13ceeeadfc
Reason for revert: Broke a lot of tests in chromium.webrtc browser_tests. See e.g. https://build.chromium.org/p/chromium.webrtc/builders/Mac%20Tester/builds/62228 and https://build.chromium.org/p/chromium.webrtc/builders/Win8%20Tester/builds/30102. [ RUN ] WebRtcVideoQualityBrowserTests/WebRtcVideoQualityBrowserTest.MANUAL_TestVideoQualityH264/1 ... # # Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\modules\rtp_rtcp\source\rtp_format_h264.cc, line 170 # last system error: 0 # Check failed: packetization_mode_ == kH264PacketizationMode1 (0 vs. 2) # Original issue's description: > Implement H.264 packetization mode 0. > > This approach extends the H.264 specific information with > a packetization mode enum. > > Status: Parameter is in code. No way to set it yet. > > Rebase of CL 2009213002 > > BUG=600254 > > Committed: https://crrev.com/3bba101f36483b8030a693dfbc93af736d1dba68 > Cr-Commit-Position: refs/heads/master@{#15032} TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,hta@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=600254 NOPRESUBMIT=true Review-Url: https://codereview.webrtc.org/2500743002 Cr-Commit-Position: refs/heads/master@{#15050}
56 lines
1.9 KiB
C++
56 lines
1.9 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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#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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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 kRtpVideoVp9:
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assert(rtp_type_header != NULL);
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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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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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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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