/* * 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/typedefs.h" namespace webrtc { class TelephoneEventHandler; // This strategy deals with media-specific RTP packet processing. // This class is not thread-safe and must be protected by its caller. class RTPReceiverStrategy { public: static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback); static RTPReceiverStrategy* CreateAudioStrategy( RtpData* data_callback, RtpAudioFeedback* incoming_messages_callback); virtual ~RTPReceiverStrategy() {} // Parses the RTP packet and calls the data callback with the payload data. // Implementations are encouraged to use the provided packet buffer and RTP // header as arguments to the callback; implementations are also allowed to // make changes in the data as necessary. The specific_payload argument // provides audio or video-specific data. The is_first_packet argument is true // if this packet is either the first packet ever or the first in its frame. virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header, const PayloadUnion& specific_payload, bool is_red, const uint8_t* payload, size_t payload_length, int64_t timestamp_ms, bool is_first_packet) = 0; virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0; // Retrieves the last known applicable frequency. virtual int GetPayloadTypeFrequency() const = 0; // Computes the current dead-or-alive state. virtual RTPAliveType ProcessDeadOrAlive( uint16_t last_payload_length) const = 0; // Returns true if we should report CSRC changes for this payload type. // TODO(phoglund): should move out of here along with other payload stuff. virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0; // Notifies the strategy that we have created a new non-RED payload type in // the payload registry. virtual int32_t OnNewPayloadTypeCreated( const char payloadName[RTP_PAYLOAD_NAME_SIZE], int8_t payloadType, uint32_t frequency) = 0; // Invokes the OnInitializeDecoder callback in a media-specific way. virtual int32_t InvokeOnInitializeDecoder( RtpFeedback* callback, int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], const PayloadUnion& specific_payload) const = 0; // Checks if the payload type has changed, and returns whether we should // reset statistics and/or discard this packet. virtual void CheckPayloadChanged(int8_t payload_type, PayloadUnion* specific_payload, bool* should_discard_changes); virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const; // Stores / retrieves the last media specific payload for later reference. void GetLastMediaSpecificPayload(PayloadUnion* payload) const; void SetLastMediaSpecificPayload(const PayloadUnion& payload); protected: // The data callback is where we should send received payload data. // See ParseRtpPacket. This class does not claim ownership of the callback. // Implementations must NOT hold any critical sections while calling the // callback. // // Note: Implementations may call the callback for other reasons than calls // to ParseRtpPacket, for instance if the implementation somehow recovers a // packet. explicit RTPReceiverStrategy(RtpData* data_callback); rtc::scoped_ptr crit_sect_; PayloadUnion last_payload_; RtpData* data_callback_; }; } // namespace webrtc #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_