Files
platform-external-webrtc/webrtc/modules/rtp_rtcp/source/rtp_format.cc
hta 243a0a7a7f Revert of H.264 packetization mode 0 (try 3) (patchset #13 id:490001 of https://codereview.webrtc.org/2528343002/ )
Reason for revert:
Failures on the Linux Memcheck bot

Original issue's description:
> This approach passes packetization mode to the encoder as part of
> a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
>
> BUG=600254
>
> Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> Cr-Commit-Position: refs/heads/master@{#15437}

TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254

Review-Url: https://codereview.webrtc.org/2558453002
Cr-Commit-Position: refs/heads/master@{#15441}
2016-12-06 12:22:05 +00:00

56 lines
1.9 KiB
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"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.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 kRtpVideoVp9:
assert(rtp_type_header != NULL);
return new RtpPacketizerVp9(rtp_type_header->VP9, 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 kRtpVideoVp9:
return new RtpDepacketizerVp9();
case kRtpVideoGeneric:
return new RtpDepacketizerGeneric();
case kRtpVideoNone:
assert(false);
}
return NULL;
}
} // namespace webrtc