1222 lines
43 KiB
C++
1222 lines
43 KiB
C++
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/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/source/rtcp_receiver.h"
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#include <string.h>
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#include <algorithm>
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#include <limits>
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#include <map>
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#include <memory>
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#include <utility>
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#include <vector>
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#include "api/video/video_bitrate_allocation.h"
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#include "api/video/video_bitrate_allocator.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "modules/rtp_rtcp/source/time_util.h"
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#include "modules/rtp_rtcp/source/tmmbr_help.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/time_utils.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/ntp_time.h"
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namespace webrtc {
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namespace {
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using rtcp::CommonHeader;
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using rtcp::ReportBlock;
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// The number of RTCP time intervals needed to trigger a timeout.
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const int kRrTimeoutIntervals = 3;
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const int64_t kTmmbrTimeoutIntervalMs = 5 * 5000;
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const int64_t kMaxWarningLogIntervalMs = 10000;
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const int64_t kRtcpMinFrameLengthMs = 17;
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// Maximum number of received RRTRs that will be stored.
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const size_t kMaxNumberOfStoredRrtrs = 300;
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constexpr TimeDelta kDefaultVideoReportInterval = TimeDelta::Seconds(1);
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constexpr TimeDelta kDefaultAudioReportInterval = TimeDelta::Seconds(5);
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std::set<uint32_t> GetRegisteredSsrcs(
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const RtpRtcpInterface::Configuration& config) {
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std::set<uint32_t> ssrcs;
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ssrcs.insert(config.local_media_ssrc);
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if (config.rtx_send_ssrc) {
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ssrcs.insert(*config.rtx_send_ssrc);
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}
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if (config.fec_generator) {
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absl::optional<uint32_t> flexfec_ssrc = config.fec_generator->FecSsrc();
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if (flexfec_ssrc) {
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ssrcs.insert(*flexfec_ssrc);
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}
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}
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return ssrcs;
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}
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// Returns true if the |timestamp| has exceeded the |interval *
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// kRrTimeoutIntervals| period and was reset (set to PlusInfinity()). Returns
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// false if the timer was either already reset or if it has not expired.
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bool ResetTimestampIfExpired(const Timestamp now,
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Timestamp& timestamp,
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TimeDelta interval) {
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if (timestamp.IsInfinite() ||
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now <= timestamp + interval * kRrTimeoutIntervals) {
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return false;
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}
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timestamp = Timestamp::PlusInfinity();
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return true;
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}
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} // namespace
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struct RTCPReceiver::PacketInformation {
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uint32_t packet_type_flags = 0; // RTCPPacketTypeFlags bit field.
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uint32_t remote_ssrc = 0;
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std::vector<uint16_t> nack_sequence_numbers;
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// TODO(hbos): Remove |report_blocks| in favor of |report_block_datas|.
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ReportBlockList report_blocks;
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std::vector<ReportBlockData> report_block_datas;
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int64_t rtt_ms = 0;
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uint32_t receiver_estimated_max_bitrate_bps = 0;
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std::unique_ptr<rtcp::TransportFeedback> transport_feedback;
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absl::optional<VideoBitrateAllocation> target_bitrate_allocation;
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absl::optional<NetworkStateEstimate> network_state_estimate;
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std::unique_ptr<rtcp::LossNotification> loss_notification;
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};
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// Structure for handing TMMBR and TMMBN rtcp messages (RFC5104, section 3.5.4).
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struct RTCPReceiver::TmmbrInformation {
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struct TimedTmmbrItem {
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rtcp::TmmbItem tmmbr_item;
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int64_t last_updated_ms;
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};
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int64_t last_time_received_ms = 0;
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bool ready_for_delete = false;
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std::vector<rtcp::TmmbItem> tmmbn;
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std::map<uint32_t, TimedTmmbrItem> tmmbr;
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};
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// Structure for storing received RRTR RTCP messages (RFC3611, section 4.4).
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struct RTCPReceiver::RrtrInformation {
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RrtrInformation(uint32_t ssrc,
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uint32_t received_remote_mid_ntp_time,
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uint32_t local_receive_mid_ntp_time)
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: ssrc(ssrc),
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received_remote_mid_ntp_time(received_remote_mid_ntp_time),
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local_receive_mid_ntp_time(local_receive_mid_ntp_time) {}
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uint32_t ssrc;
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// Received NTP timestamp in compact representation.
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uint32_t received_remote_mid_ntp_time;
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// NTP time when the report was received in compact representation.
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uint32_t local_receive_mid_ntp_time;
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};
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struct RTCPReceiver::LastFirStatus {
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LastFirStatus(int64_t now_ms, uint8_t sequence_number)
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: request_ms(now_ms), sequence_number(sequence_number) {}
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int64_t request_ms;
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uint8_t sequence_number;
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};
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RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
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ModuleRtpRtcp* owner)
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: clock_(config.clock),
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receiver_only_(config.receiver_only),
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rtp_rtcp_(owner),
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main_ssrc_(config.local_media_ssrc),
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registered_ssrcs_(GetRegisteredSsrcs(config)),
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rtcp_bandwidth_observer_(config.bandwidth_callback),
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rtcp_intra_frame_observer_(config.intra_frame_callback),
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rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
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network_state_estimate_observer_(config.network_state_estimate_observer),
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transport_feedback_observer_(config.transport_feedback_callback),
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bitrate_allocation_observer_(config.bitrate_allocation_observer),
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report_interval_(config.rtcp_report_interval_ms > 0
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? TimeDelta::Millis(config.rtcp_report_interval_ms)
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: (config.audio ? kDefaultAudioReportInterval
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: kDefaultVideoReportInterval)),
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// TODO(bugs.webrtc.org/10774): Remove fallback.
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remote_ssrc_(0),
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remote_sender_rtp_time_(0),
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xr_rrtr_status_(false),
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xr_rr_rtt_ms_(0),
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oldest_tmmbr_info_ms_(0),
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stats_callback_(config.rtcp_statistics_callback),
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cname_callback_(config.rtcp_cname_callback),
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report_block_data_observer_(config.report_block_data_observer),
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packet_type_counter_observer_(config.rtcp_packet_type_counter_observer),
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num_skipped_packets_(0),
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last_skipped_packets_warning_ms_(clock_->TimeInMilliseconds()) {
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RTC_DCHECK(owner);
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}
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RTCPReceiver::~RTCPReceiver() {}
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void RTCPReceiver::IncomingPacket(rtc::ArrayView<const uint8_t> packet) {
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if (packet.empty()) {
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RTC_LOG(LS_WARNING) << "Incoming empty RTCP packet";
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return;
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}
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PacketInformation packet_information;
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if (!ParseCompoundPacket(packet, &packet_information))
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return;
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TriggerCallbacksFromRtcpPacket(packet_information);
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}
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// This method is only used by test and legacy code, so we should be able to
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// remove it soon.
