|  | /* | 
|  | *  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/modules/rtp_rtcp/interface/rtp_rtcp.h" | 
|  | #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" | 
|  | #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 
|  | #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | 
|  | #include "webrtc/system_wrappers/interface/scoped_ptr.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( | 
|  | int32_t id, 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, | 
|  | int32_t id, | 
|  | 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_reset_statistics, | 
|  | 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. | 
|  | RTPReceiverStrategy(RtpData* data_callback); | 
|  |  | 
|  | scoped_ptr<CriticalSectionWrapper> crit_sect_; | 
|  | PayloadUnion last_payload_; | 
|  | RtpData* data_callback_; | 
|  | }; | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |