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Add a padding generator handler #794
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,217 @@ | ||
| #include "rtp/RtpPaddingGeneratorHandler.h" | ||
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| #include <algorithm> | ||
| #include <string> | ||
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| #include "./MediaDefinitions.h" | ||
| #include "./WebRtcConnection.h" | ||
| #include "./RtpUtils.h" | ||
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| namespace erizo { | ||
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| DEFINE_LOGGER(RtpPaddingGeneratorHandler, "rtp.RtpPaddingGeneratorHandler"); | ||
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| constexpr duration kStatsPeriod = std::chrono::milliseconds(100); | ||
| constexpr duration kFastStartMaxDuration = std::chrono::seconds(30); | ||
| constexpr uint8_t kMaxPaddingSize = 255; | ||
| constexpr uint64_t kMinMarkerRate = 3; | ||
| constexpr uint8_t kMaxFractionLostAllowed = .05 * 255; // 5% of packet losts | ||
| constexpr uint64_t kInitialRembValue = 300000; | ||
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| RtpPaddingGeneratorHandler::RtpPaddingGeneratorHandler(std::shared_ptr<erizo::Clock> the_clock) : | ||
| clock_{the_clock}, connection_{nullptr}, max_video_bw_{0}, higher_sequence_number_{0}, | ||
| video_sink_ssrc_{0}, audio_source_ssrc_{0}, | ||
| number_of_full_padding_packets_{0}, last_padding_packet_size_{0}, | ||
| last_rate_calculation_time_{clock_->now()}, started_at_{clock_->now()}, | ||
| enabled_{false}, first_packet_received_{false}, | ||
| marker_rate_{std::chrono::milliseconds(100), 20, 1., clock_}, | ||
| padding_bitrate_{std::chrono::milliseconds(100), 10, 8., clock_}, | ||
| rtp_header_length_{12}, remb_value_{kInitialRembValue}, fast_start_{true} {} | ||
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| void RtpPaddingGeneratorHandler::enable() { | ||
| } | ||
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| void RtpPaddingGeneratorHandler::disable() { | ||
| } | ||
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| void RtpPaddingGeneratorHandler::notifyUpdate() { | ||
| auto pipeline = getContext()->getPipelineShared(); | ||
| if (pipeline && !connection_) { | ||
| connection_ = pipeline->getService<WebRtcConnection>().get(); | ||
| video_sink_ssrc_ = connection_->getVideoSinkSSRC(); | ||
| audio_source_ssrc_ = connection_->getAudioSinkSSRC(); | ||
| stats_ = pipeline->getService<Stats>(); | ||
| } | ||
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| auto quality_manager = pipeline->getService<QualityManager>(); | ||
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| if (quality_manager->isPaddingEnabled() && !enabled_) { | ||
| enablePadding(); | ||
| } else if (!quality_manager->isPaddingEnabled() && enabled_) { | ||
| disablePadding(); | ||
| } | ||
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| auto processor = pipeline->getService<RtcpProcessor>(); | ||
| if (processor) { | ||
| max_video_bw_ = processor->getMaxVideoBW(); | ||
| } | ||
| } | ||
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| void RtpPaddingGeneratorHandler::read(Context *ctx, std::shared_ptr<dataPacket> packet) { | ||
| RtpUtils::forEachRRBlock(packet, [this](RtcpHeader *chead) { | ||
| if (chead->packettype == RTCP_PS_Feedback_PT && chead->getBlockCount() == RTCP_AFB) { | ||
| char *uniqueId = reinterpret_cast<char*>(&chead->report.rembPacket.uniqueid); | ||
| if (!strncmp(uniqueId, "REMB", 4)) { | ||
| remb_value_ = chead->getREMBBitRate(); | ||
| } | ||
| } else if (fast_start_ && chead->packettype == RTCP_Receiver_PT && chead->getSourceSSRC() == audio_source_ssrc_) { | ||
| if (chead->getFractionLost() > kMaxFractionLostAllowed) { | ||
| ELOG_DEBUG("Fast start disabled"); | ||
| fast_start_ = false; | ||
| } | ||
| } | ||
| }); | ||
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| ctx->fireRead(packet); | ||
| } | ||
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| void RtpPaddingGeneratorHandler::write(Context *ctx, std::shared_ptr<dataPacket> packet) { | ||
| RtcpHeader *chead = reinterpret_cast<RtcpHeader*>(packet->data); | ||
| bool is_higher_sequence_number = false; | ||
| if (packet->type == VIDEO_PACKET && !chead->isRtcp()) { | ||
| is_higher_sequence_number = isHigherSequenceNumber(packet); | ||
| if (!first_packet_received_) { | ||
| started_at_ = clock_->now(); | ||
| } | ||
| first_packet_received_ = true; | ||
| if (fast_start_ && clock_->now() - started_at_ > kFastStartMaxDuration) { | ||
| ELOG_DEBUG("Fast start disabled"); | ||
| fast_start_ = false; | ||
| } | ||
| } | ||
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| ctx->fireWrite(packet); | ||
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| if (is_higher_sequence_number) { | ||
| onVideoPacket(packet); | ||
| } | ||
| } | ||
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| void RtpPaddingGeneratorHandler::sendPaddingPacket(std::shared_ptr<dataPacket> packet, uint8_t padding_size) { | ||
| if (padding_size == 0) { | ||
| return; | ||
| } | ||
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| SequenceNumber sequence_number = translator_.generate(); | ||
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| auto padding_packet = RtpUtils::makePaddingPacket(packet, padding_size); | ||
