9aa96889a3
Reason for revert: Fixed timeouts in slow tests Original issue's description: > 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 > > Committed: https://crrev.com/243a0a7a7fd6b5da1e32df31f1bfbb6a68dc09f3 > Cr-Commit-Position: refs/heads/master@{#15441} 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/2558463002 Cr-Commit-Position: refs/heads/master@{#15445}
60 lines
2.0 KiB
C++
60 lines
2.0 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 <utility>
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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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#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.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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RTC_CHECK(rtp_type_header);
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return new RtpPacketizerH264(max_payload_len,
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rtp_type_header->H264.packetization_mode);
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case kRtpVideoVp8:
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RTC_CHECK(rtp_type_header);
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return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len);
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case kRtpVideoVp9:
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RTC_CHECK(rtp_type_header);
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return new RtpPacketizerVp9(rtp_type_header->VP9, 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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RTC_NOTREACHED();
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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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switch (type) {
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case kRtpVideoH264:
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return new RtpDepacketizerH264();
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case kRtpVideoVp8:
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return new RtpDepacketizerVp8();
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case kRtpVideoVp9:
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return new RtpDepacketizerVp9();
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case kRtpVideoGeneric:
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return new RtpDepacketizerGeneric();
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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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