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}
127 lines
3.3 KiB
C++
127 lines
3.3 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_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
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#include <stdio.h>
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#include <queue>
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#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RTPStream {
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public:
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virtual ~RTPStream() {
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}
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virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
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const int16_t seqNo, const uint8_t* payloadData,
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const size_t payloadSize, uint32_t frequency) = 0;
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// Returns the packet's payload size. Zero should be treated as an
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// end-of-stream (in the case that EndOfFile() is true) or an error.
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virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
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size_t payloadSize, uint32_t* offset) = 0;
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virtual bool EndOfFile() const = 0;
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protected:
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void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
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uint32_t timeStamp, uint32_t ssrc);
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void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
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};
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class RTPPacket {
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public:
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RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
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const uint8_t* payloadData, size_t payloadSize,
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uint32_t frequency);
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~RTPPacket();
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uint8_t payloadType;
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uint32_t timeStamp;
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int16_t seqNo;
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uint8_t* payloadData;
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size_t payloadSize;
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uint32_t frequency;
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};
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class RTPBuffer : public RTPStream {
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public:
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RTPBuffer();
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~RTPBuffer();
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void Write(const uint8_t payloadType,
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const uint32_t timeStamp,
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const int16_t seqNo,
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const uint8_t* payloadData,
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const size_t payloadSize,
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uint32_t frequency) override;
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size_t Read(WebRtcRTPHeader* rtpInfo,
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uint8_t* payloadData,
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size_t payloadSize,
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uint32_t* offset) override;
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bool EndOfFile() const override;
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private:
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RWLockWrapper* _queueRWLock;
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std::queue<RTPPacket *> _rtpQueue;
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};
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class RTPFile : public RTPStream {
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public:
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~RTPFile() {
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}
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RTPFile()
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: _rtpFile(NULL),
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_rtpEOF(false) {
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}
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void Open(const char *outFilename, const char *mode);
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void Close();
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void WriteHeader();
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void ReadHeader();
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void Write(const uint8_t payloadType,
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const uint32_t timeStamp,
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const int16_t seqNo,
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const uint8_t* payloadData,
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const size_t payloadSize,
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uint32_t frequency) override;
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size_t Read(WebRtcRTPHeader* rtpInfo,
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uint8_t* payloadData,
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size_t payloadSize,
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uint32_t* offset) override;
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bool EndOfFile() const override { return _rtpEOF; }
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private:
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FILE* _rtpFile;
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bool _rtpEOF;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
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