ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
103 lines
3.7 KiB
C++
103 lines
3.7 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
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#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
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#include <map>
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#include "webrtc/modules/include/module.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class Clock;
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class StreamStatistician {
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public:
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virtual ~StreamStatistician();
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virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0;
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virtual void GetDataCounters(size_t* bytes_received,
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uint32_t* packets_received) const = 0;
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// Gets received stream data counters (includes reset counter values).
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virtual void GetReceiveStreamDataCounters(
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StreamDataCounters* data_counters) const = 0;
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virtual uint32_t BitrateReceived() const = 0;
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// Returns true if the packet with RTP header |header| is likely to be a
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// retransmitted packet, false otherwise.
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virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
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int64_t min_rtt) const = 0;
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// Returns true if |sequence_number| is received in order, false otherwise.
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virtual bool IsPacketInOrder(uint16_t sequence_number) const = 0;
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};
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typedef std::map<uint32_t, StreamStatistician*> StatisticianMap;
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class ReceiveStatistics : public Module {
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public:
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virtual ~ReceiveStatistics() {}
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static ReceiveStatistics* Create(Clock* clock);
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// Updates the receive statistics with this packet.
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virtual void IncomingPacket(const RTPHeader& rtp_header,
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size_t packet_length,
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bool retransmitted) = 0;
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// Increment counter for number of FEC packets received.
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virtual void FecPacketReceived(const RTPHeader& header,
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size_t packet_length) = 0;
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// Returns a map of all statisticians which have seen an incoming packet
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// during the last two seconds.
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virtual StatisticianMap GetActiveStatisticians() const = 0;
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// Returns a pointer to the statistician of an ssrc.
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virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
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// Sets the max reordering threshold in number of packets.
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virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
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// Called on new RTCP stats creation.
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virtual void RegisterRtcpStatisticsCallback(
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RtcpStatisticsCallback* callback) = 0;
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// Called on new RTP stats creation.
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virtual void RegisterRtpStatisticsCallback(
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StreamDataCountersCallback* callback) = 0;
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};
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class NullReceiveStatistics : public ReceiveStatistics {
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public:
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void IncomingPacket(const RTPHeader& rtp_header,
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size_t packet_length,
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bool retransmitted) override;
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void FecPacketReceived(const RTPHeader& header,
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size_t packet_length) override;
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StatisticianMap GetActiveStatisticians() const override;
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StreamStatistician* GetStatistician(uint32_t ssrc) const override;
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int64_t TimeUntilNextProcess() override;
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int32_t Process() override;
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void SetMaxReorderingThreshold(int max_reordering_threshold) override;
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void RegisterRtcpStatisticsCallback(
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RtcpStatisticsCallback* callback) override;
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void RegisterRtpStatisticsCallback(
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StreamDataCountersCallback* callback) override;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
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