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int64_t RTCPReceiver::LastReceivedReportBlockMs() const {
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MutexLock lock(&rtcp_receiver_lock_);
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return last_received_rb_.IsFinite() ? last_received_rb_.ms() : 0;
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}
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void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
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MutexLock lock(&rtcp_receiver_lock_);
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// New SSRC reset old reports.
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last_received_sr_ntp_.Reset();
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remote_ssrc_ = ssrc;
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}
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uint32_t RTCPReceiver::RemoteSSRC() const {
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MutexLock lock(&rtcp_receiver_lock_);
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return remote_ssrc_;
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}
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int32_t RTCPReceiver::RTT(uint32_t remote_ssrc,
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int64_t* last_rtt_ms,
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int64_t* avg_rtt_ms,
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int64_t* min_rtt_ms,
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int64_t* max_rtt_ms) const {
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MutexLock lock(&rtcp_receiver_lock_);
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auto it = received_report_blocks_.find(main_ssrc_);
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if (it == received_report_blocks_.end())
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return -1;
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auto it_info = it->second.find(remote_ssrc);
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if (it_info == it->second.end())
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return -1;
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const ReportBlockData* report_block_data = &it_info->second;
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if (report_block_data->num_rtts() == 0)
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return -1;
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if (last_rtt_ms)
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*last_rtt_ms = report_block_data->last_rtt_ms();
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if (avg_rtt_ms) {
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*avg_rtt_ms =
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report_block_data->sum_rtt_ms() / report_block_data->num_rtts();
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}
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if (min_rtt_ms)
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*min_rtt_ms = report_block_data->min_rtt_ms();
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if (max_rtt_ms)
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*max_rtt_ms = report_block_data->max_rtt_ms();
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return 0;
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}
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void RTCPReceiver::SetRtcpXrRrtrStatus(bool enable) {
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MutexLock lock(&rtcp_receiver_lock_);
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xr_rrtr_status_ = enable;
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}
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bool RTCPReceiver::GetAndResetXrRrRtt(int64_t* rtt_ms) {
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RTC_DCHECK(rtt_ms);
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MutexLock lock(&rtcp_receiver_lock_);
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if (xr_rr_rtt_ms_ == 0) {
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return false;
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}
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*rtt_ms = xr_rr_rtt_ms_;
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xr_rr_rtt_ms_ = 0;
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return true;
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}
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// Called regularly (1/sec) on the worker thread to do rtt calculations.
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absl::optional<TimeDelta> RTCPReceiver::OnPeriodicRttUpdate(
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Timestamp newer_than,
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bool sending) {
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// Running on the worker thread (same as construction thread).
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absl::optional<TimeDelta> rtt;
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if (sending) {
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// Check if we've received a report block within the last kRttUpdateInterval
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// amount of time.
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MutexLock lock(&rtcp_receiver_lock_);
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if (last_received_rb_.IsInfinite() || last_received_rb_ > newer_than) {
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// Stow away the report block for the main ssrc. We'll use the associated
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// data map to look up each sender and check the last_rtt_ms().
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auto main_report_it = received_report_blocks_.find(main_ssrc_);
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if (main_report_it != received_report_blocks_.end()) {
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const ReportBlockDataMap& main_data_map = main_report_it->second;
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int64_t max_rtt = 0;
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for (const auto& reports_per_receiver : received_report_blocks_) {
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for (const auto& report : reports_per_receiver.second) {
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const RTCPReportBlock& block = report.second.report_block();
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auto it_info = main_data_map.find(block.sender_ssrc);
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if (it_info != main_data_map.end()) {
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const ReportBlockData* report_block_data = &it_info->second;
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if (report_block_data->num_rtts() > 0) {
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max_rtt = std::max(report_block_data->last_rtt_ms(), max_rtt);
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}
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}
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}
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}
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if (max_rtt)
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rtt.emplace(TimeDelta::Millis(max_rtt));
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}
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}
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// Check for expired timers and if so, log and reset.
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auto now = clock_->CurrentTime();
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if (RtcpRrTimeoutLocked(now)) {
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RTC_LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
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} else if (RtcpRrSequenceNumberTimeoutLocked(now)) {
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RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended "
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"highest sequence number.";
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}
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} else {
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// Report rtt from receiver.
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int64_t rtt_ms;
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if (GetAndResetXrRrRtt(&rtt_ms)) {
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rtt.emplace(TimeDelta::Millis(rtt_ms));
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}
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}
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return rtt;
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}
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bool RTCPReceiver::NTP(uint32_t* received_ntp_secs,
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uint32_t* received_ntp_frac,
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uint32_t* rtcp_arrival_time_secs,
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uint32_t* rtcp_arrival_time_frac,
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uint32_t* rtcp_timestamp) const {
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MutexLock lock(&rtcp_receiver_lock_);
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if (!last_received_sr_ntp_.Valid())
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return false;
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// NTP from incoming SenderReport.
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if (received_ntp_secs)
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*received_ntp_secs = remote_sender_ntp_time_.seconds();
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if (received_ntp_frac)
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*received_ntp_frac = remote_sender_ntp_time_.fractions();
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// Rtp time from incoming SenderReport.
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if (rtcp_timestamp)
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*rtcp_timestamp = remote_sender_rtp_time_;
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// Local NTP time when we received a RTCP packet with a send block.
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if (rtcp_arrival_time_secs)
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*rtcp_arrival_time_secs = last_received_sr_ntp_.seconds();
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if (rtcp_arrival_time_frac)
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*rtcp_arrival_time_frac = last_received_sr_ntp_.fractions();
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return true;
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}
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std::vector<rtcp::ReceiveTimeInfo>
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RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
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MutexLock lock(&rtcp_receiver_lock_);
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const size_t last_xr_rtis_size = std::min(
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received_rrtrs_.size(), rtcp::ExtendedReports::kMaxNumberOfDlrrItems);
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std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis;
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last_xr_rtis.reserve(last_xr_rtis_size);
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const uint32_t now_ntp =
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CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));
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for (size_t i = 0; i < last_xr_rtis_size; ++i) {
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RrtrInformation& rrtr = received_rrtrs_.front();
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last_xr_rtis.emplace_back(rrtr.ssrc, rrtr.received_remote_mid_ntp_time,
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now_ntp - rrtr.local_receive_mid_ntp_time);
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received_rrtrs_ssrc_it_.erase(rrtr.ssrc);
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received_rrtrs_.pop_front();
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}
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return last_xr_rtis;
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}
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// We can get multiple receive reports when we receive the report from a CE.
