tor-browser

The Tor Browser
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encoded_frame.h (3987B)

      1 /*
      2 *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
      3 *
      4 *  Use of this source code is governed by a BSD-style license
      5 *  that can be found in the LICENSE file in the root of the source
      6 *  tree. An additional intellectual property rights grant can be found
      7 *  in the file PATENTS.  All contributing project authors may
      8 *  be found in the AUTHORS file in the root of the source tree.
      9 */
     10 
     11 #ifndef API_VIDEO_ENCODED_FRAME_H_
     12 #define API_VIDEO_ENCODED_FRAME_H_
     13 
     14 #include <stddef.h>
     15 #include <stdint.h>
     16 
     17 #include <optional>
     18 
     19 #include "api/units/timestamp.h"
     20 #include "api/video/corruption_detection/frame_instrumentation_data.h"
     21 #include "api/video/encoded_image.h"
     22 #include "api/video/video_codec_type.h"
     23 #include "modules/rtp_rtcp/source/rtp_video_header.h"
     24 #include "modules/video_coding/include/video_codec_interface.h"
     25 
     26 namespace webrtc {
     27 
     28 // TODO(philipel): Move transport specific info out of EncodedFrame.
     29 // NOTE: This class is still under development and may change without notice.
     30 class EncodedFrame : public EncodedImage {
     31 public:
     32  static const uint8_t kMaxFrameReferences = 5;
     33 
     34  EncodedFrame() = default;
     35  EncodedFrame(const EncodedFrame&) = default;
     36  virtual ~EncodedFrame() {}
     37 
     38  // When this frame was received.
     39  // TODO(bugs.webrtc.org/13756): Use Timestamp instead of int.
     40  virtual int64_t ReceivedTime() const { return -1; }
     41  // Returns a Timestamp from `ReceivedTime`, or nullopt if there is no receive
     42  // time.
     43  std::optional<Timestamp> ReceivedTimestamp() const;
     44 
     45  // When this frame should be rendered.
     46  // TODO(bugs.webrtc.org/13756): Use Timestamp instead of int.
     47  virtual int64_t RenderTime() const { return _renderTimeMs; }
     48  // TODO(bugs.webrtc.org/13756): Migrate to ReceivedTimestamp.
     49  int64_t RenderTimeMs() const { return _renderTimeMs; }
     50  // Returns a Timestamp from `RenderTime`, or nullopt if there is no
     51  // render time.
     52  std::optional<Timestamp> RenderTimestamp() const;
     53 
     54  // This information is currently needed by the timing calculation class.
     55  // TODO(philipel): Remove this function when a new timing class has
     56  //                 been implemented.
     57  virtual bool delayed_by_retransmission() const;
     58 
     59  bool is_keyframe() const { return num_references == 0; }
     60 
     61  void SetId(int64_t id) { id_ = id; }
     62  int64_t Id() const { return id_; }
     63 
     64  uint8_t PayloadType() const { return _payloadType; }
     65 
     66  void SetRenderTime(const int64_t renderTimeMs) {
     67    _renderTimeMs = renderTimeMs;
     68  }
     69 
     70  const webrtc::EncodedImage& EncodedImage() const {
     71    return static_cast<const webrtc::EncodedImage&>(*this);
     72  }
     73 
     74  const CodecSpecificInfo* CodecSpecific() const { return &_codecSpecificInfo; }
     75  void SetCodecSpecific(const CodecSpecificInfo* codec_specific) {
     76    _codecSpecificInfo = *codec_specific;
     77  }
     78  void SetFrameInstrumentationData(
     79      const std::optional<FrameInstrumentationData> frame_instrumentation) {
     80    _codecSpecificInfo.frame_instrumentation_data = frame_instrumentation;
     81  }
     82 
     83  // TODO(philipel): Add simple modify/access functions to prevent adding too
     84  // many `references`.
     85  size_t num_references = 0;
     86  int64_t references[kMaxFrameReferences];
     87  // Is this subframe the last one in the superframe (In RTP stream that would
     88  // mean that the last packet has a marker bit set).
     89  bool is_last_spatial_layer = true;
     90 
     91 protected:
     92  // TODO(https://bugs.webrtc.org/9378): Move RTP specifics down into a
     93  // transport-aware subclass, eg RtpFrameObject.
     94  void CopyCodecSpecific(const RTPVideoHeader* header);
     95 
     96  // TODO(https://bugs.webrtc.org/9378): Make fields private with
     97  // getters/setters as needed.
     98  int64_t _renderTimeMs = -1;
     99  uint8_t _payloadType = 0;
    100  CodecSpecificInfo _codecSpecificInfo;
    101  VideoCodecType _codec = kVideoCodecGeneric;
    102 
    103 private:
    104  // The ID of the frame is determined from RTP level information. The IDs are
    105  // used to describe order and dependencies between frames.
    106  int64_t id_ = -1;
    107 };
    108 
    109 }  // namespace webrtc
    110 
    111 #endif  // API_VIDEO_ENCODED_FRAME_H_