mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
bbf21a3fd6
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
125 lines
3.2 KiB
C++
125 lines
3.2 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_CODING_TEST_RTPFILE_H_
|
|
#define MODULES_AUDIO_CODING_TEST_RTPFILE_H_
|
|
|
|
#include <stdio.h>
|
|
#include <queue>
|
|
|
|
#include "modules/audio_coding/include/audio_coding_module.h"
|
|
#include "rtc_base/synchronization/rw_lock_wrapper.h"
|
|
#include "typedefs.h" // NOLINT(build/include)
|
|
|
|
namespace webrtc {
|
|
|
|
class RTPStream {
|
|
public:
|
|
virtual ~RTPStream() {
|
|
}
|
|
|
|
virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
|
|
const int16_t seqNo, const uint8_t* payloadData,
|
|
const size_t payloadSize, uint32_t frequency) = 0;
|
|
|
|
// Returns the packet's payload size. Zero should be treated as an
|
|
// end-of-stream (in the case that EndOfFile() is true) or an error.
|
|
virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
|
|
size_t payloadSize, uint32_t* offset) = 0;
|
|
virtual bool EndOfFile() const = 0;
|
|
|
|
protected:
|
|
void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
|
|
uint32_t timeStamp, uint32_t ssrc);
|
|
|
|
void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
|
|
};
|
|
|
|
class RTPPacket {
|
|
public:
|
|
RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
|
|
const uint8_t* payloadData, size_t payloadSize,
|
|
uint32_t frequency);
|
|
|
|
~RTPPacket();
|
|
|
|
uint8_t payloadType;
|
|
uint32_t timeStamp;
|
|
int16_t seqNo;
|
|
uint8_t* payloadData;
|
|
size_t payloadSize;
|
|
uint32_t frequency;
|
|
};
|
|
|
|
class RTPBuffer : public RTPStream {
|
|
public:
|
|
RTPBuffer();
|
|
|
|
~RTPBuffer();
|
|
|
|
void Write(const uint8_t payloadType,
|
|
const uint32_t timeStamp,
|
|
const int16_t seqNo,
|
|
const uint8_t* payloadData,
|
|
const size_t payloadSize,
|
|
uint32_t frequency) override;
|
|
|
|
size_t Read(WebRtcRTPHeader* rtpInfo,
|
|
uint8_t* payloadData,
|
|
size_t payloadSize,
|
|
uint32_t* offset) override;
|
|
|
|
bool EndOfFile() const override;
|
|
|
|
private:
|
|
RWLockWrapper* _queueRWLock;
|
|
std::queue<RTPPacket *> _rtpQueue;
|
|
};
|
|
|
|
class RTPFile : public RTPStream {
|
|
public:
|
|
~RTPFile() {
|
|
}
|
|
|
|
RTPFile()
|
|
: _rtpFile(NULL),
|
|
_rtpEOF(false) {
|
|
}
|
|
|
|
void Open(const char *outFilename, const char *mode);
|
|
|
|
void Close();
|
|
|
|
void WriteHeader();
|
|
|
|
void ReadHeader();
|
|
|
|
void Write(const uint8_t payloadType,
|
|
const uint32_t timeStamp,
|
|
const int16_t seqNo,
|
|
const uint8_t* payloadData,
|
|
const size_t payloadSize,
|
|
uint32_t frequency) override;
|
|
|
|
size_t Read(WebRtcRTPHeader* rtpInfo,
|
|
uint8_t* payloadData,
|
|
size_t payloadSize,
|
|
uint32_t* offset) override;
|
|
|
|
bool EndOfFile() const override { return _rtpEOF; }
|
|
|
|
private:
|
|
FILE* _rtpFile;
|
|
bool _rtpEOF;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_CODING_TEST_RTPFILE_H_
|