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int32_t RTCPReceiver::StatisticsReceived(
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||
|
std::vector<RTCPReportBlock>* receive_blocks) const {
|
||
|
RTC_DCHECK(receive_blocks);
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
for (const auto& reports_per_receiver : received_report_blocks_)
|
||
|
for (const auto& report : reports_per_receiver.second)
|
||
|
receive_blocks->push_back(report.second.report_block());
|
||
|
return 0;
|
||
|
}
|
||
|
|
||
|
std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
|
||
|
std::vector<ReportBlockData> result;
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
for (const auto& reports_per_receiver : received_report_blocks_)
|
||
|
for (const auto& report : reports_per_receiver.second)
|
||
|
result.push_back(report.second);
|
||
|
return result;
|
||
|
}
|
||
|
|
||
|
bool RTCPReceiver::ParseCompoundPacket(rtc::ArrayView<const uint8_t> packet,
|
||
|
PacketInformation* packet_information) {
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
|
||
|
CommonHeader rtcp_block;
|
||
|
for (const uint8_t* next_block = packet.begin(); next_block != packet.end();
|
||
|
next_block = rtcp_block.NextPacket()) {
|
||
|
ptrdiff_t remaining_blocks_size = packet.end() - next_block;
|
||
|
RTC_DCHECK_GT(remaining_blocks_size, 0);
|
||
|
if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
|
||
|
if (next_block == packet.begin()) {
|
||
|
// Failed to parse 1st header, nothing was extracted from this packet.
|
||
|
RTC_LOG(LS_WARNING) << "Incoming invalid RTCP packet";
|
||
|
return false;
|
||
|
}
|
||
|
++num_skipped_packets_;
|
||
|
break;
|
||
|
}
|
||
|
|
||
|
if (packet_type_counter_.first_packet_time_ms == -1)
|
||
|
packet_type_counter_.first_packet_time_ms = clock_->TimeInMilliseconds();
|
||
|
|
||
|
switch (rtcp_block.type()) {
|
||
|
case rtcp::SenderReport::kPacketType:
|
||
|
HandleSenderReport(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::ReceiverReport::kPacketType:
|
||
|
HandleReceiverReport(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::Sdes::kPacketType:
|
||
|
HandleSdes(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::ExtendedReports::kPacketType:
|
||
|
HandleXr(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::Bye::kPacketType:
|
||
|
HandleBye(rtcp_block);
|
||
|
break;
|
||
|
case rtcp::App::kPacketType:
|
||
|
HandleApp(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::Rtpfb::kPacketType:
|
||
|
switch (rtcp_block.fmt()) {
|
||
|
case rtcp::Nack::kFeedbackMessageType:
|
||
|
HandleNack(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::Tmmbr::kFeedbackMessageType:
|
||
|
HandleTmmbr(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::Tmmbn::kFeedbackMessageType:
|
||
|
HandleTmmbn(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::RapidResyncRequest::kFeedbackMessageType:
|
||
|
HandleSrReq(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::TransportFeedback::kFeedbackMessageType:
|
||
|
HandleTransportFeedback(rtcp_block, packet_information);
|
||
|
break;
|
||
|
default:
|
||
|
++num_skipped_packets_;
|
||
|
break;
|
||
|
}
|
||
|
break;
|
||
|
case rtcp::Psfb::kPacketType:
|
||
|
switch (rtcp_block.fmt()) {
|
||
|
case rtcp::Pli::kFeedbackMessageType:
|
||
|
HandlePli(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::Fir::kFeedbackMessageType:
|
||
|
HandleFir(rtcp_block, packet_information);
|
||
|
break;
|
||
|
case rtcp::Psfb::kAfbMessageType:
|
||
|
HandlePsfbApp(rtcp_block, packet_information);
|
||
|
break;
|
||
|
default:
|
||
|
++num_skipped_packets_;
|
||
|
break;
|
||
|
}
|
||
|
break;
|
||
|
default:
|
||
|
++num_skipped_packets_;
|
||
|
break;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
if (packet_type_counter_observer_) {
|
||
|
packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
|
||
|
main_ssrc_, packet_type_counter_);
|
||
|
}
|
||
|
|
||
|
if (num_skipped_packets_ > 0) {
|
||
|
const int64_t now_ms = clock_->TimeInMilliseconds();
|
||
|
if (now_ms - last_skipped_packets_warning_ms_ >= kMaxWarningLogIntervalMs) {
|
||
|
last_skipped_packets_warning_ms_ = now_ms;
|
||
|
RTC_LOG(LS_WARNING)
|
||
|
<< num_skipped_packets_
|
||
|
<< " RTCP blocks were skipped due to being malformed or of "
|
||
|
"unrecognized/unsupported type, during the past "
|
||
|
<< (kMaxWarningLogIntervalMs / 1000) << " second period.";
|
||
|
}
|
||
|
}
|
||
|
|
||
|
return true;
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleSenderReport(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::SenderReport sender_report;
|
||
|
if (!sender_report.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
const uint32_t remote_ssrc = sender_report.sender_ssrc();
|
||
|
|
||
|
packet_information->remote_ssrc = remote_ssrc;
|
||
|
|
||
|
UpdateTmmbrRemoteIsAlive(remote_ssrc);
|
||
|
|
||
|
// Have I received RTP packets from this party?
|
||
|
if (remote_ssrc_ == remote_ssrc) {
|
||
|
// Only signal that we have received a SR when we accept one.
|
||
|
packet_information->packet_type_flags |= kRtcpSr;
|
||
|
|
||
|
remote_sender_ntp_time_ = sender_report.ntp();
|
||
|
remote_sender_rtp_time_ = sender_report.rtp_timestamp();
|
||
|
last_received_sr_ntp_ = TimeMicrosToNtp(clock_->TimeInMicroseconds());
|
||
|
} else {
|
||
|
// We will only store the send report from one source, but
|
||
|
// we will store all the receive blocks.
