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Delete RtpDepacketizer::Create factory

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Bug: webrtc:11152
Change-Id: I09824b97506a11f917cd71f2f0d30306538eee13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163023
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30178}
This commit is contained in:
Danil Chapovalov
2020-01-07 10:36:49 +01:00
committed by Commit Bot
parent 0b3a6e383e
commit 57218b4e22
5 changed files with 11 additions and 40 deletions
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21
modules/rtp_rtcp/source/rtp_format.cc
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@ -13,7 +13,6 @@
#include <memory>
#include "absl/types/variant.h"
#include "modules/rtp_rtcp/source/rtp_depacketizer_av1.h"
#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"
@ -143,24 +142,4 @@ std::vector<int> RtpPacketizer::SplitAboutEqually(
return result;
}
RtpDepacketizer* RtpDepacketizer::Create(absl::optional<VideoCodecType> type) {
if (!type) {
// Use raw depacketizer.
return new RtpDepacketizerGeneric(/*generic_header_enabled=*/false);
}
switch (*type) {
case kVideoCodecH264:
return new RtpDepacketizerH264();
case kVideoCodecVP8:
return new RtpDepacketizerVp8();
case kVideoCodecVP9:
return new RtpDepacketizerVp9();
case kVideoCodecAV1:
return new RtpDepacketizerAv1();
default:
return new RtpDepacketizerGeneric(/*generic_header_enabled=*/true);
}
}
} // namespace webrtc