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webrtc/modules/rtp_rtcp/source/rtp_sender.h
Steve Anton 2bac7da134 Optimize sending the MID and (R)RID header extensions 
These RTP header extensions are used for Unified Plan SDP / BUNDLE and
replace SSRC signaling.

Previously, the RTPSender would attach these header extensions to every
packet when configured. Now, the header extensions will be attached to
every packet until the an RTCP RR is received on that SSRC which
indicates the receiver knows what MID/RID the SSRC is associated with.

This should reduce overhead by 2-4 bytes per packet when the MID header
extension is used and by 4-8 bytes when both header extensions are used.

Bug: webrtc:10078
Change-Id: I5fa3ce28a75224adf11d2792bf4ff8dc76e46d99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146480
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28685}
2019-07-25 19:23:14 +00:00

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/*
* Copyright (c) 2012 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.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/transport/webrtc_key_value_config.h"
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.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_rtcp_config.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/random.h"
#include "rtc_base/rate_statistics.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class FrameEncryptorInterface;
class OverheadObserver;
class RateLimiter;
class RtcEventLog;
class RtpPacketToSend;
class RTPSender {
public:
explicit RTPSender(const RtpRtcp::Configuration& config);
// TODO(bugs.webrtc.org/10774): Remove once downstream projects are fixed.
RTPSender(bool audio,
Clock* clock,
Transport* transport,
RtpPacketSender* paced_sender,
absl::optional<uint32_t> flexfec_ssrc,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_callback,
BitrateStatisticsObserver* bitrate_callback,
SendSideDelayObserver* send_side_delay_observer,
RtcEventLog* event_log,
SendPacketObserver* send_packet_observer,
RateLimiter* nack_rate_limiter,
OverheadObserver* overhead_observer,
bool populate_network2_timestamp,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
bool extmap_allow_mixed,
const WebRtcKeyValueConfig& field_trials);
~RTPSender();
void ProcessBitrate();
uint16_t ActualSendBitrateKbit() const;
uint32_t NackOverheadRate() const;
void SetSendingMediaStatus(bool enabled);
bool SendingMedia() const;
void SetAsPartOfAllocation(bool part_of_allocation);
void GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const;
uint32_t TimestampOffset() const;
void SetTimestampOffset(uint32_t timestamp);
// TODO(bugs.webrtc.org/10774): Remove.
void SetSSRC(uint32_t ssrc);
void SetRid(const std::string& rid);
void SetMid(const std::string& mid);
uint16_t SequenceNumber() const;
void SetSequenceNumber(uint16_t seq);
void SetCsrcs(const std::vector<uint32_t>& csrcs);
void SetMaxRtpPacketSize(size_t max_packet_size);
void SetExtmapAllowMixed(bool extmap_allow_mixed);
// RTP header extension
int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
bool RegisterRtpHeaderExtension(const std::string& uri, int id);
bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) const;
int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
// Returns an RtpPacketSendResult indicating success, network unavailable,
// or packet not found.
RtpPacketSendResult TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info);
bool TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info);
bool SupportsPadding() const;
bool SupportsRtxPayloadPadding() const;
size_t TimeToSendPadding(size_t bytes, const PacedPacketInfo& pacing_info);
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes);
// NACK.
void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt);
void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
bool StorePackets() const;
int32_t ReSendPacket(uint16_t packet_id);
// ACK.
void OnReceivedAckOnSsrc(int64_t extended_highest_sequence_number);
void OnReceivedAckOnRtxSsrc(int64_t extended_highest_sequence_number);
// RTX.
void SetRtxStatus(int mode);
int RtxStatus() const;
uint32_t RtxSsrc() const;
// TODO(bugs.webrtc.org/10774): Remove.
void SetRtxSsrc(uint32_t ssrc);
void SetRtxPayloadType(int payload_type, int associated_payload_type);
// Size info for header extensions used by FEC packets.
static rtc::ArrayView<const RtpExtensionSize> FecExtensionSizes();
// Size info for header extensions used by video packets.
static rtc::ArrayView<const RtpExtensionSize> VideoExtensionSizes();
// Create empty packet, fills ssrc, csrcs and reserve place for header
// extensions RtpSender updates before sending.
std::unique_ptr<RtpPacketToSend> AllocatePacket() const;
// Allocate sequence number for provided packet.
// Save packet's fields to generate padding that doesn't break media stream.
// Return false if sending was turned off.
bool AssignSequenceNumber(RtpPacketToSend* packet);
// Used for padding and FEC packets only.
size_t RtpHeaderLength() const;
uint16_t AllocateSequenceNumber(uint16_t packets_to_send);
// Including RTP headers.
size_t MaxRtpPacketSize() const;
uint32_t SSRC() const;
absl::optional<uint32_t> FlexfecSsrc() const;
// Sends packet to |transport_| or to the pacer, depending on configuration.
bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
StorageType storage);
// Fallback that infers PacketType from Priority.
bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
StorageType storage,
RtpPacketSender::Priority priority);
// Called on update of RTP statistics.
void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
StreamDataCountersCallback* GetRtpStatisticsCallback() const;
