| /* |
| * 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/webrtcvideoengine.h" |
| |
| #include <stdio.h> |
| #include <algorithm> |
| #include <set> |
| #include <string> |
| #include <utility> |
| |
| #include "api/video/i420_buffer.h" |
| #include "api/video_codecs/sdp_video_format.h" |
| #include "api/video_codecs/video_decoder.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 "common_video/h264/profile_level_id.h" |
| #include "media/engine/constants.h" |
| #include "media/engine/internaldecoderfactory.h" |
| #include "media/engine/internalencoderfactory.h" |
| #include "media/engine/scopedvideodecoder.h" |
| #include "media/engine/scopedvideoencoder.h" |
| #include "media/engine/simulcast.h" |
| #include "media/engine/simulcast_encoder_adapter.h" |
| #include "media/engine/videodecodersoftwarefallbackwrapper.h" |
| #include "media/engine/videoencodersoftwarefallbackwrapper.h" |
| #include "media/engine/webrtcmediaengine.h" |
| #include "media/engine/webrtcvideoencoderfactory.h" |
| #include "media/engine/webrtcvoiceengine.h" |
| #include "rtc_base/copyonwritebuffer.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/stringutils.h" |
| #include "rtc_base/timeutils.h" |
| #include "rtc_base/trace_event.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "vp8_encoder_simulcast_proxy.h" |
| |
| using DegradationPreference = webrtc::VideoSendStream::DegradationPreference; |
| |
| namespace cricket { |
| // This class represents all encoders, i.e. both internal and external. It |
| // serves as a temporary adapter between WebRtcVideoEncoderFactory* and the new |
| // factory interface that is being developed. |
| // TODO(magjed): Remove once WebRtcVideoEncoderFactory* is deprecated and |
| // webrtc:7925 is fixed. |
| class EncoderFactoryAdapter { |
| public: |
| struct AllocatedEncoder { |
| AllocatedEncoder() = default; |
| AllocatedEncoder(std::unique_ptr<webrtc::VideoEncoder> encoder, |
| bool is_hardware_accelerated, |
| bool has_internal_source); |
| |
| std::unique_ptr<webrtc::VideoEncoder> encoder; |
| bool is_hardware_accelerated; |
| bool has_internal_source; |
| }; |
| |
| virtual ~EncoderFactoryAdapter() {} |
| |
| virtual AllocatedEncoder CreateVideoEncoder( |
| const VideoCodec& codec) const = 0; |
| |
| virtual std::vector<VideoCodec> GetSupportedCodecs() const = 0; |
| }; |
| |
| class DecoderFactoryAdapter { |
| public: |
| virtual ~DecoderFactoryAdapter() {} |
| |
| virtual std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder( |
| const VideoCodec& codec, |
| const VideoDecoderParams& decoder_params) const = 0; |
| }; |
| |
| namespace { |
| |
| std::vector<VideoCodec> AssignPayloadTypesAndAddAssociatedRtxCodecs( |
| const std::vector<VideoCodec>& input_codecs); |
| |
| // Wraps cricket::WebRtcVideoEncoderFactory into common EncoderFactoryAdapter |
| // interface. |
| // TODO(magjed): Remove once WebRtcVideoEncoderFactory is deprecated and |
| // webrtc:7925 is fixed. |
| class CricketEncoderFactoryAdapter : public EncoderFactoryAdapter { |
| public: |
| explicit CricketEncoderFactoryAdapter( |
| std::unique_ptr<WebRtcVideoEncoderFactory> external_encoder_factory) |
| : internal_encoder_factory_(new InternalEncoderFactory()), |
| external_encoder_factory_(std::move(external_encoder_factory)) {} |
| |
| private: |
| AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec) const override; |
| |
| std::vector<VideoCodec> GetSupportedCodecs() const override; |
| |
| const std::unique_ptr<WebRtcVideoEncoderFactory> internal_encoder_factory_; |
| const std::unique_ptr<WebRtcVideoEncoderFactory> external_encoder_factory_; |
| }; |
| |
| class CricketDecoderFactoryAdapter : public DecoderFactoryAdapter { |
| public: |
| explicit CricketDecoderFactoryAdapter( |
| std::unique_ptr<WebRtcVideoDecoderFactory> external_decoder_factory) |
| : internal_decoder_factory_(new InternalDecoderFactory()), |
| external_decoder_factory_(std::move(external_decoder_factory)) {} |
| |
| private: |
| std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder( |
| const VideoCodec& codec, |
| const VideoDecoderParams& decoder_params) const override; |
| |
| const std::unique_ptr<WebRtcVideoDecoderFactory> internal_decoder_factory_; |
| const std::unique_ptr<WebRtcVideoDecoderFactory> external_decoder_factory_; |
| }; |
| |
| // Wraps webrtc::VideoEncoderFactory into common EncoderFactoryAdapter |
| // interface. |
| class WebRtcEncoderFactoryAdapter : public EncoderFactoryAdapter { |
| public: |
| explicit WebRtcEncoderFactoryAdapter( |
| std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory) |
| : encoder_factory_(std::move(encoder_factory)) {} |
| |
| private: |
| AllocatedEncoder CreateVideoEncoder(const VideoCodec& codec) const override { |
| if (!encoder_factory_) |
| return AllocatedEncoder(); |
| const webrtc::SdpVideoFormat format(codec.name, codec.params); |
| const webrtc::VideoEncoderFactory::CodecInfo info = |
| encoder_factory_->QueryVideoEncoder(format); |
| return AllocatedEncoder(encoder_factory_->CreateVideoEncoder(format), |
| info.is_hardware_accelerated, |
| info.has_internal_source); |
| } |
| |
| std::vector<VideoCodec> GetSupportedCodecs() const override { |
| if (!encoder_factory_) |
| return std::vector<VideoCodec>(); |
| std::vector<VideoCodec> codecs; |
| for (const webrtc::SdpVideoFormat& format : |
| encoder_factory_->GetSupportedFormats()) { |
| codecs.push_back(VideoCodec(format)); |
| } |
| return AssignPayloadTypesAndAddAssociatedRtxCodecs(codecs); |
| } |
| |
| std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory_; |
| }; |
| |
| // Wraps webrtc::VideoDecoderFactory into common DecoderFactoryAdapter |
| // interface. |
| class WebRtcDecoderFactoryAdapter : public DecoderFactoryAdapter { |
| public: |
| explicit WebRtcDecoderFactoryAdapter( |
| std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory) |
| : decoder_factory_(std::move(decoder_factory)) {} |
| |
| private: |
| std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder( |
| const VideoCodec& codec, |
| const VideoDecoderParams& decoder_params) const override { |
| return decoder_factory_ |
| ? decoder_factory_->CreateVideoDecoder( |
| webrtc::SdpVideoFormat(codec.name, codec.params)) |
| : nullptr; |
| } |
| |
| std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory_; |
| }; |
| |
| // If this field trial is enabled, we will enable sending FlexFEC and disable |
| // sending ULPFEC whenever the former has been negotiated in the SDPs. |
| bool IsFlexfecFieldTrialEnabled() { |
| return webrtc::field_trial::IsEnabled("WebRTC-FlexFEC-03"); |
| } |
| |
| // If this field trial is enabled, the "flexfec-03" codec may have been |
| // advertised as being supported in the local SDP. That means that we must be |
| // ready to receive FlexFEC packets. See internalencoderfactory.cc. |
| bool IsFlexfecAdvertisedFieldTrialEnabled() { |
| return webrtc::field_trial::IsEnabled("WebRTC-FlexFEC-03-Advertised"); |
| } |
| |
| void AddDefaultFeedbackParams(VideoCodec* codec) { |
| // 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)); |
| } |
| |
| static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) { |
| std::stringstream out; |
| out << '{'; |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| out << codecs[i].ToString(); |
| if (i != codecs.size() - 1) { |
| out << ", "; |
| } |
| } |
| out << '}'; |
| return out.str(); |
| } |
| |
| static bool ValidateCodecFormats(const std::vector<VideoCodec>& codecs) { |
| bool has_video = false; |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| if (!codecs[i].ValidateCodecFormat()) { |
| return false; |
| } |
| if (codecs[i].GetCodecType() == VideoCodec::CODEC_VIDEO) { |
| has_video = true; |
| } |
| } |
| if (!has_video) { |
| LOG(LS_ERROR) << "Setting codecs without a video codec is invalid: " |
| << CodecVectorToString(codecs); |
| return false; |
| } |
| return true; |
| } |
| |
| static bool ValidateStreamParams(const StreamParams& sp) { |
| if (sp.ssrcs.empty()) { |
| LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString(); |
| return false; |
| } |
| |
| std::vector<uint32_t> primary_ssrcs; |
| sp.GetPrimarySsrcs(&primary_ssrcs); |
| std::vector<uint32_t> rtx_ssrcs; |
| sp.GetFidSsrcs(primary_ssrcs, &rtx_ssrcs); |
| for (uint32_t rtx_ssrc : rtx_ssrcs) { |
| bool rtx_ssrc_present = false; |
| for (uint32_t sp_ssrc : sp.ssrcs) { |
| if (sp_ssrc == rtx_ssrc) { |
| rtx_ssrc_present = true; |
| break; |
| } |
| } |
| if (!rtx_ssrc_present) { |
| LOG(LS_ERROR) << "RTX SSRC '" << rtx_ssrc |
| << "' missing from StreamParams ssrcs: " << sp.ToString(); |
| return false; |
| } |
| } |
| if (!rtx_ssrcs.empty() && primary_ssrcs.size() != rtx_ssrcs.size()) { |
| LOG(LS_ERROR) |
| << "RTX SSRCs exist, but don't cover all SSRCs (unsupported): " |
| << sp.ToString(); |
| return false; |
| } |
| |
| return true; |
| } |
| |
| // Returns true if the given codec is disallowed from doing simulcast. |
| bool IsCodecBlacklistedForSimulcast(const std::string& codec_name) { |
| return CodecNamesEq(codec_name, kH264CodecName) || |
| CodecNamesEq(codec_name, kVp9CodecName); |
| } |
| |
| // The selected thresholds for QVGA and VGA corresponded to a QP around 10. |
| // The change in QP declined above the selected bitrates. |
| static int GetMaxDefaultVideoBitrateKbps(int width, int height) { |
| if (width * height <= 320 * 240) { |
| return 600; |
| } else if (width * height <= 640 * 480) { |
| return 1700; |
| } else if (width * height <= 960 * 540) { |
| return 2000; |
| } else { |
| return 2500; |
| } |
| } |
| |
| bool GetVp9LayersFromFieldTrialGroup(int* num_spatial_layers, |
| int* num_temporal_layers) { |
| std::string group = webrtc::field_trial::FindFullName("WebRTC-SupportVP9SVC"); |
| if (group.empty()) |
| return false; |
| |
| if (sscanf(group.c_str(), "EnabledByFlag_%dSL%dTL", num_spatial_layers, |
| num_temporal_layers) != 2) { |
| return false; |
| } |
| const int kMaxSpatialLayers = 2; |
| if (*num_spatial_layers > kMaxSpatialLayers || *num_spatial_layers < 1) |
| return false; |
| |
| const int kMaxTemporalLayers = 3; |
| if (*num_temporal_layers > kMaxTemporalLayers || *num_temporal_layers < 1) |
| return false; |
| |
| return true; |
| } |
| |
| int GetDefaultVp9SpatialLayers() { |
| int num_sl; |
| int num_tl; |
| if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) { |
| return num_sl; |
| } |
| return 1; |
| } |
| |
| int GetDefaultVp9TemporalLayers() { |
