mirror of
https://github.com/mollyim/webrtc.git
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5daeff9c1f
This reverts commit dfbced6504.
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}
63 lines
2.3 KiB
C++
63 lines
2.3 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 "modules/rtp_rtcp/source/rtp_format.h"
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#include <utility>
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#include "modules/rtp_rtcp/source/rtp_format_h264.h"
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#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
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#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
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#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
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namespace webrtc {
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RtpPacketizer* RtpPacketizer::Create(VideoCodecType type,
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size_t max_payload_len,
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size_t last_packet_reduction_len,
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const RTPVideoHeader* rtp_video_header,
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FrameType frame_type) {
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switch (type) {
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case kVideoCodecH264:
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RTC_CHECK(rtp_video_header);
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return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
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rtp_video_header->h264().packetization_mode);
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case kVideoCodecVP8:
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RTC_CHECK(rtp_video_header);
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return new RtpPacketizerVp8(rtp_video_header->vp8(), max_payload_len,
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last_packet_reduction_len);
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case kVideoCodecVP9:
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RTC_CHECK(rtp_video_header);
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return new RtpPacketizerVp9(rtp_video_header->vp9(), max_payload_len,
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last_packet_reduction_len);
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case kVideoCodecGeneric:
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return new RtpPacketizerGeneric(frame_type, max_payload_len,
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last_packet_reduction_len);
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default:
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RTC_NOTREACHED();
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}
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return nullptr;
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}
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RtpDepacketizer* RtpDepacketizer::Create(VideoCodecType type) {
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switch (type) {
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case kVideoCodecH264:
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return new RtpDepacketizerH264();
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case kVideoCodecVP8:
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return new RtpDepacketizerVp8();
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case kVideoCodecVP9:
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return new RtpDepacketizerVp9();
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case kVideoCodecGeneric:
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return new RtpDepacketizerGeneric();
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default:
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RTC_NOTREACHED();
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}
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return nullptr;
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}
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} // namespace webrtc
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