/* * 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 MODULES_VIDEO_CODING_FRAME_OBJECT_H_ #define MODULES_VIDEO_CODING_FRAME_OBJECT_H_ #include "api/optional.h" #include "common_types.h" // NOLINT(build/include) #include "modules/include/module_common_types.h" #include "modules/video_coding/encoded_frame.h" namespace webrtc { namespace video_coding { class FrameObject : public webrtc::VCMEncodedFrame { public: static const uint8_t kMaxFrameReferences = 5; FrameObject(); virtual ~FrameObject() {} virtual bool GetBitstream(uint8_t* destination) const = 0; // The capture timestamp of this frame. virtual uint32_t Timestamp() const = 0; // When this frame was received. virtual int64_t ReceivedTime() const = 0; // When this frame should be rendered. virtual int64_t RenderTime() const = 0; // 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 { return 0; } size_t size() const { return _length; } bool is_keyframe() const { return num_references == 0; } // The tuple (|picture_id|, |spatial_layer|) uniquely identifies a frame // object. For codec types that don't necessarily have picture ids they // have to be constructed from the header data relevant to that codec. int64_t picture_id; uint8_t spatial_layer; uint32_t timestamp; // TODO(philipel): Add simple modify/access functions to prevent adding too // many |references|. size_t num_references; int64_t references[kMaxFrameReferences]; bool inter_layer_predicted; }; class PacketBuffer; class RtpFrameObject : public FrameObject { public: RtpFrameObject(PacketBuffer* packet_buffer, uint16_t first_seq_num, uint16_t last_seq_num, size_t frame_size, int times_nacked, int64_t received_time); ~RtpFrameObject(); uint16_t first_seq_num() const; uint16_t last_seq_num() const; int times_nacked() const; enum FrameType frame_type() const; VideoCodecType codec_type() const; bool GetBitstream(uint8_t* destination) const override; uint32_t Timestamp() const override; int64_t ReceivedTime() const override; int64_t RenderTime() const override; bool delayed_by_retransmission() const override; rtc::Optional GetCodecHeader() const; private: rtc::scoped_refptr packet_buffer_; enum FrameType frame_type_; VideoCodecType codec_type_; uint16_t first_seq_num_; uint16_t last_seq_num_; uint32_t timestamp_; int64_t received_time_; // Equal to times nacked of the packet with the highet times nacked // belonging to this frame. int times_nacked_; }; } // namespace video_coding } // namespace webrtc #endif // MODULES_VIDEO_CODING_FRAME_OBJECT_H_