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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 RateLimiter;
class RtcEventLog;
class RtpPacketToSend;
class RTPSender {
public:
RTPSender(const RtpRtcp::Configuration& config,
RtpPacketHistory* packet_history,
RtpPacketSender* packet_sender);
~RTPSender();
void SetSendingMediaStatus(bool enabled);
bool SendingMedia() const;
bool IsAudioConfigured() const;
uint32_t TimestampOffset() const;
void SetTimestampOffset(uint32_t timestamp);
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(absl::string_view uri, int id);
bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) const;
int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
void DeregisterRtpHeaderExtension(absl::string_view uri);
bool SupportsPadding() const;
bool SupportsRtxPayloadPadding() const;
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes,
bool media_has_been_sent);
// NACK.
void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt);
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;
absl::optional<uint32_t> RtxSsrc() const { return rtx_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();
// Size info for header extensions used by audio packets.
static rtc::ArrayView<const RtpExtensionSize> AudioExtensionSizes();
// 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);
// Maximum header overhead per fec/padding packet.
size_t FecOrPaddingPacketMaxRtpHeaderLength() const;
// Expected header overhead per media packet.
size_t ExpectedPerPacketOverhead() const;
uint16_t AllocateSequenceNumber(uint16_t packets_to_send);
// Including RTP headers.
size_t MaxRtpPacketSize() const;
uint32_t SSRC() const { return ssrc_; }
absl::optional<uint32_t> FlexfecSsrc() const { return flexfec_ssrc_; }
// 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);
void SetRtpState(const RtpState& rtp_state);
RtpState GetRtpState() const;
void SetRtxRtpState(const RtpState& rtp_state);
RtpState GetRtxRtpState() const;
int64_t LastTimestampTimeMs() const;
private:
std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
const RtpPacketToSend& packet);
bool IsFecPacket(const RtpPacketToSend& packet) const;
void UpdateHeaderSizes() RTC_EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
Clock* const clock_;
Random random_ RTC_GUARDED_BY(send_critsect_);
const bool audio_configured_;
const uint32_t ssrc_;
const absl::optional<uint32_t> rtx_ssrc_;
const absl::optional<uint32_t> flexfec_ssrc_;
// Limits GeneratePadding() outcome to <=
// |max_padding_size_factor_| * |target_size_bytes|
const double max_padding_size_factor_;
RtpPacketHistory* const packet_history_;
RtpPacketSender* const paced_sender_;
rtc::CriticalSection send_critsect_;
bool sending_media_ 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_);
size_t max_media_packet_header_ RTC_GUARDED_BY(send_critsect_);
size_t max_padding_fec_packet_header_ RTC_GUARDED_BY(send_critsect_);
// 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_);
// 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_);
// Should we send MID/RID even when ACKed? (see below).
const bool always_send_mid_and_rid_;
// 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 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_);
// Mapping rtx_payload_type_map_[associated] = rtx.
std::map<int8_t, int8_t> rtx_payload_type_map_ RTC_GUARDED_BY(send_critsect_);
bool supports_bwe_extension_ RTC_GUARDED_BY(send_critsect_);
RateLimiter* const retransmission_rate_limiter_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_