git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklase@google.com
297
modules/audio_coding/main/test/RTPFile.cpp
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297
modules/audio_coding/main/test/RTPFile.cpp
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/*
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* Copyright (c) 2011 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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#include "RTPFile.h"
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#include "rw_lock_wrapper.h"
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#include "engine_configurations.h"
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#include <stdlib.h>
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#ifdef WIN32
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# include <Winsock2.h>
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#else
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# include <arpa/inet.h>
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#endif
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#include "audio_coding_module.h"
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void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader)
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{
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rtpInfo->header.payloadType = rtpHeader[1];
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rtpInfo->header.sequenceNumber = (static_cast<WebRtc_UWord16>(rtpHeader[2])<<8) | rtpHeader[3];
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rtpInfo->header.timestamp = (static_cast<WebRtc_UWord32>(rtpHeader[4])<<24) |
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(static_cast<WebRtc_UWord32>(rtpHeader[5])<<16) |
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(static_cast<WebRtc_UWord32>(rtpHeader[6])<<8) |
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rtpHeader[7];
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rtpInfo->header.ssrc = (static_cast<WebRtc_UWord32>(rtpHeader[8])<<24) |
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(static_cast<WebRtc_UWord32>(rtpHeader[9])<<16) |
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(static_cast<WebRtc_UWord32>(rtpHeader[10])<<8) |
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rtpHeader[11];
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}
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void RTPStream::MakeRTPheader(WebRtc_UWord8* rtpHeader,
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WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
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WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc)
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{
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rtpHeader[0]=(unsigned char)0x80;
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rtpHeader[1]=(unsigned char)(payloadType & 0xFF);
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rtpHeader[2]=(unsigned char)((seqNo>>8)&0xFF);
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rtpHeader[3]=(unsigned char)((seqNo)&0xFF);
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rtpHeader[4]=(unsigned char)((timeStamp>>24)&0xFF);
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rtpHeader[5]=(unsigned char)((timeStamp>>16)&0xFF);
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rtpHeader[6]=(unsigned char)((timeStamp>>8)&0xFF);
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rtpHeader[7]=(unsigned char)(timeStamp & 0xFF);
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rtpHeader[8]=(unsigned char)((ssrc>>24)&0xFF);
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rtpHeader[9]=(unsigned char)((ssrc>>16)&0xFF);
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rtpHeader[10]=(unsigned char)((ssrc>>8)&0xFF);
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rtpHeader[11]=(unsigned char)(ssrc & 0xFF);
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}
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RTPPacket::RTPPacket(WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
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WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
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WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
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:
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payloadType(payloadType),
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timeStamp(timeStamp),
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seqNo(seqNo),
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payloadSize(payloadSize),
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frequency(frequency)
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{
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if (payloadSize > 0)
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{
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this->payloadData = new WebRtc_UWord8[payloadSize];
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memcpy(this->payloadData, payloadData, payloadSize);
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}
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}
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RTPPacket::~RTPPacket()
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{
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delete [] payloadData;
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}
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RTPBuffer::RTPBuffer()
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{
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_queueRWLock = RWLockWrapper::CreateRWLock();
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}
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RTPBuffer::~RTPBuffer()
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{
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delete _queueRWLock;
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}
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void
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RTPBuffer::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
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const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
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{
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RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, payloadSize, frequency);
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_queueRWLock->AcquireLockExclusive();
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_rtpQueue.push(packet);
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_queueRWLock->ReleaseLockExclusive();
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}
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WebRtc_UWord16
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RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
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WebRtc_Word8* payloadData,
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WebRtc_UWord16 payloadSize,
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WebRtc_UWord32* offset)
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{
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_queueRWLock->AcquireLockShared();
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RTPPacket *packet = _rtpQueue.front();
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_rtpQueue.pop();
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_queueRWLock->ReleaseLockShared();
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rtpInfo->header.markerBit = 1;
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rtpInfo->header.payloadType = packet->payloadType;
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rtpInfo->header.sequenceNumber = packet->seqNo;
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rtpInfo->header.ssrc = 0;
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rtpInfo->header.timestamp = packet->timeStamp;
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if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize)
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{
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memcpy(payloadData, packet->payloadData, packet->payloadSize);
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}
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else
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{
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throw "Payload buffer too small";
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exit(1);
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}
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/*#ifdef WEBRTC_CODEC_G722
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if(ACMCodecDB::_mycodecs[ACMCodecDB::g722].pltype == packet->payloadType)
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{
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*offset = (packet->timeStamp/(packet->frequency/1000))<<1;
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}
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else
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{
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#endif*/
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*offset = (packet->timeStamp/(packet->frequency/1000));
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/*#ifdef WEBRTC_CODEC_G722
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}
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#endif*/
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return packet->payloadSize;
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}
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bool
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RTPBuffer::EndOfFile() const
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{
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_queueRWLock->AcquireLockShared();
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bool eof = _rtpQueue.empty();
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_queueRWLock->ReleaseLockShared();
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return eof;
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}
