| Index: webrtc/modules/rtp_rtcp/source/packet_loss_stats.h
|
| diff --git a/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h b/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..2eab043c0d01a4170f85aef0192ba4f7a6b0af41
|
| --- /dev/null
|
| +++ b/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h
|
| @@ -0,0 +1,57 @@
|
| +/*
|
| + * Copyright (c) 2015 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_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
|
| +#define WEBRTC_MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
|
| +
|
| +#include <stdint.h>
|
| +#include <set>
|
| +
|
| +namespace webrtc {
|
| +
|
| +// Keeps track of statistics of packet loss including whether losses are a
|
| +// single packet or multiple packets in a row.
|
| +class PacketLossStats {
|
| + public:
|
| + PacketLossStats();
|
| + ~PacketLossStats() {}
|
| +
|
| + // Adds a lost packet to the stats by sequence number.
|
| + void AddLostPacket(uint16_t sequence_number);
|
| +
|
| + // Queries the number of packets that were lost by themselves, no neighboring
|
| + // packets were lost.
|
| + int GetSingleLossCount() const;
|
| +
|
| + // Queries the number of times that multiple packets with sequential numbers
|
| + // were lost. This is the number of events with more than one packet lost,
|
| + // regardless of the size of the event;
|
| + int GetMultipleLossEventCount() const;
|
| +
|
| + // Queries the number of packets lost in multiple packet loss events. Combined
|
| + // with the event count, this can be used to determine the average event size.
|
| + int GetMultipleLossPacketCount() const;
|
| +
|
| + private:
|
| + std::set<uint16_t> lost_packets_buffer_;
|
| + std::set<uint16_t> lost_packets_wrapped_buffer_;
|
| + int single_loss_historic_count_;
|
| + int multiple_loss_historic_event_count_;
|
| + int multiple_loss_historic_packet_count_;
|
| +
|
| + void ComputeLossCounts(int* out_single_loss_count,
|
| + int* out_multiple_loss_event_count,
|
| + int* out_multiple_loss_packet_count) const;
|
| + void PruneBuffer();
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
|
|
|