| /* |
| * 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 MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |
| #define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |
| |
| #include "modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/rtp_utility.h" |
| #include "rtc_base/criticalsection.h" |
| #include "typedefs.h" // NOLINT(build/include) |
| |
| namespace webrtc { |
| |
| struct CodecInst; |
| |
| 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); |
| |
| 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. |
| virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header, |
| const PayloadUnion& specific_payload, |
| const uint8_t* payload, |
| size_t payload_length, |
| int64_t timestamp_ms) = 0; |
| |
| virtual TelephoneEventHandler* GetTelephoneEventHandler() = 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 audio payload type |
| // in the payload registry. |
| virtual int32_t OnNewPayloadTypeCreated( |
| int payload_type, |
| const SdpAudioFormat& audio_format) = 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::CriticalSection crit_sect_; |
| rtc::Optional<PayloadUnion> last_payload_; |
| RtpData* data_callback_; |
| }; |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |