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}
63 lines
2.0 KiB
C++
63 lines
2.0 KiB
C++
/*
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* Copyright (c) 2014 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_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
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#define WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
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#include <map>
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#include <vector>
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#include "webrtc/common_types.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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// Helper class for rtcp statistics.
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class ReportBlockStats {
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public:
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typedef std::map<uint32_t, RTCPReportBlock> ReportBlockMap;
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typedef std::vector<RTCPReportBlock> ReportBlockVector;
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ReportBlockStats();
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~ReportBlockStats() {}
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// Updates stats and stores report blocks.
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// Returns an aggregate of the |report_blocks|.
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RTCPReportBlock AggregateAndStore(const ReportBlockVector& report_blocks);
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// Updates stats and stores report block.
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void Store(const RtcpStatistics& rtcp_stats,
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uint32_t remote_ssrc,
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uint32_t source_ssrc);
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// Returns the total fraction of lost packets (or -1 if less than two report
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// blocks have been stored).
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int FractionLostInPercent() const;
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private:
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// Updates the total number of packets/lost packets.
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// Stores the report block.
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// Returns the number of packets/lost packets since previous report block.
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void StoreAndAddPacketIncrement(const RTCPReportBlock& report_block,
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uint32_t* num_sequence_numbers,
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uint32_t* num_lost_sequence_numbers);
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// The total number of packets/lost packets.
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uint32_t num_sequence_numbers_;
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uint32_t num_lost_sequence_numbers_;
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// Map holding the last stored report block (mapped by the source SSRC).
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ReportBlockMap prev_report_blocks_;
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_REPORT_BLOCK_STATS_H_
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