| int num_sl; |
| int num_tl; |
| if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) { |
| return num_tl; |
| } |
| return 1; |
| } |
| } // namespace |
| |
| // Constants defined in webrtc/media/engine/constants.h |
| // TODO(pbos): Move these to a separate constants.cc file. |
| const int kMinVideoBitrateKbps = 30; |
| |
| const int kVideoMtu = 1200; |
| const int kVideoRtpBufferSize = 65536; |
| |
| // This constant is really an on/off, lower-level configurable NACK history |
| // duration hasn't been implemented. |
| static const int kNackHistoryMs = 1000; |
| |
| static const int kDefaultRtcpReceiverReportSsrc = 1; |
| |
| // Minimum time interval for logging stats. |
| static const int64_t kStatsLogIntervalMs = 10000; |
| |
| 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. |
| bool automatic_resize = |
| !is_screencast && parameters_.config.rtp.ssrcs.size() == 1; |
| bool frame_dropping = !is_screencast; |
| bool denoising; |
| bool codec_default_denoising = false; |
| if (is_screencast) { |
| denoising = false; |
| } else { |
| // Use codec default if video_noise_reduction is unset. |
| codec_default_denoising = !parameters_.options.video_noise_reduction; |
| denoising = parameters_.options.video_noise_reduction.value_or(false); |
| } |
| |
| if (CodecNamesEq(codec.name, kH264CodecName)) { |
| webrtc::VideoCodecH264 h264_settings = |
| webrtc::VideoEncoder::GetDefaultH264Settings(); |
| h264_settings.frameDroppingOn = frame_dropping; |
| return new rtc::RefCountedObject< |
| webrtc::VideoEncoderConfig::H264EncoderSpecificSettings>(h264_settings); |
| } |
| if (CodecNamesEq(codec.name, kVp8CodecName)) { |
| webrtc::VideoCodecVP8 vp8_settings = |
| webrtc::VideoEncoder::GetDefaultVp8Settings(); |
| vp8_settings.automaticResizeOn = automatic_resize; |
| // VP8 denoising is enabled by default. |
| vp8_settings.denoisingOn = codec_default_denoising ? true : denoising; |
| vp8_settings.frameDroppingOn = frame_dropping; |
| return new rtc::RefCountedObject< |
| webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings); |
| } |
| if (CodecNamesEq(codec.name, kVp9CodecName)) { |
| webrtc::VideoCodecVP9 vp9_settings = |
| webrtc::VideoEncoder::GetDefaultVp9Settings(); |
| if (is_screencast) { |
| // TODO(asapersson): Set to 2 for now since there is a DCHECK in |
| // VideoSendStream::ReconfigureVideoEncoder. |
| vp9_settings.numberOfSpatialLayers = 2; |
| } else { |
| vp9_settings.numberOfSpatialLayers = GetDefaultVp9SpatialLayers(); |
| } |
| // VP9 denoising is disabled by default. |
| vp9_settings.denoisingOn = codec_default_denoising ? true : denoising; |
| vp9_settings.frameDroppingOn = frame_dropping; |
| vp9_settings.automaticResizeOn = automatic_resize; |
| 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) { |
| rtc::Optional<uint32_t> default_recv_ssrc = |
| channel->GetDefaultReceiveStreamSsrc(); |
| |
| if (default_recv_ssrc) { |
| LOG(LS_INFO) << "Destroying old default receive stream for SSRC=" << ssrc |
| << "."; |
| channel->RemoveRecvStream(*default_recv_ssrc); |
| } |
| |
| StreamParams sp; |
| sp.ssrcs.push_back(ssrc); |
| LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << "."; |
| if (!channel->AddRecvStream(sp, true)) { |
| LOG(LS_WARNING) << "Could not create default receive stream."; |
| } |
| |
| channel->SetSink(ssrc, default_sink_); |
| 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; |
| rtc::Optional<uint32_t> default_recv_ssrc = |
| channel->GetDefaultReceiveStreamSsrc(); |
| if (default_recv_ssrc) { |
| channel->SetSink(*default_recv_ssrc, default_sink_); |
| } |
| } |
| |
| WebRtcVideoEngine::WebRtcVideoEngine( |
| std::unique_ptr<WebRtcVideoEncoderFactory> external_video_encoder_factory, |
| std::unique_ptr<WebRtcVideoDecoderFactory> external_video_decoder_factory) |
| : decoder_factory_(new CricketDecoderFactoryAdapter( |
| std::move(external_video_decoder_factory))), |
| encoder_factory_(new CricketEncoderFactoryAdapter( |
| std::move(external_video_encoder_factory))) { |
| LOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine()"; |
| } |
| |
| WebRtcVideoEngine::WebRtcVideoEngine( |
| std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory, |
| std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory) |
| : decoder_factory_( |
| new WebRtcDecoderFactoryAdapter(std::move(video_decoder_factory))), |
| encoder_factory_( |
| new WebRtcEncoderFactoryAdapter(std::move(video_encoder_factory))) { |
| LOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine()"; |
| } |
| |
| WebRtcVideoEngine::~WebRtcVideoEngine() { |
| LOG(LS_INFO) << "WebRtcVideoEngine::~WebRtcVideoEngine"; |
| } |
| |
| WebRtcVideoChannel* WebRtcVideoEngine::CreateChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options) { |
| LOG(LS_INFO) << "CreateChannel. Options: " << options.ToString(); |
| return new WebRtcVideoChannel(call, config, options, encoder_factory_.get(), |
| decoder_factory_.get()); |
| } |
| |
| std::vector<VideoCodec> WebRtcVideoEngine::codecs() const { |
| return encoder_factory_->GetSupportedCodecs(); |
| } |
| |
| RtpCapabilities WebRtcVideoEngine::GetCapabilities() const { |
| RtpCapabilities capabilities; |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kTimestampOffsetUri, |
| webrtc::RtpExtension::kTimestampOffsetDefaultId)); |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri, |
| webrtc::RtpExtension::kAbsSendTimeDefaultId)); |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kVideoRotationUri, |
| webrtc::RtpExtension::kVideoRotationDefaultId)); |
| capabilities.header_extensions.push_back(webrtc::RtpExtension( |
| webrtc::RtpExtension::kTransportSequenceNumberUri, |
| webrtc::RtpExtension::kTransportSequenceNumberDefaultId)); |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kPlayoutDelayUri, |
| webrtc::RtpExtension::kPlayoutDelayDefaultId)); |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kVideoContentTypeUri, |
| webrtc::RtpExtension::kVideoContentTypeDefaultId)); |
| capabilities.header_extensions.push_back( |
| webrtc::RtpExtension(webrtc::RtpExtension::kVideoTimingUri, |
| webrtc::RtpExtension::kVideoTimingDefaultId)); |
| return capabilities; |
| } |
| |
| namespace { |
| // This function will assign dynamic payload types (in the range [96, 127]) to |
| // the input codecs, and also add associated RTX codecs for recognized codecs |
| // (VP8, VP9, H264, and RED). It will also add default feedback params to the |
| // codecs. |
| std::vector<VideoCodec> AssignPayloadTypesAndAddAssociatedRtxCodecs( |
| const std::vector<VideoCodec>& input_codecs) { |
| static const int kFirstDynamicPayloadType = 96; |
| static const int kLastDynamicPayloadType = 127; |
| int payload_type = kFirstDynamicPayloadType; |
| std::vector<VideoCodec> output_codecs; |
| for (VideoCodec codec : input_codecs) { |
| codec.id = payload_type; |
| AddDefaultFeedbackParams(&codec); |
| output_codecs.push_back(codec); |
| |
| // Increment payload type. |
| ++payload_type; |
| if (payload_type > kLastDynamicPayloadType) |
| break; |
| |
| // Add associated RTX codec for recognized codecs. |
| // TODO(deadbeef): Should we add RTX codecs for external codecs whose names |
| // we don't recognize? |
| if (CodecNamesEq(codec.name, kVp8CodecName) || |
| CodecNamesEq(codec.name, kVp9CodecName) || |
| CodecNamesEq(codec.name, kH264CodecName) || |
| CodecNamesEq(codec.name, kRedCodecName)) { |
| output_codecs.push_back( |
| VideoCodec::CreateRtxCodec(payload_type, codec.id)); |
| |
| // Increment payload type. |
| ++payload_type; |
| if (payload_type > kLastDynamicPayloadType) |
| break; |
| } |
| } |
| return output_codecs; |
| } |
| } // namespace |
| |
| std::vector<VideoCodec> CricketEncoderFactoryAdapter::GetSupportedCodecs() |
| const { |
| std::vector<VideoCodec> codecs = InternalEncoderFactory().supported_codecs(); |
| LOG(LS_INFO) << "Internally supported codecs: " |
| << CodecVectorToString(codecs); |
| |
| // Add external codecs. |
| if (external_encoder_factory_ != nullptr) { |
| const std::vector<VideoCodec>& external_codecs = |
| external_encoder_factory_->supported_codecs(); |
| for (const VideoCodec& codec : external_codecs) { |
| // Don't add same codec twice. |
| if (!FindMatchingCodec(codecs, codec)) |
| codecs.push_back(codec); |
| } |
| LOG(LS_INFO) << "Codecs supported by the external encoder factory: " |
| << CodecVectorToString(external_codecs); |
| } |
| |
| return AssignPayloadTypesAndAddAssociatedRtxCodecs(codecs); |
| } |
| |
| WebRtcVideoChannel::WebRtcVideoChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options, |
| const EncoderFactoryAdapter* encoder_factory, |
| const DecoderFactoryAdapter* decoder_factory) |
| : VideoMediaChannel(config), |
| call_(call), |
| unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_), |
| video_config_(config.video), |
| encoder_factory_(encoder_factory), |
| decoder_factory_(decoder_factory), |
| default_send_options_(options), |
| last_stats_log_ms_(-1) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| |
| rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc; |
| sending_ = false; |
| recv_codecs_ = MapCodecs(encoder_factory_->GetSupportedCodecs()); |
| recv_flexfec_payload_type_ = recv_codecs_.front().flexfec_payload_type; |
| } |
| |
| WebRtcVideoChannel::~WebRtcVideoChannel() { |
| for (auto& kv : send_streams_) |
| delete kv.second; |
| for (auto& kv : receive_streams_) |
| delete kv.second; |
| } |
| |
| rtc::Optional<WebRtcVideoChannel::VideoCodecSettings> |
| WebRtcVideoChannel::SelectSendVideoCodec( |
| const std::vector<VideoCodecSettings>& remote_mapped_codecs) const { |
| const std::vector<VideoCodec> local_supported_codecs = |
| encoder_factory_->GetSupportedCodecs(); |
| // Select the first remote codec that is supported locally. |
| for (const VideoCodecSettings& remote_mapped_codec : remote_mapped_codecs) { |
| // 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 (FindMatchingCodec(local_supported_codecs, remote_mapped_codec.codec)) |
| return rtc::Optional<VideoCodecSettings>(remote_mapped_codec); |
| } |
| // No remote codec was supported. |
| return rtc::Optional<VideoCodecSettings>(); |
| } |
| |
| bool WebRtcVideoChannel::NonFlexfecReceiveCodecsHaveChanged( |
| std::vector<VideoCodecSettings> before, |
| std::vector<VideoCodecSettings> after) { |
| if (before.size() != after.size()) { |
| return true; |
| } |
| |
| // The receive codec order doesn't matter, so we sort the codecs before |
