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