/* * Copyright (c) 2013 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_RECEIVER_IMPL_H_ #define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ #include #include #include #include #include "api/optional.h" #include "modules/rtp_rtcp/include/rtp_receiver.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_receiver_strategy.h" #include "rtc_base/criticalsection.h" #include "typedefs.h" // NOLINT(build/include) namespace webrtc { class RtpReceiverImpl : public RtpReceiver { public: // Callbacks passed in here may not be NULL (use Null Object callbacks if you // want callbacks to do nothing). This class takes ownership of the media // receiver but nothing else. RtpReceiverImpl(Clock* clock, RtpFeedback* incoming_messages_callback, RTPPayloadRegistry* rtp_payload_registry, RTPReceiverStrategy* rtp_media_receiver); virtual ~RtpReceiverImpl(); int32_t RegisterReceivePayload(int payload_type, const SdpAudioFormat& audio_format) override; int32_t RegisterReceivePayload(const VideoCodec& video_codec) override; int32_t DeRegisterReceivePayload(const int8_t payload_type) override; bool IncomingRtpPacket(const RTPHeader& rtp_header, const uint8_t* payload, size_t payload_length, PayloadUnion payload_specific) override; bool GetLatestTimestamps(uint32_t* timestamp, int64_t* receive_time_ms) const override; uint32_t SSRC() const override; int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override; int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override; TelephoneEventHandler* GetTelephoneEventHandler() override; std::vector GetSources() const override; const std::vector& ssrc_sources_for_testing() const { return ssrc_sources_; } const std::list& csrc_sources_for_testing() const { return csrc_sources_; } private: void CheckSSRCChanged(const RTPHeader& rtp_header); void CheckCSRC(const WebRtcRTPHeader& rtp_header); int32_t CheckPayloadChanged(const RTPHeader& rtp_header, const int8_t first_payload_byte, bool* is_red, PayloadUnion* payload); void UpdateSources(const rtc::Optional& ssrc_audio_level); void RemoveOutdatedSources(int64_t now_ms); Clock* clock_; rtc::CriticalSection critical_section_rtp_receiver_; RTPPayloadRegistry* const rtp_payload_registry_ RTC_PT_GUARDED_BY(critical_section_rtp_receiver_); const std::unique_ptr rtp_media_receiver_; RtpFeedback* const cb_rtp_feedback_; // SSRCs. uint32_t ssrc_ RTC_GUARDED_BY(critical_section_rtp_receiver_); uint8_t num_csrcs_ RTC_GUARDED_BY(critical_section_rtp_receiver_); uint32_t current_remote_csrc_[kRtpCsrcSize] RTC_GUARDED_BY( critical_section_rtp_receiver_); // Sequence number and timestamps for the latest in-order packet. rtc::Optional last_received_sequence_number_ RTC_GUARDED_BY(critical_section_rtp_receiver_); uint32_t last_received_timestamp_ RTC_GUARDED_BY(critical_section_rtp_receiver_); int64_t last_received_frame_time_ms_ RTC_GUARDED_BY(critical_section_rtp_receiver_); std::unordered_map::iterator> iterator_by_csrc_; // The RtpSource objects are sorted chronologically. std::list csrc_sources_; std::vector ssrc_sources_; }; } // namespace webrtc #endif // MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_