| /* |
| * 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 "media/engine/webrtc_video_engine.h" |
| |
| #include <stdio.h> |
| |
| #include <algorithm> |
| #include <set> |
| #include <string> |
| #include <utility> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/strings/match.h" |
| #include "api/media_stream_interface.h" |
| #include "api/units/data_rate.h" |
| #include "api/video/video_codec_constants.h" |
| #include "api/video/video_codec_type.h" |
| #include "api/video_codecs/sdp_video_format.h" |
| #include "api/video_codecs/video_decoder_factory.h" |
| #include "api/video_codecs/video_encoder.h" |
| #include "api/video_codecs/video_encoder_factory.h" |
| #include "call/call.h" |
| #include "media/engine/simulcast.h" |
| #include "media/engine/webrtc_media_engine.h" |
| #include "media/engine/webrtc_voice_engine.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/experiments/field_trial_parser.h" |
| #include "rtc_base/experiments/field_trial_units.h" |
| #include "rtc_base/experiments/min_video_bitrate_experiment.h" |
| #include "rtc_base/experiments/normalize_simulcast_size_experiment.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/time_utils.h" |
| #include "rtc_base/trace_event.h" |
| |
| namespace cricket { |
| |
| namespace { |
| |
| const int kMinLayerSize = 16; |
| |
| const char* StreamTypeToString( |
| webrtc::VideoSendStream::StreamStats::StreamType type) { |
| switch (type) { |
| case webrtc::VideoSendStream::StreamStats::StreamType::kMedia: |
| return "kMedia"; |
| case webrtc::VideoSendStream::StreamStats::StreamType::kRtx: |
| return "kRtx"; |
| case webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec: |
| return "kFlexfec"; |
| } |
| return nullptr; |
| } |
| |
| bool IsEnabled(const webrtc::WebRtcKeyValueConfig& trials, |
| absl::string_view name) { |
| return absl::StartsWith(trials.Lookup(name), "Enabled"); |
| } |
| |
| bool IsDisabled(const webrtc::WebRtcKeyValueConfig& trials, |
| absl::string_view name) { |
| return absl::StartsWith(trials.Lookup(name), "Disabled"); |
| } |
| |
| bool PowerOfTwo(int value) { |
| return (value > 0) && ((value & (value - 1)) == 0); |
| } |
| |
| bool IsScaleFactorsPowerOfTwo(const webrtc::VideoEncoderConfig& config) { |
| for (const auto& layer : config.simulcast_layers) { |
| double scale = std::max(layer.scale_resolution_down_by, 1.0); |
| if (std::round(scale) != scale || !PowerOfTwo(scale)) { |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| void AddDefaultFeedbackParams(VideoCodec* codec, |
| const webrtc::WebRtcKeyValueConfig& trials) { |
| // Don't add any feedback params for RED and ULPFEC. |
| if (codec->name == kRedCodecName || codec->name == kUlpfecCodecName) |
| return; |
| codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty)); |
| codec->AddFeedbackParam( |
| FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty)); |
| // Don't add any more feedback params for FLEXFEC. |
| if (codec->name == kFlexfecCodecName) |
| return; |
| codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir)); |
| codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kParamValueEmpty)); |
| codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli)); |
| if (codec->name == kVp8CodecName && |
| IsEnabled(trials, "WebRTC-RtcpLossNotification")) { |
| codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamLntf, kParamValueEmpty)); |
| } |
| } |
| |
| // This function will assign dynamic payload types (in the range [96, 127]) to |
| // the input codecs, and also add ULPFEC, RED, FlexFEC, and associated RTX |
| // codecs for recognized codecs (VP8, VP9, H264, and RED). It will also add |
| // default feedback params to the codecs. |
| // is_decoder_factory is needed to keep track of the implict assumption that any |
| // H264 decoder also supports constrained base line profile. |
| // Also, is_decoder_factory is used to decide whether FlexFEC video format |
| // should be advertised as supported. |
| // TODO(kron): Perhaps it is better to move the implicit knowledge to the place |
| // where codecs are negotiated. |
| template <class T> |
| std::vector<VideoCodec> GetPayloadTypesAndDefaultCodecs( |
| const T* factory, |
| bool is_decoder_factory, |
| const webrtc::WebRtcKeyValueConfig& trials) { |
| if (!factory) { |
| return {}; |
| } |
| |
| std::vector<webrtc::SdpVideoFormat> supported_formats = |
| factory->GetSupportedFormats(); |
| if (is_decoder_factory) { |
| AddH264ConstrainedBaselineProfileToSupportedFormats(&supported_formats); |
| } |
| |
| if (supported_formats.empty()) |
| return std::vector<VideoCodec>(); |
| |
| // Due to interoperability issues with old Chrome/WebRTC versions only use |
| // the lower range for new codecs. |
| static const int kFirstDynamicPayloadTypeLowerRange = 35; |
| static const int kLastDynamicPayloadTypeLowerRange = 65; |
| |
| static const int kFirstDynamicPayloadTypeUpperRange = 96; |
| static const int kLastDynamicPayloadTypeUpperRange = 127; |
| int payload_type_upper = kFirstDynamicPayloadTypeUpperRange; |
| int payload_type_lower = kFirstDynamicPayloadTypeLowerRange; |
| |
| supported_formats.push_back(webrtc::SdpVideoFormat(kRedCodecName)); |
| supported_formats.push_back(webrtc::SdpVideoFormat(kUlpfecCodecName)); |
| |
| // flexfec-03 is supported as |
| // - receive codec unless WebRTC-FlexFEC-03-Advertised is disabled |
| // - send codec if WebRTC-FlexFEC-03-Advertised is enabled |
| if ((is_decoder_factory && |
| !IsDisabled(trials, "WebRTC-FlexFEC-03-Advertised")) || |
| (!is_decoder_factory && |
| IsEnabled(trials, "WebRTC-FlexFEC-03-Advertised"))) { |
| webrtc::SdpVideoFormat flexfec_format(kFlexfecCodecName); |
| // This value is currently arbitrarily set to 10 seconds. (The unit |
| // is microseconds.) This parameter MUST be present in the SDP, but |
| // we never use the actual value anywhere in our code however. |
| // TODO(brandtr): Consider honouring this value in the sender and receiver. |
| flexfec_format.parameters = {{kFlexfecFmtpRepairWindow, "10000000"}}; |
| supported_formats.push_back(flexfec_format); |
| } |
| |
| std::vector<VideoCodec> output_codecs; |
| for (const webrtc::SdpVideoFormat& format : supported_formats) { |
| VideoCodec codec(format); |
| bool isCodecValidForLowerRange = |
| absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName) || |
| absl::EqualsIgnoreCase(codec.name, kAv1CodecName); |
| if (!isCodecValidForLowerRange) { |
| codec.id = payload_type_upper++; |
| } else { |
| codec.id = payload_type_lower++; |
| } |
| AddDefaultFeedbackParams(&codec, trials); |
| output_codecs.push_back(codec); |
| |
| if (payload_type_upper > kLastDynamicPayloadTypeUpperRange) { |
| RTC_LOG(LS_ERROR) |
| << "Out of dynamic payload types [96,127], skipping the rest."; |
| // TODO(https://bugs.chromium.org/p/webrtc/issues/detail?id=12194): |
| // continue in lower range. |
| break; |
| } |
| if (payload_type_lower > kLastDynamicPayloadTypeLowerRange) { |
| // TODO(https://bugs.chromium.org/p/webrtc/issues/detail?id=12248): |
| // return an error. |
| RTC_LOG(LS_ERROR) |
| << "Out of dynamic payload types [35,65], skipping the rest."; |
| break; |
| } |
| |
| // Add associated RTX codec for non-FEC codecs. |
| if (!absl::EqualsIgnoreCase(codec.name, kUlpfecCodecName) && |
| !absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName)) { |
| if (!isCodecValidForLowerRange) { |
| output_codecs.push_back( |
| VideoCodec::CreateRtxCodec(payload_type_upper++, codec.id)); |
| } else { |
| output_codecs.push_back( |
| VideoCodec::CreateRtxCodec(payload_type_lower++, codec.id)); |
| } |
| |
| if (payload_type_upper > kLastDynamicPayloadTypeUpperRange) { |
| RTC_LOG(LS_ERROR) |
| << "Out of dynamic payload types [96,127], skipping rtx."; |
| // TODO(https://bugs.chromium.org/p/webrtc/issues/detail?id=12194): |
| // continue in lower range. |
| break; |
| } |
| if (payload_type_lower > kLastDynamicPayloadTypeLowerRange) { |
| // TODO(https://bugs.chromium.org/p/webrtc/issues/detail?id=12248): |
| // return an error. |
| RTC_LOG(LS_ERROR) |
| << "Out of dynamic payload types [35,65], skipping rtx."; |
| break; |
| } |
| } |
| } |
| return output_codecs; |
| } |
| |
| bool IsTemporalLayersSupported(const std::string& codec_name) { |
| return absl::EqualsIgnoreCase(codec_name, kVp8CodecName) || |
| absl::EqualsIgnoreCase(codec_name, kVp9CodecName); |
| } |
| |
| static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) { |
| rtc::StringBuilder out; |
| out << "{"; |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| out << codecs[i].ToString(); |
| if (i != codecs.size() - 1) { |
| out << ", "; |
| } |
| } |
| out << "}"; |
| return out.Release(); |
| } |
| |
| 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) { |
| RTC_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()) { |
| RTC_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) { |
| RTC_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()) { |
| RTC_LOG(LS_ERROR) |
| << "RTX SSRCs exist, but don't cover all SSRCs (unsupported): " |
| << sp.ToString(); |
| return false; |
| } |
| |
| return true; |
| } |
| |
| // Returns true if the given codec is disallowed from doing simulcast. |
| bool IsCodecDisabledForSimulcast(const std::string& codec_name, |
| const webrtc::WebRtcKeyValueConfig& trials) { |
| return !absl::StartsWith(trials.Lookup("WebRTC-H264Simulcast"), "Disabled") |
| ? absl::EqualsIgnoreCase(codec_name, kVp9CodecName) |
| : absl::EqualsIgnoreCase(codec_name, kH264CodecName) || |
| absl::EqualsIgnoreCase(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, |
| bool is_screenshare) { |
| int max_bitrate; |
| if (width * height <= 320 * 240) { |
| max_bitrate = 600; |
| } else if (width * height <= 640 * 480) { |
| max_bitrate = 1700; |
| } else if (width * height <= 960 * 540) { |
| max_bitrate = 2000; |
| } else { |
| max_bitrate = 2500; |
| } |
| if (is_screenshare) |
| max_bitrate = std::max(max_bitrate, 1200); |
| return max_bitrate; |
| } |
| |
| bool GetVp9LayersFromFieldTrialGroup( |
| size_t* num_spatial_layers, |
| size_t* num_temporal_layers, |
| const webrtc::WebRtcKeyValueConfig& trials) { |
| std::string group = trials.Lookup("WebRTC-SupportVP9SVC"); |
| if (group.empty()) |
| return false; |
| |
| if (sscanf(group.c_str(), "EnabledByFlag_%zuSL%zuTL", num_spatial_layers, |
| num_temporal_layers) != 2) { |
| return false; |
| } |
| if (*num_spatial_layers > webrtc::kMaxSpatialLayers || |
| *num_spatial_layers < 1) |
| return false; |
| |
| const size_t kMaxTemporalLayers = 3; |
| if (*num_temporal_layers > kMaxTemporalLayers || *num_temporal_layers < 1) |
| return false; |
| |
| return true; |
| } |
| |
| absl::optional<size_t> GetVp9SpatialLayersFromFieldTrial( |
| const webrtc::WebRtcKeyValueConfig& trials) { |
| size_t num_sl; |
| size_t num_tl; |
| if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl, trials)) { |
| return num_sl; |
| } |
| return absl::nullopt; |
| } |
| |
| absl::optional<size_t> GetVp9TemporalLayersFromFieldTrial( |
| const webrtc::WebRtcKeyValueConfig& trials) { |
| size_t num_sl; |
| size_t num_tl; |
| if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl, trials)) { |
| return num_tl; |
| } |
| return absl::nullopt; |
| } |
| |
