webrtc/modules/media_file/source/media_file_utility.cc

2672 lines
77 KiB
C++

/*
* Copyright (c) 2011 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.
*/
#include <assert.h>
#include <sys/stat.h>
#include <sys/types.h>
#include "common_types.h"
#include "engine_configurations.h"
#include "file_wrapper.h"
#include "media_file_utility.h"
#include "module_common_types.h"
#include "trace.h"
#ifdef WEBRTC_MODULE_UTILITY_VIDEO
#include "avi_file.h"
#endif
#if (defined(WIN32) || defined(WINCE))
#define STR_CASE_CMP _stricmp
#define STR_NCASE_CMP _strnicmp
#else
#define STR_CASE_CMP strcasecmp
#define STR_NCASE_CMP strncasecmp
#endif
namespace {
enum WaveFormats
{
kWaveFormatPcm = 0x0001,
kWaveFormatALaw = 0x0006,
kWaveFormatMuLaw = 0x0007
};
// First 16 bytes the WAVE header. ckID should be "RIFF", wave_ckID should be
// "WAVE" and ckSize is the chunk size (4 + n)
struct WAVE_RIFF_header
{
WebRtc_Word8 ckID[4];
WebRtc_Word32 ckSize;
WebRtc_Word8 wave_ckID[4];
};
// First 8 byte of the format chunk. fmt_ckID should be "fmt ". fmt_ckSize is
// the chunk size (16, 18 or 40 byte)
struct WAVE_CHUNK_header
{
WebRtc_Word8 fmt_ckID[4];
WebRtc_Word32 fmt_ckSize;
};
} // unnamed namespace
namespace webrtc {
ModuleFileUtility::ModuleFileUtility(const WebRtc_Word32 id)
: _wavFormatObj(),
_dataSize(0),
_readSizeBytes(0),
_id(id),
_stopPointInMs(0),
_startPointInMs(0),
_playoutPositionMs(0),
_bytesWritten(0),
codec_info_(),
_codecId(kCodecNoCodec),
_bytesPerSample(0),
_readPos(0),
_reading(false),
_writing(false),
_tempData()
#ifdef WEBRTC_MODULE_UTILITY_VIDEO
,
_aviAudioInFile(0),
_aviVideoInFile(0),
_aviOutFile(0)
#endif
{
WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
"ModuleFileUtility::ModuleFileUtility()");
memset(&codec_info_,0,sizeof(CodecInst));
codec_info_.pltype = -1;
#ifdef WEBRTC_MODULE_UTILITY_VIDEO
memset(&_videoCodec,0,sizeof(_videoCodec));
#endif
}
ModuleFileUtility::~ModuleFileUtility()
{
WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
"ModuleFileUtility::~ModuleFileUtility()");
#ifdef WEBRTC_MODULE_UTILITY_VIDEO
delete _aviAudioInFile;
delete _aviVideoInFile;
#endif
}
#ifdef WEBRTC_MODULE_UTILITY_VIDEO
WebRtc_Word32 ModuleFileUtility::InitAviWriting(
const WebRtc_Word8* filename,
const CodecInst& audioCodecInst,
const VideoCodec& videoCodecInst,
const bool videoOnly /*= false*/)
{
_writing = false;
delete _aviOutFile;
_aviOutFile = new AviFile( );
AVISTREAMHEADER videoStreamHeader;
videoStreamHeader.fccType = AviFile::MakeFourCc('v', 'i', 'd', 's');
#ifdef VIDEOCODEC_H263
if (strncmp(videoCodecInst.plName, "H263", 7) == 0)
{
videoStreamHeader.fccHandler = AviFile::MakeFourCc('H','2','6','3');
}
#endif
#ifdef VIDEOCODEC_MPEG4
if (strncmp(videoCodecInst.plName, "MP4V-ES", 7) == 0)
{
videoStreamHeader.fccHandler = AviFile::MakeFourCc('M','4','S','2');
}
#endif
#ifdef VIDEOCODEC_I420
if (strncmp(videoCodecInst.plName, "I420", 7) == 0)
{
videoStreamHeader.fccHandler = AviFile::MakeFourCc('I','4','2','0');
}
#endif
#ifdef VIDEOCODEC_VP8
if (strncmp(videoCodecInst.plName, "VP8", 7) == 0)
{
videoStreamHeader.fccHandler = AviFile::MakeFourCc('V','P','8','0');
}
#endif
if (videoStreamHeader.fccHandler == 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"InitAviWriting() Codec not supported");
return -1;
}
videoStreamHeader.dwScale = 1;
videoStreamHeader.dwRate = videoCodecInst.maxFramerate;
videoStreamHeader.dwSuggestedBufferSize = videoCodecInst.height *
(videoCodecInst.width >> 1) * 3;
videoStreamHeader.dwQuality = (WebRtc_UWord32)-1;
videoStreamHeader.dwSampleSize = 0;
videoStreamHeader.rcFrame.top = 0;
videoStreamHeader.rcFrame.bottom = videoCodecInst.height;
videoStreamHeader.rcFrame.left = 0;
videoStreamHeader.rcFrame.right = videoCodecInst.width;
BITMAPINFOHEADER bitMapInfoHeader;
bitMapInfoHeader.biSize = sizeof(BITMAPINFOHEADER);
bitMapInfoHeader.biHeight = videoCodecInst.height;
bitMapInfoHeader.biWidth = videoCodecInst.width;
bitMapInfoHeader.biPlanes = 1;
bitMapInfoHeader.biBitCount = 12;
bitMapInfoHeader.biClrImportant = 0;
bitMapInfoHeader.biClrUsed = 0;
bitMapInfoHeader.biCompression = videoStreamHeader.fccHandler;
bitMapInfoHeader.biSizeImage = bitMapInfoHeader.biWidth *
bitMapInfoHeader.biHeight * bitMapInfoHeader.biBitCount / 8;
if(videoCodecInst.codecType == kVideoCodecMPEG4)
{
if(_aviOutFile->CreateVideoStream(
videoStreamHeader,
bitMapInfoHeader,
videoCodecInst.codecSpecific.MPEG4.configParameters,
videoCodecInst.codecSpecific.MPEG4.configParametersSize) != 0)
{
return -1;
}
} else
{
if(_aviOutFile->CreateVideoStream(
videoStreamHeader,
bitMapInfoHeader,
NULL,
0) != 0)
{
return -1;
}
}
if(!videoOnly)
{
AVISTREAMHEADER audioStreamHeader;
audioStreamHeader.fccType = AviFile::MakeFourCc('a', 'u', 'd', 's');
// fccHandler is the FOURCC of the codec for decoding the stream.
// It's an optional parameter that is not used by audio streams.
