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}
47 lines
1.5 KiB
C++
47 lines
1.5 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_FEC_RECEIVER_H_
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#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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struct FecPacketCounter {
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FecPacketCounter()
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: num_packets(0),
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num_fec_packets(0),
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num_recovered_packets(0) {}
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size_t num_packets; // Number of received packets.
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size_t num_fec_packets; // Number of received FEC packets.
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size_t num_recovered_packets; // Number of recovered media packets using FEC.
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};
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class FecReceiver {
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public:
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static FecReceiver* Create(RtpData* callback);
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virtual ~FecReceiver() {}
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virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
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const uint8_t* incoming_rtp_packet,
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size_t packet_length,
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uint8_t ulpfec_payload_type) = 0;
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virtual int32_t ProcessReceivedFec() = 0;
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virtual FecPacketCounter GetPacketCounter() const = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
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