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9d8eaac4ee
And delete corresponding dependencies on :webrtc_common. After this change, common_types.h is included directly only from code in the following directories: api/ api/video/ api/video_codecs/ common_video/libyuv/include/ media/base/ modules/remote_bitrate_estimator/ modules/rtp_rtcp/source/ modules/video_coding/codecs/vp9/ There remains plenty of indirect dependencies on the types declared in common_types.h, but the fewer direct dependencies should make it easier to find the proper place for each type. Bug: webrtc:5876 Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27376}
65 lines
2.1 KiB
C++
65 lines
2.1 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 "absl/types/optional.h"
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#include "api/video/encoded_frame.h"
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
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namespace webrtc {
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namespace video_coding {
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class PacketBuffer;
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class RtpFrameObject : public EncodedFrame {
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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 first_packet_received_time,
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int64_t last_packet_received_time);
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~RtpFrameObject() override;
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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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VideoFrameType frame_type() const;
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VideoCodecType codec_type() const;
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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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absl::optional<RTPVideoHeader> GetRtpVideoHeader() const;
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absl::optional<RtpGenericFrameDescriptor> GetGenericFrameDescriptor() const;
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absl::optional<FrameMarking> GetFrameMarking() const;
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private:
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void AllocateBitstreamBuffer(size_t frame_size);
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rtc::scoped_refptr<PacketBuffer> packet_buffer_;
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VideoFrameType 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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int64_t last_packet_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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