blob: 20e0bffae493418af1dd998a40df38839b6c1dcd [file] [log] [blame]
* 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.
#include <memory>
#include <queue>
#include <set>
#include <vector>
#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 {
virtual ~OnAssembledFrameCallback() {}
virtual void OnAssembledFrame(std::unique_ptr<RtpFrameObject> frame) = 0;
class PacketBuffer {
static rtc::scoped_refptr<PacketBuffer> 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<int64_t> LastReceivedPacketMs() const;
absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
// Returns number of different frames seen in the packet buffer
int GetUniqueFramesSeen() const;
int AddRef() const;
int Release() const;
// 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);
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
// Test if all packets of a frame has arrived, and if so, creates a frame.
// Returns a vector of received frames.
std::vector<std::unique_ptr<RtpFrameObject>> FindFrames(uint16_t seq_num)
// 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)
// Mark all slots used by |frame| as not used.
// Virtual for testing.
virtual void ReturnFrame(RtpFrameObject* frame);
void UpdateMissingPackets(uint16_t seq_num)
// Counts unique received timestamps and updates |unique_frames_seen_|.
void OnTimestampReceived(uint32_t rtp_timestamp)
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<VCMPacket> 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<ContinuityInfo> 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<int64_t> last_received_packet_ms_ RTC_GUARDED_BY(crit_);
absl::optional<int64_t> last_received_keyframe_packet_ms_
int unique_frames_seen_ RTC_GUARDED_BY(crit_);
absl::optional<uint16_t> newest_inserted_seq_num_ RTC_GUARDED_BY(crit_);
std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_
// 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<uint32_t> rtp_timestamps_history_set_ RTC_GUARDED_BY(crit_);
// Stores the same unique timestamps in the order of insertion.
std::queue<uint32_t> rtp_timestamps_history_queue_ RTC_GUARDED_BY(crit_);
mutable volatile int ref_count_ = 0;
} // namespace video_coding
} // namespace webrtc