|
||
|
packet_information->packet_type_flags |= kRtcpRr;
|
||
|
}
|
||
|
|
||
|
for (const rtcp::ReportBlock& report_block : sender_report.report_blocks())
|
||
|
HandleReportBlock(report_block, packet_information, remote_ssrc);
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleReceiverReport(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::ReceiverReport receiver_report;
|
||
|
if (!receiver_report.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
const uint32_t remote_ssrc = receiver_report.sender_ssrc();
|
||
|
|
||
|
packet_information->remote_ssrc = remote_ssrc;
|
||
|
|
||
|
UpdateTmmbrRemoteIsAlive(remote_ssrc);
|
||
|
|
||
|
packet_information->packet_type_flags |= kRtcpRr;
|
||
|
|
||
|
for (const ReportBlock& report_block : receiver_report.report_blocks())
|
||
|
HandleReportBlock(report_block, packet_information, remote_ssrc);
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block,
|
||
|
PacketInformation* packet_information,
|
||
|
uint32_t remote_ssrc) {
|
||
|
// This will be called once per report block in the RTCP packet.
|
||
|
// We filter out all report blocks that are not for us.
|
||
|
// Each packet has max 31 RR blocks.
|
||
|
//
|
||
|
// We can calc RTT if we send a send report and get a report block back.
|
||
|
|
||
|
// |report_block.source_ssrc()| is the SSRC identifier of the source to
|
||
|
// which the information in this reception report block pertains.
|
||
|
|
||
|
// Filter out all report blocks that are not for us.
|
||
|
if (registered_ssrcs_.count(report_block.source_ssrc()) == 0)
|
||
|
return;
|
||
|
|
||
|
last_received_rb_ = clock_->CurrentTime();
|
||
|
|
||
|
ReportBlockData* report_block_data =
|
||
|
&received_report_blocks_[report_block.source_ssrc()][remote_ssrc];
|
||
|
RTCPReportBlock rtcp_report_block;
|
||
|
rtcp_report_block.sender_ssrc = remote_ssrc;
|
||
|
rtcp_report_block.source_ssrc = report_block.source_ssrc();
|
||
|
rtcp_report_block.fraction_lost = report_block.fraction_lost();
|
||
|
rtcp_report_block.packets_lost = report_block.cumulative_lost_signed();
|
||
|
if (report_block.extended_high_seq_num() >
|
||
|
report_block_data->report_block().extended_highest_sequence_number) {
|
||
|
// We have successfully delivered new RTP packets to the remote side after
|
||
|
// the last RR was sent from the remote side.
|
||
|
last_increased_sequence_number_ = last_received_rb_;
|
||
|
}
|
||
|
rtcp_report_block.extended_highest_sequence_number =
|
||
|
report_block.extended_high_seq_num();
|
||
|
rtcp_report_block.jitter = report_block.jitter();
|
||
|
rtcp_report_block.delay_since_last_sender_report =
|
||
|
report_block.delay_since_last_sr();
|
||
|
rtcp_report_block.last_sender_report_timestamp = report_block.last_sr();
|
||
|
report_block_data->SetReportBlock(rtcp_report_block, rtc::TimeUTCMicros());
|
||
|
|
||
|
int64_t rtt_ms = 0;
|
||
|
uint32_t send_time_ntp = report_block.last_sr();
|
||
|
// RFC3550, section 6.4.1, LSR field discription states:
|
||
|
// If no SR has been received yet, the field is set to zero.
|
||
|
// Receiver rtp_rtcp module is not expected to calculate rtt using
|
||
|
// Sender Reports even if it accidentally can.
|
||
|
|
||
|
// TODO(nisse): Use this way to determine the RTT only when |receiver_only_|
|
||
|
// is false. However, that currently breaks the tests of the
|
||
|
// googCaptureStartNtpTimeMs stat for audio receive streams. To fix, either
|
||
|
// delete all dependencies on RTT measurements for audio receive streams, or
|
||
|
// ensure that audio receive streams that need RTT and stats that depend on it
|
||
|
// are configured with an associated audio send stream.
|
||
|
if (send_time_ntp != 0) {
|
||
|
uint32_t delay_ntp = report_block.delay_since_last_sr();
|
||
|
// Local NTP time.
|
||
|
uint32_t receive_time_ntp =
|
||
|
CompactNtp(TimeMicrosToNtp(last_received_rb_.us()));
|
||
|
|
||
|
// RTT in 1/(2^16) seconds.
|
||
|
uint32_t rtt_ntp = receive_time_ntp - delay_ntp - send_time_ntp;
|
||
|
// Convert to 1/1000 seconds (milliseconds).
|
||
|
rtt_ms = CompactNtpRttToMs(rtt_ntp);
|
||
|
report_block_data->AddRoundTripTimeSample(rtt_ms);
|
||
|
|
||
|
packet_information->rtt_ms = rtt_ms;
|
||
|
}
|
||
|
|
||
|
packet_information->report_blocks.push_back(
|
||
|
report_block_data->report_block());
|
||
|
packet_information->report_block_datas.push_back(*report_block_data);
|
||
|
}
|
||
|
|
||
|
RTCPReceiver::TmmbrInformation* RTCPReceiver::FindOrCreateTmmbrInfo(
|
||
|
uint32_t remote_ssrc) {
|
||
|
// Create or find receive information.
|
||
|
TmmbrInformation* tmmbr_info = &tmmbr_infos_[remote_ssrc];
|
||
|
// Update that this remote is alive.
|
||
|
tmmbr_info->last_time_received_ms = clock_->TimeInMilliseconds();
|
||
|
return tmmbr_info;
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::UpdateTmmbrRemoteIsAlive(uint32_t remote_ssrc) {
|
||
|
auto tmmbr_it = tmmbr_infos_.find(remote_ssrc);
|
||
|
if (tmmbr_it != tmmbr_infos_.end())
|
||
|
tmmbr_it->second.last_time_received_ms = clock_->TimeInMilliseconds();
|
||
|
}
|
||
|
|
||
|
RTCPReceiver::TmmbrInformation* RTCPReceiver::GetTmmbrInformation(
|
||
|
uint32_t remote_ssrc) {
|
||
|
auto it = tmmbr_infos_.find(remote_ssrc);
|
||
|
if (it == tmmbr_infos_.end())
|
||
|
return nullptr;
|
||
|
return &it->second;
|
||
|
}
|
||
|
|
||
|
// These two methods (RtcpRrTimeout and RtcpRrSequenceNumberTimeout) only exist
|
||
|
// for tests and legacy code (rtp_rtcp_impl.cc). We should be able to to delete
|
||
|
// the methods and require that access to the locked variables only happens on
|
||
|
// the worker thread and thus no locking is needed.
