| /* |
| * 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_INCLUDE_RTP_RECEIVER_H_ |
| #define MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ |
| |
| #include <vector> |
| |
| #include "api/rtpreceiverinterface.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| |
| namespace webrtc { |
| |
| class RTPPayloadRegistry; |
| class VideoCodec; |
| |
| class RtpReceiver { |
| public: |
| // Creates a video-enabled RTP receiver. |
| static RtpReceiver* CreateVideoReceiver( |
| Clock* clock, |
| RtpData* incoming_payload_callback, |
| RTPPayloadRegistry* rtp_payload_registry); |
| |
| // Creates an audio-enabled RTP receiver. |
| static RtpReceiver* CreateAudioReceiver( |
| Clock* clock, |
| RtpData* incoming_payload_callback, |
| RTPPayloadRegistry* rtp_payload_registry); |
| |
| virtual ~RtpReceiver() {} |
| |
| // Registers a receive payload in the payload registry and notifies the media |
| // receiver strategy. |
| virtual int32_t RegisterReceivePayload( |
| int payload_type, |
| const SdpAudioFormat& audio_format) = 0; |
| |
| // Deprecated version of the above. |
| int32_t RegisterReceivePayload(const CodecInst& audio_codec); |
| |
| // Registers a receive payload in the payload registry. |
| virtual int32_t RegisterReceivePayload(const VideoCodec& video_codec) = 0; |
| |
| // De-registers |payload_type| from the payload registry. |
| virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0; |
| |
| // Parses the media specific parts of an RTP packet and updates the receiver |
| // state. This for instance means that any changes in SSRC and payload type is |
| // detected and acted upon. |
| virtual bool IncomingRtpPacket(const RTPHeader& rtp_header, |
| const uint8_t* payload, |
| size_t payload_length, |
| PayloadUnion payload_specific) = 0; |
| // TODO(nisse): Deprecated version, delete as soon as downstream |
| // applications are updated. |
| bool IncomingRtpPacket(const RTPHeader& rtp_header, |
| const uint8_t* payload, |
| size_t payload_length, |
| PayloadUnion payload_specific, |
| bool in_order /* Ignored */) { |
| return IncomingRtpPacket(rtp_header, payload, payload_length, |
| payload_specific); |
| } |
| |
| // Gets the RTP timestamp and the corresponding monotonic system |
| // time for the most recent in-order packet. Returns true on |
| // success, false if no packet has been received. |
| virtual bool GetLatestTimestamps(uint32_t* timestamp, |
| int64_t* receive_time_ms) const = 0; |
| |
| // Returns the remote SSRC of the currently received RTP stream. |
| virtual uint32_t SSRC() const = 0; |
| |
| virtual std::vector<RtpSource> GetSources() const = 0; |
| }; |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ |