modules/video_coding refactorings
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.
To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test
BUG=webrtc: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 --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417283007 .
Cr-Commit-Position: refs/heads/master@{#10694}
diff --git a/webrtc/modules/video_coding/frame_buffer.h b/webrtc/modules/video_coding/frame_buffer.h
new file mode 100644
index 0000000..f5a707e
--- /dev/null
+++ b/webrtc/modules/video_coding/frame_buffer.h
@@ -0,0 +1,92 @@
+/*
+ * Copyright (c) 2012 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 WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/encoded_frame.h"
+#include "webrtc/modules/video_coding/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/session_info.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class VCMFrameBuffer : public VCMEncodedFrame {
+ public:
+ VCMFrameBuffer();
+ virtual ~VCMFrameBuffer();
+
+ VCMFrameBuffer(const VCMFrameBuffer& rhs);
+
+ virtual void Reset();
+
+ VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
+ int64_t timeInMs,
+ VCMDecodeErrorMode decode_error_mode,
+ const FrameData& frame_data);
+
+ // State
+ // Get current state of frame
+ VCMFrameBufferStateEnum GetState() const;
+ // Get current state and timestamp of frame
+ VCMFrameBufferStateEnum GetState(uint32_t& timeStamp) const;
+ void PrepareForDecode(bool continuous);
+
+ bool IsRetransmitted() const;
+ bool IsSessionComplete() const;
+ bool HaveFirstPacket() const;
+ bool HaveLastPacket() const;
+ int NumPackets() const;
+ // Makes sure the session contain a decodable stream.
+ void MakeSessionDecodable();
+
+ // Sequence numbers
+ // Get lowest packet sequence number in frame
+ int32_t GetLowSeqNum() const;
+ // Get highest packet sequence number in frame
+ int32_t GetHighSeqNum() const;
+
+ int PictureId() const;
+ int TemporalId() const;
+ bool LayerSync() const;
+ int Tl0PicId() const;
+ bool NonReference() const;
+
+ void SetGofInfo(const GofInfoVP9& gof_info, size_t idx);
+
+ // Increments a counter to keep track of the number of packets of this frame
+ // which were NACKed before they arrived.
+ void IncrementNackCount();
+ // Returns the number of packets of this frame which were NACKed before they
+ // arrived.
+ int16_t GetNackCount() const;
+
+ int64_t LatestPacketTimeMs() const;
+
+ webrtc::FrameType FrameType() const;
+ void SetPreviousFrameLoss();
+
+ // The number of packets discarded because the decoder can't make use of them.
+ int NotDecodablePackets() const;
+
+ private:
+ void SetState(VCMFrameBufferStateEnum state); // Set state of frame
+
+ VCMFrameBufferStateEnum _state; // Current state of the frame
+ VCMSessionInfo _sessionInfo;
+ uint16_t _nackCount;
+ int64_t _latestPacketTimeMs;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_