| // comparing. This is necessary because currently the |
| // only way to change the send codec is to munge SDP, which causes |
| // the receive codec list to change order, which causes the streams |
| // to be recreates which causes a "blink" of black video. In order |
| // to support munging the SDP in this way without recreating receive |
| // streams, we ignore the order of the received codecs so that |
| // changing the order doesn't cause this "blink". |
| auto comparison = |
| [](const VideoCodecSettings& codec1, const VideoCodecSettings& codec2) { |
| return codec1.codec.id > codec2.codec.id; |
| }; |
| std::sort(before.begin(), before.end(), comparison); |
| std::sort(after.begin(), after.end(), comparison); |
| |
| // Changes in FlexFEC payload type are handled separately in |
| // WebRtcVideoChannel::GetChangedRecvParameters, so disregard FlexFEC in the |
| // comparison here. |
| return !std::equal(before.begin(), before.end(), after.begin(), |
| VideoCodecSettings::EqualsDisregardingFlexfec); |
| } |
| |
| bool WebRtcVideoChannel::GetChangedSendParameters( |
| const VideoSendParameters& params, |
| ChangedSendParameters* changed_params) const { |
| if (!ValidateCodecFormats(params.codecs) || |
| !ValidateRtpExtensions(params.extensions)) { |
| return false; |
| } |
| |
| // Select one of the remote codecs that will be used as send codec. |
| rtc::Optional<VideoCodecSettings> selected_send_codec = |
| SelectSendVideoCodec(MapCodecs(params.codecs)); |
| |
| if (!selected_send_codec) { |
| LOG(LS_ERROR) << "No video codecs supported."; |
| return false; |
| } |
| |
| // Never enable sending FlexFEC, unless we are in the experiment. |
| if (!IsFlexfecFieldTrialEnabled()) { |
| if (selected_send_codec->flexfec_payload_type != -1) { |
| LOG(LS_INFO) << "Remote supports flexfec-03, but we will not send since " |
| << "WebRTC-FlexFEC-03 field trial is not enabled."; |
| } |
| selected_send_codec->flexfec_payload_type = -1; |
| } |
| |
| if (!send_codec_ || *selected_send_codec != *send_codec_) |
| changed_params->codec = selected_send_codec; |
| |
| // Handle RTP header extensions. |
| std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( |
| params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true); |
| if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) { |
| changed_params->rtp_header_extensions = |
| rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions); |
| } |
| |
| // Handle max bitrate. |
| if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps && |
| params.max_bandwidth_bps >= -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 = rtc::Optional<int>( |
| params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps); |
| } |
| |
| // Handle conference mode. |
| if (params.conference_mode != send_params_.conference_mode) { |
| changed_params->conference_mode = |
| rtc::Optional<bool>(params.conference_mode); |
| } |
| |
| // Handle RTCP mode. |
| if (params.rtcp.reduced_size != send_params_.rtcp.reduced_size) { |
| changed_params->rtcp_mode = rtc::Optional<webrtc::RtcpMode>( |
| params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound); |
| } |
| |
| return true; |
| } |
| |
| rtc::DiffServCodePoint WebRtcVideoChannel::PreferredDscp() const { |
| return rtc::DSCP_AF41; |
| } |
| |
| bool WebRtcVideoChannel::SetSendParameters(const VideoSendParameters& params) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSendParameters"); |
| LOG(LS_INFO) << "SetSendParameters: " << params.ToString(); |
| ChangedSendParameters changed_params; |
| if (!GetChangedSendParameters(params, &changed_params)) { |
| return false; |
| } |
| |
| if (changed_params.codec) { |
| const VideoCodecSettings& codec_settings = *changed_params.codec; |
| send_codec_ = rtc::Optional<VideoCodecSettings>(codec_settings); |
| LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString(); |
| } |
| |
| if (changed_params.rtp_header_extensions) { |
| send_rtp_extensions_ = changed_params.rtp_header_extensions; |
| } |
| |
| if (changed_params.codec || changed_params.max_bandwidth_bps) { |
| if (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.codec) { |
| // If the codec isn't changing, set the start bitrate to -1 which means |
| // "unchanged" so that BWE isn't affected. |
| bitrate_config_.start_bitrate_bps = -1; |
| } |
| } |
| if (params.max_bandwidth_bps >= 0) { |
| // Note that max_bandwidth_bps intentionally takes priority over the |
| // bitrate config for the codec. This allows FEC to be applied above the |
| // codec target bitrate. |
| // TODO(pbos): Figure out whether b=AS means max bitrate for this |
| // WebRtcVideoChannel (in which case we're good), or per sender (SSRC), |
| // in which case this should not set a Call::BitrateConfig but rather |
| // reconfigure all senders. |
| bitrate_config_.max_bitrate_bps = |
| params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps; |
| } |
| call_->SetBitrateConfig(bitrate_config_); |
| } |
| |
| { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (auto& kv : send_streams_) { |
| kv.second->SetSendParameters(changed_params); |
| } |
| if (changed_params.codec || changed_params.rtcp_mode) { |
| // Update receive feedback parameters from new codec or RTCP mode. |
| LOG(LS_INFO) |
| << "SetFeedbackOptions on all the receive streams because the send " |
| "codec or RTCP mode has changed."; |
| for (auto& kv : receive_streams_) { |
| RTC_DCHECK(kv.second != nullptr); |
| kv.second->SetFeedbackParameters( |
| HasNack(send_codec_->codec), HasRemb(send_codec_->codec), |
| HasTransportCc(send_codec_->codec), |
| params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound); |
| } |
| } |
| } |
| send_params_ = params; |
| return true; |
| } |
| |
| webrtc::RtpParameters WebRtcVideoChannel::GetRtpSendParameters( |
| uint32_t ssrc) const { |
| rtc::CritScope stream_lock(&stream_crit_); |
| auto it = send_streams_.find(ssrc); |
| if (it == send_streams_.end()) { |
| LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream " |
| << "with ssrc " << ssrc << " which doesn't exist."; |
| return webrtc::RtpParameters(); |
| } |
| |
| webrtc::RtpParameters rtp_params = it->second->GetRtpParameters(); |
| // Need to add the common list of codecs to the send stream-specific |
| // RTP parameters. |
| for (const VideoCodec& codec : send_params_.codecs) { |
| rtp_params.codecs.push_back(codec.ToCodecParameters()); |
| } |
| return rtp_params; |
| } |
| |
| bool WebRtcVideoChannel::SetRtpSendParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetRtpSendParameters"); |
| rtc::CritScope stream_lock(&stream_crit_); |
| auto it = send_streams_.find(ssrc); |
| if (it == send_streams_.end()) { |
| LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream " |
| << "with ssrc " << ssrc << " which doesn't exist."; |
| return false; |
| } |
| |
| // TODO(deadbeef): Handle setting parameters with a list of codecs in a |
| // different order (which should change the send codec). |
| webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc); |
| if (current_parameters.codecs != parameters.codecs) { |
| LOG(LS_ERROR) << "Using SetParameters to change the set of codecs " |
| << "is not currently supported."; |
| return false; |
| } |
| |
| return it->second->SetRtpParameters(parameters); |
| } |
| |
| webrtc::RtpParameters WebRtcVideoChannel::GetRtpReceiveParameters( |
| uint32_t ssrc) const { |
| webrtc::RtpParameters rtp_params; |
| rtc::CritScope stream_lock(&stream_crit_); |
| // SSRC of 0 represents an unsignaled receive stream. |
| if (ssrc == 0) { |
| if (!default_unsignalled_ssrc_handler_.GetDefaultSink()) { |
| 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(); |
| } else { |
| auto it = receive_streams_.find(ssrc); |
| if (it == receive_streams_.end()) { |
| LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream " |
| << "with SSRC " << ssrc << " which doesn't exist."; |
| return webrtc::RtpParameters(); |
| } |
| // TODO(deadbeef): Return stream-specific parameters, beyond just SSRC. |
| rtp_params.encodings.emplace_back(); |
| rtp_params.encodings[0].ssrc = it->second->GetFirstPrimarySsrc(); |
| } |
| |
| // 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::SetRtpReceiveParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetRtpReceiveParameters"); |
| rtc::CritScope stream_lock(&stream_crit_); |
| |
| // SSRC of 0 represents an unsignaled receive stream. |
| if (ssrc == 0) { |
| if (!default_unsignalled_ssrc_handler_.GetDefaultSink()) { |
| LOG(LS_WARNING) << "Attempting to set RTP parameters for the default, " |
| "unsignaled video receive stream, but not yet " |
| "configured to receive such a stream."; |
| return false; |
| } |
| } else { |
| auto it = receive_streams_.find(ssrc); |
| if (it == receive_streams_.end()) { |
| LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream " |
| << "with SSRC " << ssrc << " which doesn't exist."; |
| return false; |
| } |
| } |
| |
| webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc); |
| if (current_parameters != parameters) { |
| LOG(LS_ERROR) << "Changing the RTP receive parameters is currently " |
| << "unsupported."; |
| return false; |
| } |
| return true; |
| } |
| |
| bool 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()) { |
| LOG(LS_ERROR) << "SetRecvParameters called without any video codecs."; |
| return false; |
| } |
| |
| // Verify that every mapped codec is supported locally. |
| const std::vector<VideoCodec> local_supported_codecs = |
| encoder_factory_->GetSupportedCodecs(); |
| for (const VideoCodecSettings& mapped_codec : mapped_codecs) { |
| if (!FindMatchingCodec(local_supported_codecs, mapped_codec.codec)) { |
| LOG(LS_ERROR) << "SetRecvParameters called with unsupported video codec: " |
| << mapped_codec.codec.ToString(); |
| return false; |
| } |
| } |
| |
| if (NonFlexfecReceiveCodecsHaveChanged(recv_codecs_, mapped_codecs)) { |
| changed_params->codec_settings = |
| rtc::Optional<std::vector<VideoCodecSettings>>(mapped_codecs); |
| } |
| |
| // Handle RTP header extensions. |
| std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions( |
| params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false); |
| if (filtered_extensions != recv_rtp_extensions_) { |
| changed_params->rtp_header_extensions = |
| rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions); |
| } |
| |
| int flexfec_payload_type = mapped_codecs.front().flexfec_payload_type; |
