blob: ae0916a75f3bfb4f099f2ea730dbc6d0bd2379a2 [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 <vector>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/packet.h"
#include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
#include "webrtc/modules/video_coding/sequence_number_util.h"
namespace webrtc {
namespace video_coding {
class FrameObject;
class RtpFrameObject;
class OnCompleteFrameCallback {
virtual ~OnCompleteFrameCallback() {}
virtual void OnCompleteFrame(std::unique_ptr<FrameObject> frame) = 0;
class PacketBuffer {
// 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,
OnCompleteFrameCallback* frame_callback);
bool InsertPacket(const VCMPacket& packet);
void ClearTo(uint16_t seq_num);
void Clear();
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;
// Tries to expand the buffer.
bool ExpandBufferSize() EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Test if all previous packets has arrived for the given sequence number.
bool IsContinuous(uint16_t seq_num) const EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Test if all packets of a frame has arrived, and if so, creates a frame.
// May create multiple frames per invocation.
void FindFrames(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Copy the bitstream for |frame| to |destination|.
bool GetBitstream(const RtpFrameObject& frame, uint8_t* destination);
// Get the packet with sequence number |seq_num|.
VCMPacket* GetPacket(uint16_t seq_num);
// Mark all slots used by |frame| as not used.
void ReturnFrame(RtpFrameObject* frame);
rtc::CriticalSection crit_;
// Buffer size_ and max_size_ must always be a power of two.
size_t size_ GUARDED_BY(crit_);
const size_t max_size_;
// The fist sequence number currently in the buffer.
uint16_t first_seq_num_ GUARDED_BY(crit_);
// The last sequence number currently in the buffer.
uint16_t last_seq_num_ GUARDED_BY(crit_);
// If the packet buffer has received its first packet.
bool first_packet_received_ GUARDED_BY(crit_);
// Buffer that holds the inserted packets.
std::vector<VCMPacket> data_buffer_ 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_ GUARDED_BY(crit_);
// Frames that have received all their packets are handed off to the
// |reference_finder_| which finds the dependencies between the frames.
RtpFrameReferenceFinder reference_finder_;
} // namespace video_coding
} // namespace webrtc