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}
60 lines
1.8 KiB
C++
60 lines
1.8 KiB
C++
/*
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* Copyright (c) 2012 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_SOURCE_FEC_TEST_HELPER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
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namespace webrtc {
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const uint8_t kFecPayloadType = 96;
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const uint8_t kRedPayloadType = 97;
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const uint8_t kVp8PayloadType = 120;
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typedef ForwardErrorCorrection::Packet Packet;
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struct RtpPacket : public Packet {
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WebRtcRTPHeader header;
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};
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class FrameGenerator {
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public:
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FrameGenerator();
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void NewFrame(int num_packets);
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uint16_t NextSeqNum();
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RtpPacket* NextPacket(int offset, size_t length);
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// Creates a new RtpPacket with the RED header added to the packet.
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RtpPacket* BuildMediaRedPacket(const RtpPacket* packet);
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// Creates a new RtpPacket with FEC payload and red header. Does this by
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// creating a new fake media RtpPacket, clears the marker bit and adds a RED
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// header. Finally replaces the payload with the content of |packet->data|.
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RtpPacket* BuildFecRedPacket(const Packet* packet);
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void SetRedHeader(Packet* red_packet, uint8_t payload_type,
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size_t header_length) const;
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private:
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static void BuildRtpHeader(uint8_t* data, const RTPHeader* header);
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int num_packets_;
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uint16_t seq_num_;
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uint32_t timestamp_;
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};
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}
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
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