| if (flexfec_payload_type != recv_flexfec_payload_type_) { |
| changed_params->flexfec_payload_type = |
| rtc::Optional<int>(flexfec_payload_type); |
| } |
| |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::SetRecvParameters(const VideoRecvParameters& params) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetRecvParameters"); |
| LOG(LS_INFO) << "SetRecvParameters: " << params.ToString(); |
| ChangedRecvParameters changed_params; |
| if (!GetChangedRecvParameters(params, &changed_params)) { |
| return false; |
| } |
| if (changed_params.flexfec_payload_type) { |
| LOG(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) { |
| LOG(LS_INFO) << "Changing recv codecs from " |
| << CodecSettingsVectorToString(recv_codecs_) << " to " |
| << CodecSettingsVectorToString(*changed_params.codec_settings); |
| recv_codecs_ = *changed_params.codec_settings; |
| } |
| |
| { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (auto& kv : receive_streams_) { |
| kv.second->SetRecvParameters(changed_params); |
| } |
| } |
| recv_params_ = params; |
| return true; |
| } |
| |
| std::string WebRtcVideoChannel::CodecSettingsVectorToString( |
| const std::vector<VideoCodecSettings>& codecs) { |
| std::stringstream out; |
| out << '{'; |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| out << codecs[i].codec.ToString(); |
| if (i != codecs.size() - 1) { |
| out << ", "; |
| } |
| } |
| out << '}'; |
| return out.str(); |
| } |
| |
| bool WebRtcVideoChannel::GetSendCodec(VideoCodec* codec) { |
| if (!send_codec_) { |
| LOG(LS_VERBOSE) << "GetSendCodec: No send codec set."; |
| return false; |
| } |
| *codec = send_codec_->codec; |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::SetSend(bool send) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSend"); |
| LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false"); |
| if (send && !send_codec_) { |
| LOG(LS_ERROR) << "SetSend(true) called before setting codec."; |
| return false; |
| } |
| { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (const auto& kv : send_streams_) { |
| kv.second->SetSend(send); |
| } |
| } |
| sending_ = send; |
| return true; |
| } |
| |
| // TODO(nisse): The enable argument was used for mute logic which has |
| // been moved to VideoBroadcaster. So remove the argument from this |
| // method. |
| bool WebRtcVideoChannel::SetVideoSend( |
| uint32_t ssrc, |
| bool enable, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source) { |
| TRACE_EVENT0("webrtc", "SetVideoSend"); |
| RTC_DCHECK(ssrc != 0); |
| LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", enable = " << enable |
| << ", options: " << (options ? options->ToString() : "nullptr") |
| << ", source = " << (source ? "(source)" : "nullptr") << ")"; |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| const auto& kv = send_streams_.find(ssrc); |
| if (kv == send_streams_.end()) { |
| // Allow unknown ssrc only if source is null. |
| RTC_CHECK(source == nullptr); |
| LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; |
| return false; |
| } |
| |
| return kv->second->SetVideoSend(enable, options, source); |
| } |
| |
| bool WebRtcVideoChannel::ValidateSendSsrcAvailability( |
| const StreamParams& sp) const { |
| for (uint32_t ssrc : sp.ssrcs) { |
| if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) { |
| LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists."; |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::ValidateReceiveSsrcAvailability( |
| const StreamParams& sp) const { |
| for (uint32_t ssrc : sp.ssrcs) { |
| if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) { |
| LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc |
| << "' already exists."; |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::AddSendStream(const StreamParams& sp) { |
| LOG(LS_INFO) << "AddSendStream: " << sp.ToString(); |
| if (!ValidateStreamParams(sp)) |
| return false; |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| |
| if (!ValidateSendSsrcAvailability(sp)) |
| return false; |
| |
| for (uint32_t used_ssrc : sp.ssrcs) |
| send_ssrcs_.insert(used_ssrc); |
| |
| webrtc::VideoSendStream::Config config(this); |
| config.suspend_below_min_bitrate = video_config_.suspend_below_min_bitrate; |
| config.periodic_alr_bandwidth_probing = |
| video_config_.periodic_alr_bandwidth_probing; |
| WebRtcVideoSendStream* stream = new WebRtcVideoSendStream( |
| call_, sp, std::move(config), default_send_options_, encoder_factory_, |
| video_config_.enable_cpu_overuse_detection, |
| bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_, |
| send_params_); |
| |
| uint32_t ssrc = sp.first_ssrc(); |
| RTC_DCHECK(ssrc != 0); |
| send_streams_[ssrc] = stream; |
| |
| if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) { |
| rtcp_receiver_report_ssrc_ = ssrc; |
| LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added " |
| "a send stream."; |
| for (auto& kv : receive_streams_) |
| kv.second->SetLocalSsrc(ssrc); |
| } |
| if (sending_) { |
| stream->SetSend(true); |
| } |
| |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::RemoveSendStream(uint32_t ssrc) { |
| LOG(LS_INFO) << "RemoveSendStream: " << ssrc; |
| |
| WebRtcVideoSendStream* removed_stream; |
| { |
| rtc::CritScope stream_lock(&stream_crit_); |
| std::map<uint32_t, WebRtcVideoSendStream*>::iterator it = |
| send_streams_.find(ssrc); |
| if (it == send_streams_.end()) { |
| return false; |
| } |
| |
| for (uint32_t old_ssrc : it->second->GetSsrcs()) |
| send_ssrcs_.erase(old_ssrc); |
| |
| removed_stream = it->second; |
| send_streams_.erase(it); |
| |
| // Switch receiver report SSRCs, the one in use is no longer valid. |
| if (rtcp_receiver_report_ssrc_ == ssrc) { |
| rtcp_receiver_report_ssrc_ = send_streams_.empty() |
| ? kDefaultRtcpReceiverReportSsrc |
| : send_streams_.begin()->first; |
| LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because the " |
| "previous local SSRC was removed."; |
| |
| for (auto& kv : receive_streams_) { |
| kv.second->SetLocalSsrc(rtcp_receiver_report_ssrc_); |
| } |
| } |
| } |
| |
| delete removed_stream; |
| |
| return true; |
| } |
| |
| void 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(thread_checker_.CalledOnValidThread()); |
| |
| LOG(LS_INFO) << "AddRecvStream" << (default_stream ? " (default stream)" : "") |
| << ": " << sp.ToString(); |
| if (!ValidateStreamParams(sp)) |
| return false; |
| |
| uint32_t ssrc = sp.first_ssrc(); |
| RTC_DCHECK(ssrc != 0); // TODO(pbos): Is this ever valid? |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| // Remove running stream if this was a default stream. |
| const auto& prev_stream = receive_streams_.find(ssrc); |
| if (prev_stream != receive_streams_.end()) { |
| if (default_stream || !prev_stream->second->IsDefaultStream()) { |
| LOG(LS_ERROR) << "Receive stream for SSRC '" << ssrc |
| << "' already exists."; |
| return false; |
| } |
| DeleteReceiveStream(prev_stream->second); |
| receive_streams_.erase(prev_stream); |
| } |
| |
| if (!ValidateReceiveSsrcAvailability(sp)) |
| return false; |
| |
| for (uint32_t used_ssrc : sp.ssrcs) |
| receive_ssrcs_.insert(used_ssrc); |
| |
| webrtc::VideoReceiveStream::Config config(this); |
| webrtc::FlexfecReceiveStream::Config flexfec_config(this); |
| ConfigureReceiverRtp(&config, &flexfec_config, sp); |
| |
| config.disable_prerenderer_smoothing = |
| video_config_.disable_prerenderer_smoothing; |
| config.sync_group = sp.sync_label; |
| |
| receive_streams_[ssrc] = new WebRtcVideoReceiveStream( |
| 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; |
| |
| config->rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false; |
| 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 (IsFlexfecAdvertisedFieldTrialEnabled() && |
| 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) { |
| LOG(LS_INFO) << "RemoveRecvStream: " << ssrc; |
| if (ssrc == 0) { |
| LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported."; |
| return false; |
| } |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator stream = |
| receive_streams_.find(ssrc); |
| if (stream == receive_streams_.end()) { |
| LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc; |
| return false; |
| } |
| DeleteReceiveStream(stream->second); |
| receive_streams_.erase(stream); |
| |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::SetSink( |
| uint32_t ssrc, |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) { |
| LOG(LS_INFO) << "SetSink: ssrc:" << ssrc << " " |
| << (sink ? "(ptr)" : "nullptr"); |
| if (ssrc == 0) { |
| // Do not hold |stream_crit_| here, since SetDefaultSink will call |
| // WebRtcVideoChannel::GetDefaultReceiveStreamSsrc(). |
| default_unsignalled_ssrc_handler_.SetDefaultSink(this, sink); |
| return true; |
| } |
| |
| rtc::CritScope stream_lock(&stream_crit_); |
| std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it = |
| receive_streams_.find(ssrc); |
| if (it == receive_streams_.end()) { |
| return false; |
| } |
| |
| it->second->SetSink(sink); |
| return true; |
| } |
| |
| bool WebRtcVideoChannel::GetStats(VideoMediaInfo* info) { |
| 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; |
| } |
| } |
| |
| if (log_stats) |
| LOG(LS_INFO) << stats.ToString(now_ms); |
| |
| return true; |
| } |
| |
| void WebRtcVideoChannel::FillSenderStats(VideoMediaInfo* video_media_info, |
| bool log_stats) { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it = |
| send_streams_.begin(); |
| it != send_streams_.end(); ++it) { |
| video_media_info->senders.push_back( |
| it->second->GetVideoSenderInfo(log_stats)); |
| } |
| } |
| |
| void WebRtcVideoChannel::FillReceiverStats(VideoMediaInfo* video_media_info, |
| bool log_stats) { |
| rtc::CritScope stream_lock(&stream_crit_); |
| for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it = |
| receive_streams_.begin(); |
| it != receive_streams_.end(); ++it) { |
| video_media_info->receivers.push_back( |
| it->second->GetVideoReceiverInfo(log_stats)); |
| } |
| } |
| |
| void WebRtcVideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) { |
| rtc::CritScope stream_lock(&stream_crit_); |
| 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, |
| const rtc::PacketTime& packet_time) { |
| const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, |
| packet_time.not_before); |
| const webrtc::PacketReceiver::DeliveryStatus delivery_result = |
| call_->Receiver()->DeliverPacket( |
| webrtc::MediaType::VIDEO, |
