rtspTransmit/includes/rtcp/rtc/rtppacketizer.hpp

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2024-03-21 16:36:26 +08:00
/**
* Copyright (c) 2020 Filip Klembara (in2core)
*
* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at https://mozilla.org/MPL/2.0/.
*/
#ifndef RTC_RTP_PACKETIZER_H
#define RTC_RTP_PACKETIZER_H
#if RTC_ENABLE_MEDIA
#include "mediahandler.hpp"
#include "message.hpp"
#include "rtppacketizationconfig.hpp"
namespace rtc {
/// RTP packetizer
class RTC_CPP_EXPORT RtpPacketizer : public MediaHandler {
public:
/// Constructs packetizer with given RTP configuration
/// @note RTP configuration is used in packetization process which may change some configuration
/// properties such as sequence number.
/// @param rtpConfig RTP configuration
RtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig);
virtual ~RtpPacketizer();
virtual void media(const Description::Media &desc) override;
virtual void outgoing(message_vector &messages, const message_callback &send) override;
/// RTP packetization config
const shared_ptr<RtpPacketizationConfig> rtpConfig;
protected:
/// Creates RTP packet for given payload
/// @note This function increase sequence number after packetization.
/// @param payload RTP payload
/// @param setMark Set marker flag in RTP packet if true
virtual message_ptr packetize(shared_ptr<binary> payload, bool mark);
private:
static const auto RtpHeaderSize = 12;
static const auto RtpExtHeaderCvoSize = 8;
};
// Generic audio RTP packetizer
template <uint32_t DEFAULT_CLOCK_RATE>
class RTC_CPP_EXPORT AudioRtpPacketizer final : public RtpPacketizer {
public:
inline static const uint32_t DefaultClockRate = DEFAULT_CLOCK_RATE;
inline static const uint32_t defaultClockRate [[deprecated("Use DefaultClockRate")]] =
DEFAULT_CLOCK_RATE; // for backward compatibility
AudioRtpPacketizer(shared_ptr<RtpPacketizationConfig> rtpConfig)
: RtpPacketizer(std::move(rtpConfig)) {}
};
// Audio RTP packetizers
using OpusRtpPacketizer = AudioRtpPacketizer<48000>;
using AACRtpPacketizer = AudioRtpPacketizer<48000>;
// Dummy wrapper for backward compatibility, do not use
class RTC_CPP_EXPORT PacketizationHandler final : public MediaHandler {
public:
PacketizationHandler(shared_ptr<RtpPacketizer> packetizer)
: mPacketizer(std::move(packetizer)) {}
inline void outgoing(message_vector &messages, const message_callback &send) {
return mPacketizer->outgoing(messages, send);
}
private:
shared_ptr<RtpPacketizer> mPacketizer;
};
// Audio packetization handlers for backward compatibility, do not use
using OpusPacketizationHandler [[deprecated("Add OpusRtpPacketizer directly")]] =
PacketizationHandler;
using AACPacketizationHandler [[deprecated("Add AACRtpPacketizer directly")]] =
PacketizationHandler;
} // namespace rtc
#endif /* RTC_ENABLE_MEDIA */
#endif /* RTC_RTP_PACKETIZER_H */