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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);
~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);
// Tries to send packet to transport. Also updates any timing extensions,
// calls observers waiting for packet send events, and updates stats.
// Returns true if packet belongs to this RTP module, false otherwise.
bool TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info);
bool SupportsPadding() const;
bool SupportsRtxPayloadPadding() const;
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.
// TODO(bugs.webrtc.org/XXX): Remove in favor of EnqueuePackets().
bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet);
// Pass a set of packets to RtpPacketSender instance, for paced or immediate
// sending to the network.
void EnqueuePackets(std::vector<std::unique_ptr<RtpPacketToSend>> packets);
// 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;
// Helper class that redirects packets directly to the send part of this class
// without passing through an actual paced sender.
class NonPacedPacketSender : public RtpPacketSender {
public:
explicit NonPacedPacketSender(RTPSender* rtp_sender);
virtual ~NonPacedPacketSender();
void EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;
private:
uint16_t transport_sequence_number_;
RTPSender* const rtp_sender_;
};
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);
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_;
const std::unique_ptr<NonPacedPacketSender> non_paced_packet_sender_;
RtpPacketSender* const paced_sender_;
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_;
// 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_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_