| // Returns its smallest positive argument. If neither argument is positive, |
| // returns an arbitrary nonpositive value. |
| int MinPositive(int a, int b) { |
| if (a <= 0) { |
| return b; |
| } |
| if (b <= 0) { |
| return a; |
| } |
| return std::min(a, b); |
| } |
| |
| bool IsLayerActive(const webrtc::RtpEncodingParameters& layer) { |
| return layer.active && |
| (!layer.max_bitrate_bps || *layer.max_bitrate_bps > 0) && |
| (!layer.max_framerate || *layer.max_framerate > 0); |
| } |
| |
| size_t FindRequiredActiveLayers( |
| const webrtc::VideoEncoderConfig& encoder_config) { |
| // Need enough layers so that at least the first active one is present. |
| for (size_t i = 0; i < encoder_config.number_of_streams; ++i) { |
| if (encoder_config.simulcast_layers[i].active) { |
| return i + 1; |
| } |
| } |
| return 0; |
| } |
| |
| int NumActiveStreams(const webrtc::RtpParameters& rtp_parameters) { |
| int res = 0; |
| for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) { |
| if (rtp_parameters.encodings[i].active) { |
| ++res; |
| } |
| } |
| return res; |
| } |
| |
| std::map<uint32_t, webrtc::VideoSendStream::StreamStats> |
| MergeInfoAboutOutboundRtpSubstreams( |
| const std::map<uint32_t, webrtc::VideoSendStream::StreamStats>& |
| substreams) { |
| std::map<uint32_t, webrtc::VideoSendStream::StreamStats> rtp_substreams; |
| // Add substreams for all RTP media streams. |
| for (const auto& pair : substreams) { |
| uint32_t ssrc = pair.first; |
| const webrtc::VideoSendStream::StreamStats& substream = pair.second; |
| switch (substream.type) { |
| case webrtc::VideoSendStream::StreamStats::StreamType::kMedia: |
| break; |
| case webrtc::VideoSendStream::StreamStats::StreamType::kRtx: |
| case webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec: |
| continue; |
| } |
| rtp_substreams.insert(std::make_pair(ssrc, substream)); |
| } |
| // Complement the kMedia substream stats with the associated kRtx and kFlexfec |
| // substream stats. |
| for (const auto& pair : substreams) { |
| switch (pair.second.type) { |
| case webrtc::VideoSendStream::StreamStats::StreamType::kMedia: |
| continue; |
| case webrtc::VideoSendStream::StreamStats::StreamType::kRtx: |
| case webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec: |
| break; |
| } |
| // The associated substream is an RTX or FlexFEC substream that is |
| // referencing an RTP media substream. |
| const webrtc::VideoSendStream::StreamStats& associated_substream = |
| pair.second; |
| RTC_DCHECK(associated_substream.referenced_media_ssrc.has_value()); |
| uint32_t media_ssrc = associated_substream.referenced_media_ssrc.value(); |
| if (substreams.find(media_ssrc) == substreams.end()) { |
| RTC_LOG(LS_WARNING) << "Substream [ssrc: " << pair.first << ", type: " |
| << StreamTypeToString(associated_substream.type) |
| << "] is associated with a media ssrc (" << media_ssrc |
| << ") that does not have StreamStats. Ignoring its " |
| << "RTP stats."; |
| continue; |
| } |
| webrtc::VideoSendStream::StreamStats& rtp_substream = |
| rtp_substreams[media_ssrc]; |
| |
| // We only merge |rtp_stats|. All other metrics are not applicable for RTX |
| // and FlexFEC. |
| // TODO(hbos): kRtx and kFlexfec stats should use a separate struct to make |
| // it clear what is or is not applicable. |
| rtp_substream.rtp_stats.Add(associated_substream.rtp_stats); |
| } |
| return rtp_substreams; |
| } |
| |
| } // namespace |
| |
| // 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 kDefaultRtcpReceiverReportSsrc = 1; |
| |
| // Minimum time interval for logging stats. |
| static const int64_t kStatsLogIntervalMs = 10000; |
| |
| std::map<uint32_t, webrtc::VideoSendStream::StreamStats> |
| MergeInfoAboutOutboundRtpSubstreamsForTesting( |
| const std::map<uint32_t, webrtc::VideoSendStream::StreamStats>& |
| substreams) { |
| return MergeInfoAboutOutboundRtpSubstreams(substreams); |
| } |
| |
| rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings> |
| WebRtcVideoChannel::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, or when |
| // disabled by field trial flag. |
| bool automatic_resize = !disable_automatic_resize_ && !is_screencast && |
| (parameters_.config.rtp.ssrcs.size() == 1 || |
| NumActiveStreams(rtp_parameters_) == 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 (absl::EqualsIgnoreCase(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 (absl::EqualsIgnoreCase(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 (absl::EqualsIgnoreCase(codec.name, kVp9CodecName)) { |
| webrtc::VideoCodecVP9 vp9_settings = |
| webrtc::VideoEncoder::GetDefaultVp9Settings(); |
| const size_t default_num_spatial_layers = |
| parameters_.config.rtp.ssrcs.size(); |
| const size_t num_spatial_layers = |
| GetVp9SpatialLayersFromFieldTrial(call_->trials()) |
| .value_or(default_num_spatial_layers); |
| |
| const size_t default_num_temporal_layers = |
| num_spatial_layers > 1 ? kConferenceDefaultNumTemporalLayers : 1; |
| const size_t num_temporal_layers = |
| GetVp9TemporalLayersFromFieldTrial(call_->trials()) |
| .value_or(default_num_temporal_layers); |
| |
| vp9_settings.numberOfSpatialLayers = std::min<unsigned char>( |
| num_spatial_layers, kConferenceMaxNumSpatialLayers); |
| vp9_settings.numberOfTemporalLayers = std::min<unsigned char>( |
| num_temporal_layers, kConferenceMaxNumTemporalLayers); |
| |
| // VP9 denoising is disabled by default. |
| vp9_settings.denoisingOn = codec_default_denoising ? true : denoising; |
| vp9_settings.automaticResizeOn = automatic_resize; |
| // Ensure frame dropping is always enabled. |
| RTC_DCHECK(vp9_settings.frameDroppingOn); |
| if (!is_screencast) { |
| webrtc::FieldTrialFlag interlayer_pred_experiment_enabled = |
| webrtc::FieldTrialFlag("Enabled"); |
| webrtc::FieldTrialEnum<webrtc::InterLayerPredMode> inter_layer_pred_mode( |
| "inter_layer_pred_mode", webrtc::InterLayerPredMode::kOnKeyPic, |
| {{"off", webrtc::InterLayerPredMode::kOff}, |
| {"on", webrtc::InterLayerPredMode::kOn}, |
| {"onkeypic", webrtc::InterLayerPredMode::kOnKeyPic}}); |
| webrtc::ParseFieldTrial( |
| {&interlayer_pred_experiment_enabled, &inter_layer_pred_mode}, |
| call_->trials().Lookup("WebRTC-Vp9InterLayerPred")); |
| if (interlayer_pred_experiment_enabled) { |
| vp9_settings.interLayerPred = inter_layer_pred_mode; |
| } else { |
| // Limit inter-layer prediction to key pictures by default. |
| vp9_settings.interLayerPred = webrtc::InterLayerPredMode::kOnKeyPic; |
| } |
| } else { |
| // Multiple spatial layers vp9 screenshare needs flexible mode. |
| vp9_settings.flexibleMode = vp9_settings.numberOfSpatialLayers > 1; |
| vp9_settings.interLayerPred = webrtc::InterLayerPredMode::kOn; |
| } |
| return new rtc::RefCountedObject< |
| webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings); |
| } |
| return nullptr; |
| } |
| |
| DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler() |
| : default_sink_(nullptr) {} |
| |
| UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc( |
| WebRtcVideoChannel* channel, |
| uint32_t ssrc) { |
| absl::optional<uint32_t> default_recv_ssrc = |
| channel->GetDefaultReceiveStreamSsrc(); |
| |
| if (default_recv_ssrc) { |
| RTC_LOG(LS_INFO) << "Destroying old default receive stream for SSRC=" |
| << ssrc << "."; |
| channel->RemoveRecvStream(*default_recv_ssrc); |
| } |
| |
| StreamParams sp = channel->unsignaled_stream_params(); |
| sp.ssrcs.push_back(ssrc); |
| |
| RTC_LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc |
| << "."; |
| if (!channel->AddRecvStream(sp, /*default_stream=*/true)) { |
| RTC_LOG(LS_WARNING) << "Could not create default receive stream."; |
| } |
| |
| // SSRC 0 returns default_recv_base_minimum_delay_ms. |
| const int unsignaled_ssrc = 0; |
| int default_recv_base_minimum_delay_ms = |
| channel->GetBaseMinimumPlayoutDelayMs(unsignaled_ssrc).value_or(0); |
| // Set base minimum delay if it was set before for the default receive stream. |
| channel->SetBaseMinimumPlayoutDelayMs(ssrc, |
| default_recv_base_minimum_delay_ms); |
| channel->SetSink(ssrc, default_sink_); |
| return kDeliverPacket; |
| } |
| |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* |
| DefaultUnsignalledSsrcHandler::GetDefaultSink() const { |
| return default_sink_; |
| } |
| |
| void DefaultUnsignalledSsrcHandler::SetDefaultSink( |
| WebRtcVideoChannel* channel, |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) { |
| default_sink_ = sink; |
| absl::optional<uint32_t> default_recv_ssrc = |
| channel->GetDefaultReceiveStreamSsrc(); |
| if (default_recv_ssrc) { |
| channel->SetSink(*default_recv_ssrc, default_sink_); |
| } |
| } |
| |
| WebRtcVideoEngine::WebRtcVideoEngine( |
| std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory, |
| std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory, |
| const webrtc::WebRtcKeyValueConfig& trials) |
| : decoder_factory_(std::move(video_decoder_factory)), |
| encoder_factory_(std::move(video_encoder_factory)), |
| trials_(trials) { |
| RTC_DLOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine()"; |
| } |
| |
| WebRtcVideoEngine::~WebRtcVideoEngine() { |
| RTC_DLOG(LS_INFO) << "WebRtcVideoEngine::~WebRtcVideoEngine"; |
| } |
| |
| VideoMediaChannel* WebRtcVideoEngine::CreateMediaChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options, |
| const webrtc::CryptoOptions& crypto_options, |
| webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) { |
| RTC_LOG(LS_INFO) << "CreateMediaChannel. Options: " << options.ToString(); |
| return new WebRtcVideoChannel(call, config, options, crypto_options, |
| encoder_factory_.get(), decoder_factory_.get(), |
| video_bitrate_allocator_factory); |
| } |
| std::vector<VideoCodec> WebRtcVideoEngine::send_codecs() const { |
| return GetPayloadTypesAndDefaultCodecs(encoder_factory_.get(), |
| /*is_decoder_factory=*/false, trials_); |
| } |
| |
| std::vector<VideoCodec> WebRtcVideoEngine::recv_codecs() const { |
| return GetPayloadTypesAndDefaultCodecs(decoder_factory_.get(), |
| /*is_decoder_factory=*/true, trials_); |
| } |
| |
| std::vector<webrtc::RtpHeaderExtensionCapability> |
| WebRtcVideoEngine::GetRtpHeaderExtensions() const { |
| std::vector<webrtc::RtpHeaderExtensionCapability> result; |
| int id = 1; |
| for (const auto& uri : |
| {webrtc::RtpExtension::kTimestampOffsetUri, |
| webrtc::RtpExtension::kAbsSendTimeUri, |
| webrtc::RtpExtension::kVideoRotationUri, |
| webrtc::RtpExtension::kTransportSequenceNumberUri, |
| webrtc::RtpExtension::kPlayoutDelayUri, |
| webrtc::RtpExtension::kVideoContentTypeUri, |
| webrtc::RtpExtension::kVideoTimingUri, |
| webrtc::RtpExtension::kColorSpaceUri, webrtc::RtpExtension::kMidUri, |
| webrtc::RtpExtension::kRidUri, webrtc::RtpExtension::kRepairedRidUri}) { |
| result.emplace_back(uri, id++, webrtc::RtpTransceiverDirection::kSendRecv); |
| } |
| result.emplace_back(webrtc::RtpExtension::kGenericFrameDescriptorUri00, id++, |
| IsEnabled(trials_, "WebRTC-GenericDescriptorAdvertised") |
| ? webrtc::RtpTransceiverDirection::kSendRecv |
| : webrtc::RtpTransceiverDirection::kStopped); |
| result.emplace_back( |
| webrtc::RtpExtension::kDependencyDescriptorUri, id++, |
| IsEnabled(trials_, "WebRTC-DependencyDescriptorAdvertised") |
| ? webrtc::RtpTransceiverDirection::kSendRecv |
