1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
|
/*
* Copyright (c) 2014 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.
*/
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
namespace webrtc {
RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
size_t max_payload_len,
const RTPVideoTypeHeader* rtp_type_header,
FrameType frame_type) {
switch (type) {
case kRtpVideoH264:
return new RtpPacketizerH264(frame_type, max_payload_len);
case kRtpVideoVp8:
assert(rtp_type_header != NULL);
return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len);
case kRtpVideoGeneric:
return new RtpPacketizerGeneric(frame_type, max_payload_len);
case kRtpVideoNone:
assert(false);
}
return NULL;
}
RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) {
switch (type) {
case kRtpVideoH264:
return new RtpDepacketizerH264();
case kRtpVideoVp8:
return new RtpDepacketizerVp8();
case kRtpVideoGeneric:
return new RtpDepacketizerGeneric();
case kRtpVideoNone:
assert(false);
}
return NULL;
}
} // namespace webrtc
|
