webrtc/modules/rtp_rtcp/source/transformable_encoded_frame.h
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

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2.1 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.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_TRANSFORMABLE_ENCODED_FRAME_H_
#define MODULES_RTP_RTCP_SOURCE_TRANSFORMABLE_ENCODED_FRAME_H_
#include <memory>
#include "absl/types/optional.h"
#include "api/video/encoded_frame.h"
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
namespace webrtc {
class TransformableEncodedFrame : public video_coding::EncodedFrame {
public:
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);
~TransformableEncodedFrame() override;
const RTPVideoHeader& video_header() const;
absl::optional<VideoCodecType> codec_type() const;
int64_t capture_time_ms() const { return capture_time_ms_; }
RTPFragmentationHeader* fragmentation_header() const {
return fragmentation_header_.get();
}
const absl::optional<int64_t>& expected_retransmission_time_ms() const {
return expected_retransmission_time_ms_;
}
// Implements EncodedFrame.
int64_t ReceivedTime() const override;
int64_t RenderTime() const override;
private:
RTPVideoHeader video_header_;
absl::optional<VideoCodecType> codec_type_ = absl::nullopt;
std::unique_ptr<RTPFragmentationHeader> fragmentation_header_;
absl::optional<int64_t> expected_retransmission_time_ms_ = absl::nullopt;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_TRANSFORMABLE_ENCODED_FRAME_H_