| : webrtc::RtpTransceiverDirection::kStopped); |
| |
| result.emplace_back( |
| webrtc::RtpExtension::kVideoLayersAllocationUri, id++, |
| IsEnabled(trials_, "WebRTC-VideoLayersAllocationAdvertised") |
| ? webrtc::RtpTransceiverDirection::kSendRecv |
| : webrtc::RtpTransceiverDirection::kStopped); |
| |
| result.emplace_back( |
| webrtc::RtpExtension::kVideoFrameTrackingIdUri, id++, |
| IsEnabled(trials_, "WebRTC-VideoFrameTrackingIdAdvertised") |
| ? webrtc::RtpTransceiverDirection::kSendRecv |
| : webrtc::RtpTransceiverDirection::kStopped); |
| |
| return result; |
| } |
| |
| WebRtcVideoChannel::WebRtcVideoChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options, |
| const webrtc::CryptoOptions& crypto_options, |
| webrtc::VideoEncoderFactory* encoder_factory, |
| webrtc::VideoDecoderFactory* decoder_factory, |
| webrtc::VideoBitrateAllocatorFactory* bitrate_allocator_factory) |
| : VideoMediaChannel(config), |
| worker_thread_(rtc::Thread::Current()), |
| call_(call), |
| unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_), |
| video_config_(config.video), |
| encoder_factory_(encoder_factory), |
| decoder_factory_(decoder_factory), |
| bitrate_allocator_factory_(bitrate_allocator_factory), |
| default_send_options_(options), |
| last_stats_log_ms_(-1), |
| discard_unknown_ssrc_packets_( |
| IsEnabled(call_->trials(), |
| "WebRTC-Video-DiscardPacketsWithUnknownSsrc")), |
| crypto_options_(crypto_options), |
| unknown_ssrc_packet_buffer_( |
| IsEnabled(call_->trials(), |
| "WebRTC-Video-BufferPacketsWithUnknownSsrc") |
| ? new UnhandledPacketsBuffer() |
| : nullptr) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| network_thread_checker_.Detach(); |
| |
| rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc; |
| sending_ = false; |
| recv_codecs_ = MapCodecs(GetPayloadTypesAndDefaultCodecs( |
| decoder_factory_, /*is_decoder_factory=*/true, call_->trials())); |
| recv_flexfec_payload_type_ = |
| recv_codecs_.empty() ? 0 : recv_codecs_.front().flexfec_payload_type; |
| } |
| |
| WebRtcVideoChannel::~WebRtcVideoChannel() { |
| for (auto& kv : send_streams_) |
| delete kv.second; |
| for (auto& kv : receive_streams_) |
| delete kv.second; |
| } |
| |
| std::vector<WebRtcVideoChannel::VideoCodecSettings> |
| WebRtcVideoChannel::SelectSendVideoCodecs( |
| const std::vector<VideoCodecSettings>& remote_mapped_codecs) const { |
| std::vector<webrtc::SdpVideoFormat> sdp_formats = |
| encoder_factory_ ? encoder_factory_->GetImplementations() |
| : std::vector<webrtc::SdpVideoFormat>(); |
| |
| // The returned vector holds the VideoCodecSettings in term of preference. |
| // They are orderd by receive codec preference first and local implementation |
| // preference second. |
| std::vector<VideoCodecSettings> encoders; |
| for (const VideoCodecSettings& remote_codec : remote_mapped_codecs) { |
| for (auto format_it = sdp_formats.begin(); |
| format_it != sdp_formats.end();) { |
| // For H264, we will limit the encode level to the remote offered level |
| // regardless if level asymmetry is allowed or not. This is strictly not |
| // following the spec in https://tools.ietf.org/html/rfc6184#section-8.2.2 |
| // since we should limit the encode level to the lower of local and remote |
| // level when level asymmetry is not allowed. |
| if (IsSameCodec(format_it->name, format_it->parameters, |
| remote_codec.codec.name, remote_codec.codec.params)) { |
| encoders.push_back(remote_codec); |
| |
| // To allow the VideoEncoderFactory to keep information about which |
| // implementation to instantitate when CreateEncoder is called the two |
| // parmeter sets are merged. |
| encoders.back().codec.params.insert(format_it->parameters.begin(), |
| format_it->parameters.end()); |
| |
| format_it = sdp_formats.erase(format_it); |
| } else { |
| ++format_it; |
| } |
| } |
| } |
| |
| return encoders; |
| } |
| |
| bool WebRtcVideoChannel::NonFlexfecReceiveCodecsHaveChanged( |
| std::vector<VideoCodecSettings> before, |
| std::vector<VideoCodecSettings> after) { |
| // 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; |
| }; |
| absl::c_sort(before, comparison); |
| absl::c_sort(after, comparison); |
| |
| // Changes in FlexFEC payload type are handled separately in |
| // WebRtcVideoChannel::GetChangedRecvParameters, so disregard FlexFEC in the |
| // comparison here. |
| return !absl::c_equal(before, after, |
| VideoCodecSettings::EqualsDisregardingFlexfec); |
| } |
| |
| bool WebRtcVideoChannel::GetChangedSendParameters( |
| const VideoSendParameters& params, |
| ChangedSendParameters* changed_params) const { |
| if (!ValidateCodecFormats(params.codecs) || |
| !ValidateRtpExtensions(params.extensions)) { |
| return false; |
| } |
| |
| std::vector<VideoCodecSettings> negotiated_codecs = |
| SelectSendVideoCodecs(MapCodecs(params.codecs)); |
| |
| // We should only fail here if send direction is enabled. |
| if (params.is_stream_active && negotiated_codecs.empty()) { |
| RTC_LOG(LS_ERROR) << "No video codecs supported."; |
| return false; |
| } |
| |
| // Never enable sending FlexFEC, unless we are in the experiment. |
| if (!IsEnabled(call_->trials(), "WebRTC-FlexFEC-03")) { |
| for (VideoCodecSettings& codec : negotiated_codecs) |
| codec.flexfec_payload_type = -1; |
| } |
| |
| if (negotiated_codecs_ != negotiated_codecs) { |
| if (negotiated_codecs.empty()) { |
| changed_params->send_codec = absl::nullopt; |
| } else if (send_codec_ != negotiated_codecs.front()) { |
| changed_params->send_codec = negotiated_codecs.front(); |
| } |
| changed_params->negotiated_codecs = std::move(negotiated_codecs); |
| } |
| |
| // Handle RTP header extensions. |
| if (params.extmap_allow_mixed != ExtmapAllowMixed()) { |
| changed_params->extmap_allow_mixed = params.extmap_allow_mixed; |
| } |
| std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( |
| params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true, |
| call_->trials()); |
| if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) { |
| changed_params->rtp_header_extensions = |
| absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions); |
| } |
| |
| if (params.mid != send_params_.mid) { |
| changed_params->mid = params.mid; |
| } |
| |
| // Handle max bitrate. |
| if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps && |
| params.max_bandwidth_bps >= -1) { |
| // 0 or -1 uncaps max bitrate. |
| // TODO(pbos): Reconsider how 0 should be treated. It is not mentioned as a |
| // special value and might very well be used for stopping sending. |
| changed_params->max_bandwidth_bps = |
| 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 = params.conference_mode; |
| } |
| |
| // Handle RTCP mode. |
| if (params.rtcp.reduced_size != send_params_.rtcp.reduced_size) { |
| changed_params->rtcp_mode = params.rtcp.reduced_size |
| ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound; |
| } |
| |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::SetSendParameters(const VideoSendParameters& params) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSendParameters"); |
| RTC_LOG(LS_INFO) << "SetSendParameters: " << params.ToString(); |
| ChangedSendParameters changed_params; |
| if (!GetChangedSendParameters(params, &changed_params)) { |
| return false; |
| } |
| |
| if (changed_params.negotiated_codecs) { |
| for (const auto& send_codec : *changed_params.negotiated_codecs) |
| RTC_LOG(LS_INFO) << "Negotiated codec: " << send_codec.codec.ToString(); |
| } |
| |
| send_params_ = params; |
| return ApplyChangedParams(changed_params); |
| } |
| |
| void WebRtcVideoChannel::RequestEncoderFallback() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (negotiated_codecs_.size() <= 1) { |
| RTC_LOG(LS_WARNING) << "Encoder failed but no fallback codec is available"; |
| return; |
| } |
| |
| ChangedSendParameters params; |
| params.negotiated_codecs = negotiated_codecs_; |
| params.negotiated_codecs->erase(params.negotiated_codecs->begin()); |
| params.send_codec = params.negotiated_codecs->front(); |
| ApplyChangedParams(params); |
| } |
| |
| void WebRtcVideoChannel::RequestEncoderSwitch( |
| const EncoderSwitchRequestCallback::Config& conf) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| if (!allow_codec_switching_) { |
| RTC_LOG(LS_INFO) << "Encoder switch requested but codec switching has" |
| " not been enabled yet."; |
| requested_encoder_switch_ = conf; |
| return; |
| } |
| |
| for (const VideoCodecSettings& codec_setting : negotiated_codecs_) { |
| if (codec_setting.codec.name == conf.codec_name) { |
| if (conf.param) { |
| auto it = codec_setting.codec.params.find(*conf.param); |
| if (it == codec_setting.codec.params.end()) |
| continue; |
| |
| if (conf.value && it->second != *conf.value) |
| continue; |
| } |
| |
| if (send_codec_ == codec_setting) { |
| // Already using this codec, no switch required. |
| return; |
| } |
| |
| ChangedSendParameters params; |
| params.send_codec = codec_setting; |
| ApplyChangedParams(params); |
| return; |
| } |
| } |
| |
| RTC_LOG(LS_WARNING) << "Requested encoder with codec_name:" << conf.codec_name |
| << ", param:" << conf.param.value_or("none") |
| << " and value:" << conf.value.value_or("none") |
| << "not found. No switch performed."; |
| } |
| |
| void WebRtcVideoChannel::RequestEncoderSwitch( |
| const webrtc::SdpVideoFormat& format) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| for (const VideoCodecSettings& codec_setting : negotiated_codecs_) { |
| if (IsSameCodec(format.name, format.parameters, codec_setting.codec.name, |
| codec_setting.codec.params)) { |
| VideoCodecSettings new_codec_setting = codec_setting; |
| for (const auto& kv : format.parameters) { |
| new_codec_setting.codec.params[kv.first] = kv.second; |
| } |
| |
| if (send_codec_ == new_codec_setting) { |
| // Already using this codec, no switch required. |
| return; |
| } |
| |
| ChangedSendParameters params; |
| params.send_codec = new_codec_setting; |
| ApplyChangedParams(params); |
| return; |
| } |
| } |
| |
| RTC_LOG(LS_WARNING) << "Encoder switch failed: SdpVideoFormat " |
| << format.ToString() << " not negotiated."; |
| } |
| |
| bool WebRtcVideoChannel::ApplyChangedParams( |
| const ChangedSendParameters& changed_params) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (changed_params.negotiated_codecs) |
| negotiated_codecs_ = *changed_params.negotiated_codecs; |
| |
| if (changed_params.send_codec) |
| send_codec_ = changed_params.send_codec; |
| |
| if (changed_params.extmap_allow_mixed) { |
| SetExtmapAllowMixed(*changed_params.extmap_allow_mixed); |
| } |
| if (changed_params.rtp_header_extensions) { |
| send_rtp_extensions_ = changed_params.rtp_header_extensions; |
| } |
| |
| if (changed_params.send_codec || changed_params.max_bandwidth_bps) { |
| if (send_params_.max_bandwidth_bps == -1) { |
| // Unset the global max bitrate (max_bitrate_bps) if max_bandwidth_bps is |
| // -1, which corresponds to no "b=AS" attribute in SDP. Note that the |
| // global max bitrate may be set below in GetBitrateConfigForCodec, from |
| // the codec max bitrate. |
| // TODO(pbos): This should be reconsidered (codec max bitrate should |
| // probably not affect global call max bitrate). |
| bitrate_config_.max_bitrate_bps = -1; |