audioStreamHeader.fccHandler = 0;
audioStreamHeader.dwScale = 1;
WAVEFORMATEX waveFormatHeader;
waveFormatHeader.cbSize = 0;
waveFormatHeader.nChannels = 1;
if (strncmp(audioCodecInst.plname, "PCMU", 4) == 0)
{
audioStreamHeader.dwSampleSize = 1;
audioStreamHeader.dwRate = 8000;
audioStreamHeader.dwQuality = (WebRtc_UWord32)-1;
audioStreamHeader.dwSuggestedBufferSize = 80;
waveFormatHeader.nAvgBytesPerSec = 8000;
waveFormatHeader.nSamplesPerSec = 8000;
waveFormatHeader.wBitsPerSample = 8;
waveFormatHeader.nBlockAlign = 1;
waveFormatHeader.wFormatTag = kWaveFormatMuLaw;
} else if (strncmp(audioCodecInst.plname, "PCMA", 4) == 0)
{
audioStreamHeader.dwSampleSize = 1;
audioStreamHeader.dwRate = 8000;
audioStreamHeader.dwQuality = (WebRtc_UWord32)-1;
audioStreamHeader.dwSuggestedBufferSize = 80;
waveFormatHeader.nAvgBytesPerSec = 8000;
waveFormatHeader.nSamplesPerSec = 8000;
waveFormatHeader.wBitsPerSample = 8;
waveFormatHeader.nBlockAlign = 1;
waveFormatHeader.wFormatTag = kWaveFormatALaw;
} else if (strncmp(audioCodecInst.plname, "L16", 3) == 0)
{
audioStreamHeader.dwSampleSize = 2;
audioStreamHeader.dwRate = audioCodecInst.plfreq;
audioStreamHeader.dwQuality = (WebRtc_UWord32)-1;
audioStreamHeader.dwSuggestedBufferSize =
(audioCodecInst.plfreq/100) * 2;
waveFormatHeader.nAvgBytesPerSec = audioCodecInst.plfreq * 2;
waveFormatHeader.nSamplesPerSec = audioCodecInst.plfreq;
waveFormatHeader.wBitsPerSample = 16;
waveFormatHeader.nBlockAlign = 2;
waveFormatHeader.wFormatTag = kWaveFormatPcm;
} else
{
return -1;
}
if(_aviOutFile->CreateAudioStream(
audioStreamHeader,
waveFormatHeader) != 0)
{
return -1;
}
if( InitWavCodec(waveFormatHeader.nSamplesPerSec,
waveFormatHeader.nChannels,
waveFormatHeader.wBitsPerSample,
waveFormatHeader.wFormatTag) != 0)
{
return -1;
}
}
_aviOutFile->Create(filename);
_writing = true;
return 0;
}
WebRtc_Word32 ModuleFileUtility::WriteAviAudioData(
const WebRtc_Word8* buffer,
WebRtc_UWord32 bufferLengthInBytes)
{
if( _aviOutFile != 0)
{
return _aviOutFile->WriteAudio(
reinterpret_cast<const WebRtc_UWord8*>(buffer),
bufferLengthInBytes);
}
else
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id, "AVI file not initialized");
return -1;
}
}
WebRtc_Word32 ModuleFileUtility::WriteAviVideoData(
const WebRtc_Word8* buffer,
WebRtc_UWord32 bufferLengthInBytes)
{
if( _aviOutFile != 0)
{
return _aviOutFile->WriteVideo(
reinterpret_cast<const WebRtc_UWord8*>(buffer),
bufferLengthInBytes);
}
else
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id, "AVI file not initialized");
return -1;
}
}
WebRtc_Word32 ModuleFileUtility::CloseAviFile( )
{
if( _reading && _aviAudioInFile)
{
delete _aviAudioInFile;
_aviAudioInFile = 0;
}
if( _reading && _aviVideoInFile)
{
delete _aviVideoInFile;
_aviVideoInFile = 0;
}
if( _writing && _aviOutFile)
{
delete _aviOutFile;
_aviOutFile = 0;
}
return 0;
}
WebRtc_Word32 ModuleFileUtility::InitAviReading(const WebRtc_Word8* filename,
bool videoOnly, bool loop)
{
_reading = false;
delete _aviVideoInFile;
_aviVideoInFile = new AviFile( );
if ((_aviVideoInFile != 0) && _aviVideoInFile->Open(AviFile::AVI_VIDEO,
filename, loop) == -1)
{
WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
"Unable to open AVI file (video)");
return -1;
}
AVISTREAMHEADER videoInStreamHeader;
BITMAPINFOHEADER bitmapInfo;
char codecConfigParameters[AviFile::CODEC_CONFIG_LENGTH] = {};
WebRtc_Word32 configLength = 0;
if( _aviVideoInFile->GetVideoStreamInfo(videoInStreamHeader, bitmapInfo,
codecConfigParameters,
configLength) != 0)
{
return -1;
}
_videoCodec.width = static_cast<WebRtc_UWord16>(
videoInStreamHeader.rcFrame.right);
_videoCodec.height = static_cast<WebRtc_UWord16>(
videoInStreamHeader.rcFrame.bottom);
_videoCodec.maxFramerate = static_cast<WebRtc_UWord8>(
videoInStreamHeader.dwRate);
const size_t plnameLen = sizeof(_videoCodec.plName) / sizeof(char);
if (bitmapInfo.biCompression == AviFile::MakeFourCc('M','4','S','2'))
{
strncpy(_videoCodec.plName, "MP4V-ES", plnameLen);
if (configLength > 0)
{
if (configLength < kConfigParameterSize)
{
_videoCodec.codecSpecific.MPEG4.configParametersSize =
(WebRtc_UWord8)configLength;
memcpy(_videoCodec.codecSpecific.MPEG4.configParameters,
&codecConfigParameters,
_videoCodec.codecSpecific.MPEG4.configParametersSize);
}
else
{
return -1;
}
}
}
else if (bitmapInfo.biCompression == AviFile::MakeFourCc('I','4','2','0'))
{
strncpy(_videoCodec.plName, "I420", plnameLen);
_videoCodec.codecType = kVideoCodecI420;
}
else if (bitmapInfo.biCompression == AviFile::MakeFourCc('H','2','6','3'))
{
strncpy(_videoCodec.plName, "H263", plnameLen);
_videoCodec.codecType = kVideoCodecH263;
}
else if (bitmapInfo.biCompression ==
AviFile::MakeFourCc('V', 'P', '8', '0'))
{
strncpy(_videoCodec.plName, "VP8", plnameLen);
_videoCodec.codecType = kVideoCodecVP8;
}
else
{
return -1;
}
if(!videoOnly)
{
delete _aviAudioInFile;
_aviAudioInFile = new AviFile();
if ( (_aviAudioInFile != 0) &&
_aviAudioInFile->Open(AviFile::AVI_AUDIO, filename, loop) == -1)
{
WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
"Unable to open AVI file (audio)");
return -1;
}
WAVEFORMATEX waveHeader;
if(_aviAudioInFile->GetAudioStreamInfo(waveHeader) != 0)
{
return -1;
}
if(InitWavCodec(waveHeader.nSamplesPerSec, waveHeader.nChannels,
waveHeader.wBitsPerSample, waveHeader.wFormatTag) != 0)
{
return -1;
}
}
_reading = true;
return 0;
}
WebRtc_Word32 ModuleFileUtility::ReadAviAudioData(
WebRtc_Word8* outBuffer,
const WebRtc_UWord32 bufferLengthInBytes)
{
if(_aviAudioInFile == 0)
{
WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "AVI file not opened.");
return -1;
}
WebRtc_Word32 length = bufferLengthInBytes;
if(_aviAudioInFile->ReadAudio(
reinterpret_cast<WebRtc_UWord8*>(outBuffer),
length) != 0)
{
return -1;
}
else
{
return length;
}
}
WebRtc_Word32 ModuleFileUtility::ReadAviVideoData(
WebRtc_Word8* outBuffer,
const WebRtc_UWord32 bufferLengthInBytes)
{
if(_aviVideoInFile == 0)
{
WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "AVI file not opened.");
return -1;
}
WebRtc_Word32 length = bufferLengthInBytes;
if( _aviVideoInFile->ReadVideo(
reinterpret_cast<WebRtc_UWord8*>(outBuffer),
length) != 0)
{
return -1;
} else {
return length;
}
}
WebRtc_Word32 ModuleFileUtility::VideoCodecInst(VideoCodec& codecInst)
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
"ModuleFileUtility::CodecInst(codecInst= 0x%x)", &codecInst);
if(!_reading)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"CodecInst: not currently reading audio file!");
return -1;
}
memcpy(&codecInst,&_videoCodec,sizeof(VideoCodec));
return 0;
}
#endif
WebRtc_Word32 ModuleFileUtility::ReadWavHeader(InStream& wav)
{
WAVE_RIFF_header RIFFheaderObj;
WAVE_CHUNK_header CHUNKheaderObj;
// TODO (hellner): tmpStr and tmpStr2 seems unnecessary here.
