webrtc_m130/webrtc/modules/rtp_rtcp/include/receive_statistics.h
sprang cd349d9743 Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
Reason for revert:
Upstream fixes in place, should be OK now.

Original issue's description:
> Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
>
> Reason for revert:
> Breaks upstream code.
>
> Original issue's description:
> > Refactor NACK bitrate allocation
> >
> > Nack bitrate allocation should not be done on a per-rtp-module basis,
> > but rather shared bitrate pool per call. This CL moves allocation to the
> > pacer and cleans up a bunch if bitrate stats handling.
> >
> > BUG=
> > R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
> >
> > Committed: 5fc59e810b
>
> TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/e5dd44101eca485f5ad12e5f7ce6f6b0d204116b
> Cr-Commit-Position: refs/heads/master@{#13417}

TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=

Review-Url: https://codereview.webrtc.org/2146013002
Cr-Commit-Position: refs/heads/master@{#13465}
2016-07-13 16:11:38 +00:00

101 lines
3.6 KiB
C++

/*
* Copyright (c) 2013 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_INCLUDE_RECEIVE_STATISTICS_H_
#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
#include <map>
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class Clock;
class StreamStatistician {
public:
virtual ~StreamStatistician();
virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0;
virtual void GetDataCounters(size_t* bytes_received,
uint32_t* packets_received) const = 0;
// Gets received stream data counters (includes reset counter values).
virtual void GetReceiveStreamDataCounters(
StreamDataCounters* data_counters) const = 0;
virtual uint32_t BitrateReceived() const = 0;
// Returns true if the packet with RTP header |header| is likely to be a
// retransmitted packet, false otherwise.
virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
int64_t min_rtt) const = 0;
// Returns true if |sequence_number| is received in order, false otherwise.
virtual bool IsPacketInOrder(uint16_t sequence_number) const = 0;
};
typedef std::map<uint32_t, StreamStatistician*> StatisticianMap;
class ReceiveStatistics {
public:
virtual ~ReceiveStatistics() {}
static ReceiveStatistics* Create(Clock* clock);
// Updates the receive statistics with this packet.
virtual void IncomingPacket(const RTPHeader& rtp_header,
size_t packet_length,
bool retransmitted) = 0;
// Increment counter for number of FEC packets received.
virtual void FecPacketReceived(const RTPHeader& header,
size_t packet_length) = 0;
// Returns a map of all statisticians which have seen an incoming packet
// during the last two seconds.
virtual StatisticianMap GetActiveStatisticians() const = 0;
// Returns a pointer to the statistician of an ssrc.
virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
// Sets the max reordering threshold in number of packets.
virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
// Called on new RTCP stats creation.
virtual void RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) = 0;
// Called on new RTP stats creation.
virtual void RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) = 0;
};
class NullReceiveStatistics : public ReceiveStatistics {
public:
void IncomingPacket(const RTPHeader& rtp_header,
size_t packet_length,
bool retransmitted) override;
void FecPacketReceived(const RTPHeader& header,
size_t packet_length) override;
StatisticianMap GetActiveStatisticians() const override;
StreamStatistician* GetStatistician(uint32_t ssrc) const override;
void SetMaxReorderingThreshold(int max_reordering_threshold) override;
void RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) override;
void RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) override;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_