| packet->cdata(), packet->size(), |
| webrtc_packet_time); |
| switch (delivery_result) { |
| case webrtc::PacketReceiver::DELIVERY_OK: |
| return; |
| case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR: |
| return; |
| case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC: |
| break; |
| } |
| |
| uint32_t ssrc = 0; |
| if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) { |
| return; |
| } |
| |
| int payload_type = 0; |
| if (!GetRtpPayloadType(packet->cdata(), packet->size(), &payload_type)) { |
| return; |
| } |
| |
| // See if this payload_type is registered as one that usually gets its own |
| // SSRC (RTX) or at least is safe to drop either way (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; |
| } |
| |
| switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) { |
| case UnsignalledSsrcHandler::kDropPacket: |
| return; |
| case UnsignalledSsrcHandler::kDeliverPacket: |
| break; |
| } |
| |
| if (call_->Receiver()->DeliverPacket( |
| webrtc::MediaType::VIDEO, |
| packet->cdata(), packet->size(), |
| webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) { |
| LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery."; |
| return; |
| } |
| } |
| |
| void WebRtcVideoChannel::OnRtcpReceived( |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time) { |
| const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, |
| packet_time.not_before); |
| // TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver |
| // for both audio and video on the same path. Since BundleFilter doesn't |
| // filter RTCP anymore incoming RTCP packets could've been going to audio (so |
| // logging failures spam the log). |
| call_->Receiver()->DeliverPacket( |
| webrtc::MediaType::VIDEO, |
| packet->cdata(), packet->size(), |
| webrtc_packet_time); |
| } |
| |
| void WebRtcVideoChannel::OnReadyToSend(bool ready) { |
| 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) { |
| call_->OnNetworkRouteChanged(transport_name, network_route); |
| } |
| |
| void WebRtcVideoChannel::OnTransportOverheadChanged( |
| int transport_overhead_per_packet) { |
| call_->OnTransportOverheadChanged(webrtc::MediaType::VIDEO, |
| transport_overhead_per_packet); |
| } |
| |
| void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) { |
| MediaChannel::SetInterface(iface); |
| // Set the RTP recv/send buffer to a bigger size |
| MediaChannel::SetOption(NetworkInterface::ST_RTP, |
| rtc::Socket::OPT_RCVBUF, |
| kVideoRtpBufferSize); |
| |
| // Speculative change to increase the outbound socket buffer size. |
| // In b/15152257, we are seeing a significant number of packets discarded |
| // due to lack of socket buffer space, although it's not yet clear what the |
| // ideal value should be. |
| MediaChannel::SetOption(NetworkInterface::ST_RTP, |
| rtc::Socket::OPT_SNDBUF, |
| kVideoRtpBufferSize); |
| } |
| |
| rtc::Optional<uint32_t> WebRtcVideoChannel::GetDefaultReceiveStreamSsrc() { |
| rtc::CritScope stream_lock(&stream_crit_); |
| rtc::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; |
| } |
| |
| 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; |
| return MediaChannel::SendPacket(&packet, rtc_options); |
| } |
| |
| bool WebRtcVideoChannel::SendRtcp(const uint8_t* data, size_t len) { |
| rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen); |
| return MediaChannel::SendRtcp(&packet, rtc::PacketOptions()); |
| } |
| |
| WebRtcVideoChannel::WebRtcVideoSendStream::VideoSendStreamParameters:: |
| VideoSendStreamParameters( |
| webrtc::VideoSendStream::Config config, |
| const VideoOptions& options, |
| int max_bitrate_bps, |
| const rtc::Optional<VideoCodecSettings>& codec_settings) |
| : config(std::move(config)), |
| options(options), |
| max_bitrate_bps(max_bitrate_bps), |
| conference_mode(false), |
| codec_settings(codec_settings) {} |
| |
| EncoderFactoryAdapter::AllocatedEncoder::AllocatedEncoder( |
| std::unique_ptr<webrtc::VideoEncoder> encoder, |
| bool is_hardware_accelerated, |
| bool has_internal_source) |
| : encoder(std::move(encoder)), |
| is_hardware_accelerated(is_hardware_accelerated), |
| has_internal_source(has_internal_source) {} |
| |
| WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream( |
| webrtc::Call* call, |
| const StreamParams& sp, |
| webrtc::VideoSendStream::Config config, |
| const VideoOptions& options, |
| const EncoderFactoryAdapter* encoder_factory, |
| bool enable_cpu_overuse_detection, |
| int max_bitrate_bps, |
| const rtc::Optional<VideoCodecSettings>& codec_settings, |
| const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions, |
| // TODO(deadbeef): Don't duplicate information between send_params, |
| // rtp_extensions, options, etc. |
| const VideoSendParameters& send_params) |
| : worker_thread_(rtc::Thread::Current()), |
| ssrcs_(sp.ssrcs), |
| ssrc_groups_(sp.ssrc_groups), |
| call_(call), |
| enable_cpu_overuse_detection_(enable_cpu_overuse_detection), |
| source_(nullptr), |
| encoder_factory_(encoder_factory), |
| stream_(nullptr), |
| encoder_sink_(nullptr), |
| parameters_(std::move(config), options, max_bitrate_bps, codec_settings), |
| rtp_parameters_(CreateRtpParametersWithOneEncoding()), |
| sending_(false) { |
| 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 = |
| rtc::Optional<uint32_t>(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 (IsFlexfecFieldTrialEnabled()) { |
| 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) { |
| 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; |
| parameters_.config.track_id = sp.id; |
| if (rtp_extensions) { |
| parameters_.config.rtp.extensions = *rtp_extensions; |
| } |
| parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size |
| ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound; |
| if (codec_settings) { |
| bool force_encoder_allocation = false; |
| SetCodec(*codec_settings, force_encoder_allocation); |
| } |
| } |
| |
| WebRtcVideoChannel::WebRtcVideoSendStream::~WebRtcVideoSendStream() { |
| if (stream_ != NULL) { |
| call_->DestroyVideoSendStream(stream_); |
| } |
| // Release |allocated_encoder_|. |
| allocated_encoder_.reset(); |
| } |
| |
| bool WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend( |
| bool enable, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source) { |
| TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetVideoSend"); |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| // Ignore |options| pointer if |enable| is false. |
| bool options_present = enable && options; |
| |
| if (options_present) { |
| 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. |
| |
| bool force_encoder_allocation = true; |
| SetCodec(*parameters_.codec_settings, force_encoder_allocation); |
| // 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, DegradationPreference::kDegradationDisabled); |
| } |
| // Switch to the new source. |
| source_ = source; |
| if (source && stream_) { |
| stream_->SetSource(this, GetDegradationPreference()); |
| } |
| return true; |
| } |
| |
| webrtc::VideoSendStream::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. |
| DegradationPreference degradation_preference; |
| if (!enable_cpu_overuse_detection_) { |
| degradation_preference = DegradationPreference::kDegradationDisabled; |
| } else { |
| if (parameters_.options.is_screencast.value_or(false)) { |
| degradation_preference = DegradationPreference::kMaintainResolution; |
| } else if (webrtc::field_trial::IsEnabled( |
| "WebRTC-Video-BalancedDegradation")) { |
| degradation_preference = DegradationPreference::kBalanced; |
| } else { |
| degradation_preference = DegradationPreference::kMaintainFramerate; |
| } |
| } |
| return degradation_preference; |
| } |
| |
| const std::vector<uint32_t>& |
| WebRtcVideoChannel::WebRtcVideoSendStream::GetSsrcs() const { |
| return ssrcs_; |
| } |
| |
| EncoderFactoryAdapter::AllocatedEncoder |
| CricketEncoderFactoryAdapter::CreateVideoEncoder( |
| const VideoCodec& codec) const { |
| // Try creating external encoder. |
| if (external_encoder_factory_ != nullptr && |
| FindMatchingCodec(external_encoder_factory_->supported_codecs(), codec)) { |
| std::unique_ptr<webrtc::VideoEncoder> external_encoder; |
| if (CodecNamesEq(codec.name.c_str(), kVp8CodecName)) { |
| // If it's a codec type we can simulcast, create a wrapped encoder. |
| external_encoder = std::unique_ptr<webrtc::VideoEncoder>( |
| new webrtc::SimulcastEncoderAdapter(external_encoder_factory_.get())); |
| } else { |
| external_encoder = |
| CreateScopedVideoEncoder(external_encoder_factory_.get(), codec); |
| } |
| if (external_encoder) { |
| std::unique_ptr<webrtc::VideoEncoder> internal_encoder( |
| new webrtc::VideoEncoderSoftwareFallbackWrapper( |
| codec, std::move(external_encoder))); |
| const webrtc::VideoCodecType codec_type = |
| webrtc::PayloadStringToCodecType(codec.name); |
| const bool has_internal_source = |
| external_encoder_factory_->EncoderTypeHasInternalSource(codec_type); |
| return AllocatedEncoder(std::move(internal_encoder), |
| true /* is_hardware_accelerated */, |
| has_internal_source); |
| } |
| } |
| |
| // Try creating internal encoder. |
| std::unique_ptr<webrtc::VideoEncoder> internal_encoder; |
| if (FindMatchingCodec(internal_encoder_factory_->supported_codecs(), codec)) { |
| if (CodecNamesEq(codec.name.c_str(), kVp8CodecName)) { |
| internal_encoder = std::unique_ptr<webrtc::VideoEncoder>( |
| new webrtc::VP8EncoderSimulcastProxy( |
| internal_encoder_factory_.get())); |
| } else { |
| internal_encoder = std::unique_ptr<webrtc::VideoEncoder>( |
| internal_encoder_factory_->CreateVideoEncoder(codec)); |
| } |
| return AllocatedEncoder(std::move(internal_encoder), |
| false /* is_hardware_accelerated */, |
| false /* has_internal_source */); |
| } |
| |
| // This shouldn't happen, we should not be trying to create something we don't |
| // support. |
| RTC_NOTREACHED(); |
| return AllocatedEncoder(); |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoSendStream::SetCodec( |
| const VideoCodecSettings& codec_settings, |
| bool force_encoder_allocation) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| parameters_.encoder_config = CreateVideoEncoderConfig(codec_settings.codec); |
| RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0); |
| |
| // Do not re-create encoders of the same type. We can't overwrite |
| // |allocated_encoder_| immediately, because we need to release it after the |
| // RecreateWebRtcStream() call. |
