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- /**
- * Copyright (c) 2024 Paul-Louis Ageneau
- *
- * 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_DEPACKETIZER_H
- #define RTC_RTP_DEPACKETIZER_H
- #if RTC_ENABLE_MEDIA
- #include "mediahandler.hpp"
- #include "message.hpp"
- #include <set>
- namespace rtc {
- // Base RTP depacketizer class
- class RTC_CPP_EXPORT RtpDepacketizer : public MediaHandler {
- public:
- RtpDepacketizer();
- RtpDepacketizer(uint32_t clockRate);
- virtual ~RtpDepacketizer();
- virtual void incoming(message_vector &messages, const message_callback &send) override;
- protected:
- shared_ptr<FrameInfo> createFrameInfo(uint32_t timestamp, uint8_t payloadType) const;
- private:
- const uint32_t mClockRate;
- };
- // Base class for video RTP depacketizer
- class RTC_CPP_EXPORT VideoRtpDepacketizer : public RtpDepacketizer {
- public:
- inline static const uint32_t ClockRate = 90000;
- VideoRtpDepacketizer();
- virtual ~VideoRtpDepacketizer();
- protected:
- struct sequence_cmp {
- bool operator()(message_ptr a, message_ptr b) const;
- };
- using message_buffer = std::set<message_ptr, sequence_cmp>;
- virtual message_ptr reassemble(message_buffer &messages) = 0;
- private:
- void incoming(message_vector &messages, const message_callback &send) override;
- message_buffer mBuffer;
- };
- // Generic audio RTP depacketizer
- template <uint32_t DEFAULT_CLOCK_RATE>
- class RTC_CPP_EXPORT AudioRtpDepacketizer final : public RtpDepacketizer {
- public:
- inline static const uint32_t DefaultClockRate = DEFAULT_CLOCK_RATE;
- AudioRtpDepacketizer(uint32_t clockRate = DefaultClockRate) : RtpDepacketizer(clockRate) {}
- };
- // Audio RTP depacketizers
- using OpusRtpDepacketizer = AudioRtpDepacketizer<48000>;
- using AACRtpDepacketizer = AudioRtpDepacketizer<48000>;
- using PCMARtpDepacketizer = AudioRtpDepacketizer<8000>;
- using PCMURtpDepacketizer = AudioRtpDepacketizer<8000>;
- using G722RtpDepacketizer = AudioRtpDepacketizer<8000>;
- } // namespace rtc
- #endif /* RTC_ENABLE_MEDIA */
- #endif /* RTC_RTP_DEPACKETIZER_H */
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