blob: 69ab6f03c8120e808a40c1b87a6e5cc15617e740 [file] [log] [blame]
* 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.
#include "webrtc/api/optional.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/encoded_frame.h"
namespace webrtc {
namespace video_coding {
class FrameObject : public webrtc::VCMEncodedFrame {
static const uint8_t kMaxFrameReferences = 5;
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 {
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);
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<RTPVideoTypeHeader> GetCodecHeader() const;
rtc::scoped_refptr<PacketBuffer> 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