ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
93 lines
3.0 KiB
C++
93 lines
3.0 KiB
C++
/*
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* Copyright (c) 2012 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 WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
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#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/video_coding/main/interface/video_coding.h"
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#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
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#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
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#include "webrtc/modules/video_coding/main/source/session_info.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class VCMFrameBuffer : public VCMEncodedFrame {
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public:
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VCMFrameBuffer();
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virtual ~VCMFrameBuffer();
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VCMFrameBuffer(const VCMFrameBuffer& rhs);
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virtual void Reset();
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VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
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int64_t timeInMs,
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VCMDecodeErrorMode decode_error_mode,
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const FrameData& frame_data);
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// State
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// Get current state of frame
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VCMFrameBufferStateEnum GetState() const;
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// Get current state and timestamp of frame
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VCMFrameBufferStateEnum GetState(uint32_t& timeStamp) const;
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void PrepareForDecode(bool continuous);
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bool IsRetransmitted() const;
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bool IsSessionComplete() const;
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bool HaveFirstPacket() const;
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bool HaveLastPacket() const;
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int NumPackets() const;
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// Makes sure the session contain a decodable stream.
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void MakeSessionDecodable();
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// Sequence numbers
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// Get lowest packet sequence number in frame
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int32_t GetLowSeqNum() const;
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// Get highest packet sequence number in frame
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int32_t GetHighSeqNum() const;
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int PictureId() const;
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int TemporalId() const;
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bool LayerSync() const;
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int Tl0PicId() const;
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bool NonReference() const;
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void SetGofInfo(const GofInfoVP9& gof_info, size_t idx);
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// Increments a counter to keep track of the number of packets of this frame
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// which were NACKed before they arrived.
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void IncrementNackCount();
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// Returns the number of packets of this frame which were NACKed before they
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// arrived.
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int16_t GetNackCount() const;
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int64_t LatestPacketTimeMs() const;
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webrtc::FrameType FrameType() const;
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void SetPreviousFrameLoss();
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// The number of packets discarded because the decoder can't make use of them.
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int NotDecodablePackets() const;
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private:
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void SetState(VCMFrameBufferStateEnum state); // Set state of frame
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VCMFrameBufferStateEnum _state; // Current state of the frame
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VCMSessionInfo _sessionInfo;
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uint16_t _nackCount;
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int64_t _latestPacketTimeMs;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
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