|
||
|
bool RTCPReceiver::RtcpRrTimeout() {
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
return RtcpRrTimeoutLocked(clock_->CurrentTime());
|
||
|
}
|
||
|
|
||
|
bool RTCPReceiver::RtcpRrSequenceNumberTimeout() {
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
return RtcpRrSequenceNumberTimeoutLocked(clock_->CurrentTime());
|
||
|
}
|
||
|
|
||
|
bool RTCPReceiver::UpdateTmmbrTimers() {
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
|
||
|
int64_t now_ms = clock_->TimeInMilliseconds();
|
||
|
int64_t timeout_ms = now_ms - kTmmbrTimeoutIntervalMs;
|
||
|
|
||
|
if (oldest_tmmbr_info_ms_ >= timeout_ms)
|
||
|
return false;
|
||
|
|
||
|
bool update_bounding_set = false;
|
||
|
oldest_tmmbr_info_ms_ = -1;
|
||
|
for (auto tmmbr_it = tmmbr_infos_.begin(); tmmbr_it != tmmbr_infos_.end();) {
|
||
|
TmmbrInformation* tmmbr_info = &tmmbr_it->second;
|
||
|
if (tmmbr_info->last_time_received_ms > 0) {
|
||
|
if (tmmbr_info->last_time_received_ms < timeout_ms) {
|
||
|
// No rtcp packet for the last 5 regular intervals, reset limitations.
|
||
|
tmmbr_info->tmmbr.clear();
|
||
|
// Prevent that we call this over and over again.
|
||
|
tmmbr_info->last_time_received_ms = 0;
|
||
|
// Send new TMMBN to all channels using the default codec.
|
||
|
update_bounding_set = true;
|
||
|
} else if (oldest_tmmbr_info_ms_ == -1 ||
|
||
|
tmmbr_info->last_time_received_ms < oldest_tmmbr_info_ms_) {
|
||
|
oldest_tmmbr_info_ms_ = tmmbr_info->last_time_received_ms;
|
||
|
}
|
||
|
++tmmbr_it;
|
||
|
} else if (tmmbr_info->ready_for_delete) {
|
||
|
// When we dont have a last_time_received_ms and the object is marked
|
||
|
// ready_for_delete it's removed from the map.
|
||
|
tmmbr_it = tmmbr_infos_.erase(tmmbr_it);
|
||
|
} else {
|
||
|
++tmmbr_it;
|
||
|
}
|
||
|
}
|
||
|
return update_bounding_set;
|
||
|
}
|
||
|
|
||
|
std::vector<rtcp::TmmbItem> RTCPReceiver::BoundingSet(bool* tmmbr_owner) {
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
TmmbrInformation* tmmbr_info = GetTmmbrInformation(remote_ssrc_);
|
||
|
if (!tmmbr_info)
|
||
|
return std::vector<rtcp::TmmbItem>();
|
||
|
|
||
|
*tmmbr_owner = TMMBRHelp::IsOwner(tmmbr_info->tmmbn, main_ssrc_);
|
||
|
return tmmbr_info->tmmbn;
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleSdes(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::Sdes sdes;
|
||
|
if (!sdes.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
for (const rtcp::Sdes::Chunk& chunk : sdes.chunks()) {
|
||
|
received_cnames_[chunk.ssrc] = chunk.cname;
|
||
|
if (cname_callback_)
|
||
|
cname_callback_->OnCname(chunk.ssrc, chunk.cname);
|
||
|
}
|
||
|
packet_information->packet_type_flags |= kRtcpSdes;
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleNack(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::Nack nack;
|
||
|
if (!nack.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
if (receiver_only_ || main_ssrc_ != nack.media_ssrc()) // Not to us.
|
||
|
return;
|
||
|
|
||
|
packet_information->nack_sequence_numbers.insert(
|
||
|
packet_information->nack_sequence_numbers.end(),
|
||
|
nack.packet_ids().begin(), nack.packet_ids().end());
|
||
|
for (uint16_t packet_id : nack.packet_ids())
|
||
|
nack_stats_.ReportRequest(packet_id);
|
||
|
|
||
|
if (!nack.packet_ids().empty()) {
|
||
|
packet_information->packet_type_flags |= kRtcpNack;
|
||
|
++packet_type_counter_.nack_packets;
|
||
|
packet_type_counter_.nack_requests = nack_stats_.requests();
|
||
|
packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
|
||
|
}
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleApp(const rtcp::CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::App app;
|
||
|
if (app.Parse(rtcp_block)) {
|
||
|
if (app.name() == rtcp::RemoteEstimate::kName &&
|
||
|
app.sub_type() == rtcp::RemoteEstimate::kSubType) {
|
||
|
rtcp::RemoteEstimate estimate(std::move(app));
|
||
|
if (estimate.ParseData()) {
|
||
|
packet_information->network_state_estimate = estimate.estimate();
|
||
|
return;
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
++num_skipped_packets_;
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleBye(const CommonHeader& rtcp_block) {
|
||
|
rtcp::Bye bye;
|
||
|
if (!bye.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
// Clear our lists.