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| RtpHeader *rtp_header = reinterpret_cast<RtpHeader*>(padding_packet->data); | ||
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| rtp_header->setSeqNumber(sequence_number.output); | ||
| padding_bitrate_ += padding_packet->length; | ||
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| getContext()->fireWrite(padding_packet); | ||
| } | ||
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| void RtpPaddingGeneratorHandler::onPacketWithMarkerSet(std::shared_ptr<dataPacket> packet) { | ||
| marker_rate_++; | ||
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| for (int i = 0; i < number_of_full_padding_packets_; i++) { | ||
| sendPaddingPacket(packet, kMaxPaddingSize); | ||
| } | ||
| sendPaddingPacket(packet, last_padding_packet_size_); | ||
| } | ||
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| bool RtpPaddingGeneratorHandler::isHigherSequenceNumber(std::shared_ptr<dataPacket> packet) { | ||
| RtpHeader *rtp_header = reinterpret_cast<RtpHeader*>(packet->data); | ||
| rtp_header_length_ = rtp_header->getHeaderLength(); | ||
| uint16_t new_sequence_number = rtp_header->getSeqNumber(); | ||
| SequenceNumber sequence_number = translator_.get(new_sequence_number, false); | ||
| rtp_header->setSeqNumber(sequence_number.output); | ||
| if (first_packet_received_ && RtpUtils::sequenceNumberLessThan(new_sequence_number, higher_sequence_number_)) { | ||
| return false; | ||
| } | ||
| higher_sequence_number_ = new_sequence_number; | ||
| return true; | ||
| } | ||
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| void RtpPaddingGeneratorHandler::onVideoPacket(std::shared_ptr<dataPacket> packet) { | ||
| if (!enabled_) { | ||
| return; | ||
| } | ||
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| recalculatePaddingRate(); | ||
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| RtpHeader *rtp_header = reinterpret_cast<RtpHeader*>(packet->data); | ||
| if (rtp_header->getMarker()) { | ||
| onPacketWithMarkerSet(packet); | ||
| } | ||
| } | ||
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| uint64_t RtpPaddingGeneratorHandler::getStat(std::string stat_name) { | ||
| if (stats_->getNode()["total"].hasChild(stat_name)) { | ||
| StatNode & stat = stats_->getNode()["total"][stat_name]; | ||
| return static_cast<MovingIntervalRateStat&>(stat).value(); | ||
| } | ||
| return 0; | ||
| } | ||
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| bool RtpPaddingGeneratorHandler::isTimeToCalculateBitrate() { | ||
| return (clock_->now() - last_rate_calculation_time_) >= kStatsPeriod; | ||
| } | ||
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| void RtpPaddingGeneratorHandler::recalculatePaddingRate() { | ||
| if (!isTimeToCalculateBitrate()) { | ||
| return; | ||
| } | ||
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| last_rate_calculation_time_ = clock_->now(); | ||
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| int64_t total_bitrate = getStat("bitrateCalculated"); | ||
| int64_t padding_bitrate = padding_bitrate_.value(); | ||
| int64_t media_bitrate = std::max(total_bitrate - padding_bitrate, int64_t(0)); | ||
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| uint64_t target_bitrate = getTargetBitrate(); | ||
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| int64_t target_padding_bitrate = target_bitrate - media_bitrate; | ||
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| if (target_padding_bitrate <= 0) { | ||
| number_of_full_padding_packets_ = 0; | ||
| last_padding_packet_size_ = 0; | ||
| return; | ||
| } | ||
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| uint64_t marker_rate = marker_rate_.value(std::chrono::seconds(2)); | ||
| marker_rate = std::max(marker_rate, kMinMarkerRate); | ||
| uint64_t bytes_per_marker = target_padding_bitrate / (marker_rate * 8); | ||
| number_of_full_padding_packets_ = bytes_per_marker / (kMaxPaddingSize + rtp_header_length_); | ||
| last_padding_packet_size_ = bytes_per_marker % (kMaxPaddingSize + rtp_header_length_) - rtp_header_length_; | ||
| } | ||
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| uint64_t RtpPaddingGeneratorHandler::getTargetBitrate() { | ||
| uint64_t target_bitrate = remb_value_; | ||
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| if (!fast_start_) { | ||
| target_bitrate = getStat("senderBitrateEstimation"); | ||
| } | ||
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| if (max_video_bw_ > 0) { | ||
| target_bitrate = std::min(target_bitrate, max_video_bw_); | ||
| } | ||
| return target_bitrate; | ||
| } | ||
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| void RtpPaddingGeneratorHandler::enablePadding() { | ||
| enabled_ = true; | ||
| number_of_full_padding_packets_ = 0; | ||
| last_padding_packet_size_ = 0; | ||
| last_rate_calculation_time_ = clock_->now(); | ||
| } | ||
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| void RtpPaddingGeneratorHandler::disablePadding() { | ||
| enabled_ = false; | ||
| } | ||
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| } // namespace erizo | ||
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I'm thinking we should move as much as we can to the stats. Here it makes sense to react as soon as we get the info but I keep thinking we should avoid parsing the same packets n times.
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👍