| /* |
| * 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_VIDEO_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_ |
| |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| #include "webrtc/rtc_base/onetimeevent.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| class RTPReceiverVideo : public RTPReceiverStrategy { |
| public: |
| explicit RTPReceiverVideo(RtpData* data_callback); |
| |
| virtual ~RTPReceiverVideo(); |
| |
| int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header, |
| const PayloadUnion& specific_payload, |
| bool is_red, |
| const uint8_t* packet, |
| size_t packet_length, |
| int64_t timestamp, |
| bool is_first_packet) override; |
| |
| TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; } |
| |
| RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override; |
| |
| bool ShouldReportCsrcChanges(uint8_t payload_type) const override; |
| |
| int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) override; |
| |
| int32_t InvokeOnInitializeDecoder( |
| RtpFeedback* callback, |
| int8_t payload_type, |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| const PayloadUnion& specific_payload) const override; |
| |
| void SetPacketOverHead(uint16_t packet_over_head); |
| |
| private: |
| OneTimeEvent first_packet_received_; |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_ |