5daeff9c1f
This reverts commit dfbced6504720d2c0807d7b92798eb80ba3f8be9. Reason for revert: Crashes when making a video call. #9 0x00000001043dd8d8 in webrtc::RTPVideoHeaderH264& absl::variant_internal::TypedThrowBadVariantAccess<webrtc::RTPVideoHeaderH264&>() at /third_party/absl/types/internal/variant.h:315 #10 0x00000001043dd8ac in absl::variant_internal::VariantAccessResultImpl<2ul, absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&&&>::type absl::variant_internal::VariantCoreAccess::CheckedAccess<2ul, absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&>(absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&&&) at /third_party/absl/types/internal/variant.h:597 #11 0x00000001043db778 in webrtc::RTPVideoHeaderH264& absl::get<webrtc::RTPVideoHeaderH264, webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>(absl::variant<webrtc::RTPVideoHeaderVP8, webrtc::RTPVideoHeaderVP9, webrtc::RTPVideoHeaderH264>&) at /third_party/absl/types/variant.h:299 #12 0x0000000104558bcc in webrtc::RtpPacketizer::Create(webrtc::VideoCodecType, unsigned long, unsigned long, webrtc::RTPVideoHeader const*, webrtc::FrameType) at webrtc/modules/rtp_rtcp/source/rtp_format.cc:30 Original change's description: > Remove RTPVideoHeader::h264() accessors. > > Bug: none > Change-Id: I043bcaf358575688b223bc3631506e148b47fd58 > Reviewed-on: https://webrtc-review.googlesource.com/88220 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23971} TBR=danilchap@webrtc.org,stefan@webrtc.org,philipel@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: none Change-Id: If99bcabdfe3cae7094f24e407bbe2f47233e46e3 Reviewed-on: https://webrtc-review.googlesource.com/88820 Commit-Queue: JT Teh <jtteh@webrtc.org> Reviewed-by: Zeke Chin <tkchin@webrtc.org> Reviewed-by: JT Teh <jtteh@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23993}
89 lines
3.3 KiB
C++
89 lines
3.3 KiB
C++
/*
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* Copyright (c) 2018 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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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
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#include "absl/types/variant.h"
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#include "api/video/video_content_type.h"
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#include "api/video/video_rotation.h"
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#include "api/video/video_timing.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/video_coding/codecs/h264/include/h264_globals.h"
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#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
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#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
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namespace webrtc {
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using RTPVideoTypeHeader =
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absl::variant<RTPVideoHeaderVP8, RTPVideoHeaderVP9, RTPVideoHeaderH264>;
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struct RTPVideoHeader {
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RTPVideoHeader();
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RTPVideoHeader(const RTPVideoHeader& other);
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// TODO(philipel): Remove when downstream projects have been updated.
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RTPVideoHeaderVP8& vp8() {
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if (!absl::holds_alternative<RTPVideoHeaderVP8>(video_type_header))
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video_type_header.emplace<RTPVideoHeaderVP8>();
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return absl::get<RTPVideoHeaderVP8>(video_type_header);
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}
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// TODO(philipel): Remove when downstream projects have been updated.
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const RTPVideoHeaderVP8& vp8() const {
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if (!absl::holds_alternative<RTPVideoHeaderVP8>(video_type_header))
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video_type_header.emplace<RTPVideoHeaderVP8>();
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return absl::get<RTPVideoHeaderVP8>(video_type_header);
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}
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// TODO(philipel): Remove when downstream projects have been updated.
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RTPVideoHeaderVP9& vp9() {
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if (!absl::holds_alternative<RTPVideoHeaderVP9>(video_type_header))
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video_type_header.emplace<RTPVideoHeaderVP9>();
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return absl::get<RTPVideoHeaderVP9>(video_type_header);
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}
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// TODO(philipel): Remove when downstream projects have been updated.
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const RTPVideoHeaderVP9& vp9() const {
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if (!absl::holds_alternative<RTPVideoHeaderVP9>(video_type_header))
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video_type_header.emplace<RTPVideoHeaderVP9>();
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return absl::get<RTPVideoHeaderVP9>(video_type_header);
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}
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// TODO(philipel): Remove when downstream projects have been updated.
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RTPVideoHeaderH264& h264() {
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if (!absl::holds_alternative<RTPVideoHeaderH264>(video_type_header))
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video_type_header.emplace<RTPVideoHeaderH264>();
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return absl::get<RTPVideoHeaderH264>(video_type_header);
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}
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// TODO(philipel): Remove when downstream projects have been updated.
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const RTPVideoHeaderH264& h264() const {
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if (!absl::holds_alternative<RTPVideoHeaderH264>(video_type_header))
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video_type_header.emplace<RTPVideoHeaderH264>();
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return absl::get<RTPVideoHeaderH264>(video_type_header);
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}
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uint16_t width;
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uint16_t height;
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VideoRotation rotation;
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PlayoutDelay playout_delay;
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VideoContentType content_type;
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VideoSendTiming video_timing;
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bool is_first_packet_in_frame;
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uint8_t simulcastIdx;
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VideoCodecType codec;
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// TODO(philipel): remove mutable when downstream projects have been updated.
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mutable RTPVideoTypeHeader video_type_header;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
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