Repo created

TMessagesProj/jni/voip/webrtc/api/video/encoded_frame.h

@@ -0,0 +1,106 @@
+/*
+ *  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_