtp_history_ms = nack_history_ms; |
| config_.rtp.transport_cc = transport_cc_enabled; |
| config_.rtp.rtcp_mode = rtcp_mode; |
| LOG(LS_INFO) |
| << "RecreateWebRtcStream (recv) because of SetFeedbackParameters; nack=" |
| << nack_enabled << ", remb=" << remb_enabled |
| << ", transport_cc=" << transport_cc_enabled; |
| RecreateWebRtcStream(); |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvParameters( |
| const ChangedRecvParameters& params) { |
| bool needs_recreation = false; |
| std::vector<AllocatedDecoder> old_decoders; |
| if (params.codec_settings) { |
| ConfigureCodecs(*params.codec_settings, &old_decoders); |
| needs_recreation = true; |
| } |
| if (params.rtp_header_extensions) { |
| config_.rtp.extensions = *params.rtp_header_extensions; |
| needs_recreation = true; |
| } |
| if (needs_recreation) { |
| LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetRecvParameters"; |
| RecreateWebRtcStream(); |
| ClearDecoders(&old_decoders); |
| } |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() { |
| if (stream_ != NULL) { |
| call_->DestroyVideoReceiveStream(stream_); |
| } |
| webrtc::VideoReceiveStream::Config config = config_.Copy(); |
| if (red_disabled_by_remote_side_) { |
| config.rtp.fec.red_payload_type = -1; |
| config.rtp.fec.ulpfec_payload_type = -1; |
| config.rtp.fec.red_rtx_payload_type = -1; |
| } |
| stream_ = call_->CreateVideoReceiveStream(std::move(config)); |
| stream_->Start(); |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ClearDecoders( |
| std::vector<AllocatedDecoder>* allocated_decoders) { |
| for (size_t i = 0; i < allocated_decoders->size(); ++i) { |
| if ((*allocated_decoders)[i].external) { |
| external_decoder_factory_->DestroyVideoDecoder( |
| (*allocated_decoders)[i].external_decoder); |
| } |
| delete (*allocated_decoders)[i].decoder; |
| } |
| allocated_decoders->clear(); |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoReceiveStream::OnFrame( |
| const webrtc::VideoFrame& frame) { |
| rtc::CritScope crit(&sink_lock_); |
| |
| if (first_frame_timestamp_ < 0) |
| first_frame_timestamp_ = frame.timestamp(); |
| int64_t rtp_time_elapsed_since_first_frame = |
| (timestamp_wraparound_handler_.Unwrap(frame.timestamp()) - |
| first_frame_timestamp_); |
| int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame / |
| (cricket::kVideoCodecClockrate / 1000); |
| if (frame.ntp_time_ms() > 0) |
| estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms; |
| |
| if (sink_ == NULL) { |
| LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoSink."; |
| return; |
| } |
| |
| WebRtcVideoFrame render_frame( |
| frame.video_frame_buffer(), frame.rotation(), |
| frame.render_time_ms() * rtc::kNumNanosecsPerMicrosec, frame.timestamp()); |
| sink_->OnFrame(render_frame); |
| } |
| |
| bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsDefaultStream() const { |
| return default_stream_; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetSink( |
| rtc::VideoSinkInterface<cricket::VideoFrame>* sink) { |
| rtc::CritScope crit(&sink_lock_); |
| sink_ = sink; |
| } |
| |
| std::string |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetCodecNameFromPayloadType( |
| int payload_type) { |
| for (const webrtc::VideoReceiveStream::Decoder& decoder : config_.decoders) { |
| if (decoder.payload_type == payload_type) { |
| return decoder.payload_name; |
| } |
| } |
| return ""; |
| } |
| |
| VideoReceiverInfo |
| WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo( |
| bool log_stats) { |
| VideoReceiverInfo info; |
| info.ssrc_groups = stream_params_.ssrc_groups; |
| info.add_ssrc(config_.rtp.remote_ssrc); |
| webrtc::VideoReceiveStream::Stats stats = stream_->GetStats(); |
| info.decoder_implementation_name = stats.decoder_implementation_name; |
| info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes + |
| stats.rtp_stats.transmitted.header_bytes + |
| stats.rtp_stats.transmitted.padding_bytes; |
| info.packets_rcvd = stats.rtp_stats.transmitted.packets; |
| info.packets_lost = stats.rtcp_stats.cumulative_lost; |
| info.fraction_lost = |
| static_cast<float>(stats.rtcp_stats.fraction_lost) / (1 << 8); |
| |
| info.framerate_rcvd = stats.network_frame_rate; |
| info.framerate_decoded = stats.decode_frame_rate; |
| info.framerate_output = stats.render_frame_rate; |
| info.frame_width = stats.width; |
| info.frame_height = stats.height; |
| |
| { |
| rtc::CritScope frame_cs(&sink_lock_); |
| info.capture_start_ntp_time_ms = estimated_remote_start_ntp_time_ms_; |
| } |
| |
| info.decode_ms = stats.decode_ms; |
| info.max_decode_ms = stats.max_decode_ms; |
| info.current_delay_ms = stats.current_delay_ms; |