| } |
| |
| 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.send_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 (send_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 |
| // WebRtcVideoChannel (in which case we're good), or per sender (SSRC), |
| // in which case this should not set a BitrateConstraints but rather |
| // reconfigure all senders. |
| bitrate_config_.max_bitrate_bps = send_params_.max_bandwidth_bps == 0 |
| ? -1 |
| : send_params_.max_bandwidth_bps; |
| } |
| |
| call_->GetTransportControllerSend()->SetSdpBitrateParameters( |
| bitrate_config_); |
| } |
| |
| for (auto& kv : send_streams_) { |
| kv.second->SetSendParameters(changed_params); |
| } |
| if (changed_params.send_codec || changed_params.rtcp_mode) { |
| // Update receive feedback parameters from new codec or RTCP mode. |
| RTC_LOG(LS_INFO) |
| << "SetFeedbackParameters 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( |
| HasLntf(send_codec_->codec), HasNack(send_codec_->codec), |
| HasTransportCc(send_codec_->codec), |
| send_params_.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound, |
| send_codec_->rtx_time); |
| } |
| } |
| return true; |
| } |
| |
| webrtc::RtpParameters WebRtcVideoChannel::GetRtpSendParameters( |
| uint32_t ssrc) const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| auto it = send_streams_.find(ssrc); |
| if (it == send_streams_.end()) { |
| RTC_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; |
| } |
| |
| webrtc::RTCError WebRtcVideoChannel::SetRtpSendParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetRtpSendParameters"); |
| auto it = send_streams_.find(ssrc); |
| if (it == send_streams_.end()) { |
| RTC_LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream " |
| "with ssrc " |
| << ssrc << " which doesn't exist."; |
| return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR); |
| } |
| |
| // 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) { |
| RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs " |
| "is not currently supported."; |
| return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR); |
| } |
| |
| if (!parameters.encodings.empty()) { |
| // Note that these values come from: |
| // https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-16#section-5 |
| // TODO(deadbeef): Change values depending on whether we are sending a |
| // keyframe or non-keyframe. |
| rtc::DiffServCodePoint new_dscp = rtc::DSCP_DEFAULT; |
| switch (parameters.encodings[0].network_priority) { |
| case webrtc::Priority::kVeryLow: |
| new_dscp = rtc::DSCP_CS1; |
| break; |
| case webrtc::Priority::kLow: |
| new_dscp = rtc::DSCP_DEFAULT; |
| break; |
| case webrtc::Priority::kMedium: |
| new_dscp = rtc::DSCP_AF42; |
| break; |
| case webrtc::Priority::kHigh: |
| new_dscp = rtc::DSCP_AF41; |
| break; |
| } |
| SetPreferredDscp(new_dscp); |
| } |
| |
| return it->second->SetRtpParameters(parameters); |
| } |
| |
| webrtc::RtpParameters WebRtcVideoChannel::GetRtpReceiveParameters( |
| uint32_t ssrc) const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| webrtc::RtpParameters rtp_params; |
| auto it = receive_streams_.find(ssrc); |
| if (it == receive_streams_.end()) { |
| RTC_LOG(LS_WARNING) |
| << "Attempting to get RTP receive parameters for stream " |
| "with SSRC " |
| << ssrc << " which doesn't exist."; |
| return webrtc::RtpParameters(); |
| } |
| rtp_params = it->second->GetRtpParameters(); |
| |
| // Add codecs, which any stream is prepared to receive. |
| for (const VideoCodec& codec : recv_params_.codecs) { |
| rtp_params.codecs.push_back(codec.ToCodecParameters()); |
| } |
| |
| return rtp_params; |
| } |
| |
| webrtc::RtpParameters WebRtcVideoChannel::GetDefaultRtpReceiveParameters() |
| const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| webrtc::RtpParameters rtp_params; |
| if (!default_unsignalled_ssrc_handler_.GetDefaultSink()) { |
| RTC_LOG(LS_WARNING) << "Attempting to get RTP parameters for the default, " |
| "unsignaled video receive stream, but not yet " |
| "configured to receive such a stream."; |
| return rtp_params; |
| } |
| rtp_params.encodings.emplace_back(); |
| |
| // Add codecs, which any stream is prepared to receive. |
| for (const VideoCodec& codec : recv_params_.codecs) { |
| rtp_params.codecs.push_back(codec.ToCodecParameters()); |
| } |
| |
| return rtp_params; |
| } |
| |
| bool WebRtcVideoChannel::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()) { |
| RTC_LOG(LS_ERROR) |
| << "GetChangedRecvParameters called without any video codecs."; |
| return false; |
| } |
| |
| // Verify that every mapped codec is supported locally. |
| if (params.is_stream_active) { |
| const std::vector<VideoCodec> local_supported_codecs = |
| GetPayloadTypesAndDefaultCodecs(decoder_factory_, |
| /*is_decoder_factory=*/true, |
| call_->trials()); |
| for (const VideoCodecSettings& mapped_codec : mapped_codecs) { |
| if (!FindMatchingCodec(local_supported_codecs, mapped_codec.codec)) { |
| RTC_LOG(LS_ERROR) |
| << "GetChangedRecvParameters called with unsupported video codec: " |
| << mapped_codec.codec.ToString(); |
| return false; |
| } |
| } |
| } |
| |
| if (NonFlexfecReceiveCodecsHaveChanged(recv_codecs_, mapped_codecs)) { |
| changed_params->codec_settings = |
| absl::optional<std::vector<VideoCodecSettings>>(mapped_codecs); |
| } |
| |
| // Handle RTP header extensions. |
| std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( |
| params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false, |
| call_->trials()); |
| if (filtered_extensions != recv_rtp_extensions_) { |
| changed_params->rtp_header_extensions = |
| absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions); |
| } |
| |
| int flexfec_payload_type = mapped_codecs.front().flexfec_payload_type; |
| if (flexfec_payload_type != recv_flexfec_payload_type_) { |
| changed_params->flexfec_payload_type = flexfec_payload_type; |
| } |
| |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::SetRecvParameters(const VideoRecvParameters& params) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetRecvParameters"); |
| RTC_DLOG(LS_INFO) << "SetRecvParameters: " << params.ToString(); |
| ChangedRecvParameters changed_params; |
| if (!GetChangedRecvParameters(params, &changed_params)) { |
| return false; |
| } |
| if (changed_params.flexfec_payload_type) { |
| RTC_DLOG(LS_INFO) << "Changing FlexFEC payload type (recv) from " |
| << recv_flexfec_payload_type_ << " to " |
| << *changed_params.flexfec_payload_type; |
| recv_flexfec_payload_type_ = *changed_params.flexfec_payload_type; |
| } |
| if (changed_params.rtp_header_extensions) { |
| recv_rtp_extensions_ = *changed_params.rtp_header_extensions; |
| } |
| if (changed_params.codec_settings) { |
| RTC_DLOG(LS_INFO) << "Changing recv codecs from " |
| << CodecSettingsVectorToString(recv_codecs_) << " to " |
| << CodecSettingsVectorToString( |
| *changed_params.codec_settings); |
| recv_codecs_ = *changed_params.codec_settings; |
| } |
| |
| for (auto& kv : receive_streams_) { |
| kv.second->SetRecvParameters(changed_params); |
| } |
| recv_params_ = params; |
| return true; |
| } |
| |
| std::string WebRtcVideoChannel::CodecSettingsVectorToString( |
| const std::vector<VideoCodecSettings>& codecs) { |
| rtc::StringBuilder 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.Release(); |
| } |
| |
| bool WebRtcVideoChannel::GetSendCodec(VideoCodec* codec) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (!send_codec_) { |
| RTC_LOG(LS_VERBOSE) << "GetSendCodec: No send codec set."; |
| return false; |
| } |
| *codec = send_codec_->codec; |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::SetSend(bool send) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSend"); |
| RTC_LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false"); |
| if (send && !send_codec_) { |
| RTC_DLOG(LS_ERROR) << "SetSend(true) called before setting codec."; |
| return false; |
| } |
| for (const auto& kv : send_streams_) { |
| kv.second->SetSend(send); |
| } |
| sending_ = send; |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::SetVideoSend( |
| uint32_t ssrc, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| TRACE_EVENT0("webrtc", "SetVideoSend"); |
| RTC_DCHECK(ssrc != 0); |
| RTC_LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", options: " |
| << (options ? options->ToString() : "nullptr") |
| << ", source = " << (source ? "(source)" : "nullptr") << ")"; |
| |
| const auto& kv = send_streams_.find(ssrc); |
| if (kv == send_streams_.end()) { |
| // Allow unknown ssrc only if source is null. |
| RTC_CHECK(source == nullptr); |
| RTC_LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; |
| return false; |
| } |
| |
| return kv->second->SetVideoSend(options, source); |
| } |
| |
| bool WebRtcVideoChannel::ValidateSendSsrcAvailability( |
| const StreamParams& sp) const { |
| for (uint32_t ssrc : sp.ssrcs) { |
| if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) { |
| RTC_LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc |
| << "' already exists."; |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::ValidateReceiveSsrcAvailability( |
| const StreamParams& sp) const { |
| for (uint32_t ssrc : sp.ssrcs) { |
| if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) { |
| RTC_LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc |
| << "' already exists."; |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::AddSendStream(const StreamParams& sp) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_LOG(LS_INFO) << "AddSendStream: " << sp.ToString(); |
| if (!ValidateStreamParams(sp)) |
| return false; |
| |
| if (!ValidateSendSsrcAvailability(sp)) |
| return false; |
| |
| for (uint32_t used_ssrc : sp.ssrcs) |
| send_ssrcs_.insert(used_ssrc); |
| |
| webrtc::VideoSendStream::Config config(this); |
| |
| for (const RidDescription& rid : sp.rids()) { |
| config.rtp.rids.push_back(rid.rid); |
| } |
| |
| config.suspend_below_min_bitrate = video_config_.suspend_below_min_bitrate; |
| config.periodic_alr_bandwidth_probing = |
| video_config_.periodic_alr_bandwidth_probing; |
| config.encoder_settings.experiment_cpu_load_estimator = |
| video_config_.experiment_cpu_load_estimator; |
| config.encoder_settings.encoder_factory = encoder_factory_; |
| config.encoder_settings.bitrate_allocator_factory = |
| bitrate_allocator_factory_; |
| config.encoder_settings.encoder_switch_request_callback = this; |
| config.crypto_options = crypto_options_; |
| config.rtp.extmap_allow_mixed = ExtmapAllowMixed(); |
| config.rtcp_report_interval_ms = video_config_.rtcp_report_interval_ms; |
| |
| WebRtcVideoSendStream* stream = new WebRtcVideoSendStream( |
| call_, sp, std::move(config), default_send_options_, |
| video_config_.enable_cpu_adaptation, 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; |
| RTC_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 WebRtcVideoChannel::RemoveSendStream(uint32_t ssrc) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_LOG(LS_INFO) << "RemoveSendStream: " << ssrc; |
| |
| WebRtcVideoSendStream* removed_stream; |
| 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; |
| RTC_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 WebRtcVideoChannel::DeleteReceiveStream( |