WebRtc_Word8 tmpStr[6] = "FOUR";
WebRtc_UWord8 tmpStr2[4];
WebRtc_Word32 i, len;
bool dataFound = false;
bool fmtFound = false;
WebRtc_Word8 dummyRead;
WEBRTC_TRACE(kTraceModuleCall, kTraceFile, _id,
"ModuleFileUtility::ReadWavHeader(wav= 0x%x)", &wav);
_dataSize = 0;
len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
if(len != sizeof(WAVE_RIFF_header))
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"Not a wave file (too short)");
return -1;
}
for (i = 0; i < 4; i++)
{
tmpStr[i] = RIFFheaderObj.ckID[i];
}
if(strcmp(tmpStr, "RIFF") != 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"Not a wave file (does not have RIFF)");
return -1;
}
for (i = 0; i < 4; i++)
{
tmpStr[i] = RIFFheaderObj.wave_ckID[i];
}
if(strcmp(tmpStr, "WAVE") != 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"Not a wave file (does not have WAVE)");
return -1;
}
len = wav.Read(&CHUNKheaderObj, sizeof(WAVE_CHUNK_header));
// WAVE files are stored in little endian byte order. Make sure that the
// data can be read on big endian as well.
// TODO (hellner): little endian to system byte order should be done in
// in a subroutine.
memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
CHUNKheaderObj.fmt_ckSize =
(WebRtc_Word32) ((WebRtc_UWord32) tmpStr2[0] +
(((WebRtc_UWord32)tmpStr2[1])<<8) +
(((WebRtc_UWord32)tmpStr2[2])<<16) +
(((WebRtc_UWord32)tmpStr2[3])<<24));
memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
while ((len == sizeof(WAVE_CHUNK_header)) && (!fmtFound || !dataFound))
{
if(strcmp(tmpStr, "fmt ") == 0)
{
len = wav.Read(&_wavFormatObj, sizeof(WAVE_FMTINFO_header));
memcpy(tmpStr2, &_wavFormatObj.formatTag, 2);
_wavFormatObj.formatTag =
(WaveFormats) ((WebRtc_UWord32)tmpStr2[0] +
(((WebRtc_UWord32)tmpStr2[1])<<8));
memcpy(tmpStr2, &_wavFormatObj.nChannels, 2);
_wavFormatObj.nChannels =
(WebRtc_Word16) ((WebRtc_UWord32)tmpStr2[0] +
(((WebRtc_UWord32)tmpStr2[1])<<8));
memcpy(tmpStr2, &_wavFormatObj.nSamplesPerSec, 4);
_wavFormatObj.nSamplesPerSec =
(WebRtc_Word32) ((WebRtc_UWord32)tmpStr2[0] +
(((WebRtc_UWord32)tmpStr2[1])<<8) +
(((WebRtc_UWord32)tmpStr2[2])<<16) +
(((WebRtc_UWord32)tmpStr2[3])<<24));
memcpy(tmpStr2, &_wavFormatObj.nAvgBytesPerSec, 4);
_wavFormatObj.nAvgBytesPerSec =
(WebRtc_Word32) ((WebRtc_UWord32)tmpStr2[0] +
(((WebRtc_UWord32)tmpStr2[1])<<8) +
(((WebRtc_UWord32)tmpStr2[2])<<16) +
(((WebRtc_UWord32)tmpStr2[3])<<24));
memcpy(tmpStr2, &_wavFormatObj.nBlockAlign, 2);
_wavFormatObj.nBlockAlign =
(WebRtc_Word16) ((WebRtc_UWord32)tmpStr2[0] +
(((WebRtc_UWord32)tmpStr2[1])<<8));
memcpy(tmpStr2, &_wavFormatObj.nBitsPerSample, 2);
_wavFormatObj.nBitsPerSample =
(WebRtc_Word16) ((WebRtc_UWord32)tmpStr2[0] +
(((WebRtc_UWord32)tmpStr2[1])<<8));
for (i = 0;
i < (CHUNKheaderObj.fmt_ckSize -
(WebRtc_Word32)sizeof(WAVE_FMTINFO_header));
i++)
{
len = wav.Read(&dummyRead, 1);
if(len != 1)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"File corrupted, reached EOF (reading fmt)");
return -1;
}
}
fmtFound = true;
}
else if(strcmp(tmpStr, "data") == 0)
{
_dataSize = CHUNKheaderObj.fmt_ckSize;
dataFound = true;
break;
}
else
{
for (i = 0; i < (CHUNKheaderObj.fmt_ckSize); i++)
{
len = wav.Read(&dummyRead, 1);
if(len != 1)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"File corrupted, reached EOF (reading other)");
return -1;
}
}
}
len = wav.Read(&CHUNKheaderObj, sizeof(WAVE_CHUNK_header));
memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
CHUNKheaderObj.fmt_ckSize =
(WebRtc_Word32) ((WebRtc_UWord32)tmpStr2[0] +
(((WebRtc_UWord32)tmpStr2[1])<<8) +
(((WebRtc_UWord32)tmpStr2[2])<<16) +
(((WebRtc_UWord32)tmpStr2[3])<<24));
memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
}
// Either a proper format chunk has been read or a data chunk was come
// across.
if( (_wavFormatObj.formatTag != kWaveFormatPcm) &&
(_wavFormatObj.formatTag != kWaveFormatALaw) &&
(_wavFormatObj.formatTag != kWaveFormatMuLaw))
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"Coding formatTag value=%d not supported!",
_wavFormatObj.formatTag);
return -1;
}
if((_wavFormatObj.nChannels < 1) ||
(_wavFormatObj.nChannels > 2))
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"nChannels value=%d not supported!",
_wavFormatObj.nChannels);
return -1;
}
if((_wavFormatObj.nBitsPerSample != 8) &&
(_wavFormatObj.nBitsPerSample != 16))
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"nBitsPerSample value=%d not supported!",
_wavFormatObj.nBitsPerSample);
return -1;
}
// Calculate the number of bytes that 10 ms of audio data correspond to.
if(_wavFormatObj.formatTag == kWaveFormatPcm)
{
// TODO (hellner): integer division for 22050 and 11025 would yield
// the same result as the else statement. Remove those
// special cases?
if(_wavFormatObj.nSamplesPerSec == 44100)
{
_readSizeBytes = 440 * _wavFormatObj.nChannels *
(_wavFormatObj.nBitsPerSample / 8);
} else if(_wavFormatObj.nSamplesPerSec == 22050) {
_readSizeBytes = 220 * _wavFormatObj.nChannels *
(_wavFormatObj.nBitsPerSample / 8);
} else if(_wavFormatObj.nSamplesPerSec == 11025) {
_readSizeBytes = 110 * _wavFormatObj.nChannels *
(_wavFormatObj.nBitsPerSample / 8);
} else {
_readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
_wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
}
} else {
_readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
_wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
}
WEBRTC_TRACE(
kTraceModuleCall,
kTraceFile,
_id,
"ModuleFileUtility::ReadWavHeader: format=PCM %d KHz, sampleSize=%d,\
nChannels=%d, readSize=%d, dataSize=%d, rate=%d",
_wavFormatObj.nSamplesPerSec/1000,
_wavFormatObj.nBitsPerSample,
_wavFormatObj.nChannels,
_readSizeBytes,
_dataSize,
_wavFormatObj.nAvgBytesPerSec * 8);
return 0;
}
WebRtc_Word32 ModuleFileUtility::InitWavCodec(WebRtc_UWord32 samplesPerSec,
WebRtc_UWord32 channels,
WebRtc_UWord32 bitsPerSample,
WebRtc_UWord32 formatTag)
{
codec_info_.pltype = -1;
codec_info_.plfreq = samplesPerSec;
codec_info_.channels = channels;
codec_info_.rate = bitsPerSample * samplesPerSec;
// Calculate the packet size for 10ms frames
switch(formatTag)
{
case kWaveFormatALaw:
strcpy(codec_info_.plname, "PCMA");
_codecId = kCodecPcma;
codec_info_.pltype = 8;
codec_info_.pacsize = codec_info_.plfreq / 100;
break;
case kWaveFormatMuLaw:
strcpy(codec_info_.plname, "PCMU");
_codecId = kCodecPcmu;
codec_info_.pltype = 0;
codec_info_.pacsize = codec_info_.plfreq / 100;
break;
case kWaveFormatPcm:
codec_info_.pacsize = (bitsPerSample * (codec_info_.plfreq / 100)) / 8;
if(samplesPerSec == 8000)
{
strcpy(codec_info_.plname, "L16");
_codecId = kCodecL16_8Khz;
}
else if(samplesPerSec == 16000)
{
strcpy(codec_info_.plname, "L16");
_codecId = kCodecL16_16kHz;
}
else if(samplesPerSec == 32000)
{
strcpy(codec_info_.plname, "L16");
_codecId = kCodecL16_32Khz;
}
// Set the packet size for "odd" sampling frequencies so that it
// properly corresponds to _readSizeBytes.