| info.target_delay_ms = stats.target_delay_ms; |
| info.jitter_buffer_ms = stats.jitter_buffer_ms; |
| info.min_playout_delay_ms = stats.min_playout_delay_ms; |
| info.render_delay_ms = stats.render_delay_ms; |
| |
| info.codec_name = GetCodecNameFromPayloadType(stats.current_payload_type); |
| |
| info.firs_sent = stats.rtcp_packet_type_counts.fir_packets; |
| info.plis_sent = stats.rtcp_packet_type_counts.pli_packets; |
| info.nacks_sent = stats.rtcp_packet_type_counts.nack_packets; |
| |
| if (log_stats) |
| LOG(LS_INFO) << stats.ToString(rtc::TimeMillis()); |
| |
| return info; |
| } |
| |
| void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetFecDisabledRemotely( |
| bool disable) { |
| red_disabled_by_remote_side_ = disable; |
| RecreateWebRtcStream(); |
| } |
| |
| WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings() |
| : rtx_payload_type(-1) {} |
| |
| bool WebRtcVideoChannel2::VideoCodecSettings::operator==( |
| const WebRtcVideoChannel2::VideoCodecSettings& other) const { |
| return codec == other.codec && |
| fec.ulpfec_payload_type == other.fec.ulpfec_payload_type && |
| fec.red_payload_type == other.fec.red_payload_type && |
| fec.red_rtx_payload_type == other.fec.red_rtx_payload_type && |
| rtx_payload_type == other.rtx_payload_type; |
| } |
| |
| bool WebRtcVideoChannel2::VideoCodecSettings::operator!=( |
| const WebRtcVideoChannel2::VideoCodecSettings& other) const { |
| return !(*this == other); |
| } |
| |
| std::vector<WebRtcVideoChannel2::VideoCodecSettings> |
| WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) { |
| RTC_DCHECK(!codecs.empty()); |
| |
| std::vector<VideoCodecSettings> video_codecs; |
| std::map<int, bool> payload_used; |
| std::map<int, VideoCodec::CodecType> payload_codec_type; |
| // |rtx_mapping| maps video payload type to rtx payload type. |
| std::map<int, int> rtx_mapping; |
| |
| webrtc::FecConfig fec_settings; |
| |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| const VideoCodec& in_codec = codecs[i]; |
| int payload_type = in_codec.id; |
| |
| if (payload_used[payload_type]) { |
| LOG(LS_ERROR) << "Payload type already registered: " |
| << in_codec.ToString(); |
| return std::vector<VideoCodecSettings>(); |
| } |
| payload_used[payload_type] = true; |
| payload_codec_type[payload_type] = in_codec.GetCodecType(); |
| |
| switch (in_codec.GetCodecType()) { |
| case VideoCodec::CODEC_RED: { |
| // RED payload type, should not have duplicates. |
| RTC_DCHECK(fec_settings.red_payload_type == -1); |
| fec_settings.red_payload_type = in_codec.id; |
| continue; |
| } |
| |
| case VideoCodec::CODEC_ULPFEC: { |
| // ULPFEC payload type, should not have duplicates. |
| RTC_DCHECK(fec_settings.ulpfec_payload_type == -1); |
| fec_settings.ulpfec_payload_type = in_codec.id; |
| continue; |
| } |
| |
| case VideoCodec::CODEC_RTX: { |
| int associated_payload_type; |
| if (!in_codec.GetParam(kCodecParamAssociatedPayloadType, |
| &associated_payload_type) || |
| !IsValidRtpPayloadType(associated_payload_type)) { |
| LOG(LS_ERROR) |
| << "RTX codec with invalid or no associated payload type: " |
| << in_codec.ToString(); |
| return std::vector<VideoCodecSettings>(); |
| } |
| rtx_mapping[associated_payload_type] = in_codec.id; |
| continue; |
| } |
| |
| case VideoCodec::CODEC_VIDEO: |
| break; |
| } |
| |
| video_codecs.push_back(VideoCodecSettings()); |
| video_codecs.back().codec = in_codec; |
| } |
| |
| // One of these codecs should have been a video codec. Only having FEC |
| // parameters into this code is a logic error. |
| RTC_DCHECK(!video_codecs.empty()); |
| |
| for (std::map<int, int>::const_iterator it = rtx_mapping.begin(); |
| it != rtx_mapping.end(); |
| ++it) { |
| if (!payload_used[it->first]) { |
| LOG(LS_ERROR) << "RTX mapped to payload not in codec list."; |
| return std::vector<VideoCodecSettings>(); |
| } |
| if (payload_codec_type[it->first] != VideoCodec::CODEC_VIDEO && |
| payload_codec_type[it->first] != VideoCodec::CODEC_RED) { |
| LOG(LS_ERROR) << "RTX not mapped to regular video codec or RED codec."; |
| return std::vector<VideoCodecSettings>(); |
| } |
| |
| if (it->first == fec_settings.red_payload_type) { |
| fec_settings.red_rtx_payload_type = it->second; |
| } |
| } |
| |
| for (size_t i = 0; i < video_codecs.size(); ++i) { |
| video_codecs[i].fec = fec_settings; |
| if (rtx_mapping[video_codecs[i].codec.id] != 0 && |
| rtx_mapping[video_codecs[i].codec.id] != |
| fec_settings.red_payload_type) { |
| video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; |
| } |
| } |
| |
| return video_codecs; |
| } |
| |
| } // namespace cricket |