| WebRtcVideoChannel::WebRtcVideoReceiveStream* stream) { |
| for (uint32_t old_ssrc : stream->GetSsrcs()) |
| receive_ssrcs_.erase(old_ssrc); |
| delete stream; |
| } |
| |
| bool WebRtcVideoChannel::AddRecvStream(const StreamParams& sp) { |
| return AddRecvStream(sp, false); |
| } |
| |
| bool WebRtcVideoChannel::AddRecvStream(const StreamParams& sp, |
| bool default_stream) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| RTC_LOG(LS_INFO) << "AddRecvStream" |
| << (default_stream ? " (default stream)" : "") << ": " |
| << sp.ToString(); |
| if (!sp.has_ssrcs()) { |
| // This is a StreamParam with unsignaled SSRCs. Store it, so it can be used |
| // later when we know the SSRC on the first packet arrival. |
| unsignaled_stream_params_ = sp; |
| return true; |
| } |
| |
| if (!ValidateStreamParams(sp)) |
| return false; |
| |
| uint32_t ssrc = sp.first_ssrc(); |
| |
| // 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()) { |
| RTC_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); |
| webrtc::FlexfecReceiveStream::Config flexfec_config(this); |
| ConfigureReceiverRtp(&config, &flexfec_config, sp); |
| |
| config.crypto_options = crypto_options_; |
| config.enable_prerenderer_smoothing = |
| video_config_.enable_prerenderer_smoothing; |
| if (!sp.stream_ids().empty()) { |
| config.sync_group = sp.stream_ids()[0]; |
| } |
| |
| if (unsignaled_frame_transformer_ && !config.frame_transformer) |
| config.frame_transformer = unsignaled_frame_transformer_; |
| |
| receive_streams_[ssrc] = new WebRtcVideoReceiveStream( |
| this, call_, sp, std::move(config), decoder_factory_, default_stream, |
| recv_codecs_, flexfec_config); |
| |
| return true; |
| } |
| |
| void WebRtcVideoChannel::ConfigureReceiverRtp( |
| webrtc::VideoReceiveStream::Config* config, |
| webrtc::FlexfecReceiveStream::Config* flexfec_config, |
| const StreamParams& sp) const { |
| uint32_t ssrc = sp.first_ssrc(); |
| |
| config->rtp.remote_ssrc = ssrc; |
| config->rtp.local_ssrc = rtcp_receiver_report_ssrc_; |
| |
| // 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; |
| } |
| } |
| |
| // 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; |
| |
| // rtx-time (RFC 4588) is a declarative attribute similar to rtcp-rsize and |
| // determined by the sender / send codec. |
| if (send_codec_ && send_codec_->rtx_time != -1) { |
| config->rtp.nack.rtp_history_ms = send_codec_->rtx_time; |
| } |
| |
| config->rtp.transport_cc = |
| send_codec_ ? HasTransportCc(send_codec_->codec) : false; |
| |
| sp.GetFidSsrc(ssrc, &config->rtp.rtx_ssrc); |
| |
| config->rtp.extensions = recv_rtp_extensions_; |
| |
| // TODO(brandtr): Generalize when we add support for multistream protection. |
| flexfec_config->payload_type = recv_flexfec_payload_type_; |
| if (!IsDisabled(call_->trials(), "WebRTC-FlexFEC-03-Advertised") && |
| sp.GetFecFrSsrc(ssrc, &flexfec_config->remote_ssrc)) { |
| flexfec_config->protected_media_ssrcs = {ssrc}; |
| flexfec_config->local_ssrc = config->rtp.local_ssrc; |
| flexfec_config->rtcp_mode = config->rtp.rtcp_mode; |
| // TODO(brandtr): We should be spec-compliant and set |transport_cc| here |
| // based on the rtcp-fb for the FlexFEC codec, not the media codec. |
| flexfec_config->transport_cc = config->rtp.transport_cc; |
| flexfec_config->rtp_header_extensions = config->rtp.extensions; |
| } |
| } |
| |
| bool WebRtcVideoChannel::RemoveRecvStream(uint32_t ssrc) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_LOG(LS_INFO) << "RemoveRecvStream: " << ssrc; |
| |
| std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator stream = |
| receive_streams_.find(ssrc); |
| if (stream == receive_streams_.end()) { |
| RTC_LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc; |
| return false; |
| } |
| DeleteReceiveStream(stream->second); |
| receive_streams_.erase(stream); |
| |
| return true; |
| } |
| |
| void WebRtcVideoChannel::ResetUnsignaledRecvStream() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_LOG(LS_INFO) << "ResetUnsignaledRecvStream."; |
| unsignaled_stream_params_ = StreamParams(); |
| |
| // Delete any created default streams. This is needed to avoid SSRC collisions |
| // in Call's RtpDemuxer, in the case that |this| has created a default video |
| // receiver, and then some other WebRtcVideoChannel gets the SSRC signaled |
| // in the corresponding Unified Plan "m=" section. |
| auto it = receive_streams_.begin(); |
| while (it != receive_streams_.end()) { |
| if (it->second->IsDefaultStream()) { |
| DeleteReceiveStream(it->second); |
| receive_streams_.erase(it++); |
| } else { |
| ++it; |
| } |
| } |
| } |
| |
| void WebRtcVideoChannel::OnDemuxerCriteriaUpdatePending() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| ++demuxer_criteria_id_; |
| } |
| |
| void WebRtcVideoChannel::OnDemuxerCriteriaUpdateComplete() { |
| RTC_DCHECK_RUN_ON(&network_thread_checker_); |
| worker_thread_->PostTask(ToQueuedTask(task_safety_, [this] { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| ++demuxer_criteria_completed_id_; |
| })); |
| } |
| |
| bool WebRtcVideoChannel::SetSink( |
| uint32_t ssrc, |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_LOG(LS_INFO) << "SetSink: ssrc:" << ssrc << " " |
| << (sink ? "(ptr)" : "nullptr"); |
| |
| std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it = |
| receive_streams_.find(ssrc); |
| if (it == receive_streams_.end()) { |
| return false; |
| } |
| |
| it->second->SetSink(sink); |
| return true; |
| } |
| |
| void WebRtcVideoChannel::SetDefaultSink( |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_LOG(LS_INFO) << "SetDefaultSink: " << (sink ? "(ptr)" : "nullptr"); |
| default_unsignalled_ssrc_handler_.SetDefaultSink(this, sink); |
| } |
| |
| bool WebRtcVideoChannel::GetStats(VideoMediaInfo* info) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::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); |
| FillSendAndReceiveCodecStats(info); |
| // TODO(holmer): We should either have rtt available as a metric on |
| // VideoSend/ReceiveStreams, or we should remove rtt from VideoSenderInfo. |
| webrtc::Call::Stats stats = call_->GetStats(); |
| if (stats.rtt_ms != -1) { |
| for (size_t i = 0; i < info->senders.size(); ++i) { |
| info->senders[i].rtt_ms = stats.rtt_ms; |
| } |
| for (size_t i = 0; i < info->aggregated_senders.size(); ++i) { |
| info->aggregated_senders[i].rtt_ms = stats.rtt_ms; |
| } |
| } |
| |
| if (log_stats) |
| RTC_LOG(LS_INFO) << stats.ToString(now_ms); |
| |
| return true; |
| } |
| |
| void WebRtcVideoChannel::FillSenderStats(VideoMediaInfo* video_media_info, |
| bool log_stats) { |
| for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it = |
| send_streams_.begin(); |
| it != send_streams_.end(); ++it) { |
| auto infos = it->second->GetPerLayerVideoSenderInfos(log_stats); |
| if (infos.empty()) |
| continue; |
| video_media_info->aggregated_senders.push_back( |
| it->second->GetAggregatedVideoSenderInfo(infos)); |
| for (auto&& info : infos) { |
| video_media_info->senders.push_back(info); |
| } |
| } |
| } |
| |
| void WebRtcVideoChannel::FillReceiverStats(VideoMediaInfo* video_media_info, |
| bool log_stats) { |
| 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 WebRtcVideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream = |
| send_streams_.begin(); |
| stream != send_streams_.end(); ++stream) { |
| stream->second->FillBitrateInfo(bwe_info); |
| } |
| } |
| |
| void WebRtcVideoChannel::FillSendAndReceiveCodecStats( |
| VideoMediaInfo* video_media_info) { |
| for (const VideoCodec& codec : send_params_.codecs) { |
| webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters(); |
| video_media_info->send_codecs.insert( |
| std::make_pair(codec_params.payload_type, std::move(codec_params))); |
| } |
| for (const VideoCodec& codec : recv_params_.codecs) { |
| webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters(); |
| video_media_info->receive_codecs.insert( |
| std::make_pair(codec_params.payload_type, std::move(codec_params))); |
| } |
| } |
| |
| void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer packet, |
| int64_t packet_time_us) { |
| RTC_DCHECK_RUN_ON(&network_thread_checker_); |
| // TODO(bugs.webrtc.org/11993): This code is very similar to what |
| // WebRtcVoiceMediaChannel::OnPacketReceived does. For maintainability and |
| // consistency it would be good to move the interaction with call_->Receiver() |
| // to a common implementation and provide a callback on the worker thread |
| // for the exception case (DELIVERY_UNKNOWN_SSRC) and how retry is attempted. |
| worker_thread_->PostTask( |
| ToQueuedTask(task_safety_, [this, packet, packet_time_us] { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| const webrtc::PacketReceiver::DeliveryStatus delivery_result = |
| call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, packet, |
| packet_time_us); |
| 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; |
| } |
| |
| if (unknown_ssrc_packet_buffer_) { |
| unknown_ssrc_packet_buffer_->AddPacket(ssrc, packet_time_us, packet); |
| return; |
| } |
| |
| if (discard_unknown_ssrc_packets_) { |
| 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 (FEC). 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.ulpfec.red_rtx_payload_type || |
| payload_type == codec.ulpfec.ulpfec_payload_type) { |
| return; |
| } |
| } |
| if (payload_type == recv_flexfec_payload_type_) { |
| return; |
| } |
| |
| // Ignore unknown ssrcs if there is a demuxer criteria update pending. |
| // During a demuxer update we may receive ssrcs that were recently |
| // removed or we may receve ssrcs that were recently configured for a |
| // different video channel. |
| if (demuxer_criteria_id_ != demuxer_criteria_completed_id_) { |
| return; |
| } |
| |
| switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) { |
| case UnsignalledSsrcHandler::kDropPacket: |
| return; |
| case UnsignalledSsrcHandler::kDeliverPacket: |
| break; |
| } |
| |
| if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, packet, |
| packet_time_us) != |
| webrtc::PacketReceiver::DELIVERY_OK) { |
| RTC_LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery."; |
| } |
| })); |
| } |
| |
| void WebRtcVideoChannel::BackfillBufferedPackets( |
| rtc::ArrayView<const uint32_t> ssrcs) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (!unknown_ssrc_packet_buffer_) { |
| return; |
| } |
| |
| int delivery_ok_cnt = 0; |
| int delivery_unknown_ssrc_cnt = 0; |
| int delivery_packet_error_cnt = 0; |
| webrtc::PacketReceiver* receiver = this->call_->Receiver(); |
| unknown_ssrc_packet_buffer_->BackfillPackets( |
| ssrcs, [&](uint32_t /*ssrc*/, int64_t packet_time_us, |
| rtc::CopyOnWriteBuffer packet) { |
| switch (receiver->DeliverPacket(webrtc::MediaType::VIDEO, packet, |
| packet_time_us)) { |
| case webrtc::PacketReceiver::DELIVERY_OK: |
| delivery_ok_cnt++; |
| break; |
| case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC: |
| delivery_unknown_ssrc_cnt++; |
| break; |
| case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR: |
| delivery_packet_error_cnt++; |
| break; |
| } |
| }); |
| rtc::StringBuilder out; |
| out << "[ "; |
| for (uint32_t ssrc : ssrcs) { |
| out << std::to_string(ssrc) << " "; |