else if(samplesPerSec == 11025)
{
strcpy(codec_info_.plname, "L16");
_codecId = kCodecL16_16kHz;
codec_info_.pacsize = 110;
codec_info_.plfreq = 11000;
}
else if(samplesPerSec == 22050)
{
strcpy(codec_info_.plname, "L16");
_codecId = kCodecL16_16kHz;
codec_info_.pacsize = 220;
codec_info_.plfreq = 22000;
}
else if(samplesPerSec == 44100)
{
strcpy(codec_info_.plname, "L16");
_codecId = kCodecL16_16kHz;
codec_info_.pacsize = 440;
codec_info_.plfreq = 44000;
}
else if(samplesPerSec == 48000)
{
strcpy(codec_info_.plname, "L16");
_codecId = kCodecL16_16kHz;
codec_info_.pacsize = 480;
codec_info_.plfreq = 48000;
}
else
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"Unsupported PCM frequency!");
return -1;
}
break;
default:
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"unknown WAV format TAG!");
return -1;
break;
}
return 0;
}
WebRtc_Word32 ModuleFileUtility::InitWavReading(InStream& wav,
const WebRtc_UWord32 start,
const WebRtc_UWord32 stop)
{
WEBRTC_TRACE(
kTraceModuleCall,
kTraceFile,
_id,
"ModuleFileUtility::InitWavReading(wav= 0x%x, start= %d, stop=%d)",
&wav,
start,
stop);
_reading = false;
if(ReadWavHeader(wav) == -1)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"failed to read WAV header!");
return -1;
}
_playoutPositionMs = 0;
_readPos = 0;
if(start > 0)
{
WebRtc_UWord8 dummy[WAV_MAX_BUFFER_SIZE];
WebRtc_Word32 readLength;
if(_readSizeBytes <= WAV_MAX_BUFFER_SIZE)
{
while (_playoutPositionMs < start)
{
readLength = wav.Read(dummy, _readSizeBytes);
if(readLength == _readSizeBytes)
{
_readPos += readLength;
_playoutPositionMs += 10;
}
else // Must have reached EOF before start position!
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"InitWavReading(), EOF before start position");
return -1;
}
}
}
else
{
return -1;
}
}
if( InitWavCodec(_wavFormatObj.nSamplesPerSec, _wavFormatObj.nChannels,
_wavFormatObj.nBitsPerSample,
_wavFormatObj.formatTag) != 0)
{
return -1;
}
_bytesPerSample = _wavFormatObj.nBitsPerSample / 8;
WEBRTC_TRACE(kTraceModuleCall, kTraceFile, _id,
"WAV header: codecName= %s, sampleSize= %d, freq= %d",
codec_info_.plname, _bytesPerSample, codec_info_.plfreq);
_startPointInMs = start;
_stopPointInMs = stop;
_reading = true;
return 0;
}
WebRtc_Word32 ModuleFileUtility::ReadWavDataAsMono(
InStream& wav,
WebRtc_Word8* outData,
const WebRtc_UWord32 bufferSize)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::ReadWavDataAsMono(wav= 0x%x, outData= 0x%d,\
bufSize= %ld)",
&wav,
outData,
bufferSize);
// The number of bytes that should be read from file.
const WebRtc_UWord32 totalBytesNeeded = _readSizeBytes;
// The number of bytes that will be written to outData.
const WebRtc_UWord32 bytesRequested = (codec_info_.channels == 2) ?
totalBytesNeeded >> 1 : totalBytesNeeded;
if(bufferSize < bytesRequested)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavDataAsMono: output buffer is too short!");
return -1;
}
if(outData == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavDataAsMono: output buffer NULL!");
return -1;
}
if(!_reading)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavDataAsMono: no longer reading file.");
return -1;
}
WebRtc_Word32 bytesRead = ReadWavData(
wav,
(codec_info_.channels == 2) ? _tempData : (WebRtc_UWord8*)outData,
totalBytesNeeded);
if(bytesRead == 0)
{
return 0;
}
if(bytesRead < 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavDataAsMono: failed to read data from WAV file.");
return -1;
}
// Output data is should be mono.
if(codec_info_.channels == 2)
{
for (WebRtc_UWord32 i = 0; i < bytesRequested; i++)
{
// Sample value is the average of left and right buffer rounded to
// closest integer value. Note samples can be either 1 or 2 byte.
if(_bytesPerSample == 1)
{
_tempData[i] = ((_tempData[2 * i] + _tempData[(2 * i) + 1] +
1) >> 1);
}
else
{
WebRtc_Word16* sampleData = (WebRtc_Word16*) _tempData;
sampleData[i] = ((sampleData[2 * i] + sampleData[(2 * i) + 1] +
1) >> 1);
}
}
memcpy(outData, _tempData, bytesRequested);
}
return bytesRequested;
}
WebRtc_Word32 ModuleFileUtility::ReadWavDataAsStereo(
InStream& wav,
WebRtc_Word8* outDataLeft,
WebRtc_Word8* outDataRight,
const WebRtc_UWord32 bufferSize)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::ReadWavDataAsStereo(wav= 0x%x, outLeft= 0x%x,\
outRight= 0x%x, bufSize= %ld)",
&wav,
outDataLeft,
outDataRight,
bufferSize);
if((outDataLeft == NULL) ||
(outDataRight == NULL))
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavDataAsMono: an input buffer is NULL!");
return -1;
}
if(codec_info_.channels != 2)
{
WEBRTC_TRACE(
kTraceError,
kTraceFile,
_id,
"ReadWavDataAsStereo: WAV file does not contain stereo data!");
return -1;
}
if(! _reading)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavDataAsStereo: no longer reading file.");
return -1;
}
// The number of bytes that should be read from file.
const WebRtc_UWord32 totalBytesNeeded = _readSizeBytes;
// The number of bytes that will be written to the left and the right
// buffers.
const WebRtc_UWord32 bytesRequested = totalBytesNeeded >> 1;
if(bufferSize < bytesRequested)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavData: Output buffers are too short!");
assert(false);
return -1;
}
WebRtc_Word32 bytesRead = ReadWavData(wav, _tempData, totalBytesNeeded);
if(bytesRead <= 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavDataAsStereo: failed to read data from WAV file.");
return -1;
}
// Turn interleaved audio to left and right buffer. Note samples can be
// either 1 or 2 bytes
if(_bytesPerSample == 1)
{
for (WebRtc_UWord32 i = 0; i < bytesRequested; i++)
{
outDataLeft[i] = _tempData[2 * i];
outDataRight[i] = _tempData[(2 * i) + 1];
}
}
else if(_bytesPerSample == 2)
{
WebRtc_Word16* sampleData = reinterpret_cast<WebRtc_Word16*>(_tempData);
WebRtc_Word16* outLeft = reinterpret_cast<WebRtc_Word16*>(outDataLeft);
WebRtc_Word16* outRight = reinterpret_cast<WebRtc_Word16*>(
outDataRight);
// Bytes requested to samples requested.
