stefan@webrtc.org 2ec560606b Add H.264 packetization.
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
2014-07-31 14:59:24 +00:00

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C++

/*
* 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"
namespace webrtc {
RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
size_t max_payload_len) {
switch (type) {
case kRtpVideoH264:
return new RtpPacketizerH264(max_payload_len);
case kRtpVideoNone:
case kRtpVideoGeneric:
case kRtpVideoVp8:
assert(false);
}
return NULL;
}
RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type,
RtpData* const callback) {
switch (type) {
case kRtpVideoH264:
return new RtpDepacketizerH264(callback);
case kRtpVideoNone:
case kRtpVideoGeneric:
case kRtpVideoVp8:
assert(false);
}
return NULL;
}
} // namespace webrtc