webrtc/modules/rtp_rtcp/source/rtp_format.cc
Niels Möller 520ca4e3b8 Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
Bug: webrtc:8995
Change-Id: I0b44aa26f2f6a81aec7ca1281b8513d8e03228b8
Reviewed-on: https://webrtc-review.googlesource.com/79561
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23507}
2018-06-04 11:53:17 +00:00

63 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 RTPVideoTypeHeader* rtp_type_header,
FrameType frame_type) {
switch (type) {
case kVideoCodecH264:
RTC_CHECK(rtp_type_header);
return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
rtp_type_header->H264.packetization_mode);
case kVideoCodecVP8:
RTC_CHECK(rtp_type_header);
return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len,
last_packet_reduction_len);
case kVideoCodecVP9:
RTC_CHECK(rtp_type_header);
return new RtpPacketizerVp9(rtp_type_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