Repo created

TMessagesProj/jni/voip/webrtc/modules/video_coding/packet_buffer.h

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+/*
+ *  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_PACKET_BUFFER_H_
+#define MODULES_VIDEO_CODING_PACKET_BUFFER_H_
+
+#include <memory>
+#include <queue>
+#include <set>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "api/rtp_packet_info.h"
+#include "api/units/timestamp.h"
+#include "api/video/encoded_image.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class PacketBuffer {
+ public:
+  struct Packet {
+    Packet() = default;
+    Packet(const RtpPacketReceived& rtp_packet,
+           const RTPVideoHeader& video_header);
+    Packet(const Packet&) = delete;
+    Packet(Packet&&) = delete;
+    Packet& operator=(const Packet&) = delete;
+    Packet& operator=(Packet&&) = delete;
+    ~Packet() = default;
+
+    VideoCodecType codec() const { return video_header.codec; }
+    int width() const { return video_header.width; }
+    int height() const { return video_header.height; }
+
+    bool is_first_packet_in_frame() const {
+      return video_header.is_first_packet_in_frame;
+    }
+    bool is_last_packet_in_frame() const {
+      return video_header.is_last_packet_in_frame;
+    }
+
+    // If all its previous packets have been inserted into the packet buffer.
+    // Set and used internally by the PacketBuffer.
+    bool continuous = false;
+    bool marker_bit = false;
+    uint8_t payload_type = 0;
+    uint16_t seq_num = 0;
+    uint32_t timestamp = 0;
+    int times_nacked = -1;
+
+    rtc::CopyOnWriteBuffer video_payload;
+    RTPVideoHeader video_header;
+  };
+  struct InsertResult {
+    std::vector<std::unique_ptr<Packet>> packets;
+    // Indicates if the packet buffer was cleared, which means that a key
+    // frame request should be sent.
+    bool buffer_cleared = false;
+  };
+
+  // Both `start_buffer_size` and `max_buffer_size` must be a power of 2.
+  PacketBuffer(size_t start_buffer_size, size_t max_buffer_size);
+  ~PacketBuffer();
+
+  ABSL_MUST_USE_RESULT InsertResult
+  InsertPacket(std::unique_ptr<Packet> packet);
+  ABSL_MUST_USE_RESULT InsertResult InsertPadding(uint16_t seq_num);
+  void ClearTo(uint16_t seq_num);
+  void Clear();
+
+  void ForceSpsPpsIdrIsH264Keyframe();
+  void ResetSpsPpsIdrIsH264Keyframe();
+
+ private:
+  void ClearInternal();
+
+  // Tries to expand the buffer.
+  bool ExpandBufferSize();
+
+  // Test if all previous packets has arrived for the given sequence number.
+  bool PotentialNewFrame(uint16_t seq_num) const;
+
+  // Test if all packets of a frame has arrived, and if so, returns packets to
+  // create frames.
+  std::vector<std::unique_ptr<Packet>> FindFrames(uint16_t seq_num);
+
+  void UpdateMissingPackets(uint16_t seq_num);
+
+  // buffer_.size() and max_size_ must always be a power of two.
+  const size_t max_size_;
+
+  // The fist sequence number currently in the buffer.
+  uint16_t first_seq_num_;
+
+  // If the packet buffer has received its first packet.
+  bool first_packet_received_;
+
+  // If the buffer is cleared to `first_seq_num_`.
+  bool is_cleared_to_first_seq_num_;
+
+  // Buffer that holds the the inserted packets and information needed to
+  // determine continuity between them.
+  std::vector<std::unique_ptr<Packet>> buffer_;
+
+  absl::optional<uint16_t> newest_inserted_seq_num_;
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_;
+
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> received_padding_;
+
+  // Indicates if we should require SPS, PPS, and IDR for a particular
+  // RTP timestamp to treat the corresponding frame as a keyframe.
+  bool sps_pps_idr_is_h264_keyframe_;
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_PACKET_BUFFER_H_