mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-19 00:27:51 +01:00
29d8846df9
TBR=stefan@webrtc.org Bug: none Change-Id: Ia2f728ea3377754a16a0b081e25c4479fe211b3e Reviewed-on: https://webrtc-review.googlesource.com/93024 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24243}
67 lines
2.4 KiB
C++
67 lines
2.4 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 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:
|
|
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
|