webrtc/modules/rtp_rtcp/source/transformable_encoded_frame.cc
Marina Ciocea 3a087a839a Transform encoded frame in RTPSenderVideo.
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I491ecefc60d184b75128799274c7d7efcf907d2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169128
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30666}
2020-03-03 08:17:49 +00:00

55 lines
1.7 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/transformable_encoded_frame.h"
namespace webrtc {
TransformableEncodedFrame::TransformableEncodedFrame(
rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data,
const RTPVideoHeader& video_header,
int payload_type,
absl::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
int64_t capture_time_ms,
const RTPFragmentationHeader* fragmentation,
absl::optional<int64_t> expected_retransmission_time_ms)
: video_header_(video_header),
codec_type_(codec_type),
expected_retransmission_time_ms_(expected_retransmission_time_ms) {
SetEncodedData(encoded_data);
_payloadType = payload_type;
SetTimestamp(rtp_timestamp);
capture_time_ms_ = capture_time_ms;
if (fragmentation) {
fragmentation_header_ = std::make_unique<RTPFragmentationHeader>();
fragmentation_header_->CopyFrom(*fragmentation);
}
}
TransformableEncodedFrame::~TransformableEncodedFrame() = default;
const RTPVideoHeader& TransformableEncodedFrame::video_header() const {
return video_header_;
}
absl::optional<VideoCodecType> TransformableEncodedFrame::codec_type() const {
return codec_type_;
}
int64_t TransformableEncodedFrame::ReceivedTime() const {
return 0;
}
int64_t TransformableEncodedFrame::RenderTime() const {
return _renderTimeMs;
}
} // namespace webrtc