This also includes: - Creating new packetizer and depacketizer interfaces. - Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition. - Created a Create() factory method for packetizers and depacketizers. R=niklas.enbom@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
42 lines
1.2 KiB
C++
42 lines
1.2 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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namespace webrtc {
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RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
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size_t max_payload_len) {
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switch (type) {
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case kRtpVideoH264:
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return new RtpPacketizerH264(max_payload_len);
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case kRtpVideoNone:
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case kRtpVideoGeneric:
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case kRtpVideoVp8:
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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 kRtpVideoNone:
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case kRtpVideoGeneric:
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case kRtpVideoVp8:
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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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