Upstream version 9.38.198.0
[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / rtp_rtcp / source / rtp_receiver_video.h
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/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
15 #include "webrtc/modules/rtp_rtcp/source/bitrate.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/system_wrappers/interface/scoped_ptr.h"
19 #include "webrtc/typedefs.h"
20
21 namespace webrtc {
22
23 class RTPReceiverVideo : public RTPReceiverStrategy {
24  public:
25   RTPReceiverVideo(RtpData* data_callback);
26
27   virtual ~RTPReceiverVideo();
28
29   virtual int32_t ParseRtpPacket(
30       WebRtcRTPHeader* rtp_header,
31       const PayloadUnion& specific_payload,
32       bool is_red,
33       const uint8_t* packet,
34       uint16_t packet_length,
35       int64_t timestamp,
36       bool is_first_packet) OVERRIDE;
37
38   TelephoneEventHandler* GetTelephoneEventHandler() {
39     return NULL;
40   }
41
42   int GetPayloadTypeFrequency() const OVERRIDE;
43
44   virtual RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const
45       OVERRIDE;
46
47   virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const OVERRIDE;
48
49   virtual int32_t OnNewPayloadTypeCreated(
50       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
51       int8_t payload_type,
52       uint32_t frequency) OVERRIDE;
53
54   virtual int32_t InvokeOnInitializeDecoder(
55       RtpFeedback* callback,
56       int32_t id,
57       int8_t payload_type,
58       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
59       const PayloadUnion& specific_payload) const OVERRIDE;
60
61   void SetPacketOverHead(uint16_t packet_over_head);
62
63  private:
64   int32_t ReceiveGenericCodec(WebRtcRTPHeader* rtp_header,
65                               const uint8_t* payload_data,
66                               uint16_t payload_data_length);
67
68   int32_t ReceiveVp8Codec(WebRtcRTPHeader* rtp_header,
69                           const uint8_t* payload_data,
70                           uint16_t payload_data_length);
71
72   int32_t ReceiveH264Codec(WebRtcRTPHeader* rtp_header,
73                            const uint8_t* payload_data,
74                            uint16_t payload_data_length);
75
76   int32_t BuildRTPheader(const WebRtcRTPHeader* rtp_header,
77                          uint8_t* data_buffer) const;
78
79   int32_t ParseVideoCodecSpecific(
80       WebRtcRTPHeader* rtp_header,
81       const uint8_t* payload_data,
82       uint16_t payload_data_length,
83       RtpVideoCodecTypes video_type,
84       int64_t now_ms,
85       bool is_first_packet);
86 };
87 }  // namespace webrtc
88
89 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_