mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
b6bf0b2546
To free up RtpVideoHeader::generic field for codec agnostic details from an rtp header extension. Bug: webrtc:10342 Change-Id: I7b9d869b2ecfedb96dfd860be47ed8dffa058749 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166175 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30396}
72 lines
2.5 KiB
C++
72 lines
2.5 KiB
C++
/*
|
|
* Copyright (c) 2019 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/video_rtp_depacketizer_generic.h"
|
|
|
|
#include <stddef.h>
|
|
#include <stdint.h>
|
|
|
|
#include <utility>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "modules/rtp_rtcp/source/rtp_video_header.h"
|
|
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
|
|
#include "rtc_base/copy_on_write_buffer.h"
|
|
#include "rtc_base/logging.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
constexpr uint8_t kKeyFrameBit = 0b0000'0001;
|
|
constexpr uint8_t kFirstPacketBit = 0b0000'0010;
|
|
// If this bit is set, there will be an extended header contained in this
|
|
// packet. This was added later so old clients will not send this.
|
|
constexpr uint8_t kExtendedHeaderBit = 0b0000'0100;
|
|
|
|
constexpr size_t kGenericHeaderLength = 1;
|
|
constexpr size_t kExtendedHeaderLength = 2;
|
|
} // namespace
|
|
|
|
absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
|
|
VideoRtpDepacketizerGeneric::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
|
|
if (rtp_payload.size() == 0) {
|
|
RTC_LOG(LS_WARNING) << "Empty payload.";
|
|
return absl::nullopt;
|
|
}
|
|
absl::optional<ParsedRtpPayload> parsed(absl::in_place);
|
|
const uint8_t* payload_data = rtp_payload.cdata();
|
|
|
|
uint8_t generic_header = payload_data[0];
|
|
size_t offset = kGenericHeaderLength;
|
|
|
|
parsed->video_header.frame_type = (generic_header & kKeyFrameBit)
|
|
? VideoFrameType::kVideoFrameKey
|
|
: VideoFrameType::kVideoFrameDelta;
|
|
parsed->video_header.is_first_packet_in_frame =
|
|
(generic_header & kFirstPacketBit) != 0;
|
|
parsed->video_header.codec = kVideoCodecGeneric;
|
|
parsed->video_header.width = 0;
|
|
parsed->video_header.height = 0;
|
|
|
|
if (generic_header & kExtendedHeaderBit) {
|
|
if (rtp_payload.size() < offset + kExtendedHeaderLength) {
|
|
RTC_LOG(LS_WARNING) << "Too short payload for generic header.";
|
|
return absl::nullopt;
|
|
}
|
|
parsed->video_header.video_type_header
|
|
.emplace<RTPVideoHeaderLegacyGeneric>()
|
|
.picture_id = ((payload_data[1] & 0x7F) << 8) | payload_data[2];
|
|
offset += kExtendedHeaderLength;
|
|
}
|
|
|
|
parsed->video_payload =
|
|
rtp_payload.Slice(offset, rtp_payload.size() - offset);
|
|
return parsed;
|
|
}
|
|
} // namespace webrtc
|