| std::unique_ptr<webrtc::VideoEncoder> new_encoder; |
| if (force_encoder_allocation || !allocated_encoder_ || |
| allocated_codec_ != codec_settings.codec) { |
| EncoderFactoryAdapter::AllocatedEncoder new_allocated_encoder = |
| encoder_factory_->CreateVideoEncoder(codec_settings.codec); |
| new_encoder = std::unique_ptr<webrtc::VideoEncoder>( |
| std::move(new_allocated_encoder.encoder)); |
| parameters_.config.encoder_settings.encoder = new_encoder.get(); |
| parameters_.config.encoder_settings.full_overuse_time = |
| new_allocated_encoder.is_hardware_accelerated; |
| parameters_.config.encoder_settings.internal_source = |
| new_allocated_encoder.has_internal_source; |
| } else { |
| new_encoder = std::move(allocated_encoder_); |
| } |
| parameters_.config.encoder_settings.payload_name = codec_settings.codec.name; |
| parameters_.config.encoder_settings.payload_type = codec_settings.codec.id; |
| 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) { |
| LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX " |
| "payload type. Ignoring."; |
| parameters_.config.rtp.rtx.ssrcs.clear(); |
| } else { |
| parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type; |
| } |
| } |
| |
| parameters_.config.rtp.nack.rtp_history_ms = |
| HasNack(codec_settings.codec) ? kNackHistoryMs : 0; |
| |
| parameters_.codec_settings = |
| rtc::Optional<WebRtcVideoChannel::VideoCodecSettings>(codec_settings); |
| |
| LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetCodec."; |
| RecreateWebRtcStream(); |
| allocated_encoder_ = std::move(new_encoder); |
| allocated_codec_ = codec_settings.codec; |
| } |
| |
| 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; |
| recreate_stream = true; |
| } |
| if (params.rtp_header_extensions) { |
| parameters_.config.rtp.extensions = *params.rtp_header_extensions; |
| recreate_stream = true; |
| } |
| if (params.max_bandwidth_bps) { |
| parameters_.max_bitrate_bps = *params.max_bandwidth_bps; |
| ReconfigureEncoder(); |
| } |
| if (params.conference_mode) { |
| parameters_.conference_mode = *params.conference_mode; |
| } |
| |
| // Set codecs and options. |
| if (params.codec) { |
| bool force_encoder_allocation = false; |
| SetCodec(*params.codec, force_encoder_allocation); |
| recreate_stream = false; // SetCodec has already recreated the stream. |
| } else if (params.conference_mode && parameters_.codec_settings) { |
| bool force_encoder_allocation = false; |
| SetCodec(*parameters_.codec_settings, force_encoder_allocation); |
| recreate_stream = false; // SetCodec has already recreated the stream. |
| } |
| if (recreate_stream) { |
| LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters"; |
| RecreateWebRtcStream(); |
| } |
| } |
| |
| bool WebRtcVideoChannel::WebRtcVideoSendStream::SetRtpParameters( |
| const webrtc::RtpParameters& new_parameters) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (!ValidateRtpParameters(new_parameters)) { |
| return false; |
| } |
| |
| bool reconfigure_encoder = new_parameters.encodings[0].max_bitrate_bps != |
| rtp_parameters_.encodings[0].max_bitrate_bps; |
| rtp_parameters_ = new_parameters; |
| // Codecs are currently handled at the WebRtcVideoChannel level. |
| rtp_parameters_.codecs.clear(); |
| if (reconfigure_encoder) { |
| ReconfigureEncoder(); |
| } |
| // Encoding may have been activated/deactivated. |
| UpdateSendState(); |
| return true; |
| } |
| |
| webrtc::RtpParameters |
| WebRtcVideoChannel::WebRtcVideoSendStream::GetRtpParameters() const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| return rtp_parameters_; |
| } |
| |
| bool WebRtcVideoChannel::WebRtcVideoSendStream::ValidateRtpParameters( |
| const webrtc::RtpParameters& rtp_parameters) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (rtp_parameters.encodings.size() != 1) { |
| LOG(LS_ERROR) |
| << "Attempted to set RtpParameters without exactly one encoding"; |
| return false; |
| } |
| if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) { |
| LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC"; |
| return false; |
| } |
| return true; |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoSendStream::UpdateSendState() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| // TODO(deadbeef): Need to handle more than one encoding in the future. |
| RTC_DCHECK(rtp_parameters_.encodings.size() == 1u); |
| if (sending_ && rtp_parameters_.encodings[0].active) { |
| RTC_DCHECK(stream_ != nullptr); |
| stream_->Start(); |
| } else { |
| if (stream_ != nullptr) { |
| stream_->Stop(); |
| } |
| } |
| } |
| |
| webrtc::VideoEncoderConfig |
| WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig( |
| const VideoCodec& codec) const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| webrtc::VideoEncoderConfig encoder_config; |
| bool is_screencast = parameters_.options.is_screencast.value_or(false); |
| if (is_screencast) { |
| encoder_config.min_transmit_bitrate_bps = |
| 1000 * parameters_.options.screencast_min_bitrate_kbps.value_or(0); |
| encoder_config.content_type = |
| webrtc::VideoEncoderConfig::ContentType::kScreen; |
| } else { |
| encoder_config.min_transmit_bitrate_bps = 0; |
| encoder_config.content_type = |
| webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo; |
| } |
| |
| // By default, the stream count for the codec configuration should match the |
| // number of negotiated ssrcs. But if the codec is blacklisted for simulcast |
| // or a screencast (and not in simulcast screenshare experiment), only |
| // configure a single stream. |
| encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size(); |
| if (IsCodecBlacklistedForSimulcast(codec.name) || |
| (is_screencast && |
| (!UseSimulcastScreenshare() || !parameters_.conference_mode))) { |
| encoder_config.number_of_streams = 1; |
| } |
| |
| int stream_max_bitrate = parameters_.max_bitrate_bps; |
| if (rtp_parameters_.encodings[0].max_bitrate_bps) { |
| stream_max_bitrate = |
| webrtc::MinPositive(*(rtp_parameters_.encodings[0].max_bitrate_bps), |
| parameters_.max_bitrate_bps); |
| } |
| |
| int codec_max_bitrate_kbps; |
| if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps)) { |
| stream_max_bitrate = codec_max_bitrate_kbps * 1000; |
| } |
| encoder_config.max_bitrate_bps = stream_max_bitrate; |
| |
| int max_qp = kDefaultQpMax; |
| codec.GetParam(kCodecParamMaxQuantization, &max_qp); |
| encoder_config.video_stream_factory = |
| new rtc::RefCountedObject<EncoderStreamFactory>( |
| codec.name, max_qp, kDefaultVideoMaxFramerate, 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(); |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoSendStream::RemoveSink( |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_DCHECK(encoder_sink_ == sink); |
| encoder_sink_ = nullptr; |
| source_->RemoveSink(sink); |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoSendStream::AddOrUpdateSink( |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink, |
| const rtc::VideoSinkWants& wants) { |
| if (worker_thread_ == rtc::Thread::Current()) { |
| // AddOrUpdateSink is called on |worker_thread_| if this is the first |
| // registration of |sink|. |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| encoder_sink_ = sink; |
| source_->AddOrUpdateSink(encoder_sink_, wants); |
| } else { |
| // Subsequent calls to AddOrUpdateSink will happen on the encoder task |
| // queue. |
| invoker_.AsyncInvoke<void>( |
| RTC_FROM_HERE, worker_thread_, [this, sink, wants] { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| // |sink| may be invalidated after this task was posted since |
| // RemoveSink is called on the worker thread. |
| bool encoder_sink_valid = (sink == encoder_sink_); |
| if (source_ && encoder_sink_valid) { |
| source_->AddOrUpdateSink(encoder_sink_, wants); |
| } |
| }); |
| } |
| } |
| |
| VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo( |
| bool log_stats) { |
| VideoSenderInfo info; |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| for (uint32_t ssrc : parameters_.config.rtp.ssrcs) |
| info.add_ssrc(ssrc); |
| |
| if (parameters_.codec_settings) { |
| info.codec_name = parameters_.codec_settings->codec.name; |
| info.codec_payload_type = rtc::Optional<int>( |
| parameters_.codec_settings->codec.id); |
| } |
| |
| if (stream_ == NULL) |
| return info; |
| |
| webrtc::VideoSendStream::Stats stats = stream_->GetStats(); |
| |
| if (log_stats) |
| LOG(LS_INFO) << stats.ToString(rtc::TimeMillis()); |
| |
| info.adapt_changes = stats.number_of_cpu_adapt_changes; |
| info.adapt_reason = |
| stats.cpu_limited_resolution ? ADAPTREASON_CPU : ADAPTREASON_NONE; |
| |
| // Get bandwidth limitation info from stream_->GetStats(). |
| // Input resolution (output from video_adapter) can be further scaled down or |
| // higher video layer(s) can be dropped due to bitrate constraints. |
| // Note, adapt_changes only include changes from the video_adapter. |
| if (stats.bw_limited_resolution) |
| info.adapt_reason |= ADAPTREASON_BANDWIDTH; |
| |
| info.encoder_implementation_name = stats.encoder_implementation_name; |
| info.ssrc_groups = ssrc_groups_; |
| info.framerate_input = stats.input_frame_rate; |
| info.framerate_sent = stats.encode_frame_rate; |
| info.avg_encode_ms = stats.avg_encode_time_ms; |
| info.encode_usage_percent = stats.encode_usage_percent; |
| info.frames_encoded = stats.frames_encoded; |
| info.qp_sum = stats.qp_sum; |
| |
| info.nominal_bitrate = stats.media_bitrate_bps; |
| info.preferred_bitrate = stats.preferred_media_bitrate_bps; |
| |
| info.content_type = stats.content_type; |
| |
| info.send_frame_width = 0; |
| info.send_frame_height = 0; |
| for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it = |
| stats.substreams.begin(); |
| it != stats.substreams.end(); ++it) { |
| // TODO(pbos): Wire up additional stats, such as padding bytes. |
| webrtc::VideoSendStream::StreamStats stream_stats = it->second; |
| info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes + |
| stream_stats.rtp_stats.transmitted.header_bytes + |
| stream_stats.rtp_stats.transmitted.padding_bytes; |
| info.packets_sent += stream_stats.rtp_stats.transmitted.packets; |
| info.packets_lost += stream_stats.rtcp_stats.packets_lost; |
| if (stream_stats.width > info.send_frame_width) |
| info.send_frame_width = stream_stats.width; |
| if (stream_stats.height > info.send_frame_height) |
| info.send_frame_height = stream_stats.height; |
| info.firs_rcvd += stream_stats.rtcp_packet_type_counts.fir_packets; |
