Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
      12             : #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
      13             : 
      14             : #include "webrtc/base/onetimeevent.h"
      15             : #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
      16             : #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
      17             : #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
      18             : #include "webrtc/typedefs.h"
      19             : 
      20             : namespace webrtc {
      21             : 
      22             : class RTPReceiverVideo : public RTPReceiverStrategy {
      23             :  public:
      24             :   explicit RTPReceiverVideo(RtpData* data_callback);
      25             : 
      26             :   virtual ~RTPReceiverVideo();
      27             : 
      28             :   int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
      29             :                          const PayloadUnion& specific_payload,
      30             :                          bool is_red,
      31             :                          const uint8_t* packet,
      32             :                          size_t packet_length,
      33             :                          int64_t timestamp,
      34             :                          bool is_first_packet) override;
      35             : 
      36           0 :   TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; }
      37             : 
      38             :   RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
      39             : 
      40             :   bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
      41             : 
      42             :   int32_t OnNewPayloadTypeCreated(const CodecInst& audio_codec) override;
      43             : 
      44             :   int32_t InvokeOnInitializeDecoder(
      45             :       RtpFeedback* callback,
      46             :       int8_t payload_type,
      47             :       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
      48             :       const PayloadUnion& specific_payload) const override;
      49             : 
      50             :   void SetPacketOverHead(uint16_t packet_over_head);
      51             : 
      52             :  private:
      53             :   OneTimeEvent first_packet_received_;
      54             : };
      55             : }  // namespace webrtc
      56             : 
      57             : #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_