WebRtc_UWord32 sampleCount = bytesRequested >> 1;
for (WebRtc_UWord32 i = 0; i < sampleCount; i++)
{
outLeft[i] = sampleData[2 * i];
outRight[i] = sampleData[(2 * i) + 1];
}
} else {
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavStereoData: unsupported sample size %d!",
_bytesPerSample);
assert(false);
return -1;
}
return bytesRequested;
}
WebRtc_Word32 ModuleFileUtility::ReadWavData(
InStream& wav,
WebRtc_UWord8* buffer,
const WebRtc_UWord32 dataLengthInBytes)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::ReadWavData(wav= 0x%x, buffer= 0x%x, dataLen= %ld)",
&wav,
buffer,
dataLengthInBytes);
if(buffer == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadWavDataAsMono: output buffer NULL!");
return -1;
}
// Make sure that a read won't return too few samples.
// TODO (hellner): why not read the remaining bytes needed from the start
// of the file?
if((_dataSize - _readPos) < (WebRtc_Word32)dataLengthInBytes)
{
// Rewind() being -1 may be due to the file not supposed to be looped.
if(wav.Rewind() == -1)
{
_reading = false;
return 0;
}
if(InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1)
{
_reading = false;
return -1;
}
}
WebRtc_Word32 bytesRead = wav.Read(buffer, dataLengthInBytes);
if(bytesRead < 0)
{
_reading = false;
return -1;
}
// This should never happen due to earlier sanity checks.
// TODO (hellner): change to an assert and fail here since this should
// never happen...
if(bytesRead < (WebRtc_Word32)dataLengthInBytes)
{
if((wav.Rewind() == -1) ||
(InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1))
{
_reading = false;
return -1;
}
else
{
bytesRead = wav.Read(buffer, dataLengthInBytes);
if(bytesRead < (WebRtc_Word32)dataLengthInBytes)
{
_reading = false;
return -1;
}
}
}
_readPos += bytesRead;
// TODO (hellner): Why is dataLengthInBytes let dictate the number of bytes
// to read when exactly 10ms should be read?!
_playoutPositionMs += 10;
if((_stopPointInMs > 0) &&
(_playoutPositionMs >= _stopPointInMs))
{
if((wav.Rewind() == -1) ||
(InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1))
{
_reading = false;
}
}
return bytesRead;
}
WebRtc_Word32 ModuleFileUtility::InitWavWriting(OutStream& wav,
const CodecInst& codecInst)
{
WEBRTC_TRACE(kTraceModuleCall, kTraceFile, _id,
"ModuleFileUtility::InitWavWriting(wav= 0x%x, codec=%s)",
&wav, codecInst.plname);
if(set_codec_info(codecInst) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"codecInst identifies unsupported codec!");
return -1;
}
_writing = false;
WebRtc_UWord32 channels = (codecInst.channels == 0) ?
1 : codecInst.channels;
if(STR_CASE_CMP(codecInst.plname, "PCMU") == 0)
{
_bytesPerSample = 1;
if(WriteWavHeader(wav, 8000, _bytesPerSample, channels,
kWaveFormatMuLaw, 0) == -1)
{
return -1;
}
}else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
{
_bytesPerSample = 1;
if(WriteWavHeader(wav, 8000, _bytesPerSample, channels, kWaveFormatALaw,
0) == -1)
{
return -1;
}
}
else if(STR_CASE_CMP(codecInst.plname, "L16") == 0)
{
_bytesPerSample = 2;
if(WriteWavHeader(wav, codecInst.plfreq, _bytesPerSample, channels,
kWaveFormatPcm, 0) == -1)
{
return -1;
}
}
else
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"codecInst identifies unsupported codec for WAV file!");
return -1;
}
_writing = true;
_bytesWritten = 0;
return 0;
}
WebRtc_Word32 ModuleFileUtility::WriteWavData(OutStream& out,
const WebRtc_Word8* buffer,
const WebRtc_UWord32 dataLength)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::WriteWavData(out= 0x%x, buf= 0x%x, dataLen= %d)",
&out,
buffer,
dataLength);
if(buffer == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"WriteWavData: input buffer NULL!");
return -1;
}
if(!out.Write(buffer, dataLength))
{
return -1;
}
_bytesWritten += dataLength;
return dataLength;
}
WebRtc_Word32 ModuleFileUtility::WriteWavHeader(
OutStream& wav,
const WebRtc_UWord32 freq,
const WebRtc_UWord32 bytesPerSample,
const WebRtc_UWord32 channels,
const WebRtc_UWord32 format,
const WebRtc_UWord32 lengthInBytes)
{
WEBRTC_TRACE(
kTraceModuleCall,
kTraceFile,
_id,
"ModuleFileUtility::WriteWavHeader(format= PCM %d KHz,\
bytesPerSample= %d, channels= %d, format= %d, dataLength= %d)",
freq / 1000,
bytesPerSample,
channels,
format,
lengthInBytes);
// Frame size in bytes for 10 ms of audio.
// TODO (hellner): 44.1 kHz has 440 samples frame size. Doesn't seem to
// be taken into consideration here!
WebRtc_Word32 frameSize = (freq / 100) * bytesPerSample * channels;
// Calculate the number of full frames that the wave file contain.
const WebRtc_Word32 dataLengthInBytes = frameSize *
(lengthInBytes / frameSize);
WebRtc_Word8 tmpStr[4];
WebRtc_Word8 tmpChar;
WebRtc_UWord32 tmpLong;
memcpy(tmpStr, "RIFF", 4);
wav.Write(tmpStr, 4);
tmpLong = dataLengthInBytes + 36;
tmpChar = (WebRtc_Word8)(tmpLong);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 8);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 16);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 24);
wav.Write(&tmpChar, 1);
memcpy(tmpStr, "WAVE", 4);
wav.Write(tmpStr, 4);
memcpy(tmpStr, "fmt ", 4);
wav.Write(tmpStr, 4);
tmpChar = 16;
wav.Write(&tmpChar, 1);
tmpChar = 0;
wav.Write(&tmpChar, 1);
tmpChar = 0;
wav.Write(&tmpChar, 1);
tmpChar = 0;
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(format);
wav.Write(&tmpChar, 1);
tmpChar = 0;
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(channels);
wav.Write(&tmpChar, 1);
tmpChar = 0;
wav.Write(&tmpChar, 1);
tmpLong = freq;
tmpChar = (WebRtc_Word8)(tmpLong);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 8);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 16);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 24);
wav.Write(&tmpChar, 1);
// nAverageBytesPerSec = Sample rate * Bytes per sample * Channels
tmpLong = bytesPerSample * freq * channels;
tmpChar = (WebRtc_Word8)(tmpLong);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 8);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 16);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 24);
wav.Write(&tmpChar, 1);
// nBlockAlign = Bytes per sample * Channels
tmpChar = (WebRtc_Word8)(bytesPerSample * channels);
wav.Write(&tmpChar, 1);
tmpChar = 0;
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(bytesPerSample*8);
wav.Write(&tmpChar, 1);
tmpChar = 0;
wav.Write(&tmpChar, 1);
memcpy(tmpStr, "data", 4);
wav.Write(tmpStr, 4);
tmpLong = dataLengthInBytes;
tmpChar = (WebRtc_Word8)(tmpLong);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 8);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 16);
wav.Write(&tmpChar, 1);
tmpChar = (WebRtc_Word8)(tmpLong >> 24);
wav.Write(&tmpChar, 1);
return 0;
}
WebRtc_Word32 ModuleFileUtility::UpdateWavHeader(OutStream& wav)
{
WebRtc_Word32 res = -1;
if(wav.Rewind() == -1)
{
return -1;
}
WebRtc_UWord32 channels = (codec_info_.channels == 0) ?
1 : codec_info_.channels;
if(STR_CASE_CMP(codec_info_.plname, "L16") == 0)
{
res = WriteWavHeader(wav, codec_info_.plfreq, 2, channels,
kWaveFormatPcm, _bytesWritten);
} else if(STR_CASE_CMP(codec_info_.plname, "PCMU") == 0) {
res = WriteWavHeader(wav, 8000, 1, channels, kWaveFormatMuLaw,
_bytesWritten);
} else if(STR_CASE_CMP(codec_info_.plname, "PCMU") == 0) {
res = WriteWavHeader(wav, 8000, 1, channels, kWaveFormatALaw,
_bytesWritten);
} else {
// Allow calling this API even if not writing to a WAVE file.
