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/*
* Copyright (c) 2016 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 API_VIDEO_ENCODED_FRAME_H_
#define API_VIDEO_ENCODED_FRAME_H_
#include <stddef.h>
#include <stdint.h>
#include "absl/types/optional.h"
#include "api/units/timestamp.h"
#include "api/video/encoded_image.h"
#include "api/video/video_codec_type.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "modules/video_coding/include/video_coding_defines.h"
namespace webrtc {
// TODO(philipel): Move transport specific info out of EncodedFrame.
// NOTE: This class is still under development and may change without notice.
class EncodedFrame : public EncodedImage {
public:
static const uint8_t kMaxFrameReferences = 5;
EncodedFrame() = default;
EncodedFrame(const EncodedFrame&) = default;
virtual ~EncodedFrame() {}
// When this frame was received.
// TODO(bugs.webrtc.org/13756): Use Timestamp instead of int.
virtual int64_t ReceivedTime() const { return -1; }
// Returns a Timestamp from `ReceivedTime`, or nullopt if there is no receive
// time.
absl::optional<webrtc::Timestamp> ReceivedTimestamp() const;
// When this frame should be rendered.
// TODO(bugs.webrtc.org/13756): Use Timestamp instead of int.
virtual int64_t RenderTime() const { return _renderTimeMs; }
// TODO(bugs.webrtc.org/13756): Migrate to ReceivedTimestamp.
int64_t RenderTimeMs() const { return _renderTimeMs; }
// Returns a Timestamp from `RenderTime`, or nullopt if there is no
// render time.
absl::optional<webrtc::Timestamp> RenderTimestamp() const;
// This information is currently needed by the timing calculation class.
// TODO(philipel): Remove this function when a new timing class has
// been implemented.
virtual bool delayed_by_retransmission() const;
bool is_keyframe() const { return num_references == 0; }
void SetId(int64_t id) { id_ = id; }
int64_t Id() const { return id_; }
uint8_t PayloadType() const { return _payloadType; }
void SetRenderTime(const int64_t renderTimeMs) {
_renderTimeMs = renderTimeMs;
}
const webrtc::EncodedImage& EncodedImage() const {
return static_cast<const webrtc::EncodedImage&>(*this);
}
const CodecSpecificInfo* CodecSpecific() const { return &_codecSpecificInfo; }
void SetCodecSpecific(const CodecSpecificInfo* codec_specific) {
_codecSpecificInfo = *codec_specific;
}
// TODO(philipel): Add simple modify/access functions to prevent adding too
// many `references`.
size_t num_references = 0;
int64_t references[kMaxFrameReferences];
// Is this subframe the last one in the superframe (In RTP stream that would
// mean that the last packet has a marker bit set).
bool is_last_spatial_layer = true;
protected:
// TODO(https://bugs.webrtc.org/9378): Move RTP specifics down into a
// transport-aware subclass, eg RtpFrameObject.
void CopyCodecSpecific(const RTPVideoHeader* header);
// TODO(https://bugs.webrtc.org/9378): Make fields private with
// getters/setters as needed.
int64_t _renderTimeMs = -1;
uint8_t _payloadType = 0;
CodecSpecificInfo _codecSpecificInfo;
VideoCodecType _codec = kVideoCodecGeneric;
private:
// The ID of the frame is determined from RTP level information. The IDs are
// used to describe order and dependencies between frames.
int64_t id_ = -1;
};
} // namespace webrtc
#endif // API_VIDEO_ENCODED_FRAME_H_
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