|
||
|
for (auto& reports_per_receiver : received_report_blocks_)
|
||
|
reports_per_receiver.second.erase(bye.sender_ssrc());
|
||
|
|
||
|
TmmbrInformation* tmmbr_info = GetTmmbrInformation(bye.sender_ssrc());
|
||
|
if (tmmbr_info)
|
||
|
tmmbr_info->ready_for_delete = true;
|
||
|
|
||
|
last_fir_.erase(bye.sender_ssrc());
|
||
|
received_cnames_.erase(bye.sender_ssrc());
|
||
|
auto it = received_rrtrs_ssrc_it_.find(bye.sender_ssrc());
|
||
|
if (it != received_rrtrs_ssrc_it_.end()) {
|
||
|
received_rrtrs_.erase(it->second);
|
||
|
received_rrtrs_ssrc_it_.erase(it);
|
||
|
}
|
||
|
xr_rr_rtt_ms_ = 0;
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleXr(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::ExtendedReports xr;
|
||
|
if (!xr.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
if (xr.rrtr())
|
||
|
HandleXrReceiveReferenceTime(xr.sender_ssrc(), *xr.rrtr());
|
||
|
|
||
|
for (const rtcp::ReceiveTimeInfo& time_info : xr.dlrr().sub_blocks())
|
||
|
HandleXrDlrrReportBlock(time_info);
|
||
|
|
||
|
if (xr.target_bitrate()) {
|
||
|
HandleXrTargetBitrate(xr.sender_ssrc(), *xr.target_bitrate(),
|
||
|
packet_information);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleXrReceiveReferenceTime(uint32_t sender_ssrc,
|
||
|
const rtcp::Rrtr& rrtr) {
|
||
|
uint32_t received_remote_mid_ntp_time = CompactNtp(rrtr.ntp());
|
||
|
uint32_t local_receive_mid_ntp_time =
|
||
|
CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));
|
||
|
|
||
|
auto it = received_rrtrs_ssrc_it_.find(sender_ssrc);
|
||
|
if (it != received_rrtrs_ssrc_it_.end()) {
|
||
|
it->second->received_remote_mid_ntp_time = received_remote_mid_ntp_time;
|
||
|
it->second->local_receive_mid_ntp_time = local_receive_mid_ntp_time;
|
||
|
} else {
|
||
|
if (received_rrtrs_.size() < kMaxNumberOfStoredRrtrs) {
|
||
|
received_rrtrs_.emplace_back(sender_ssrc, received_remote_mid_ntp_time,
|
||
|
local_receive_mid_ntp_time);
|
||
|
received_rrtrs_ssrc_it_[sender_ssrc] = std::prev(received_rrtrs_.end());
|
||
|
} else {
|
||
|
RTC_LOG(LS_WARNING) << "Discarding received RRTR for ssrc " << sender_ssrc
|
||
|
<< ", reached maximum number of stored RRTRs.";
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleXrDlrrReportBlock(const rtcp::ReceiveTimeInfo& rti) {
|
||
|
if (registered_ssrcs_.count(rti.ssrc) == 0) // Not to us.
|
||
|
return;
|
||
|
|
||
|
// Caller should explicitly enable rtt calculation using extended reports.
|
||
|
if (!xr_rrtr_status_)
|
||
|
return;
|
||
|
|
||
|
// The send_time and delay_rr fields are in units of 1/2^16 sec.
|
||
|
uint32_t send_time_ntp = rti.last_rr;
|
||
|
// RFC3611, section 4.5, LRR field discription states:
|
||
|
// If no such block has been received, the field is set to zero.
|
||
|
if (send_time_ntp == 0)
|
||
|
return;
|
||
|
|
||
|
uint32_t delay_ntp = rti.delay_since_last_rr;
|
||
|
uint32_t now_ntp = CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));
|
||
|
|
||
|
uint32_t rtt_ntp = now_ntp - delay_ntp - send_time_ntp;
|
||
|
xr_rr_rtt_ms_ = CompactNtpRttToMs(rtt_ntp);
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleXrTargetBitrate(
|
||
|
uint32_t ssrc,
|
||
|
const rtcp::TargetBitrate& target_bitrate,
|
||
|
PacketInformation* packet_information) {
|
||
|
if (ssrc != remote_ssrc_) {
|
||
|
return; // Not for us.
|
||
|
}
|
||
|
|
||
|
VideoBitrateAllocation bitrate_allocation;
|
||
|
for (const auto& item : target_bitrate.GetTargetBitrates()) {
|
||
|
if (item.spatial_layer >= kMaxSpatialLayers ||
|
||
|
item.temporal_layer >= kMaxTemporalStreams) {
|
||
|
RTC_LOG(LS_WARNING)
|
||
|
<< "Invalid layer in XR target bitrate pack: spatial index "
|
||
|
<< item.spatial_layer << ", temporal index " << item.temporal_layer
|
||
|
<< ", dropping.";
|
||
|
} else {
|
||
|
bitrate_allocation.SetBitrate(item.spatial_layer, item.temporal_layer,
|
||
|
item.target_bitrate_kbps * 1000);
|
||
|
}
|
||
|
}
|
||
|
packet_information->target_bitrate_allocation.emplace(bitrate_allocation);
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandlePli(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::Pli pli;
|
||
|
if (!pli.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
if (main_ssrc_ == pli.media_ssrc()) {
|
||
|
++packet_type_counter_.pli_packets;
|
||
|
// Received a signal that we need to send a new key frame.
|
||
|
packet_information->packet_type_flags |= kRtcpPli;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleTmmbr(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::Tmmbr tmmbr;
|
||
|
if (!tmmbr.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
uint32_t sender_ssrc = tmmbr.sender_ssrc();
|
||
|
if (tmmbr.media_ssrc()) {
|
||
|
// media_ssrc() SHOULD be 0 if same as SenderSSRC.
|
||
|
// In relay mode this is a valid number.
|
||
|
sender_ssrc = tmmbr.media_ssrc();
|
||
|
}
|
||
|
|
||
|
for (const rtcp::TmmbItem& request : tmmbr.requests()) {
|
||
|
if (main_ssrc_ != request.ssrc() || request.bitrate_bps() == 0)
|
||
|
continue;
|
||
|
|
||
|
TmmbrInformation* tmmbr_info = FindOrCreateTmmbrInfo(tmmbr.sender_ssrc());
|
||
|
auto* entry = &tmmbr_info->tmmbr[sender_ssrc];
|
||
|
entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc, request.bitrate_bps(),
|
||
|
request.packet_overhead());
|
||
|
// FindOrCreateTmmbrInfo always sets |last_time_received_ms| to
|
||
|
// |clock_->TimeInMilliseconds()|.
