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55b593fb6b
Remove EncodedFrame::MissingFrame, as it was always false in actual in-use code anyway, and remove usages of the Decode missing_frames param within WebRTC. Uses/overrides in other projects will be cleaned up shortly, allowing that variant to be removed from the interface. Bug: webrtc:15444 Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Tony Herre <herre@google.com> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#40662}
106 lines
3.7 KiB
C++
106 lines
3.7 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_VIDEO_ENCODED_FRAME_H_
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#define API_VIDEO_ENCODED_FRAME_H_
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#include <stddef.h>
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "api/units/timestamp.h"
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#include "api/video/encoded_image.h"
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#include "api/video/video_codec_type.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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#include "modules/video_coding/include/video_codec_interface.h"
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#include "modules/video_coding/include/video_coding_defines.h"
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namespace webrtc {
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// TODO(philipel): Move transport specific info out of EncodedFrame.
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// NOTE: This class is still under development and may change without notice.
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class EncodedFrame : public EncodedImage {
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public:
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static const uint8_t kMaxFrameReferences = 5;
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EncodedFrame() = default;
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EncodedFrame(const EncodedFrame&) = default;
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virtual ~EncodedFrame() {}
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// When this frame was received.
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// TODO(bugs.webrtc.org/13756): Use Timestamp instead of int.
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virtual int64_t ReceivedTime() const { return -1; }
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// Returns a Timestamp from `ReceivedTime`, or nullopt if there is no receive
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// time.
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absl::optional<webrtc::Timestamp> ReceivedTimestamp() const;
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// When this frame should be rendered.
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// TODO(bugs.webrtc.org/13756): Use Timestamp instead of int.
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virtual int64_t RenderTime() const { return _renderTimeMs; }
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// TODO(bugs.webrtc.org/13756): Migrate to ReceivedTimestamp.
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int64_t RenderTimeMs() const { return _renderTimeMs; }
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// Returns a Timestamp from `RenderTime`, or nullopt if there is no
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// render time.
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absl::optional<webrtc::Timestamp> RenderTimestamp() const;
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// This information is currently needed by the timing calculation class.
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// TODO(philipel): Remove this function when a new timing class has
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// been implemented.
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virtual bool delayed_by_retransmission() const;
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bool is_keyframe() const { return num_references == 0; }
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void SetId(int64_t id) { id_ = id; }
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int64_t Id() const { return id_; }
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uint8_t PayloadType() const { return _payloadType; }
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void SetRenderTime(const int64_t renderTimeMs) {
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_renderTimeMs = renderTimeMs;
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}
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const webrtc::EncodedImage& EncodedImage() const {
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return static_cast<const webrtc::EncodedImage&>(*this);
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}
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const CodecSpecificInfo* CodecSpecific() const { return &_codecSpecificInfo; }
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void SetCodecSpecific(const CodecSpecificInfo* codec_specific) {
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_codecSpecificInfo = *codec_specific;
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}
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// TODO(philipel): Add simple modify/access functions to prevent adding too
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// many `references`.
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size_t num_references = 0;
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int64_t references[kMaxFrameReferences];
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// Is this subframe the last one in the superframe (In RTP stream that would
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// mean that the last packet has a marker bit set).
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bool is_last_spatial_layer = true;
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protected:
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// TODO(https://bugs.webrtc.org/9378): Move RTP specifics down into a
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// transport-aware subclass, eg RtpFrameObject.
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void CopyCodecSpecific(const RTPVideoHeader* header);
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// TODO(https://bugs.webrtc.org/9378): Make fields private with
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// getters/setters as needed.
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int64_t _renderTimeMs = -1;
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uint8_t _payloadType = 0;
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CodecSpecificInfo _codecSpecificInfo;
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VideoCodecType _codec = kVideoCodecGeneric;
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private:
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// The ID of the frame is determined from RTP level information. The IDs are
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// used to describe order and dependencies between frames.
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int64_t id_ = -1;
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};
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} // namespace webrtc
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#endif // API_VIDEO_ENCODED_FRAME_H_
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