| } |
| out << "]"; |
| auto level = rtc::LS_INFO; |
| if (delivery_unknown_ssrc_cnt > 0 || delivery_packet_error_cnt > 0) { |
| level = rtc::LS_ERROR; |
| } |
| int total = |
| delivery_ok_cnt + delivery_unknown_ssrc_cnt + delivery_packet_error_cnt; |
| RTC_LOG_V(level) << "Backfilled " << total |
| << " packets for ssrcs: " << out.Release() |
| << " ok: " << delivery_ok_cnt |
| << " error: " << delivery_packet_error_cnt |
| << " unknown: " << delivery_unknown_ssrc_cnt; |
| } |
| |
| void WebRtcVideoChannel::OnReadyToSend(bool ready) { |
| RTC_DCHECK_RUN_ON(&network_thread_checker_); |
| RTC_LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready."); |
| call_->SignalChannelNetworkState( |
| webrtc::MediaType::VIDEO, |
| ready ? webrtc::kNetworkUp : webrtc::kNetworkDown); |
| } |
| |
| void WebRtcVideoChannel::OnNetworkRouteChanged( |
| const std::string& transport_name, |
| const rtc::NetworkRoute& network_route) { |
| RTC_DCHECK_RUN_ON(&network_thread_checker_); |
| worker_thread_->PostTask(ToQueuedTask( |
| task_safety_, [this, name = transport_name, route = network_route] { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| webrtc::RtpTransportControllerSendInterface* transport = |
| call_->GetTransportControllerSend(); |
| transport->OnNetworkRouteChanged(name, route); |
| transport->OnTransportOverheadChanged(route.packet_overhead); |
| })); |
| } |
| |
| void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| MediaChannel::SetInterface(iface); |
| // Set the RTP recv/send buffer to a bigger size. |
| |
| // The group should be a positive integer with an explicit size, in |
| // which case that is used as UDP recevie buffer size. All other values shall |
| // result in the default value being used. |
| const std::string group_name_recv_buf_size = |
| call_->trials().Lookup("WebRTC-IncreasedReceivebuffers"); |
| int recv_buffer_size = kVideoRtpRecvBufferSize; |
| if (!group_name_recv_buf_size.empty() && |
| (sscanf(group_name_recv_buf_size.c_str(), "%d", &recv_buffer_size) != 1 || |
| recv_buffer_size <= 0)) { |
| RTC_LOG(LS_WARNING) << "Invalid receive buffer size: " |
| << group_name_recv_buf_size; |
| recv_buffer_size = kVideoRtpRecvBufferSize; |
| } |
| |
| MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_RCVBUF, |
| recv_buffer_size); |
| |
| // 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. |
| const std::string group_name_send_buf_size = |
| call_->trials().Lookup("WebRTC-SendBufferSizeBytes"); |
| int send_buffer_size = kVideoRtpSendBufferSize; |
| if (!group_name_send_buf_size.empty() && |
| (sscanf(group_name_send_buf_size.c_str(), "%d", &send_buffer_size) != 1 || |
| send_buffer_size <= 0)) { |
| RTC_LOG(LS_WARNING) << "Invalid send buffer size: " |
| << group_name_send_buf_size; |
| send_buffer_size = kVideoRtpSendBufferSize; |
| } |
| |
| MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_SNDBUF, |
| send_buffer_size); |
| } |
| |
| void WebRtcVideoChannel::SetFrameDecryptor( |
| uint32_t ssrc, |
| rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| auto matching_stream = receive_streams_.find(ssrc); |
| if (matching_stream != receive_streams_.end()) { |
| matching_stream->second->SetFrameDecryptor(frame_decryptor); |
| } |
| } |
| |
| void WebRtcVideoChannel::SetFrameEncryptor( |
| uint32_t ssrc, |
| rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| auto matching_stream = send_streams_.find(ssrc); |
| if (matching_stream != send_streams_.end()) { |
| matching_stream->second->SetFrameEncryptor(frame_encryptor); |
| } else { |
| RTC_LOG(LS_ERROR) << "No stream found to attach frame encryptor"; |
| } |
| } |
| |
| void WebRtcVideoChannel::SetVideoCodecSwitchingEnabled(bool enabled) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| allow_codec_switching_ = enabled; |
| if (allow_codec_switching_) { |
| RTC_LOG(LS_INFO) << "Encoder switching enabled."; |
| if (requested_encoder_switch_) { |
| RTC_LOG(LS_INFO) << "Executing cached video encoder switch request."; |
| RequestEncoderSwitch(*requested_encoder_switch_); |
| requested_encoder_switch_.reset(); |
| } |
| } |
| } |
| |
| bool WebRtcVideoChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, |
| int delay_ms) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| absl::optional<uint32_t> default_ssrc = GetDefaultReceiveStreamSsrc(); |
| |
| // SSRC of 0 represents the default receive stream. |
| if (ssrc == 0) { |
| default_recv_base_minimum_delay_ms_ = delay_ms; |
| } |
| |
| if (ssrc == 0 && !default_ssrc) { |
| return true; |
| } |
| |
| if (ssrc == 0 && default_ssrc) { |
| ssrc = default_ssrc.value(); |
| } |
| |
| auto stream = receive_streams_.find(ssrc); |
| if (stream != receive_streams_.end()) { |
| stream->second->SetBaseMinimumPlayoutDelayMs(delay_ms); |
| return true; |
| } else { |
| RTC_LOG(LS_ERROR) << "No stream found to set base minimum playout delay"; |
| return false; |
| } |
| } |
| |
| absl::optional<int> WebRtcVideoChannel::GetBaseMinimumPlayoutDelayMs( |
| uint32_t ssrc) const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| // SSRC of 0 represents the default receive stream. |
| if (ssrc == 0) { |
| return default_recv_base_minimum_delay_ms_; |
| } |
| |
| auto stream = receive_streams_.find(ssrc); |
| if (stream != receive_streams_.end()) { |
| return stream->second->GetBaseMinimumPlayoutDelayMs(); |
| } else { |
| RTC_LOG(LS_ERROR) << "No stream found to get base minimum playout delay"; |
| return absl::nullopt; |
| } |
| } |
| |
| absl::optional<uint32_t> WebRtcVideoChannel::GetDefaultReceiveStreamSsrc() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| absl::optional<uint32_t> ssrc; |
| for (auto it = receive_streams_.begin(); it != receive_streams_.end(); ++it) { |
| if (it->second->IsDefaultStream()) { |
| ssrc.emplace(it->first); |
| break; |
| } |
| } |
| return ssrc; |
| } |
| |
| std::vector<webrtc::RtpSource> WebRtcVideoChannel::GetSources( |
| uint32_t ssrc) const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| auto it = receive_streams_.find(ssrc); |
| if (it == receive_streams_.end()) { |
| // TODO(bugs.webrtc.org/9781): Investigate standard compliance |
| // with sources for streams that has been removed. |
| RTC_LOG(LS_ERROR) << "Attempting to get contributing sources for SSRC:" |
| << ssrc << " which doesn't exist."; |
| return {}; |
| } |
| return it->second->GetSources(); |
| } |
| |
| bool WebRtcVideoChannel::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; |
| if (DscpEnabled()) { |
| rtc_options.dscp = PreferredDscp(); |
| } |
| rtc_options.info_signaled_after_sent.included_in_feedback = |
| options.included_in_feedback; |
| rtc_options.info_signaled_after_sent.included_in_allocation = |
| options.included_in_allocation; |
| return MediaChannel::SendPacket(&packet, rtc_options); |
| } |
| |
| bool WebRtcVideoChannel::SendRtcp(const uint8_t* data, size_t len) { |
| rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen); |
| rtc::PacketOptions rtc_options; |
| if (DscpEnabled()) { |
| rtc_options.dscp = PreferredDscp(); |
| } |
| |
| return MediaChannel::SendRtcp(&packet, rtc_options); |
| } |
| |
| WebRtcVideoChannel::WebRtcVideoSendStream::VideoSendStreamParameters:: |
| VideoSendStreamParameters( |
| webrtc::VideoSendStream::Config config, |
| const VideoOptions& options, |
| int max_bitrate_bps, |
| const absl::optional<VideoCodecSettings>& codec_settings) |
| : config(std::move(config)), |
| options(options), |
| max_bitrate_bps(max_bitrate_bps), |
| conference_mode(false), |
| codec_settings(codec_settings) {} |
| |
| WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream( |
| webrtc::Call* call, |
| const StreamParams& sp, |
| webrtc::VideoSendStream::Config config, |
| const VideoOptions& options, |
| bool enable_cpu_overuse_detection, |
| int max_bitrate_bps, |
| const absl::optional<VideoCodecSettings>& codec_settings, |
| const absl::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), |
| enable_cpu_overuse_detection_(enable_cpu_overuse_detection), |
| source_(nullptr), |
| stream_(nullptr), |
| parameters_(std::move(config), options, max_bitrate_bps, codec_settings), |
| rtp_parameters_(CreateRtpParametersWithEncodings(sp)), |
| sending_(false), |
| disable_automatic_resize_( |
| IsEnabled(call->trials(), "WebRTC-Video-DisableAutomaticResize")) { |
| // Maximum packet size may come in RtpConfig from external transport, for |
| // example from QuicTransportInterface implementation, so do not exceed |
| // given max_packet_size. |
| parameters_.config.rtp.max_packet_size = |
| std::min<size_t>(parameters_.config.rtp.max_packet_size, kVideoMtu); |
| parameters_.conference_mode = send_params.conference_mode; |
| |
| sp.GetPrimarySsrcs(¶meters_.config.rtp.ssrcs); |
| |
| // ValidateStreamParams should prevent this from happening. |
| RTC_CHECK(!parameters_.config.rtp.ssrcs.empty()); |
| rtp_parameters_.encodings[0].ssrc = parameters_.config.rtp.ssrcs[0]; |
| |
| // RTX. |
| sp.GetFidSsrcs(parameters_.config.rtp.ssrcs, |
| ¶meters_.config.rtp.rtx.ssrcs); |
| |
| // FlexFEC SSRCs. |
| // TODO(brandtr): This code needs to be generalized when we add support for |
| // multistream protection. |
| if (IsEnabled(call_->trials(), "WebRTC-FlexFEC-03")) { |
| uint32_t flexfec_ssrc; |
| bool flexfec_enabled = false; |
| for (uint32_t primary_ssrc : parameters_.config.rtp.ssrcs) { |
| if (sp.GetFecFrSsrc(primary_ssrc, &flexfec_ssrc)) { |
| if (flexfec_enabled) { |
| RTC_LOG(LS_INFO) |
| << "Multiple FlexFEC streams in local SDP, but " |
| "our implementation only supports a single FlexFEC " |
| "stream. Will not enable FlexFEC for proposed " |
| "stream with SSRC: " |
| << flexfec_ssrc << "."; |
| continue; |
| } |
| |
| flexfec_enabled = true; |
| parameters_.config.rtp.flexfec.ssrc = flexfec_ssrc; |
| parameters_.config.rtp.flexfec.protected_media_ssrcs = {primary_ssrc}; |
| } |
| } |
| } |
| |
| parameters_.config.rtp.c_name = sp.cname; |
| if (rtp_extensions) { |
| parameters_.config.rtp.extensions = *rtp_extensions; |
| rtp_parameters_.header_extensions = *rtp_extensions; |
| } |
| parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size |
| ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound; |
| parameters_.config.rtp.mid = send_params.mid; |
| rtp_parameters_.rtcp.reduced_size = send_params.rtcp.reduced_size; |
| |
| if (codec_settings) { |
| SetCodec(*codec_settings); |
| } |
| } |
| |
| WebRtcVideoChannel::WebRtcVideoSendStream::~WebRtcVideoSendStream() { |
| if (stream_ != NULL) { |
| call_->DestroyVideoSendStream(stream_); |
| } |
| } |
| |
| bool WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend( |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetVideoSend"); |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| if (options) { |
| VideoOptions old_options = parameters_.options; |
| parameters_.options.SetAll(*options); |
| if (parameters_.options.is_screencast.value_or(false) != |
| old_options.is_screencast.value_or(false) && |
| parameters_.codec_settings) { |
| // If screen content settings change, we may need to recreate the codec |
| // instance so that the correct type is used. |
| |
| SetCodec(*parameters_.codec_settings); |
| // Mark screenshare parameter as being updated, then test for any other |
