Sami Kalliomäki 426a80ce08 Add extended header containing frame ID to the generic packetizer.
Also changes default value of frame ID in RTPVideoHeader to
kNoPictureId. Special care should be take so that picture ID will not
be set in RTPVideoHeader unless the client on the end supports
deserializing extended generic header.

Bug: webrtc:9582
Change-Id: Ib096373ed187f31e51d481193a2bda56de68f167
Reviewed-on: https://webrtc-review.googlesource.com/92084
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24250}
2018-08-09 14:05:39 +00:00

70 lines
2.5 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:
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();
case kVideoCodecGeneric:
return new RtpDepacketizerGeneric();
default:
RTC_NOTREACHED();
}
return nullptr;
}
} // namespace webrtc