|
||
|
entry->last_updated_ms = tmmbr_info->last_time_received_ms;
|
||
|
|
||
|
packet_information->packet_type_flags |= kRtcpTmmbr;
|
||
|
break;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleTmmbn(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::Tmmbn tmmbn;
|
||
|
if (!tmmbn.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
TmmbrInformation* tmmbr_info = FindOrCreateTmmbrInfo(tmmbn.sender_ssrc());
|
||
|
|
||
|
packet_information->packet_type_flags |= kRtcpTmmbn;
|
||
|
|
||
|
tmmbr_info->tmmbn = tmmbn.items();
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleSrReq(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::RapidResyncRequest sr_req;
|
||
|
if (!sr_req.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
packet_information->packet_type_flags |= kRtcpSrReq;
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandlePsfbApp(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
{
|
||
|
rtcp::Remb remb;
|
||
|
if (remb.Parse(rtcp_block)) {
|
||
|
packet_information->packet_type_flags |= kRtcpRemb;
|
||
|
packet_information->receiver_estimated_max_bitrate_bps =
|
||
|
remb.bitrate_bps();
|
||
|
return;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
{
|
||
|
auto loss_notification = std::make_unique<rtcp::LossNotification>();
|
||
|
if (loss_notification->Parse(rtcp_block)) {
|
||
|
packet_information->packet_type_flags |= kRtcpLossNotification;
|
||
|
packet_information->loss_notification = std::move(loss_notification);
|
||
|
return;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
RTC_LOG(LS_WARNING) << "Unknown PSFB-APP packet.";
|
||
|
|
||
|
++num_skipped_packets_;
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleFir(const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
rtcp::Fir fir;
|
||
|
if (!fir.Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
if (fir.requests().empty())
|
||
|
return;
|
||
|
|
||
|
const int64_t now_ms = clock_->TimeInMilliseconds();
|
||
|
for (const rtcp::Fir::Request& fir_request : fir.requests()) {
|
||
|
// Is it our sender that is requested to generate a new keyframe.
|
||
|
if (main_ssrc_ != fir_request.ssrc)
|
||
|
continue;
|
||
|
|
||
|
++packet_type_counter_.fir_packets;
|
||
|
|
||
|
auto inserted = last_fir_.insert(std::make_pair(
|
||
|
fir.sender_ssrc(), LastFirStatus(now_ms, fir_request.seq_nr)));
|
||
|
if (!inserted.second) { // There was already an entry.
|
||
|
LastFirStatus* last_fir = &inserted.first->second;
|
||
|
|
||
|
// Check if we have reported this FIRSequenceNumber before.
|
||
|
if (fir_request.seq_nr == last_fir->sequence_number)
|
||
|
continue;
|
||
|
|
||
|
// Sanity: don't go crazy with the callbacks.
|
||
|
if (now_ms - last_fir->request_ms < kRtcpMinFrameLengthMs)
|
||
|
continue;
|
||
|
|
||
|
last_fir->request_ms = now_ms;
|
||
|
last_fir->sequence_number = fir_request.seq_nr;
|
||
|
}
|
||
|
// Received signal that we need to send a new key frame.
|
||
|
packet_information->packet_type_flags |= kRtcpFir;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::HandleTransportFeedback(
|
||
|
const CommonHeader& rtcp_block,
|
||
|
PacketInformation* packet_information) {
|
||
|
std::unique_ptr<rtcp::TransportFeedback> transport_feedback(
|
||
|
new rtcp::TransportFeedback());
|
||
|
if (!transport_feedback->Parse(rtcp_block)) {
|
||
|
++num_skipped_packets_;
|
||
|
return;
|
||
|
}
|
||
|
|
||
|
packet_information->packet_type_flags |= kRtcpTransportFeedback;
|
||
|
packet_information->transport_feedback = std::move(transport_feedback);
|
||
|
}
|
||
|
|
||
|
void RTCPReceiver::NotifyTmmbrUpdated() {
|
||
|
// Find bounding set.
|
||
|
std::vector<rtcp::TmmbItem> bounding =
|
||
|
TMMBRHelp::FindBoundingSet(TmmbrReceived());
|
||
|
|
||
|
if (!bounding.empty() && rtcp_bandwidth_observer_) {
|
||
|
// We have a new bandwidth estimate on this channel.
|
||
|
uint64_t bitrate_bps = TMMBRHelp::CalcMinBitrateBps(bounding);
|
||
|
if (bitrate_bps <= std::numeric_limits<uint32_t>::max())
|
||
|
rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate_bps);
|
||
|
}
|
||
|
|
||
|
// Send tmmbn to inform remote clients about the new bandwidth.
|
||
|
rtp_rtcp_->SetTmmbn(std::move(bounding));
|
||
|
}
|
||
|
|
||
|
// Holding no Critical section.
|
||
|
void RTCPReceiver::TriggerCallbacksFromRtcpPacket(
|
||
|
const PacketInformation& packet_information) {
|
||
|
// Process TMMBR and REMB first to avoid multiple callbacks
|
||
|
// to OnNetworkChanged.
|
||
|
if (packet_information.packet_type_flags & kRtcpTmmbr) {
|
||
|
// Might trigger a OnReceivedBandwidthEstimateUpdate.
|
||
|
NotifyTmmbrUpdated();
|
||
|
}
|
||
|
uint32_t local_ssrc;
|
||
|
std::set<uint32_t> registered_ssrcs;
|
||
|
{
|
||
|
// We don't want to hold this critsect when triggering the callbacks below.
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
local_ssrc = main_ssrc_;
|
||
|
registered_ssrcs = registered_ssrcs_;
|
||
|
}
|
||
|
if (!receiver_only_ && (packet_information.packet_type_flags & kRtcpSrReq)) {
|
||
|
rtp_rtcp_->OnRequestSendReport();
|
||
|
}
|
||
|
if (!receiver_only_ && (packet_information.packet_type_flags & kRtcpNack)) {
|
||
|
if (!packet_information.nack_sequence_numbers.empty()) {
|
||
|
RTC_LOG(LS_VERBOSE) << "Incoming NACK length: "
|
||
|
<< packet_information.nack_sequence_numbers.size();
|
||
|
rtp_rtcp_->OnReceivedNack(packet_information.nack_sequence_numbers);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
// We need feedback that we have received a report block(s) so that we
|
||
|
// can generate a new packet in a conference relay scenario, one received
|
||
|
// report can generate several RTCP packets, based on number relayed/mixed
|
||
|
// a send report block should go out to all receivers.