| info.nacks_rcvd += stream_stats.rtcp_packet_type_counts.nack_packets; |
| info.plis_rcvd += stream_stats.rtcp_packet_type_counts.pli_packets; |
| } |
| |
| if (!stats.substreams.empty()) { |
| // TODO(pbos): Report fraction lost per SSRC. |
| webrtc::VideoSendStream::StreamStats first_stream_stats = |
| stats.substreams.begin()->second; |
| info.fraction_lost = |
| static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) / |
| (1 << 8); |
| } |
| |
| return info; |
| } |
| |
| void 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::RecreateWebRtcStream() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (stream_ != NULL) { |
| call_->DestroyVideoSendStream(stream_); |
| } |
| |
| RTC_CHECK(parameters_.codec_settings); |
| RTC_DCHECK_EQ((parameters_.encoder_config.content_type == |
| webrtc::VideoEncoderConfig::ContentType::kScreen), |
| parameters_.options.is_screencast.value_or(false)) |
| << "encoder content type inconsistent with screencast option"; |
| parameters_.encoder_config.encoder_specific_settings = |
| ConfigureVideoEncoderSettings(parameters_.codec_settings->codec); |
| |
| webrtc::VideoSendStream::Config config = parameters_.config.Copy(); |
| if (!config.rtp.rtx.ssrcs.empty() && config.rtp.rtx.payload_type == -1) { |
| LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX " |
| "payload type the set codec. Ignoring RTX."; |
| config.rtp.rtx.ssrcs.clear(); |
| } |
| stream_ = call_->CreateVideoSendStream(std::move(config), |
| parameters_.encoder_config.Copy()); |
| |
| parameters_.encoder_config.encoder_specific_settings = NULL; |
| |
| if (source_) { |
| stream_->SetSource(this, GetDegradationPreference()); |
| } |
| |
| // Call stream_->Start() if necessary conditions are met. |
| UpdateSendState(); |
| } |
| |
| WebRtcVideoChannel::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream( |
| webrtc::Call* call, |
| const StreamParams& sp, |
| webrtc::VideoReceiveStream::Config config, |
| const DecoderFactoryAdapter* decoder_factory, |
| bool default_stream, |
| const std::vector<VideoCodecSettings>& recv_codecs, |
| const webrtc::FlexfecReceiveStream::Config& flexfec_config) |
| : call_(call), |
| stream_params_(sp), |
| stream_(NULL), |
| default_stream_(default_stream), |
| config_(std::move(config)), |
| flexfec_config_(flexfec_config), |
| flexfec_stream_(nullptr), |
| decoder_factory_(decoder_factory), |
| sink_(NULL), |
| first_frame_timestamp_(-1), |
| estimated_remote_start_ntp_time_ms_(0) { |
| config_.renderer = this; |
| DecoderMap old_decoders; |
| ConfigureCodecs(recv_codecs, &old_decoders); |
| ConfigureFlexfecCodec(flexfec_config.payload_type); |
| MaybeRecreateWebRtcFlexfecStream(); |
| RecreateWebRtcVideoStream(); |
| RTC_DCHECK(old_decoders.empty()); |
| } |
| |
| WebRtcVideoChannel::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() { |
| if (flexfec_stream_) { |
| MaybeDissociateFlexfecFromVideo(); |
| call_->DestroyFlexfecReceiveStream(flexfec_stream_); |
| } |
| call_->DestroyVideoReceiveStream(stream_); |
| allocated_decoders_.clear(); |
| } |
| |
| const std::vector<uint32_t>& |
| WebRtcVideoChannel::WebRtcVideoReceiveStream::GetSsrcs() const { |
| return stream_params_.ssrcs; |
| } |
| |
| rtc::Optional<uint32_t> |
| WebRtcVideoChannel::WebRtcVideoReceiveStream::GetFirstPrimarySsrc() const { |
| std::vector<uint32_t> primary_ssrcs; |
| stream_params_.GetPrimarySsrcs(&primary_ssrcs); |
| |
| if (primary_ssrcs.empty()) { |
| LOG(LS_WARNING) << "Empty primary ssrcs vector, returning empty optional"; |
| return rtc::Optional<uint32_t>(); |
| } else { |
| return rtc::Optional<uint32_t>(primary_ssrcs[0]); |
| } |
| } |
| |
| std::unique_ptr<webrtc::VideoDecoder> |
| CricketDecoderFactoryAdapter::CreateVideoDecoder( |
| const VideoCodec& codec, |
| const VideoDecoderParams& decoder_params) const { |
| if (external_decoder_factory_ != nullptr) { |
| std::unique_ptr<webrtc::VideoDecoder> external_decoder = |
| CreateScopedVideoDecoder(external_decoder_factory_.get(), codec, |
| decoder_params); |
| if (external_decoder) { |
| webrtc::VideoCodecType type = |
| webrtc::PayloadStringToCodecType(codec.name); |
| std::unique_ptr<webrtc::VideoDecoder> internal_decoder( |
| new webrtc::VideoDecoderSoftwareFallbackWrapper( |
| type, std::move(external_decoder))); |
| return internal_decoder; |
| } |
| } |
| |
| std::unique_ptr<webrtc::VideoDecoder> internal_decoder( |
| internal_decoder_factory_->CreateVideoDecoderWithParams(codec, |
| decoder_params)); |
| return internal_decoder; |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream::ConfigureCodecs( |
| const std::vector<VideoCodecSettings>& recv_codecs, |
| DecoderMap* old_decoders) { |
| RTC_DCHECK(!recv_codecs.empty()); |
| *old_decoders = std::move(allocated_decoders_); |
| config_.decoders.clear(); |
| config_.rtp.rtx_associated_payload_types.clear(); |
| for (const auto& recv_codec : recv_codecs) { |
| webrtc::SdpVideoFormat video_format(recv_codec.codec.name, |
| recv_codec.codec.params); |
| std::unique_ptr<webrtc::VideoDecoder> new_decoder; |
| |
| auto it = old_decoders->find(video_format); |
| if (it != old_decoders->end()) { |
| new_decoder = std::move(it->second); |
| old_decoders->erase(it); |
| } |
| |
| if (!new_decoder) { |
| new_decoder = decoder_factory_->CreateVideoDecoder(recv_codec.codec, |
| {stream_params_.id}); |
| } |
| |
| webrtc::VideoReceiveStream::Decoder decoder; |
| decoder.decoder = new_decoder.get(); |
| decoder.payload_type = recv_codec.codec.id; |
| decoder.payload_name = recv_codec.codec.name; |
| decoder.codec_params = recv_codec.codec.params; |
| config_.decoders.push_back(decoder); |
| config_.rtp.rtx_associated_payload_types[recv_codec.rtx_payload_type] = |
| recv_codec.codec.id; |
| |
| const bool did_insert = |
| allocated_decoders_ |
| .insert(std::make_pair(video_format, std::move(new_decoder))) |
| .second; |
| RTC_CHECK(did_insert); |
| } |
| |
| const auto& codec = recv_codecs.front(); |
| config_.rtp.ulpfec_payload_type = codec.ulpfec.ulpfec_payload_type; |
| config_.rtp.red_payload_type = codec.ulpfec.red_payload_type; |
| |
| config_.rtp.nack.rtp_history_ms = HasNack(codec.codec) ? kNackHistoryMs : 0; |
| if (codec.ulpfec.red_rtx_payload_type != -1) { |
| config_.rtp |
| .rtx_associated_payload_types[codec.ulpfec.red_rtx_payload_type] = |
| codec.ulpfec.red_payload_type; |
| } |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream::ConfigureFlexfecCodec( |
| int flexfec_payload_type) { |
| flexfec_config_.payload_type = flexfec_payload_type; |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetLocalSsrc( |
| uint32_t local_ssrc) { |
| // TODO(pbos): Consider turning this sanity check into a RTC_DCHECK. You |
| // should not be able to create a sender with the same SSRC as a receiver, but |
| // right now this can't be done due to unittests depending on receiving what |
| // they are sending from the same MediaChannel. |
| if (local_ssrc == config_.rtp.remote_ssrc) { |
| LOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are " |
| "unchanged; local_ssrc=" << local_ssrc; |
| return; |
| } |
| |
| config_.rtp.local_ssrc = local_ssrc; |
| flexfec_config_.local_ssrc = local_ssrc; |
| LOG(LS_INFO) |
| << "RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=" |
| << local_ssrc; |
| MaybeRecreateWebRtcFlexfecStream(); |
| RecreateWebRtcVideoStream(); |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetFeedbackParameters( |
| bool nack_enabled, |
| bool remb_enabled, |
| bool transport_cc_enabled, |
| webrtc::RtcpMode rtcp_mode) { |
| int nack_history_ms = nack_enabled ? kNackHistoryMs : 0; |
| if (config_.rtp.nack.rtp_history_ms == nack_history_ms && |
| config_.rtp.remb == remb_enabled && |
| config_.rtp.transport_cc == transport_cc_enabled && |
| config_.rtp.rtcp_mode == rtcp_mode) { |
| LOG(LS_INFO) |
| << "Ignoring call to SetFeedbackParameters because parameters are " |
| "unchanged; nack=" |
| << nack_enabled << ", remb=" << remb_enabled |
| << ", transport_cc=" << transport_cc_enabled; |
| return; |
| } |
| config_.rtp.remb = remb_enabled; |
| config_.rtp.nack.rtp_history_ms = nack_history_ms; |
| config_.rtp.transport_cc = transport_cc_enabled; |
| config_.rtp.rtcp_mode = rtcp_mode; |
| // 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_.rtcp_mode = config_.rtp.rtcp_mode; |
| LOG(LS_INFO) |
| << "RecreateWebRtcStream (recv) because of SetFeedbackParameters; nack=" |
| << nack_enabled << ", remb=" << remb_enabled |
| << ", transport_cc=" << transport_cc_enabled; |
| MaybeRecreateWebRtcFlexfecStream(); |
| RecreateWebRtcVideoStream(); |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetRecvParameters( |
| const ChangedRecvParameters& params) { |
| bool video_needs_recreation = false; |
| bool flexfec_needs_recreation = false; |
| DecoderMap old_decoders; |
| if (params.codec_settings) { |
| ConfigureCodecs(*params.codec_settings, &old_decoders); |
| video_needs_recreation = true; |
| } |
| if (params.rtp_header_extensions) { |
| config_.rtp.extensions = *params.rtp_header_extensions; |
| flexfec_config_.rtp_header_extensions = *params.rtp_header_extensions; |
| video_needs_recreation = true; |
| flexfec_needs_recreation = true; |
| } |
| if (params.flexfec_payload_type) { |
| ConfigureFlexfecCodec(*params.flexfec_payload_type); |
| flexfec_needs_recreation = true; |
| } |
| if (flexfec_needs_recreation) { |
| LOG(LS_INFO) << "MaybeRecreateWebRtcFlexfecStream (recv) because of " |
| "SetRecvParameters"; |
| MaybeRecreateWebRtcFlexfecStream(); |
| } |
| if (video_needs_recreation) { |
| LOG(LS_INFO) |
| << "RecreateWebRtcVideoStream (recv) because of SetRecvParameters"; |
| RecreateWebRtcVideoStream(); |
| } |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream:: |
| RecreateWebRtcVideoStream() { |
| if (stream_) { |
| MaybeDissociateFlexfecFromVideo(); |
| call_->DestroyVideoReceiveStream(stream_); |
| stream_ = nullptr; |
| } |
| webrtc::VideoReceiveStream::Config config = config_.Copy(); |
| config.rtp.protected_by_flexfec = (flexfec_stream_ != nullptr); |
| stream_ = call_->CreateVideoReceiveStream(std::move(config)); |
| MaybeAssociateFlexfecWithVideo(); |
| stream_->Start(); |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream:: |
| MaybeRecreateWebRtcFlexfecStream() { |
| if (flexfec_stream_) { |
| MaybeDissociateFlexfecFromVideo(); |
| call_->DestroyFlexfecReceiveStream(flexfec_stream_); |
| flexfec_stream_ = nullptr; |
| } |
