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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/field_trials_view.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_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/random.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class FrameEncryptorInterface;
class RateLimiter;
class RtcEventLog;
class RtpPacketToSend;
// Maximum amount of padding in RFC 3550 is 255 bytes.
constexpr size_t kMaxPaddingLength = 255;
class RTPSender {
public:
RTPSender(const RtpRtcpInterface::Configuration& config,
RtpPacketHistory* packet_history,
RtpPacketSender* packet_sender);
RTPSender(const RTPSender&) = delete;
RTPSender& operator=(const RTPSender&) = delete;
~RTPSender();
void SetSendingMediaStatus(bool enabled) RTC_LOCKS_EXCLUDED(send_mutex_);
bool SendingMedia() const RTC_LOCKS_EXCLUDED(send_mutex_);
bool IsAudioConfigured() const RTC_LOCKS_EXCLUDED(send_mutex_);
uint32_t TimestampOffset() const RTC_LOCKS_EXCLUDED(send_mutex_);
void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(send_mutex_);
void SetMid(absl::string_view mid) RTC_LOCKS_EXCLUDED(send_mutex_);
uint16_t SequenceNumber() const RTC_LOCKS_EXCLUDED(send_mutex_);
void SetSequenceNumber(uint16_t seq) RTC_LOCKS_EXCLUDED(send_mutex_);
void SetMaxRtpPacketSize(size_t max_packet_size)
RTC_LOCKS_EXCLUDED(send_mutex_);
void SetExtmapAllowMixed(bool extmap_allow_mixed)
RTC_LOCKS_EXCLUDED(send_mutex_);
// RTP header extension
bool RegisterRtpHeaderExtension(absl::string_view uri, int id)
RTC_LOCKS_EXCLUDED(send_mutex_);
bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) const
RTC_LOCKS_EXCLUDED(send_mutex_);
void DeregisterRtpHeaderExtension(absl::string_view uri)
RTC_LOCKS_EXCLUDED(send_mutex_);
bool SupportsPadding() const RTC_LOCKS_EXCLUDED(send_mutex_);
bool SupportsRtxPayloadPadding() const RTC_LOCKS_EXCLUDED(send_mutex_);
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes,
bool media_has_been_sent,
bool can_send_padding_on_media_ssrc) RTC_LOCKS_EXCLUDED(send_mutex_);
// NACK.
void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt) RTC_LOCKS_EXCLUDED(send_mutex_);
int32_t ReSendPacket(uint16_t packet_id) RTC_LOCKS_EXCLUDED(send_mutex_);
// ACK.
void OnReceivedAckOnSsrc(int64_t extended_highest_sequence_number)
RTC_LOCKS_EXCLUDED(send_mutex_);
void OnReceivedAckOnRtxSsrc(int64_t extended_highest_sequence_number)
RTC_LOCKS_EXCLUDED(send_mutex_);
// RTX.
void SetRtxStatus(int mode) RTC_LOCKS_EXCLUDED(send_mutex_);
int RtxStatus() const RTC_LOCKS_EXCLUDED(send_mutex_);
absl::optional<uint32_t> RtxSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
return rtx_ssrc_;
}
// Returns expected size difference between an RTX packet and media packet
// that RTX packet is created from. Returns 0 if RTX is disabled.
size_t RtxPacketOverhead() const;
void SetRtxPayloadType(int payload_type, int associated_payload_type)
RTC_LOCKS_EXCLUDED(send_mutex_);
// Size info for header extensions used by FEC packets.
static rtc::ArrayView<const RtpExtensionSize> FecExtensionSizes()
RTC_LOCKS_EXCLUDED(send_mutex_);
// Size info for header extensions used by video packets.
static rtc::ArrayView<const RtpExtensionSize> VideoExtensionSizes()
RTC_LOCKS_EXCLUDED(send_mutex_);
// Size info for header extensions used by audio packets.