|
||
|
if (rtcp_intra_frame_observer_) {
|
||
|
RTC_DCHECK(!receiver_only_);
|
||
|
if ((packet_information.packet_type_flags & kRtcpPli) ||
|
||
|
(packet_information.packet_type_flags & kRtcpFir)) {
|
||
|
if (packet_information.packet_type_flags & kRtcpPli) {
|
||
|
RTC_LOG(LS_VERBOSE)
|
||
|
<< "Incoming PLI from SSRC " << packet_information.remote_ssrc;
|
||
|
} else {
|
||
|
RTC_LOG(LS_VERBOSE)
|
||
|
<< "Incoming FIR from SSRC " << packet_information.remote_ssrc;
|
||
|
}
|
||
|
rtcp_intra_frame_observer_->OnReceivedIntraFrameRequest(local_ssrc);
|
||
|
}
|
||
|
}
|
||
|
if (rtcp_loss_notification_observer_ &&
|
||
|
(packet_information.packet_type_flags & kRtcpLossNotification)) {
|
||
|
rtcp::LossNotification* loss_notification =
|
||
|
packet_information.loss_notification.get();
|
||
|
RTC_DCHECK(loss_notification);
|
||
|
if (loss_notification->media_ssrc() == local_ssrc) {
|
||
|
rtcp_loss_notification_observer_->OnReceivedLossNotification(
|
||
|
loss_notification->media_ssrc(), loss_notification->last_decoded(),
|
||
|
loss_notification->last_received(),
|
||
|
loss_notification->decodability_flag());
|
||
|
}
|
||
|
}
|
||
|
if (rtcp_bandwidth_observer_) {
|
||
|
RTC_DCHECK(!receiver_only_);
|
||
|
if (packet_information.packet_type_flags & kRtcpRemb) {
|
||
|
RTC_LOG(LS_VERBOSE)
|
||
|
<< "Incoming REMB: "
|
||
|
<< packet_information.receiver_estimated_max_bitrate_bps;
|
||
|
rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(
|
||
|
packet_information.receiver_estimated_max_bitrate_bps);
|
||
|
}
|
||
|
if ((packet_information.packet_type_flags & kRtcpSr) ||
|
||
|
(packet_information.packet_type_flags & kRtcpRr)) {
|
||
|
int64_t now_ms = clock_->TimeInMilliseconds();
|
||
|
rtcp_bandwidth_observer_->OnReceivedRtcpReceiverReport(
|
||
|
packet_information.report_blocks, packet_information.rtt_ms, now_ms);
|
||
|
}
|
||
|
}
|
||
|
if ((packet_information.packet_type_flags & kRtcpSr) ||
|
||
|
(packet_information.packet_type_flags & kRtcpRr)) {
|
||
|
rtp_rtcp_->OnReceivedRtcpReportBlocks(packet_information.report_blocks);
|
||
|
}
|
||
|
|
||
|
if (transport_feedback_observer_ &&
|
||
|
(packet_information.packet_type_flags & kRtcpTransportFeedback)) {
|
||
|
uint32_t media_source_ssrc =
|
||
|
packet_information.transport_feedback->media_ssrc();
|
||
|
if (media_source_ssrc == local_ssrc ||
|
||
|
registered_ssrcs.find(media_source_ssrc) != registered_ssrcs.end()) {
|
||
|
transport_feedback_observer_->OnTransportFeedback(
|
||
|
*packet_information.transport_feedback);
|
||
|
}
|
||
|
}
|
||
|
|
||
|
if (network_state_estimate_observer_ &&
|
||
|
packet_information.network_state_estimate) {
|
||
|
network_state_estimate_observer_->OnRemoteNetworkEstimate(
|
||
|
*packet_information.network_state_estimate);
|
||
|
}
|
||
|
|
||
|
if (bitrate_allocation_observer_ &&
|
||
|
packet_information.target_bitrate_allocation) {
|
||
|
bitrate_allocation_observer_->OnBitrateAllocationUpdated(
|
||
|
*packet_information.target_bitrate_allocation);
|
||
|
}
|
||
|
|
||
|
if (!receiver_only_) {
|
||
|
if (stats_callback_) {
|
||
|
for (const auto& report_block : packet_information.report_blocks) {
|
||
|
RtcpStatistics stats;
|
||
|
stats.packets_lost = report_block.packets_lost;
|
||
|
stats.extended_highest_sequence_number =
|
||
|
report_block.extended_highest_sequence_number;
|
||
|
stats.fraction_lost = report_block.fraction_lost;
|
||
|
stats.jitter = report_block.jitter;
|
||
|
|
||
|
stats_callback_->StatisticsUpdated(stats, report_block.source_ssrc);
|
||
|
}
|
||
|
}
|
||
|
if (report_block_data_observer_) {
|
||
|
for (const auto& report_block_data :
|
||
|
packet_information.report_block_datas) {
|
||
|
report_block_data_observer_->OnReportBlockDataUpdated(
|
||
|
report_block_data);
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
|
||
|
int32_t RTCPReceiver::CNAME(uint32_t remoteSSRC,
|
||
|
char cName[RTCP_CNAME_SIZE]) const {
|
||
|
RTC_DCHECK(cName);
|
||
|
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
auto received_cname_it = received_cnames_.find(remoteSSRC);
|
||
|
if (received_cname_it == received_cnames_.end())
|
||
|
return -1;
|
||
|
|
||
|
size_t length = received_cname_it->second.copy(cName, RTCP_CNAME_SIZE - 1);
|
||
|
cName[length] = 0;
|
||
|
return 0;
|
||
|
}
|
||
|
|
||
|
std::vector<rtcp::TmmbItem> RTCPReceiver::TmmbrReceived() {
|
||
|
MutexLock lock(&rtcp_receiver_lock_);
|
||
|
std::vector<rtcp::TmmbItem> candidates;
|
||
|
|
||
|
int64_t now_ms = clock_->TimeInMilliseconds();
|
||
|
int64_t timeout_ms = now_ms - kTmmbrTimeoutIntervalMs;
|
||
|
|
||
|
for (auto& kv : tmmbr_infos_) {
|
||
|
for (auto it = kv.second.tmmbr.begin(); it != kv.second.tmmbr.end();) {
|
||
|
if (it->second.last_updated_ms < timeout_ms) {
|
||
|
// Erase timeout entries.
|
||
|
it = kv.second.tmmbr.erase(it);
|
||
|
} else {
|
||
|
candidates.push_back(it->second.tmmbr_item);
|
||
|
++it;
|
||
|
}
|
||
|
}
|
||
|
}
|
||
|
return candidates;
|
||
|
}
|
||
|
|
||
|
bool RTCPReceiver::RtcpRrTimeoutLocked(Timestamp now) {
|
||
|
return ResetTimestampIfExpired(now, last_received_rb_, report_interval_);
|
||
|
}
|
||
|
|
||
|
bool RTCPReceiver::RtcpRrSequenceNumberTimeoutLocked(Timestamp now) {
|
||
|
return ResetTimestampIfExpired(now, last_increased_sequence_number_,
|
||
|
report_interval_);
|
||
|
}
|
||
|
|
||
|
} // namespace webrtc
|