| if (flexfec_config_.IsCompleteAndEnabled()) { |
| flexfec_stream_ = call_->CreateFlexfecReceiveStream(flexfec_config_); |
| MaybeAssociateFlexfecWithVideo(); |
| } |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream:: |
| MaybeAssociateFlexfecWithVideo() { |
| if (stream_ && flexfec_stream_) { |
| stream_->AddSecondarySink(flexfec_stream_); |
| } |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream:: |
| MaybeDissociateFlexfecFromVideo() { |
| if (stream_ && flexfec_stream_) { |
| stream_->RemoveSecondarySink(flexfec_stream_); |
| } |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream::OnFrame( |
| const webrtc::VideoFrame& frame) { |
| rtc::CritScope crit(&sink_lock_); |
| |
| if (first_frame_timestamp_ < 0) |
| first_frame_timestamp_ = frame.timestamp(); |
| int64_t rtp_time_elapsed_since_first_frame = |
| (timestamp_wraparound_handler_.Unwrap(frame.timestamp()) - |
| first_frame_timestamp_); |
| int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame / |
| (cricket::kVideoCodecClockrate / 1000); |
| if (frame.ntp_time_ms() > 0) |
| estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms; |
| |
| if (sink_ == NULL) { |
| LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoSink."; |
| return; |
| } |
| |
| sink_->OnFrame(frame); |
| } |
| |
| bool WebRtcVideoChannel::WebRtcVideoReceiveStream::IsDefaultStream() const { |
| return default_stream_; |
| } |
| |
| void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetSink( |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) { |
| rtc::CritScope crit(&sink_lock_); |
| sink_ = sink; |
| } |
| |
| std::string |
| WebRtcVideoChannel::WebRtcVideoReceiveStream::GetCodecNameFromPayloadType( |
| int payload_type) { |
| for (const webrtc::VideoReceiveStream::Decoder& decoder : config_.decoders) { |
| if (decoder.payload_type == payload_type) { |
| return decoder.payload_name; |
| } |
| } |
| return ""; |
| } |
| |
| VideoReceiverInfo |
| WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo( |
| bool log_stats) { |
| VideoReceiverInfo info; |
| info.ssrc_groups = stream_params_.ssrc_groups; |
| info.add_ssrc(config_.rtp.remote_ssrc); |
| webrtc::VideoReceiveStream::Stats stats = stream_->GetStats(); |
| info.decoder_implementation_name = stats.decoder_implementation_name; |
| if (stats.current_payload_type != -1) { |
| info.codec_payload_type = rtc::Optional<int>( |
| stats.current_payload_type); |
| } |
| info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes + |
| stats.rtp_stats.transmitted.header_bytes + |
| stats.rtp_stats.transmitted.padding_bytes; |
| info.packets_rcvd = stats.rtp_stats.transmitted.packets; |
| info.packets_lost = stats.rtcp_stats.packets_lost; |
| info.fraction_lost = |
| static_cast<float>(stats.rtcp_stats.fraction_lost) / (1 << 8); |
| |
| info.framerate_rcvd = stats.network_frame_rate; |
| info.framerate_decoded = stats.decode_frame_rate; |
| info.framerate_output = stats.render_frame_rate; |
| info.frame_width = stats.width; |
| info.frame_height = stats.height; |
| |
| { |
| rtc::CritScope frame_cs(&sink_lock_); |
| info.capture_start_ntp_time_ms = estimated_remote_start_ntp_time_ms_; |
| } |
| |
| info.decode_ms = stats.decode_ms; |
| info.max_decode_ms = stats.max_decode_ms; |
| info.current_delay_ms = stats.current_delay_ms; |
| info.target_delay_ms = stats.target_delay_ms; |
| info.jitter_buffer_ms = stats.jitter_buffer_ms; |
| info.min_playout_delay_ms = stats.min_playout_delay_ms; |
| info.render_delay_ms = stats.render_delay_ms; |
| info.frames_received = stats.frame_counts.key_frames + |
| stats.frame_counts.delta_frames; |
| info.frames_decoded = stats.frames_decoded; |
| info.frames_rendered = stats.frames_rendered; |
| info.qp_sum = stats.qp_sum; |
| |
| info.interframe_delay_max_ms = stats.interframe_delay_max_ms; |
| |
| info.content_type = stats.content_type; |
| |
| info.codec_name = GetCodecNameFromPayloadType(stats.current_payload_type); |
| |
| info.firs_sent = stats.rtcp_packet_type_counts.fir_packets; |
| info.plis_sent = stats.rtcp_packet_type_counts.pli_packets; |
| info.nacks_sent = stats.rtcp_packet_type_counts.nack_packets; |
| |
| info.timing_frame_info = stats.timing_frame_info; |
| |
| if (log_stats) |
| LOG(LS_INFO) << stats.ToString(rtc::TimeMillis()); |
| |
| return info; |
| } |
| |
| WebRtcVideoChannel::VideoCodecSettings::VideoCodecSettings() |
| : flexfec_payload_type(-1), rtx_payload_type(-1) {} |
| |
| bool WebRtcVideoChannel::VideoCodecSettings::operator==( |
| const WebRtcVideoChannel::VideoCodecSettings& other) const { |
| return codec == other.codec && ulpfec == other.ulpfec && |
| flexfec_payload_type == other.flexfec_payload_type && |
| rtx_payload_type == other.rtx_payload_type; |
| } |
| |
| bool WebRtcVideoChannel::VideoCodecSettings::EqualsDisregardingFlexfec( |
| const WebRtcVideoChannel::VideoCodecSettings& a, |
| const WebRtcVideoChannel::VideoCodecSettings& b) { |
| return a.codec == b.codec && a.ulpfec == b.ulpfec && |
| a.rtx_payload_type == b.rtx_payload_type; |
| } |
| |
| bool WebRtcVideoChannel::VideoCodecSettings::operator!=( |
| const WebRtcVideoChannel::VideoCodecSettings& other) const { |
| return !(*this == other); |
| } |
| |
| std::vector<WebRtcVideoChannel::VideoCodecSettings> |
| WebRtcVideoChannel::MapCodecs(const std::vector<VideoCodec>& codecs) { |
| RTC_DCHECK(!codecs.empty()); |
| |
| std::vector<VideoCodecSettings> video_codecs; |
| std::map<int, bool> payload_used; |
| std::map<int, VideoCodec::CodecType> payload_codec_type; |
| // |rtx_mapping| maps video payload type to rtx payload type. |
| std::map<int, int> rtx_mapping; |
| |
| webrtc::UlpfecConfig ulpfec_config; |
| int flexfec_payload_type = -1; |
| |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| const VideoCodec& in_codec = codecs[i]; |
| int payload_type = in_codec.id; |
| |
| if (payload_used[payload_type]) { |
| LOG(LS_ERROR) << "Payload type already registered: " |
| << in_codec.ToString(); |
| return std::vector<VideoCodecSettings>(); |
| } |
| payload_used[payload_type] = true; |
| payload_codec_type[payload_type] = in_codec.GetCodecType(); |
| |
| switch (in_codec.GetCodecType()) { |
| case VideoCodec::CODEC_RED: { |
| // RED payload type, should not have duplicates. |
| RTC_DCHECK_EQ(-1, ulpfec_config.red_payload_type); |
| ulpfec_config.red_payload_type = in_codec.id; |
| continue; |
| } |
| |
| case VideoCodec::CODEC_ULPFEC: { |
| // ULPFEC payload type, should not have duplicates. |
| RTC_DCHECK_EQ(-1, ulpfec_config.ulpfec_payload_type); |
| ulpfec_config.ulpfec_payload_type = in_codec.id; |
| continue; |
| } |
| |
| case VideoCodec::CODEC_FLEXFEC: { |
| // FlexFEC payload type, should not have duplicates. |
| RTC_DCHECK_EQ(-1, flexfec_payload_type); |
| flexfec_payload_type = in_codec.id; |
| continue; |
| } |
| |
| case VideoCodec::CODEC_RTX: { |
| int associated_payload_type; |
| if (!in_codec.GetParam(kCodecParamAssociatedPayloadType, |
| &associated_payload_type) || |
| !IsValidRtpPayloadType(associated_payload_type)) { |
| LOG(LS_ERROR) |
| << "RTX codec with invalid or no associated payload type: " |
| << in_codec.ToString(); |
| return std::vector<VideoCodecSettings>(); |
| } |
| rtx_mapping[associated_payload_type] = in_codec.id; |
| continue; |
| } |
| |
| case VideoCodec::CODEC_VIDEO: |
| break; |
| } |
| |
| video_codecs.push_back(VideoCodecSettings()); |
| video_codecs.back().codec = in_codec; |
| } |
| |
| // One of these codecs should have been a video codec. Only having FEC |
| // parameters into this code is a logic error. |
| RTC_DCHECK(!video_codecs.empty()); |
| |
| for (std::map<int, int>::const_iterator it = rtx_mapping.begin(); |
| it != rtx_mapping.end(); |
| ++it) { |
| if (!payload_used[it->first]) { |
| LOG(LS_ERROR) << "RTX mapped to payload not in codec list."; |
| return std::vector<VideoCodecSettings>(); |
| } |
| if (payload_codec_type[it->first] != VideoCodec::CODEC_VIDEO && |
| payload_codec_type[it->first] != VideoCodec::CODEC_RED) { |
| LOG(LS_ERROR) << "RTX not mapped to regular video codec or RED codec."; |
| return std::vector<VideoCodecSettings>(); |
| } |
| |
| if (it->first == ulpfec_config.red_payload_type) { |
| ulpfec_config.red_rtx_payload_type = it->second; |
| } |
| } |
| |
| for (size_t i = 0; i < video_codecs.size(); ++i) { |
| video_codecs[i].ulpfec = ulpfec_config; |
| video_codecs[i].flexfec_payload_type = flexfec_payload_type; |
| if (rtx_mapping[video_codecs[i].codec.id] != 0 && |
| rtx_mapping[video_codecs[i].codec.id] != |
| ulpfec_config.red_payload_type) { |
| video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; |
| } |
| } |
| |
| return video_codecs; |
| } |
| |
| EncoderStreamFactory::EncoderStreamFactory(std::string codec_name, |
| int max_qp, |
| int max_framerate, |
| bool is_screencast, |
| bool conference_mode) |
| : codec_name_(codec_name), |
| max_qp_(max_qp), |
| max_framerate_(max_framerate), |
| is_screencast_(is_screencast), |
| conference_mode_(conference_mode) {} |
| |
| std::vector<webrtc::VideoStream> EncoderStreamFactory::CreateEncoderStreams( |
| int width, |
| int height, |
| const webrtc::VideoEncoderConfig& encoder_config) { |
| if (is_screencast_ && |
| (!conference_mode_ || !cricket::UseSimulcastScreenshare())) { |
| RTC_DCHECK_EQ(1, encoder_config.number_of_streams); |
| } |
| if (encoder_config.number_of_streams > 1 || |
| (CodecNamesEq(codec_name_, kVp8CodecName) && is_screencast_ && |
| conference_mode_)) { |
| return GetSimulcastConfig(encoder_config.number_of_streams, width, height, |
| encoder_config.max_bitrate_bps, max_qp_, |
| max_framerate_, is_screencast_); |
| } |
| |
| // For unset max bitrates set default bitrate for non-simulcast. |
| int max_bitrate_bps = |
| (encoder_config.max_bitrate_bps > 0) |
| ? encoder_config.max_bitrate_bps |
| : GetMaxDefaultVideoBitrateKbps(width, height) * 1000; |
| |
| webrtc::VideoStream stream; |
| stream.width = width; |
| stream.height = height; |
| stream.max_framerate = max_framerate_; |
| stream.min_bitrate_bps = kMinVideoBitrateKbps * 1000; |
| stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate_bps; |
| stream.max_qp = max_qp_; |
| |
| if (CodecNamesEq(codec_name_, kVp9CodecName) && !is_screencast_) { |
| stream.temporal_layer_thresholds_bps.resize(GetDefaultVp9TemporalLayers() - |
| 1); |
| } |
| |
| std::vector<webrtc::VideoStream> streams; |
| streams.push_back(stream); |
| return streams; |
| } |
| |
| } // namespace cricket |