// TODO (hellner): why?!
return 0;
}
return res;
}
WebRtc_Word32 ModuleFileUtility::InitPreEncodedReading(InStream& in,
const CodecInst& cinst)
{
WEBRTC_TRACE(kTraceModuleCall, kTraceFile, _id,
"ModuleFileUtility::InitPreEncodedReading(in=0x%x, codec='%s')",
&in, cinst.plname);
WebRtc_UWord8 preEncodedID;
in.Read(&preEncodedID, 1);
MediaFileUtility_CodecType codecType =
(MediaFileUtility_CodecType)preEncodedID;
if(set_codec_info(cinst) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"Pre-encoded file send codec mismatch!");
return -1;
}
if(codecType != _codecId)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"Pre-encoded file format codec mismatch!");
return -1;
}
memcpy(&codec_info_,&cinst,sizeof(CodecInst));
_reading = true;
return 0;
}
WebRtc_Word32 ModuleFileUtility::ReadPreEncodedData(
InStream& in,
WebRtc_Word8* outData,
const WebRtc_UWord32 bufferSize)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::ReadPreEncodedData(in= 0x%x, outData= 0x%x,\
bufferSize= %d)",
&in,
outData,
bufferSize);
if(outData == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id, "output buffer NULL");
}
WebRtc_UWord32 frameLen;
WebRtc_UWord8 buf[64];
// Each frame has a two byte header containing the frame length.
WebRtc_Word32 res = in.Read(buf, 2);
if(res != 2)
{
if(!in.Rewind())
{
// The first byte is the codec identifier.
in.Read(buf, 1);
res = in.Read(buf, 2);
}
else
{
return -1;
}
}
frameLen = buf[0] + buf[1] * 256;
if(bufferSize < frameLen)
{
WEBRTC_TRACE(
kTraceError,
kTraceFile,
_id,
"buffer not large enough to read %d bytes of pre-encoded data!",
frameLen);
return -1;
}
return in.Read(outData, frameLen);
}
WebRtc_Word32 ModuleFileUtility::InitPreEncodedWriting(
OutStream& out,
const CodecInst& codecInst)
{
WEBRTC_TRACE(
kTraceModuleCall,
kTraceFile,
_id,
"ModuleFileUtility::InitFormatedWriting(out=0x%x, codecInst= %s)",
&out,
codecInst.plname);
if(set_codec_info(codecInst) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id, "CodecInst not recognized!");
return -1;
}
_writing = true;
_bytesWritten = 1;
out.Write(&_codecId, 1);
return 0;
}
WebRtc_Word32 ModuleFileUtility::WritePreEncodedData(
OutStream& out,
const WebRtc_Word8* buffer,
const WebRtc_UWord32 dataLength)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::WritePreEncodedData(out= 0x%x, inData= 0x%x,\
dataLen= %d)",
&out,
buffer,
dataLength);
if(buffer == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
}
WebRtc_Word32 bytesWritten = 0;
// The first two bytes is the size of the frame.
WebRtc_Word16 lengthBuf;
lengthBuf = (WebRtc_Word16)dataLength;
if(!out.Write(&lengthBuf, 2))
{
return -1;
}
bytesWritten = 2;
if(!out.Write(buffer, dataLength))
{
return -1;
}
bytesWritten += dataLength;
return bytesWritten;
}
WebRtc_Word32 ModuleFileUtility::InitCompressedReading(
InStream& in,
const WebRtc_UWord32 start,
const WebRtc_UWord32 stop)
{
WEBRTC_TRACE(
kTraceDebug,
kTraceFile,
_id,
"ModuleFileUtility::InitCompressedReading(in= 0x%x, start= %d,\
stop= %d)",
&in,
start,
stop);
WebRtc_Word16 read_len = 0;
_codecId = kCodecNoCodec;
_playoutPositionMs = 0;
_reading = false;
_startPointInMs = start;
_stopPointInMs = stop;
#ifdef WEBRTC_CODEC_GSMAMR
WebRtc_Word32 AMRmode2bytes[9]={12,13,15,17,19,20,26,31,5};
#endif
#ifdef WEBRTC_CODEC_GSMAMRWB
WebRtc_Word32 AMRWBmode2bytes[10]={17,23,32,36,40,46,50,58,60,6};
#endif
// Read the codec name
WebRtc_Word32 cnt = 0;
WebRtc_Word8 buf[64];
do
{
in.Read(&buf[cnt++], 1);
} while ((buf[cnt-1] != '\n') && (64 > cnt));
if(cnt==64)
{
return -1;
} else {
buf[cnt]=0;
}
#ifdef WEBRTC_CODEC_GSMAMR
if(!strcmp("#!AMR\n", buf))
{
strcpy(codec_info_.plname, "amr");
codec_info_.pacsize = 160;
_codecId = kCodecAmr;
codec_info_.pltype = 112;
codec_info_.rate = 12200;
codec_info_.plfreq = 8000;
codec_info_.channels = 1;
WebRtc_Word16 mode = 0;
if(_startPointInMs > 0)
{
while (_playoutPositionMs <= _startPointInMs)
{
// First read byte contain the AMR mode.
read_len = in.Read(buf, 1);
if(read_len != 1)
{
return -1;
}
mode = (buf[0]>>3)&0xF;
if((mode < 0) || (mode > 8))
{
if(mode != 15)
{
return -1;
}
}
if(mode != 15)
{
read_len = in.Read(&buf[1], AMRmode2bytes[mode]);
if(read_len != AMRmode2bytes[mode])
{
return -1;
}
}
_playoutPositionMs += 20;
}
}
}
#endif
#ifdef WEBRTC_CODEC_GSMAMRWB
if(!strcmp("#!AMRWB\n", buf))
{
strcpy(codec_info_.plname, "amr-wb");
codec_info_.pacsize = 320;
_codecId = kCodecAmrWb;
codec_info_.pltype = 120;
codec_info_.rate = 20000;
codec_info_.plfreq = 16000;
codec_info_.channels = 1;
WebRtc_Word16 mode = 0;
if(_startPointInMs > 0)
{
while (_playoutPositionMs <= _startPointInMs)
{
// First read byte contain the AMR mode.
