/* * 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 #include #include #include #include "api/scoped_refptr.h" #include "modules/include/module_common_types.h" #include "modules/video_coding/packet.h" #include "modules/video_coding/rtp_frame_reference_finder.h" #include "rtc_base/critical_section.h" #include "rtc_base/numerics/sequence_number_util.h" #include "rtc_base/thread_annotations.h" namespace webrtc { class Clock; namespace video_coding { class RtpFrameObject; // A frame is assembled when all of its packets have been received. class OnAssembledFrameCallback { public: virtual ~OnAssembledFrameCallback() {} virtual void OnAssembledFrame(std::unique_ptr frame) = 0; }; class PacketBuffer { public: static rtc::scoped_refptr Create( Clock* clock, size_t start_buffer_size, size_t max_buffer_size, OnAssembledFrameCallback* frame_callback); virtual ~PacketBuffer(); // Returns true if |packet| is inserted into the packet buffer, false // otherwise. The PacketBuffer will always take ownership of the // |packet.dataPtr| when this function is called. Made virtual for testing. virtual bool InsertPacket(VCMPacket* packet); void ClearTo(uint16_t seq_num); void Clear(); void PaddingReceived(uint16_t seq_num); // Timestamp (not RTP timestamp) of the last received packet/keyframe packet. absl::optional LastReceivedPacketMs() const; absl::optional LastReceivedKeyframePacketMs() const; // Returns number of different frames seen in the packet buffer int GetUniqueFramesSeen() const; int AddRef() const; int Release() const; protected: // Both |start_buffer_size| and |max_buffer_size| must be a power of 2. PacketBuffer(Clock* clock, size_t start_buffer_size, size_t max_buffer_size, OnAssembledFrameCallback* frame_callback); private: friend RtpFrameObject; // Since we want the packet buffer to be as packet type agnostic // as possible we extract only the information needed in order // to determine whether a sequence of packets is continuous or not. struct ContinuityInfo { // The sequence number of the packet. uint16_t seq_num = 0; // If this is the first packet of the frame. bool frame_begin = false; // If this is the last packet of the frame. bool frame_end = false; // If this slot is currently used. bool used = false; // If all its previous packets have been inserted into the packet buffer. bool continuous = false; // If this packet has been used to create a frame already. bool frame_created = false; }; Clock* const clock_; // Tries to expand the buffer. bool ExpandBufferSize() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); // Test if all previous packets has arrived for the given sequence number. bool PotentialNewFrame(uint16_t seq_num) const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); // Test if all packets of a frame has arrived, and if so, creates a frame. // Returns a vector of received frames. std::vector> FindFrames(uint16_t seq_num) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); // Copy the bitstream for |frame| to |destination|. // Virtual for testing. virtual bool GetBitstream(const RtpFrameObject& frame, uint8_t* destination); // Get the packet with sequence number |seq_num|. // Virtual for testing. virtual VCMPacket* GetPacket(uint16_t seq_num) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); // Mark all slots used by |frame| as not used. // Virtual for testing. virtual void ReturnFrame(RtpFrameObject* frame); void UpdateMissingPackets(uint16_t seq_num) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); // Counts unique received timestamps and updates |unique_frames_seen_|. void OnTimestampReceived(uint32_t rtp_timestamp) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_); rtc::CriticalSection crit_; // Buffer size_ and max_size_ must always be a power of two. size_t size_ RTC_GUARDED_BY(crit_); const size_t max_size_; // The fist sequence number currently in the buffer. uint16_t first_seq_num_ RTC_GUARDED_BY(crit_); // If the packet buffer has received its first packet. bool first_packet_received_ RTC_GUARDED_BY(crit_); // If the buffer is cleared to |first_seq_num_|. bool is_cleared_to_first_seq_num_ RTC_GUARDED_BY(crit_); // Buffer that holds the inserted packets. std::vector data_buffer_ RTC_GUARDED_BY(crit_); // Buffer that holds the information about which slot that is currently in use // and information needed to determine the continuity between packets. std::vector sequence_buffer_ RTC_GUARDED_BY(crit_); // Called when all packets in a frame are received, allowing the frame // to be assembled. OnAssembledFrameCallback* const assembled_frame_callback_; // Timestamp (not RTP timestamp) of the last received packet/keyframe packet. absl::optional last_received_packet_ms_ RTC_GUARDED_BY(crit_); absl::optional last_received_keyframe_packet_ms_ RTC_GUARDED_BY(crit_); int unique_frames_seen_ RTC_GUARDED_BY(crit_); absl::optional newest_inserted_seq_num_ RTC_GUARDED_BY(crit_); std::set> missing_packets_ RTC_GUARDED_BY(crit_); // Indicates if we should require SPS, PPS, and IDR for a particular // RTP timestamp to treat the corresponding frame as a keyframe. const bool sps_pps_idr_is_h264_keyframe_; // Stores several last seen unique timestamps for quick search. std::set rtp_timestamps_history_set_ RTC_GUARDED_BY(crit_); // Stores the same unique timestamps in the order of insertion. std::queue rtp_timestamps_history_queue_ RTC_GUARDED_BY(crit_); mutable volatile int ref_count_ = 0; }; } // namespace video_coding } // namespace webrtc #endif // MODULES_VIDEO_CODING_PACKET_BUFFER_H_