| // changes that may require codec reconfiguration. |
| old_options.is_screencast = options->is_screencast; |
| } |
| if (parameters_.options != old_options) { |
| ReconfigureEncoder(); |
| } |
| } |
| |
| if (source_ && stream_) { |
| stream_->SetSource(nullptr, webrtc::DegradationPreference::DISABLED); |
| } |
| // Switch to the new source. |
| source_ = source; |
| if (source && stream_) { |
| stream_->SetSource(source_, GetDegradationPreference()); |
| } |
| return true; |
| } |
| |
| webrtc::DegradationPreference |
| WebRtcVideoChannel::WebRtcVideoSendStream::GetDegradationPreference() const { |
| // Do not adapt resolution for screen content as this will likely |
| // result in blurry and unreadable text. |
| // |this| acts like a VideoSource to make sure SinkWants are handled on the |
| // correct thread. |
| if (!enable_cpu_overuse_detection_) { |
| return webrtc::DegradationPreference::DISABLED; |
| } |
| |
| webrtc::DegradationPreference degradation_preference; |
| if (rtp_parameters_.degradation_preference.has_value()) { |
| degradation_preference = *rtp_parameters_.degradation_preference; |
| } else { |
| if (parameters_.options.content_hint == |
| webrtc::VideoTrackInterface::ContentHint::kFluid) { |
| degradation_preference = |
| webrtc::DegradationPreference::MAINTAIN_FRAMERATE; |
| } else if (parameters_.options.is_screencast.value_or(false) || |
| parameters_.options.content_hint == |
| webrtc::VideoTrackInterface::ContentHint::kDetailed || |
| parameters_.options.content_hint == |
| webrtc::VideoTrackInterface::ContentHint::kText) { |
| degradation_preference = |
| webrtc::DegradationPreference::MAINTAIN_RESOLUTION; |
| } else if (IsEnabled(call_->trials(), "WebRTC-Video-BalancedDegradation")) { |
| // Standard wants balanced by default, but it needs to be tuned first. |
| degradation_preference = webrtc::DegradationPreference::BALANCED; |
| } else { |
| // Keep MAINTAIN_FRAMERATE by default until BALANCED has been tuned for |
| // all codecs and launched. |
| degradation_preference = |
| webrtc::DegradationPreference::MAINTAIN_FRAMERATE; |
| } |
| } |
| |
| return degradation_preference; |
| } |
| |
| const std::vector<uint32_t>& |
| WebRtcVideoChannel::WebRtcVideoSendStream::GetSsrcs() const { |
| return ssrcs_; |
| } |
| |
| void WebRtcVideoChannel::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, 0); |
| |
| parameters_.config.rtp.payload_name = codec_settings.codec.name; |
| parameters_.config.rtp.payload_type = codec_settings.codec.id; |
| parameters_.config.rtp.raw_payload = |
| codec_settings.codec.packetization == kPacketizationParamRaw; |
| parameters_.config.rtp.ulpfec = codec_settings.ulpfec; |
| parameters_.config.rtp.flexfec.payload_type = |
| codec_settings.flexfec_payload_type; |
| |
| // Set RTX payload type if RTX is enabled. |
| if (!parameters_.config.rtp.rtx.ssrcs.empty()) { |
| if (codec_settings.rtx_payload_type == -1) { |
| RTC_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; |
| } |
| } |
| |
| const bool has_lntf = HasLntf(codec_settings.codec); |
| parameters_.config.rtp.lntf.enabled = has_lntf; |
| parameters_.config.encoder_settings.capabilities.loss_notification = has_lntf; |
| |
| parameters_.config.rtp.nack.rtp_history_ms = |
| HasNack(codec_settings.codec) ? kNackHistoryMs : 0; |
| |
| parameters_.codec_settings = codec_settings; |
| |
| // TODO(nisse): Avoid recreation, it should be enough to call |
| // ReconfigureEncoder. |
| RTC_LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetCodec."; |
| RecreateWebRtcStream(); |
| } |
| |
| void WebRtcVideoChannel::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; |
| rtp_parameters_.rtcp.reduced_size = |
| parameters_.config.rtp.rtcp_mode == webrtc::RtcpMode::kReducedSize; |
| recreate_stream = true; |
| } |
| if (params.extmap_allow_mixed) { |
| parameters_.config.rtp.extmap_allow_mixed = *params.extmap_allow_mixed; |
| recreate_stream = true; |
| } |
| if (params.rtp_header_extensions) { |
| parameters_.config.rtp.extensions = *params.rtp_header_extensions; |
| rtp_parameters_.header_extensions = *params.rtp_header_extensions; |
| recreate_stream = true; |
| } |
| if (params.mid) { |
| parameters_.config.rtp.mid = *params.mid; |
| 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.send_codec) { |
| SetCodec(*params.send_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) { |
| RTC_LOG(LS_INFO) |
| << "RecreateWebRtcStream (send) because of SetSendParameters"; |
| RecreateWebRtcStream(); |
| } |
| } |
| |
| webrtc::RTCError WebRtcVideoChannel::WebRtcVideoSendStream::SetRtpParameters( |
| const webrtc::RtpParameters& new_parameters) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| webrtc::RTCError error = CheckRtpParametersInvalidModificationAndValues( |
| rtp_parameters_, new_parameters); |
| if (!error.ok()) { |
| return error; |
| } |
| |
| bool new_param = false; |
| for (size_t i = 0; i < rtp_parameters_.encodings.size(); ++i) { |
| if ((new_parameters.encodings[i].min_bitrate_bps != |
| rtp_parameters_.encodings[i].min_bitrate_bps) || |
| (new_parameters.encodings[i].max_bitrate_bps != |
| rtp_parameters_.encodings[i].max_bitrate_bps) || |
| (new_parameters.encodings[i].max_framerate != |
| rtp_parameters_.encodings[i].max_framerate) || |
| (new_parameters.encodings[i].scale_resolution_down_by != |
| rtp_parameters_.encodings[i].scale_resolution_down_by) || |
| (new_parameters.encodings[i].num_temporal_layers != |
| rtp_parameters_.encodings[i].num_temporal_layers)) { |
| new_param = true; |
| break; |
| } |
| } |
| |
| bool new_degradation_preference = false; |
| if (new_parameters.degradation_preference != |
| rtp_parameters_.degradation_preference) { |
| new_degradation_preference = true; |
| } |
| |
| // TODO(bugs.webrtc.org/8807): The bitrate priority really doesn't require an |
| // entire encoder reconfiguration, it just needs to update the bitrate |
| // allocator. |
| bool reconfigure_encoder = new_param || |
| (new_parameters.encodings[0].bitrate_priority != |
| rtp_parameters_.encodings[0].bitrate_priority) || |
| new_parameters.encodings[0].scalability_mode != |
| rtp_parameters_.encodings[0].scalability_mode; |
| |
| // TODO(bugs.webrtc.org/8807): The active field as well should not require |
| // a full encoder reconfiguration, but it needs to update both the bitrate |
| // allocator and the video bitrate allocator. |
| // |
| // Note that the simulcast encoder adapter relies on the fact that layers |
| // de/activation triggers encoder reinitialization. |
| bool new_send_state = false; |
| for (size_t i = 0; i < rtp_parameters_.encodings.size(); ++i) { |
| bool new_active = IsLayerActive(new_parameters.encodings[i]); |
| bool old_active = IsLayerActive(rtp_parameters_.encodings[i]); |
| if (new_active != old_active) { |
| new_send_state = true; |
| } |
| } |
| rtp_parameters_ = new_parameters; |
| // Codecs are currently handled at the WebRtcVideoChannel level. |
| rtp_parameters_.codecs.clear(); |
| if (reconfigure_encoder || new_send_state) { |
| ReconfigureEncoder(); |
| } |
| if (new_send_state) { |
| UpdateSendState(); |
| } |
| if (new_degradation_preference) { |
| if (source_ && stream_) { |
| stream_->SetSource(source_, GetDegradationPreference()); |
| } |
| } |
| return webrtc::RTCError::OK(); |
| } |
| |
| webrtc::RtpParameters |
| WebRtcVideoChannel::WebRtcVideoSendStream::GetRtpParameters() const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| return rtp_parameters_; |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoSendStream::SetFrameEncryptor( |
| rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| parameters_.config.frame_encryptor = frame_encryptor; |
| if (stream_) { |
| RTC_LOG(LS_INFO) |
| << "RecreateWebRtcStream (send) because of SetFrameEncryptor, ssrc=" |
| << parameters_.config.rtp.ssrcs[0]; |
| RecreateWebRtcStream(); |
| } |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoSendStream::UpdateSendState() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (sending_) { |
| RTC_DCHECK(stream_ != nullptr); |
| size_t num_layers = rtp_parameters_.encodings.size(); |
| if (parameters_.encoder_config.number_of_streams == 1) { |
| // SVC is used. Only one simulcast layer is present. |
| num_layers = 1; |
| } |
| std::vector<bool> active_layers(num_layers); |
| for (size_t i = 0; i < num_layers; ++i) { |
| active_layers[i] = IsLayerActive(rtp_parameters_.encodings[i]); |
| } |
| if (parameters_.encoder_config.number_of_streams == 1 && |
| rtp_parameters_.encodings.size() > 1) { |
| // SVC is used. |
| // The only present simulcast layer should be active if any of the |
| // configured SVC layers is active. |
| active_layers[0] = |
| absl::c_any_of(rtp_parameters_.encodings, |
| [](const auto& encoding) { return encoding.active; }); |
| } |
| // This updates what simulcast layers are sending, and possibly starts |
| // or stops the VideoSendStream. |
| stream_->UpdateActiveSimulcastLayers(active_layers); |
| } else { |
| if (stream_ != nullptr) { |
| stream_->Stop(); |
| } |
| } |
| } |
| |
| webrtc::VideoEncoderConfig |
| WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig( |
| const VideoCodec& codec) const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| webrtc::VideoEncoderConfig encoder_config; |
| encoder_config.codec_type = webrtc::PayloadStringToCodecType(codec.name); |
| encoder_config.video_format = |
| webrtc::SdpVideoFormat(codec.name, codec.params); |
| |
| 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 disabled for simulcast |
| // or a screencast (and not in simulcast screenshare experiment), only |
| // configure a single stream. |
| encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size(); |
| if (IsCodecDisabledForSimulcast(codec.name, call_->trials())) { |
| encoder_config.number_of_streams = 1; |
| } |
| |
| // parameters_.max_bitrate comes from the max bitrate set at the SDP |
| // (m-section) level with the attribute "b=AS." Note that we override this |
| // value below if the RtpParameters max bitrate set with |
| // RtpSender::SetParameters has a lower value. |
| int stream_max_bitrate = parameters_.max_bitrate_bps; |
| // When simulcast is enabled (when there are multiple encodings), |
| // encodings[i].max_bitrate_bps will be enforced by |
| // encoder_config.simulcast_layers[i].max_bitrate_bps. Otherwise, it's |
| // enforced by stream_max_bitrate, taking the minimum of the two maximums |
| // (one coming from SDP, the other coming from RtpParameters). |
| if (rtp_parameters_.encodings[0].max_bitrate_bps && |
| rtp_parameters_.encodings.size() == 1) { |
| stream_max_bitrate = |
| MinPositive(*(rtp_parameters_.encodings[0].max_bitrate_bps), |
| parameters_.max_bitrate_bps); |
| } |
| |
| // The codec max bitrate comes from the "x-google-max-bitrate" parameter |
| // attribute set in the SDP for a specific codec. As done in |
| // WebRtcVideoChannel::SetSendParameters, this value does not override the |
| // stream max_bitrate set above. |
| int codec_max_bitrate_kbps; |
| if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps) && |
| stream_max_bitrate == -1) { |
| stream_max_bitrate = codec_max_bitrate_kbps * 1000; |
| } |
| encoder_config.max_bitrate_bps = stream_max_bitrate; |
| |