read_len = in.Read(buf, 1);
if(read_len != 1)
{
return -1;
}
mode = (buf[0]>>3)&0xF;
if((mode < 0) || (mode > 9))
{
if(mode != 15)
{
return -1;
}
}
if(mode != 15)
{
read_len = in.Read(&buf[1], AMRWBmode2bytes[mode]);
if(read_len != AMRWBmode2bytes[mode])
{
return -1;
}
}
_playoutPositionMs += 20;
}
}
}
#endif
#ifdef WEBRTC_CODEC_ILBC
if(!strcmp("#!iLBC20\n", buf))
{
codec_info_.pltype = 102;
strcpy(codec_info_.plname, "ilbc");
codec_info_.plfreq = 8000;
codec_info_.pacsize = 160;
codec_info_.channels = 1;
codec_info_.rate = 13300;
_codecId = kCodecIlbc20Ms;
if(_startPointInMs > 0)
{
while (_playoutPositionMs <= _startPointInMs)
{
read_len = in.Read(buf, 38);
if(read_len == 38)
{
_playoutPositionMs += 20;
}
else
{
return -1;
}
}
}
}
if(!strcmp("#!iLBC30\n", buf))
{
codec_info_.pltype = 102;
strcpy(codec_info_.plname, "ilbc");
codec_info_.plfreq = 8000;
codec_info_.pacsize = 240;
codec_info_.channels = 1;
codec_info_.rate = 13300;
_codecId = kCodecIlbc30Ms;
if(_startPointInMs > 0)
{
while (_playoutPositionMs <= _startPointInMs)
{
read_len = in.Read(buf, 50);
if(read_len == 50)
{
_playoutPositionMs += 20;
}
else
{
return -1;
}
}
}
}
#endif
if(_codecId == kCodecNoCodec)
{
return -1;
}
_reading = true;
return 0;
}
WebRtc_Word32 ModuleFileUtility::ReadCompressedData(InStream& in,
WebRtc_Word8* outData,
WebRtc_UWord32 bufferSize)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::ReadCompressedData(in=0x%x, outData=0x%x,\
bytes=%ld)",
&in,
outData,
bufferSize);
#ifdef WEBRTC_CODEC_GSMAMR
WebRtc_UWord32 AMRmode2bytes[9]={12,13,15,17,19,20,26,31,5};
#endif
#ifdef WEBRTC_CODEC_GSMAMRWB
WebRtc_UWord32 AMRWBmode2bytes[10]={17,23,32,36,40,46,50,58,60,6};
#endif
WebRtc_UWord32 bytesRead = 0;
if(! _reading)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id, "not currently reading!");
return -1;
}
#ifdef WEBRTC_CODEC_GSMAMR
if(_codecId == kCodecAmr)
{
WebRtc_Word32 res = in.Read(outData, 1);
if(res != 1)
{
if(!in.Rewind())
{
InitCompressedReading(in, _startPointInMs, _stopPointInMs);
res = in.Read(outData, 1);
if(res != 1)
{
_reading = false;
return -1;
}
}
else
{
_reading = false;
return -1;
}
}
const WebRtc_Word16 mode = (outData[0]>>3)&0xF;
if((mode < 0) ||
(mode > 8))
{
if(mode != 15)
{
return -1;
}
}
if(mode != 15)
{
if(bufferSize < AMRmode2bytes[mode] + 1)
{
WEBRTC_TRACE(
kTraceError,
kTraceFile,
_id,
"output buffer is too short to read AMR compressed data.");
assert(false);
return -1;
}
bytesRead = in.Read(&outData[1], AMRmode2bytes[mode]);
if(bytesRead != AMRmode2bytes[mode])
{
_reading = false;
return -1;
}
// Count the mode byte to bytes read.
bytesRead++;
}
else
{
bytesRead = 1;
}
}
#endif
#ifdef WEBRTC_CODEC_GSMAMRWB
if(_codecId == kCodecAmrWb)
{
WebRtc_Word32 res = in.Read(outData, 1);
if(res != 1)
{
if(!in.Rewind())
{
InitCompressedReading(in, _startPointInMs, _stopPointInMs);
res = in.Read(outData, 1);
if(res != 1)
{
_reading = false;
return -1;
}
}
else
{
_reading = false;
return -1;
}
}
WebRtc_Word16 mode = (outData[0]>>3)&0xF;
if((mode < 0) ||
(mode > 8))
{
if(mode != 15)
{
return -1;
}
}
if(mode != 15)
{
if(bufferSize < AMRWBmode2bytes[mode] + 1)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"output buffer is too short to read AMRWB\
compressed.");
assert(false);
return -1;
}
bytesRead = in.Read(&outData[1], AMRWBmode2bytes[mode]);
if(bytesRead != AMRWBmode2bytes[mode])
{
_reading = false;
return -1;
}
bytesRead++;
}
else
{
bytesRead = 1;
}
}
#endif
#ifdef WEBRTC_CODEC_ILBC
if((_codecId == kCodecIlbc20Ms) ||
(_codecId == kCodecIlbc30Ms))
{
WebRtc_UWord32 byteSize = 0;
if(_codecId == kCodecIlbc30Ms)
{
byteSize = 50;
}
if(_codecId == kCodecIlbc20Ms)
{
byteSize = 38;
}
if(bufferSize < byteSize)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"output buffer is too short to read ILBC compressed\
data.");
assert(false);
return -1;
}
bytesRead = in.Read(outData, byteSize);
if(bytesRead != byteSize)
{
if(!in.Rewind())
{
InitCompressedReading(in, _startPointInMs, _stopPointInMs);
bytesRead = in.Read(outData, byteSize);
if(bytesRead != byteSize)
{
_reading = false;
return -1;
}
}
else
{
_reading = false;
return -1;
}
}
}
#endif
if(bytesRead == 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadCompressedData() no bytes read, codec not supported");
return -1;
}
_playoutPositionMs += 20;
if((_stopPointInMs > 0) &&
(_playoutPositionMs >= _stopPointInMs))
{
if(!in.Rewind())
{
InitCompressedReading(in, _startPointInMs, _stopPointInMs);
}
else
{
_reading = false;
}
}
return bytesRead;
}
WebRtc_Word32 ModuleFileUtility::InitCompressedWriting(
OutStream& out,
const CodecInst& codecInst)
{
WEBRTC_TRACE(kTraceDebug, kTraceFile, _id,
"ModuleFileUtility::InitCompressedWriting(out= 0x%x,\
codecName= %s)",
&out, codecInst.plname);
_writing = false;
#ifdef WEBRTC_CODEC_GSMAMR
if(STR_CASE_CMP(codecInst.plname, "amr") == 0)
{
if(codecInst.pacsize == 160)
{
memcpy(&codec_info_,&codecInst,sizeof(CodecInst));
_codecId = kCodecAmr;
out.Write("#!AMR\n",6);
_writing = true;
return 0;
}
}
#endif
#ifdef WEBRTC_CODEC_GSMAMRWB
if(STR_CASE_CMP(codecInst.plname, "amr-wb") == 0)
{
if(codecInst.pacsize == 320)
{
memcpy(&codec_info_,&codecInst,sizeof(CodecInst));
_codecId = kCodecAmrWb;
out.Write("#!AMRWB\n",8);
_writing = true;
return 0;
}
}
#endif
#ifdef WEBRTC_CODEC_ILBC
if(STR_CASE_CMP(codecInst.plname, "ilbc") == 0)
{
if(codecInst.pacsize == 160)
{
_codecId = kCodecIlbc20Ms;
out.Write("#!iLBC20\n",9);
}
else if(codecInst.pacsize == 240)
{
_codecId = kCodecIlbc30Ms;
out.Write("#!iLBC30\n",9);
}
else
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"codecInst defines unsupported compression codec!");
return -1;
}
memcpy(&codec_info_,&codecInst,sizeof(CodecInst));
_writing = true;
return 0;
}
#endif
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"codecInst defines unsupported compression codec!");
return -1;
}
WebRtc_Word32 ModuleFileUtility::WriteCompressedData(
OutStream& out,
const WebRtc_Word8* buffer,
const WebRtc_UWord32 dataLength)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::WriteCompressedData(out= 0x%x, buf= 0x%x,\
dataLen= %d)",
&out,
buffer,
dataLength);
if(buffer == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
}
if(!out.Write(buffer, dataLength))
{
return -1;
}
return dataLength;
}
WebRtc_Word32 ModuleFileUtility::InitPCMReading(InStream& pcm,
const WebRtc_UWord32 start,
const WebRtc_UWord32 stop,
WebRtc_UWord32 freq)
{
WEBRTC_TRACE(
kTraceInfo,
kTraceFile,
_id,
"ModuleFileUtility::InitPCMReading(pcm= 0x%x, start=%d, stop=%d,\
freq=%d)",
&pcm,
start,
stop,
freq);
WebRtc_Word8 dummy[320];
WebRtc_Word32 read_len;
_playoutPositionMs = 0;
_startPointInMs = start;
_stopPointInMs = stop;
_reading = false;
if(freq == 8000)
{
strcpy(codec_info_.plname, "L16");
codec_info_.pltype = -1;
codec_info_.plfreq = 8000;
codec_info_.pacsize = 160;
codec_info_.channels = 1;
codec_info_.rate = 128000;
_codecId = kCodecL16_8Khz;
}
else if(freq == 16000)
{
strcpy(codec_info_.plname, "L16");
codec_info_.pltype = -1;
codec_info_.plfreq = 16000;
codec_info_.pacsize = 320;
codec_info_.channels = 1;
codec_info_.rate = 256000;
_codecId = kCodecL16_16kHz;
}
else if(freq == 32000)
{
strcpy(codec_info_.plname, "L16");
codec_info_.pltype = -1;
codec_info_.plfreq = 32000;
codec_info_.pacsize = 320;
codec_info_.channels = 1;
codec_info_.rate = 512000;
_codecId = kCodecL16_32Khz;
}
// Readsize for 10ms of audio data (2 bytes per sample).
