b5e6bfc76a
Changes include, 1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric. 2) Introduce class RtpDepacketizerVp8. 3) Make RTPSenderVideo::SendH264 generic and used by all packetizers. 4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to RtpPacketizer/RtpDePacketizer sub-classes. R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26399004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
51 lines
1.8 KiB
C++
51 lines
1.8 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
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namespace webrtc {
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RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
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size_t max_payload_len,
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const RTPVideoTypeHeader* rtp_type_header,
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FrameType frame_type) {
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switch (type) {
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case kRtpVideoH264:
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return new RtpPacketizerH264(frame_type, max_payload_len);
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case kRtpVideoVp8:
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assert(rtp_type_header != NULL);
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return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len);
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case kRtpVideoGeneric:
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return new RtpPacketizerGeneric(frame_type, max_payload_len);
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case kRtpVideoNone:
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assert(false);
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}
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return NULL;
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}
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RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type,
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RtpData* const callback) {
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switch (type) {
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case kRtpVideoH264:
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return new RtpDepacketizerH264(callback);
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case kRtpVideoVp8:
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return new RtpDepacketizerVp8(callback);
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case kRtpVideoGeneric:
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return new RtpDepacketizerGeneric(callback);
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case kRtpVideoNone:
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assert(false);
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}
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return NULL;
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}
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} // namespace webrtc
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