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7120742701
Now that we have moved WebRTC from src/webrtc to src/, common_types.h and typedefs.h are triggering a cpplint error. The cpplint complaint is: Include the directory when naming .h files [build/include] [4] This CL disables the error but we have to remove these two headers from the root directory. NOPRESUBMIT=true Bug: webrtc:5876 Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333 Reviewed-on: https://webrtc-review.googlesource.com/1577 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19859}
104 lines
3.2 KiB
C++
104 lines
3.2 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 MODULES_VIDEO_CODING_FRAME_OBJECT_H_
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#define MODULES_VIDEO_CODING_FRAME_OBJECT_H_
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#include "api/optional.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/include/module_common_types.h"
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#include "modules/video_coding/encoded_frame.h"
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namespace webrtc {
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namespace video_coding {
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class FrameObject : public webrtc::VCMEncodedFrame {
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public:
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static const uint8_t kMaxFrameReferences = 5;
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FrameObject();
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virtual ~FrameObject() {}
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virtual bool GetBitstream(uint8_t* destination) const = 0;
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// The capture timestamp of this frame.
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virtual uint32_t Timestamp() const = 0;
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// When this frame was received.
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virtual int64_t ReceivedTime() const = 0;
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// When this frame should be rendered.
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virtual int64_t RenderTime() const = 0;
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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 { return 0; }
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size_t size() const { return _length; }
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bool is_keyframe() const { return num_references == 0; }
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// The tuple (|picture_id|, |spatial_layer|) uniquely identifies a frame
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// object. For codec types that don't necessarily have picture ids they
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// have to be constructed from the header data relevant to that codec.
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int64_t picture_id;
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uint8_t spatial_layer;
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uint32_t timestamp;
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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;
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int64_t references[kMaxFrameReferences];
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bool inter_layer_predicted;
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};
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class PacketBuffer;
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class RtpFrameObject : public FrameObject {
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public:
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RtpFrameObject(PacketBuffer* packet_buffer,
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uint16_t first_seq_num,
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uint16_t last_seq_num,
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size_t frame_size,
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int times_nacked,
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int64_t received_time);
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~RtpFrameObject();
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uint16_t first_seq_num() const;
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uint16_t last_seq_num() const;
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int times_nacked() const;
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enum FrameType frame_type() const;
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VideoCodecType codec_type() const;
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bool GetBitstream(uint8_t* destination) const override;
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uint32_t Timestamp() const override;
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int64_t ReceivedTime() const override;
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int64_t RenderTime() const override;
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bool delayed_by_retransmission() const override;
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rtc::Optional<RTPVideoTypeHeader> GetCodecHeader() const;
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private:
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rtc::scoped_refptr<PacketBuffer> packet_buffer_;
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enum FrameType frame_type_;
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VideoCodecType codec_type_;
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uint16_t first_seq_num_;
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uint16_t last_seq_num_;
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uint32_t timestamp_;
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int64_t received_time_;
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// Equal to times nacked of the packet with the highet times nacked
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// belonging to this frame.
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int times_nacked_;
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};
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} // namespace video_coding
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_FRAME_OBJECT_H_
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