mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 22:30:40 +01:00
0040b66ad3
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}
63 lines
1.9 KiB
C++
63 lines
1.9 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"
|
|
|
|
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 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;
|
|
absl::optional<RTPVideoTypeHeader> GetCodecHeader() const;
|
|
|
|
private:
|
|
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
|
|
|
|
#endif // MODULES_VIDEO_CODING_FRAME_OBJECT_H_
|