static rtc::ArrayView<const RtpExtensionSize> AudioExtensionSizes()
RTC_LOCKS_EXCLUDED(send_mutex_);
// Create empty packet, fills ssrc, csrcs and reserve place for header
// extensions RtpSender updates before sending.
std::unique_ptr<RtpPacketToSend> AllocatePacket(
rtc::ArrayView<const uint32_t> csrcs = {})
RTC_LOCKS_EXCLUDED(send_mutex_);
// Maximum header overhead per fec/padding packet.
size_t FecOrPaddingPacketMaxRtpHeaderLength() const
RTC_LOCKS_EXCLUDED(send_mutex_);
// Expected header overhead per media packet.
size_t ExpectedPerPacketOverhead() const RTC_LOCKS_EXCLUDED(send_mutex_);
// Including RTP headers.
size_t MaxRtpPacketSize() const RTC_LOCKS_EXCLUDED(send_mutex_);
uint32_t SSRC() const RTC_LOCKS_EXCLUDED(send_mutex_) { return ssrc_; }
const std::string& Rid() const RTC_LOCKS_EXCLUDED(send_mutex_) {
return rid_;
}
absl::optional<uint32_t> FlexfecSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
return flexfec_ssrc_;
}
// 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)
RTC_LOCKS_EXCLUDED(send_mutex_);
void SetRtpState(const RtpState& rtp_state) RTC_LOCKS_EXCLUDED(send_mutex_);
RtpState GetRtpState() const RTC_LOCKS_EXCLUDED(send_mutex_);
void SetRtxRtpState(const RtpState& rtp_state)
RTC_LOCKS_EXCLUDED(send_mutex_);
RtpState GetRtxRtpState() const RTC_LOCKS_EXCLUDED(send_mutex_);
private:
std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
const RtpPacketToSend& packet);
bool IsFecPacket(const RtpPacketToSend& packet) const;
void UpdateHeaderSizes() RTC_EXCLUSIVE_LOCKS_REQUIRED(send_mutex_);
void UpdateLastPacketState(const RtpPacketToSend& packet)
RTC_EXCLUSIVE_LOCKS_REQUIRED(send_mutex_);
Clock* const clock_;
Random random_ RTC_GUARDED_BY(send_mutex_);
const bool audio_configured_;
const uint32_t ssrc_;
const absl::optional<uint32_t> rtx_ssrc_;
const absl::optional<uint32_t> flexfec_ssrc_;
RtpPacketHistory* const packet_history_;
RtpPacketSender* const paced_sender_;
mutable Mutex send_mutex_;
bool sending_media_ RTC_GUARDED_BY(send_mutex_);
size_t max_packet_size_;
RtpHeaderExtensionMap rtp_header_extension_map_ RTC_GUARDED_BY(send_mutex_);
size_t max_media_packet_header_ RTC_GUARDED_BY(send_mutex_);
size_t max_padding_fec_packet_header_ RTC_GUARDED_BY(send_mutex_);
// RTP variables
uint32_t timestamp_offset_ RTC_GUARDED_BY(send_mutex_);
// RID value to send in the RID or RepairedRID header extension.
const std::string rid_;
// MID value to send in the MID header extension.
std::string mid_ RTC_GUARDED_BY(send_mutex_);
// 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_mutex_);
bool rtx_ssrc_has_acked_ RTC_GUARDED_BY(send_mutex_);
// Maximum number of csrcs this sender is used with.
size_t max_num_csrcs_ RTC_GUARDED_BY(send_mutex_) = 0;
int rtx_ RTC_GUARDED_BY(send_mutex_);
// Mapping rtx_payload_type_map_[associated] = rtx.
std::map<int8_t, int8_t> rtx_payload_type_map_ RTC_GUARDED_BY(send_mutex_);
bool supports_bwe_extension_ RTC_GUARDED_BY(send_mutex_);
RateLimiter* const retransmission_rate_limiter_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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