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Remove deprecated type alias for RtpVideoCodecTypes.

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First phase of this removal landed with cl https://webrtc-review.googlesource.com/79561

Bug: webrtc:8995
Change-Id: I9dc152e2f1bac17e2959af7e18106760ca5435c8
Reviewed-on: https://webrtc-review.googlesource.com/95720
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24447}
This commit is contained in:
Kári Tristan Helgason
2018-08-27 13:26:52 +02:00
committed by Commit Bot
parent 638d4d375f
commit 01a89904c0
4 changed files with 3 additions and 27 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
modules/rtp_rtcp/source/rtp_format.cc
View File

@ -23,7 +23,7 @@ RtpPacketizer* RtpPacketizer::Create(VideoCodecType type,
size_t last_packet_reduction_len,
const RTPVideoHeader* rtp_video_header,
FrameType frame_type) {
RTC_CHECK(type == kVideoCodecGeneric || rtp_video_header);
RTC_CHECK(rtp_video_header);
switch (type) {
case kVideoCodecH264: {
const auto& h264 =
@ -40,13 +40,10 @@ RtpPacketizer* RtpPacketizer::Create(VideoCodecType type,
return new RtpPacketizerVp9(vp9, max_payload_len,
last_packet_reduction_len);
}
case kVideoCodecGeneric:
RTC_CHECK(rtp_video_header);
default:
return new RtpPacketizerGeneric(*rtp_video_header, frame_type,
max_payload_len,
last_packet_reduction_len);
default:
RTC_NOTREACHED();
}
return nullptr;
}