blob: 1078a668100059e70ead3f9a6bcf49ecbeb78dab [file] [log] [blame]
/*
* 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) {
RTC_CHECK(type == kVideoCodecGeneric || rtp_video_header);
switch (type) {
case kVideoCodecH264: {
const auto& h264 =
absl::get<RTPVideoHeaderH264>(rtp_video_header->video_type_header);
return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
h264.packetization_mode);
}
case kVideoCodecVP8:
return new RtpPacketizerVp8(rtp_video_header->vp8(), max_payload_len,
last_packet_reduction_len);
case kVideoCodecVP9: {
const auto& vp9 =
absl::get<RTPVideoHeaderVP9>(rtp_video_header->video_type_header);
return new RtpPacketizerVp9(vp9, max_payload_len,
last_packet_reduction_len);
}
case kVideoCodecGeneric:
RTC_CHECK(rtp_video_header);
return new RtpPacketizerGeneric(*rtp_video_header, 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();
default:
return new RtpDepacketizerGeneric();
}
}
} // namespace webrtc