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/*
* 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
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