blob: c9ecde3be8b6d80acc6e2937058106221e421d58 [file] [log] [blame]
* Copyright (c) 2019 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 <map>
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/units/data_rate.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/rate_statistics.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class RtpSenderEgress {
// Helper class that redirects packets directly to the send part of this class
// without passing through an actual paced sender.
class NonPacedPacketSender : public RtpPacketSender {
explicit NonPacedPacketSender(RtpSenderEgress* sender);
virtual ~NonPacedPacketSender();
void EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;
uint16_t transport_sequence_number_;
RtpSenderEgress* const sender_;
RtpSenderEgress(const RtpRtcp::Configuration& config,
RtpPacketHistory* packet_history);
~RtpSenderEgress() = default;
void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info)
uint32_t Ssrc() const { return ssrc_; }
absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
void ProcessBitrateAndNotifyObservers() RTC_LOCKS_EXCLUDED(lock_);
RtpSendRates GetSendRates() const RTC_LOCKS_EXCLUDED(lock_);
void GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const
void ForceIncludeSendPacketsInAllocation(bool part_of_allocation)
bool MediaHasBeenSent() const RTC_LOCKS_EXCLUDED(lock_);
void SetMediaHasBeenSent(bool media_sent) RTC_LOCKS_EXCLUDED(lock_);
void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(lock_);
// For each sequence number in |sequence_number|, recall the last RTP packet
// which bore it - its timestamp and whether it was the first and/or last
// packet in that frame. If all of the given sequence numbers could be
// recalled, return a vector with all of them (in corresponding order).
// If any could not be recalled, return an empty vector.
std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
rtc::ArrayView<const uint16_t> sequence_numbers) const
// Maps capture time in milliseconds to send-side delay in milliseconds.
// Send-side delay is the difference between transmission time and capture
// time.
typedef std::map<int64_t, int> SendDelayMap;
RtpSendRates GetSendRatesLocked() const RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
bool HasCorrectSsrc(const RtpPacketToSend& packet) const;
void AddPacketToTransportFeedback(uint16_t packet_id,
const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info);
void UpdateDelayStatistics(int64_t capture_time_ms,
int64_t now_ms,
uint32_t ssrc);
void RecomputeMaxSendDelay() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
void UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc);
// Sends packet on to |transport_|, leaving the RTP module.
bool SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info);
void UpdateRtpStats(const RtpPacketToSend& packet)
const uint32_t ssrc_;
const absl::optional<uint32_t> rtx_ssrc_;
const absl::optional<uint32_t> flexfec_ssrc_;
const bool populate_network2_timestamp_;
const bool send_side_bwe_with_overhead_;
Clock* const clock_;
RtpPacketHistory* const packet_history_;
Transport* const transport_;
RtcEventLog* const event_log_;
const bool is_audio_;
const bool need_rtp_packet_infos_;
TransportFeedbackObserver* const transport_feedback_observer_;
SendSideDelayObserver* const send_side_delay_observer_;
SendPacketObserver* const send_packet_observer_;
StreamDataCountersCallback* const rtp_stats_callback_;
BitrateStatisticsObserver* const bitrate_callback_;
rtc::CriticalSection lock_;
bool media_has_been_sent_ RTC_GUARDED_BY(lock_);
bool force_part_of_allocation_ RTC_GUARDED_BY(lock_);
uint32_t timestamp_offset_ RTC_GUARDED_BY(lock_);
SendDelayMap send_delays_ RTC_GUARDED_BY(lock_);
SendDelayMap::const_iterator max_delay_it_ RTC_GUARDED_BY(lock_);
// The sum of delays over a kSendSideDelayWindowMs sliding window.
int64_t sum_delays_ms_ RTC_GUARDED_BY(lock_);
uint64_t total_packet_send_delay_ms_ RTC_GUARDED_BY(lock_);
StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_);
StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
// One element per value in RtpPacketMediaType, with index matching value.
std::vector<RateStatistics> send_rates_ RTC_GUARDED_BY(lock_);
// Maps sent packets' sequence numbers to a tuple consisting of:
// 1. The timestamp, without the randomizing offset mandated by the RFC.
// 2. Whether the packet was the first in its frame.
// 3. Whether the packet was the last in its frame.
const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_
} // namespace webrtc