webrtc/modules/video_coding/frame_object.h
Niels Möller 87e2d785a0 Prepare for splitting FrameType into AudioFrameType and VideoFrameType
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.

After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.

Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
2019-03-07 10:12:57 +00:00

66 lines
2.1 KiB
C++

/*
* 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 "absl/types/optional.h"
#include "api/video/encoded_frame.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
namespace webrtc {
namespace video_coding {
class PacketBuffer;
class RtpFrameObject : public EncodedFrame {
public:
RtpFrameObject(PacketBuffer* packet_buffer,
uint16_t first_seq_num,
uint16_t last_seq_num,
size_t frame_size,
int times_nacked,
int64_t first_packet_received_time,
int64_t last_packet_received_time);
~RtpFrameObject() override;
uint16_t first_seq_num() const;
uint16_t last_seq_num() const;
int times_nacked() const;
VideoFrameType frame_type() const;
VideoCodecType codec_type() const;
int64_t ReceivedTime() const override;
int64_t RenderTime() const override;
bool delayed_by_retransmission() const override;
absl::optional<RTPVideoHeader> GetRtpVideoHeader() const;
absl::optional<RtpGenericFrameDescriptor> GetGenericFrameDescriptor() const;
absl::optional<FrameMarking> GetFrameMarking() const;
private:
void AllocateBitstreamBuffer(size_t frame_size);
rtc::scoped_refptr<PacketBuffer> packet_buffer_;
VideoFrameType frame_type_;
VideoCodecType codec_type_;
uint16_t first_seq_num_;
uint16_t last_seq_num_;
int64_t last_packet_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_