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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
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(RtpVideoCodecTypes 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 RTPVideoTypeHeader* rtp_type_header,
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FrameType frame_type) {
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switch (type) {
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case kRtpVideoH264:
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RTC_CHECK(rtp_type_header);
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return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
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rtp_type_header->H264.packetization_mode);
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case kRtpVideoVp8:
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RTC_CHECK(rtp_type_header);
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return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len,
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last_packet_reduction_len);
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case kRtpVideoVp9:
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RTC_CHECK(rtp_type_header);
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return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len,
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last_packet_reduction_len);
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case kRtpVideoGeneric:
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return new RtpPacketizerGeneric(frame_type, max_payload_len,
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last_packet_reduction_len);
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case kRtpVideoNone:
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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(RtpVideoCodecTypes type) {
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switch (type) {
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case kRtpVideoH264:
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return new RtpDepacketizerH264();
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case kRtpVideoVp8:
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return new RtpDepacketizerVp8();
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case kRtpVideoVp9:
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return new RtpDepacketizerVp9();
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case kRtpVideoGeneric:
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return new RtpDepacketizerGeneric();
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case kRtpVideoNone:
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assert(false);
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}
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return nullptr;
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}
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} // namespace webrtc
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