uint32_t BitrateSent() const;
void SetRtpState(const RtpState& rtp_state);
RtpState GetRtpState() const;
void SetRtxRtpState(const RtpState& rtp_state);
RtpState GetRtxRtpState() const;
int64_t LastTimestampTimeMs() const;
void SetRtt(int64_t rtt_ms);
void OnPacketsAcknowledged(rtc::ArrayView<const uint16_t> sequence_numbers);
private:
// 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;
size_t SendPadData(size_t bytes, const PacedPacketInfo& pacing_info);
bool PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
bool send_over_rtx,
bool is_retransmit,
const PacedPacketInfo& pacing_info);
// Return the number of bytes sent. Note that both of these functions may
// return a larger value that their argument.
size_t TrySendRedundantPayloads(size_t bytes,
const PacedPacketInfo& pacing_info);
std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
const RtpPacketToSend& packet);
// Sends packet on to |transport_|, leaving the RTP module.
bool SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info);
void RecomputeMaxSendDelay() RTC_EXCLUSIVE_LOCKS_REQUIRED(statistics_crit_);
void UpdateDelayStatistics(int64_t capture_time_ms,
int64_t now_ms,
uint32_t ssrc);
void UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc);
bool UpdateTransportSequenceNumber(RtpPacketToSend* packet, int* packet_id)
RTC_EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
void UpdateRtpStats(const RtpPacketToSend& packet,
bool is_rtx,
bool is_retransmit);
bool IsFecPacket(const RtpPacketToSend& packet) const;
void AddPacketToTransportFeedback(uint16_t packet_id,
const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info);
void UpdateRtpOverhead(const RtpPacketToSend& packet);
Clock* const clock_;
Random random_ RTC_GUARDED_BY(send_critsect_);
const bool audio_configured_;
const absl::optional<uint32_t> flexfec_ssrc_;
RtpPacketSender* const paced_sender_;
TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
TransportFeedbackObserver* const transport_feedback_observer_;
rtc::CriticalSection send_critsect_;
Transport* transport_;
bool sending_media_ RTC_GUARDED_BY(send_critsect_);
bool force_part_of_allocation_ RTC_GUARDED_BY(send_critsect_);
size_t max_packet_size_;
int8_t last_payload_type_ RTC_GUARDED_BY(send_critsect_);
RtpHeaderExtensionMap rtp_header_extension_map_
RTC_GUARDED_BY(send_critsect_);
RtpPacketHistory packet_history_;
// TODO(brandtr): Remove |flexfec_packet_history_| when the FlexfecSender
// is hooked up to the PacedSender.
RtpPacketHistory flexfec_packet_history_;
// Statistics
rtc::CriticalSection statistics_crit_;
SendDelayMap send_delays_ RTC_GUARDED_BY(statistics_crit_);
SendDelayMap::const_iterator max_delay_it_ RTC_GUARDED_BY(statistics_crit_);
// The sum of delays over a kSendSideDelayWindowMs sliding window.
int64_t sum_delays_ms_ RTC_GUARDED_BY(statistics_crit_);
// The sum of delays of all packets sent.
uint64_t total_packet_send_delay_ms_ RTC_GUARDED_BY(statistics_crit_);
StreamDataCounters rtp_stats_ RTC_GUARDED_BY(statistics_crit_);
StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(statistics_crit_);
StreamDataCountersCallback* rtp_stats_callback_
RTC_GUARDED_BY(statistics_crit_);
RateStatistics total_bitrate_sent_ RTC_GUARDED_BY(statistics_crit_);
RateStatistics nack_bitrate_sent_ RTC_GUARDED_BY(statistics_crit_);
SendSideDelayObserver* const send_side_delay_observer_;
RtcEventLog* const event_log_;
SendPacketObserver* const send_packet_observer_;
BitrateStatisticsObserver* const bitrate_callback_;
// RTP variables
uint32_t timestamp_offset_ RTC_GUARDED_BY(send_critsect_);
bool sequence_number_forced_ RTC_GUARDED_BY(send_critsect_);
uint16_t sequence_number_ RTC_GUARDED_BY(send_critsect_);
uint16_t sequence_number_rtx_ RTC_GUARDED_BY(send_critsect_);
// Must be explicitly set by the application, use of absl::optional
// only to keep track of correct use.
absl::optional<uint32_t> ssrc_ RTC_GUARDED_BY(send_critsect_);
// RID value to send in the RID or RepairedRID header extension.
std::string rid_ RTC_GUARDED_BY(send_critsect_);
// MID value to send in the MID header extension.
std::string mid_ RTC_GUARDED_BY(send_critsect_);
// Track if any ACK has been received on the SSRC and RTX SSRC to indicate
// when to stop sending the MID and RID header extensions.
bool ssrc_has_acked_ RTC_GUARDED_BY(send_critsect_);
bool rtx_ssrc_has_acked_ RTC_GUARDED_BY(send_critsect_);
uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(send_critsect_);
int64_t capture_time_ms_ RTC_GUARDED_BY(send_critsect_);
int64_t last_timestamp_time_ms_ RTC_GUARDED_BY(send_critsect_);
bool media_has_been_sent_ RTC_GUARDED_BY(send_critsect_);
bool last_packet_marker_bit_ RTC_GUARDED_BY(send_critsect_);
std::vector<uint32_t> csrcs_ RTC_GUARDED_BY(send_critsect_);
int rtx_ RTC_GUARDED_BY(send_critsect_);
absl::optional<uint32_t> ssrc_rtx_ RTC_GUARDED_BY(send_critsect_);
// Mapping rtx_payload_type_map_[associated] = rtx.
std::map<int8_t, int8_t> rtx_payload_type_map_ RTC_GUARDED_BY(send_critsect_);
size_t rtp_overhead_bytes_per_packet_ RTC_GUARDED_BY(send_critsect_);
bool supports_bwe_extension_ RTC_GUARDED_BY(send_critsect_);
RateLimiter* const retransmission_rate_limiter_;
OverheadObserver* overhead_observer_;
const bool populate_network2_timestamp_;
const bool send_side_bwe_with_overhead_;
// If true, PacedSender should only reference packets as in legacy mode.
// If false, PacedSender may have direct ownership of RtpPacketToSend objects.
// Defaults to true, will be changed to default false soon.
const bool pacer_legacy_packet_referencing_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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