| // The encoder config's default bitrate priority is set to 1.0, |
| // unless it is set through the sender's encoding parameters. |
| // The bitrate priority, which is used in the bitrate allocation, is done |
| // on a per sender basis, so we use the first encoding's value. |
| encoder_config.bitrate_priority = |
| rtp_parameters_.encodings[0].bitrate_priority; |
| |
| // Application-controlled state is held in the encoder_config's |
| // simulcast_layers. Currently this is used to control which simulcast layers |
| // are active and for configuring the min/max bitrate and max framerate. |
| // The encoder_config's simulcast_layers is also used for non-simulcast (when |
| // there is a single layer). |
| RTC_DCHECK_GE(rtp_parameters_.encodings.size(), |
| encoder_config.number_of_streams); |
| RTC_DCHECK_GT(encoder_config.number_of_streams, 0); |
| |
| // Copy all provided constraints. |
| encoder_config.simulcast_layers.resize(rtp_parameters_.encodings.size()); |
| for (size_t i = 0; i < encoder_config.simulcast_layers.size(); ++i) { |
| encoder_config.simulcast_layers[i].active = |
| rtp_parameters_.encodings[i].active; |
| encoder_config.simulcast_layers[i].scalability_mode = |
| rtp_parameters_.encodings[i].scalability_mode; |
| if (rtp_parameters_.encodings[i].min_bitrate_bps) { |
| encoder_config.simulcast_layers[i].min_bitrate_bps = |
| *rtp_parameters_.encodings[i].min_bitrate_bps; |
| } |
| if (rtp_parameters_.encodings[i].max_bitrate_bps) { |
| encoder_config.simulcast_layers[i].max_bitrate_bps = |
| *rtp_parameters_.encodings[i].max_bitrate_bps; |
| } |
| if (rtp_parameters_.encodings[i].max_framerate) { |
| encoder_config.simulcast_layers[i].max_framerate = |
| *rtp_parameters_.encodings[i].max_framerate; |
| } |
| if (rtp_parameters_.encodings[i].scale_resolution_down_by) { |
| encoder_config.simulcast_layers[i].scale_resolution_down_by = |
| *rtp_parameters_.encodings[i].scale_resolution_down_by; |
| } |
| if (rtp_parameters_.encodings[i].num_temporal_layers) { |
| encoder_config.simulcast_layers[i].num_temporal_layers = |
| *rtp_parameters_.encodings[i].num_temporal_layers; |
| } |
| } |
| |
| encoder_config.legacy_conference_mode = parameters_.conference_mode; |
| |
| encoder_config.is_quality_scaling_allowed = |
| !disable_automatic_resize_ && !is_screencast && |
| (parameters_.config.rtp.ssrcs.size() == 1 || |
| NumActiveStreams(rtp_parameters_) == 1); |
| |
| int max_qp = kDefaultQpMax; |
| codec.GetParam(kCodecParamMaxQuantization, &max_qp); |
| encoder_config.video_stream_factory = |
| new rtc::RefCountedObject<EncoderStreamFactory>( |
| codec.name, max_qp, is_screencast, parameters_.conference_mode); |
| |
| return encoder_config; |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoSendStream::ReconfigureEncoder() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (!stream_) { |
| // The webrtc::VideoSendStream |stream_| has not yet been created but other |
| // parameters has changed. |
| return; |
| } |
| |
| RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0); |
| |
| 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 WebRtcVideoChannel::WebRtcVideoSendStream::SetSend(bool send) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| sending_ = send; |
| UpdateSendState(); |
| } |
| |
| std::vector<VideoSenderInfo> |
| WebRtcVideoChannel::WebRtcVideoSendStream::GetPerLayerVideoSenderInfos( |
| bool log_stats) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| VideoSenderInfo common_info; |
| if (parameters_.codec_settings) { |
| common_info.codec_name = parameters_.codec_settings->codec.name; |
| common_info.codec_payload_type = parameters_.codec_settings->codec.id; |
| } |
| std::vector<VideoSenderInfo> infos; |
| webrtc::VideoSendStream::Stats stats; |
| if (stream_ == nullptr) { |
| for (uint32_t ssrc : parameters_.config.rtp.ssrcs) { |
| common_info.add_ssrc(ssrc); |
| } |
| infos.push_back(common_info); |
| return infos; |
| } else { |
| stats = stream_->GetStats(); |
| if (log_stats) |
| RTC_LOG(LS_INFO) << stats.ToString(rtc::TimeMillis()); |
| |
| // Metrics that are in common for all substreams. |
| common_info.adapt_changes = stats.number_of_cpu_adapt_changes; |
| common_info.adapt_reason = |
| stats.cpu_limited_resolution ? ADAPTREASON_CPU : ADAPTREASON_NONE; |
| common_info.has_entered_low_resolution = stats.has_entered_low_resolution; |
| |
| // 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) |
| common_info.adapt_reason |= ADAPTREASON_BANDWIDTH; |
| |
| common_info.quality_limitation_reason = stats.quality_limitation_reason; |
| common_info.quality_limitation_durations_ms = |
| stats.quality_limitation_durations_ms; |
| common_info.quality_limitation_resolution_changes = |
| stats.quality_limitation_resolution_changes; |
| common_info.encoder_implementation_name = stats.encoder_implementation_name; |
| common_info.ssrc_groups = ssrc_groups_; |
| common_info.frames = stats.frames; |
| common_info.framerate_input = stats.input_frame_rate; |
| common_info.avg_encode_ms = stats.avg_encode_time_ms; |
| common_info.encode_usage_percent = stats.encode_usage_percent; |
| common_info.nominal_bitrate = stats.media_bitrate_bps; |
| common_info.content_type = stats.content_type; |
| common_info.aggregated_framerate_sent = stats.encode_frame_rate; |
| common_info.aggregated_huge_frames_sent = stats.huge_frames_sent; |
| |
| // If we don't have any substreams, get the remaining metrics from |stats|. |
| // Otherwise, these values are obtained from |sub_stream| below. |
| if (stats.substreams.empty()) { |
| for (uint32_t ssrc : parameters_.config.rtp.ssrcs) { |
| common_info.add_ssrc(ssrc); |
| } |
| common_info.framerate_sent = stats.encode_frame_rate; |
| common_info.frames_encoded = stats.frames_encoded; |
| common_info.total_encode_time_ms = stats.total_encode_time_ms; |
| common_info.total_encoded_bytes_target = stats.total_encoded_bytes_target; |
| common_info.frames_sent = stats.frames_encoded; |
| common_info.huge_frames_sent = stats.huge_frames_sent; |
| infos.push_back(common_info); |
| return infos; |
| } |
| } |
| auto outbound_rtp_substreams = |
| MergeInfoAboutOutboundRtpSubstreams(stats.substreams); |
| for (const auto& pair : outbound_rtp_substreams) { |
| auto info = common_info; |
| info.add_ssrc(pair.first); |
| info.rid = parameters_.config.rtp.GetRidForSsrc(pair.first); |
| auto stream_stats = pair.second; |
| RTC_DCHECK_EQ(stream_stats.type, |
| webrtc::VideoSendStream::StreamStats::StreamType::kMedia); |
| info.payload_bytes_sent = stream_stats.rtp_stats.transmitted.payload_bytes; |
| info.header_and_padding_bytes_sent = |
| stream_stats.rtp_stats.transmitted.header_bytes + |
| stream_stats.rtp_stats.transmitted.padding_bytes; |
| info.packets_sent = stream_stats.rtp_stats.transmitted.packets; |
| info.total_packet_send_delay_ms += stream_stats.total_packet_send_delay_ms; |
| info.send_frame_width = stream_stats.width; |
| info.send_frame_height = stream_stats.height; |
| info.key_frames_encoded = stream_stats.frame_counts.key_frames; |
| info.framerate_sent = stream_stats.encode_frame_rate; |
| info.frames_encoded = stream_stats.frames_encoded; |
| info.frames_sent = stream_stats.frames_encoded; |
| info.retransmitted_bytes_sent = |
| stream_stats.rtp_stats.retransmitted.payload_bytes; |
| info.retransmitted_packets_sent = |
| stream_stats.rtp_stats.retransmitted.packets; |
| info.packets_lost = stream_stats.rtcp_stats.packets_lost; |
| 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 (stream_stats.report_block_data.has_value()) { |
| info.report_block_datas.push_back(stream_stats.report_block_data.value()); |
| } |
| info.fraction_lost = |
| static_cast<float>(stream_stats.rtcp_stats.fraction_lost) / (1 << 8); |
| info.qp_sum = stream_stats.qp_sum; |
| info.total_encode_time_ms = stream_stats.total_encode_time_ms; |
| info.total_encoded_bytes_target = stream_stats.total_encoded_bytes_target; |
| info.huge_frames_sent = stream_stats.huge_frames_sent; |
| infos.push_back(info); |
| } |
| return infos; |
| } |
| |
| VideoSenderInfo |
| WebRtcVideoChannel::WebRtcVideoSendStream::GetAggregatedVideoSenderInfo( |
| const std::vector<VideoSenderInfo>& infos) const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_CHECK(!infos.empty()); |
| if (infos.size() == 1) { |
| return infos[0]; |
| } |
| VideoSenderInfo info = infos[0]; |
| info.local_stats.clear(); |
| for (uint32_t ssrc : parameters_.config.rtp.ssrcs) { |
| info.add_ssrc(ssrc); |
| } |
| info.framerate_sent = info.aggregated_framerate_sent; |
| info.huge_frames_sent = info.aggregated_huge_frames_sent; |
| |
| for (size_t i = 1; i < infos.size(); i++) { |
| info.key_frames_encoded += infos[i].key_frames_encoded; |
| info.payload_bytes_sent += infos[i].payload_bytes_sent; |
| info.header_and_padding_bytes_sent += |
| infos[i].header_and_padding_bytes_sent; |
| info.packets_sent += infos[i].packets_sent; |
| info.total_packet_send_delay_ms += infos[i].total_packet_send_delay_ms; |
| info.retransmitted_bytes_sent += infos[i].retransmitted_bytes_sent; |
| info.retransmitted_packets_sent += infos[i].retransmitted_packets_sent; |
| info.packets_lost += infos[i].packets_lost; |
| if (infos[i].send_frame_width > info.send_frame_width) |
| info.send_frame_width = infos[i].send_frame_width; |
| if (infos[i].send_frame_height > info.send_frame_height) |
| info.send_frame_height = infos[i].send_frame_height; |
| info.firs_rcvd += infos[i].firs_rcvd; |
| info.nacks_rcvd += infos[i].nacks_rcvd; |
| info.plis_rcvd += infos[i].plis_rcvd; |
| if (infos[i].report_block_datas.size()) |
| info.report_block_datas.push_back(infos[i].report_block_datas[0]); |
| if (infos[i].qp_sum) { |
| if (!info.qp_sum) { |
| info.qp_sum = 0; |
| } |
| info.qp_sum = *info.qp_sum + *infos[i].qp_sum; |
| } |
| info.frames_encoded += infos[i].frames_encoded; |
| info.frames_sent += infos[i].frames_sent; |
| info.total_encode_time_ms += infos[i].total_encode_time_ms; |
| info.total_encoded_bytes_target += infos[i].total_encoded_bytes_target; |
| } |
| return info; |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoSendStream::FillBitrateInfo( |
| 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 WebRtcVideoChannel::WebRtcVideoSendStream:: |
| SetEncoderToPacketizerFrameTransformer( |
| rtc::scoped_refptr<webrtc::FrameTransformerInterface> |
| frame_transformer) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| parameters_.config.frame_transformer = std::move(frame_transformer); |
| if (stream_) |
| RecreateWebRtcStream(); |
| } |
| |
| void WebRtcVideoChannel::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) { |
| RTC_LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX " |
| "payload type the set codec. Ignoring RTX."; |
| config.rtp.rtx.ssrcs.clear(); |
| } |
| if (parameters_.encoder_config.number_of_streams == 1) { |
| // SVC is used instead of simulcast. Remove unnecessary SSRCs. |
| if (config.rtp.ssrcs.size() > 1) { |
| config.rtp.ssrcs.resize(1); |
| if (config.rtp.rtx.ssrcs.size() > 1) { |
| config.rtp.rtx.ssrcs.resize(1); |
| } |
| } |
| } |
| stream_ = call_->CreateVideoSendStream(std::move(config), |
| parameters_.encoder_config.Copy()); |
| |
| parameters_.encoder_config.encoder_specific_settings = NULL; |
| |
|