webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer.cc
Peter Thatcher 9aeaa25d4c
Update to WebRTC 4103 (M83) (#12)
* Merge in branch-heads/4103 (M83)

* Disable legacy DTLS protocols (before 1.2)

* Update sdk/objc modifications for upstream changes

* Update ios and mac deployment targets

Co-authored-by: Jim Gustafson <jim@signal.org>
2020-06-25 11:14:34 -07:00

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1.3 KiB
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/*
* Copyright (c) 2020 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.h"
#include <stddef.h>
#include <stdint.h>
#include "api/array_view.h"
#include "api/scoped_refptr.h"
#include "api/video/encoded_image.h"
#include "rtc_base/checks.h"
namespace webrtc {
rtc::scoped_refptr<EncodedImageBuffer> VideoRtpDepacketizer::AssembleFrame(
rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads) {
size_t frame_size = 0;
for (rtc::ArrayView<const uint8_t> payload : rtp_payloads) {
frame_size += payload.size();
}
rtc::scoped_refptr<EncodedImageBuffer> bitstream =
EncodedImageBuffer::Create(frame_size);
uint8_t* write_at = bitstream->data();
for (rtc::ArrayView<const uint8_t> payload : rtp_payloads) {
memcpy(write_at, payload.data(), payload.size());
write_at += payload.size();
}
RTC_DCHECK_EQ(write_at - bitstream->data(), bitstream->size());
return bitstream;
}
} // namespace webrtc