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void RTPFile::Open(char *filename, const char *mode)
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{
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if ((_rtpFile = fopen(filename, mode)) == NULL)
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{
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printf("Cannot write file %s.\n", filename);
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throw "Unable to write file";
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exit(1);
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}
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}
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void RTPFile::Close()
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{
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if (_rtpFile != NULL)
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{
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fclose(_rtpFile);
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_rtpFile = NULL;
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}
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}
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void RTPFile::WriteHeader()
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{
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// Write data in a format that NetEQ and RTP Play can parse
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fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
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WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
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fwrite(&dummy_variable, 4, 1, _rtpFile);
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fwrite(&dummy_variable, 4, 1, _rtpFile);
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fwrite(&dummy_variable, 4, 1, _rtpFile);
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fwrite(&dummy_variable, 2, 1, _rtpFile);
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fwrite(&dummy_variable, 2, 1, _rtpFile);
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fflush(_rtpFile);
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}
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void RTPFile::ReadHeader()
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{
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WebRtc_UWord32 start_sec, start_usec, source;
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WebRtc_UWord16 port, padding;
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char fileHeader[40];
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fgets(fileHeader, 40, _rtpFile);
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fread(&start_sec, 4, 1, _rtpFile);
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start_sec=ntohl(start_sec);
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fread(&start_usec, 4, 1, _rtpFile);
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start_usec=ntohl(start_usec);
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fread(&source, 4, 1, _rtpFile);
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source=ntohl(source);
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fread(&port, 2, 1, _rtpFile);
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port=ntohs(port);
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fread(&padding, 2, 1, _rtpFile);
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padding=ntohs(padding);
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}
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void RTPFile::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
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const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
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{
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/* write RTP packet to file */
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WebRtc_UWord8 rtpHeader[12];
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MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
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WebRtc_UWord16 lengthBytes = htons(12 + payloadSize + 8);
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WebRtc_UWord16 plen = htons(12 + payloadSize);
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WebRtc_UWord32 offsetMs;
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/*#ifdef WEBRTC_CODEC_G722
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if(ACMCodecDB::_mycodecs[ACMCodecDB::g722].pltype == payloadType)
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{
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offsetMs = (timeStamp/(frequency/1000))<<1;
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}
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else
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{
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#endif*/
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offsetMs = (timeStamp/(frequency/1000));
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/*#ifdef WEBRTC_CODEC_G722
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}
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#endif*/
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offsetMs = htonl(offsetMs);
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fwrite(&lengthBytes, 2, 1, _rtpFile);
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fwrite(&plen, 2, 1, _rtpFile);
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fwrite(&offsetMs, 4, 1, _rtpFile);
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fwrite(rtpHeader, 12, 1, _rtpFile);
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fwrite(payloadData, 1, payloadSize, _rtpFile);
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}
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WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
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WebRtc_Word8* payloadData,
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WebRtc_UWord16 payloadSize,
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WebRtc_UWord32* offset)
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{
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WebRtc_UWord16 lengthBytes;
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WebRtc_UWord16 plen;
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WebRtc_UWord8 rtpHeader[12];
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fread(&lengthBytes, 2, 1, _rtpFile);
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if (feof(_rtpFile))
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{
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_rtpEOF = true;
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return 0;
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}
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fread(&plen, 2, 1, _rtpFile);
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if (feof(_rtpFile))
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{
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_rtpEOF = true;
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return 0;
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}
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fread(offset, 4, 1, _rtpFile);
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if (feof(_rtpFile))
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{
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_rtpEOF = true;
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return 0;
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}
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lengthBytes = ntohs(lengthBytes);
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plen = ntohs(plen);
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*offset = ntohl(*offset);
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if (plen < 12)
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{
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throw "Unable to read RTP file";
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exit(1);
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}
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fread(rtpHeader, 12, 1, _rtpFile);
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if (feof(_rtpFile))
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{
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_rtpEOF = true;
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return 0;
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}
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ParseRTPHeader(rtpInfo, rtpHeader);
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rtpInfo->type.Audio.isCNG = false;
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rtpInfo->type.Audio.channel = 1;
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if (lengthBytes != plen + 8)
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{
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throw "Length parameters in RTP file doesn't match";
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exit(1);
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}
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if (plen == 0)
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{
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return 0;
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}
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else if (lengthBytes - 20 > payloadSize)
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{
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throw "Payload buffer too small";
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exit(1);
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}
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lengthBytes -= 20;
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fread(payloadData, 1, lengthBytes, _rtpFile);
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if (feof(_rtpFile))
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{
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_rtpEOF = true;
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}
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return lengthBytes;
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}
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