_readSizeBytes = 2 * codec_info_. plfreq / 100;
if(_startPointInMs > 0)
{
while (_playoutPositionMs < _startPointInMs)
{
read_len = pcm.Read(dummy, _readSizeBytes);
if(read_len == _readSizeBytes)
{
_playoutPositionMs += 10;
}
else // Must have reached EOF before start position!
{
return -1;
}
}
}
_reading = true;
return 0;
}
WebRtc_Word32 ModuleFileUtility::ReadPCMData(InStream& pcm,
WebRtc_Word8* outData,
WebRtc_UWord32 bufferSize)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::ReadPCMData(pcm= 0x%x, outData= 0x%x, bufSize= %d)",
&pcm,
outData,
bufferSize);
if(outData == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
}
// Readsize for 10ms of audio data (2 bytes per sample).
WebRtc_UWord32 bytesRequested = 2 * codec_info_.plfreq / 100;
if(bufferSize < bytesRequested)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadPCMData: buffer not long enough for a 10ms frame.");
assert(false);
return -1;
}
WebRtc_UWord32 bytesRead = pcm.Read(outData, bytesRequested);
if(bytesRead < bytesRequested)
{
if(pcm.Rewind() == -1)
{
_reading = false;
}
else
{
if(InitPCMReading(pcm, _startPointInMs, _stopPointInMs,
codec_info_.plfreq) == -1)
{
_reading = false;
}
else
{
WebRtc_Word32 rest = bytesRequested - bytesRead;
WebRtc_Word32 len = pcm.Read(&(outData[bytesRead]), rest);
if(len == rest)
{
bytesRead += len;
}
else
{
_reading = false;
}
}
if(bytesRead <= 0)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"ReadPCMData: Failed to rewind audio file.");
return -1;
}
}
}
if(bytesRead <= 0)
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
"ReadPCMData: end of file");
return -1;
}
_playoutPositionMs += 10;
if(_stopPointInMs && _playoutPositionMs >= _stopPointInMs)
{
if(!pcm.Rewind())
{
if(InitPCMReading(pcm, _startPointInMs, _stopPointInMs,
codec_info_.plfreq) == -1)
{
_reading = false;
}
}
}
return bytesRead;
}
WebRtc_Word32 ModuleFileUtility::InitPCMWriting(OutStream& out,
WebRtc_UWord32 freq)
{
WEBRTC_TRACE(kTraceModuleCall, kTraceFile, _id,
"ModuleFileUtility::InitPCMWriting(out=0x%x, freq= %ld)", &out,
freq);
if(freq == 8000)
{
strcpy(codec_info_.plname, "L16");
codec_info_.pltype = -1;
codec_info_.plfreq = 8000;
codec_info_.pacsize = 160;
codec_info_.channels = 1;
codec_info_.rate = 128000;
_codecId = kCodecL16_8Khz;
}
else if(freq == 16000)
{
strcpy(codec_info_.plname, "L16");
codec_info_.pltype = -1;
codec_info_.plfreq = 16000;
codec_info_.pacsize = 320;
codec_info_.channels = 1;
codec_info_.rate = 256000;
_codecId = kCodecL16_16kHz;
}
else if(freq == 32000)
{
strcpy(codec_info_.plname, "L16");
codec_info_.pltype = -1;
codec_info_.plfreq = 32000;
codec_info_.pacsize = 320;
codec_info_.channels = 1;
codec_info_.rate = 512000;
_codecId = kCodecL16_32Khz;
}
if((_codecId != kCodecL16_8Khz) &&
(_codecId != kCodecL16_16kHz) &&
(_codecId != kCodecL16_32Khz))
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"CodecInst is not 8KHz PCM or 16KHz PCM!");
return -1;
}
_writing = true;
_bytesWritten = 0;
return 0;
}
WebRtc_Word32 ModuleFileUtility::WritePCMData(OutStream& out,
const WebRtc_Word8* buffer,
const WebRtc_UWord32 dataLength)
{
WEBRTC_TRACE(
kTraceStream,
kTraceFile,
_id,
"ModuleFileUtility::WritePCMData(out= 0x%x, buf= 0x%x, dataLen= %d)",
&out,
buffer,
dataLength);
if(buffer == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id, "buffer NULL");
}
if(!out.Write(buffer, dataLength))
{
return -1;
}
_bytesWritten += dataLength;
return dataLength;
}
WebRtc_Word32 ModuleFileUtility::codec_info(CodecInst& codecInst)
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
"ModuleFileUtility::codec_info(codecInst= 0x%x)", &codecInst);
if(!_reading && !_writing)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"CodecInst: not currently reading audio file!");
return -1;
}
memcpy(&codecInst,&codec_info_,sizeof(CodecInst));
return 0;
}
WebRtc_Word32 ModuleFileUtility::set_codec_info(const CodecInst& codecInst)
{
WEBRTC_TRACE(kTraceModuleCall, kTraceFile, _id,
"ModuleFileUtility::set_codec_info(codecName= %s)",
codecInst.plname);
_codecId = kCodecNoCodec;
if(STR_CASE_CMP(codecInst.plname, "PCMU") == 0)
{
_codecId = kCodecPcmu;
}
else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
{
_codecId = kCodecPcma;
}
else if(STR_CASE_CMP(codecInst.plname, "L16") == 0)
{
if(codecInst.plfreq == 8000)
{
_codecId = kCodecL16_8Khz;
}
else if(codecInst.plfreq == 16000)
{
_codecId = kCodecL16_16kHz;
}
else if(codecInst.plfreq == 32000)
{
_codecId = kCodecL16_32Khz;
}
}
#ifdef WEBRTC_CODEC_GSMAMR
else if(STR_CASE_CMP(codecInst.plname, "amr") == 0)
{
_codecId = kCodecAmr;
}
#endif
#ifdef WEBRTC_CODEC_GSMAMRWB
else if(STR_CASE_CMP(codecInst.plname, "amr-wb") == 0)
{
_codecId = kCodecAmrWb;
}
#endif
#ifdef WEBRTC_CODEC_ILBC
else if(STR_CASE_CMP(codecInst.plname, "ilbc") == 0)
{
if(codecInst.pacsize == 160)
{
_codecId = kCodecIlbc20Ms;
}
else if(codecInst.pacsize == 240)
{
_codecId = kCodecIlbc30Ms;
}
}
#endif
#if(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
else if(STR_CASE_CMP(codecInst.plname, "isac") == 0)
{
if(codecInst.plfreq == 16000)
{
_codecId = kCodecIsac;
}
else if(codecInst.plfreq == 32000)
{
_codecId = kCodecIsacSwb;
}
}
#endif
#ifdef WEBRTC_CODEC_ISACLC
else if(STR_CASE_CMP(codecInst.plname, "isaclc") == 0)
{
_codecId = kCodecIsacLc;
}
#endif
#ifdef WEBRTC_CODEC_G722
else if(STR_CASE_CMP(codecInst.plname, "G722") == 0)
{
_codecId = kCodecG722;
}
#endif
else if(STR_CASE_CMP(codecInst.plname, "G7221") == 0)
{
#ifdef WEBRTC_CODEC_G722_1
if(codecInst.plfreq == 16000)
{
if(codecInst.rate == 16000)
{
_codecId = kCodecG722_1_16Kbps;
}
else if(codecInst.rate == 24000)
{
_codecId = kCodecG722_1_24Kbps;
}
else if(codecInst.rate == 32000)
{
_codecId = kCodecG722_1_32Kbps;
}
}
#endif
#ifdef WEBRTC_CODEC_G722_1C
if(codecInst.plfreq == 32000)
{
if(codecInst.rate == 48000)
{
_codecId = kCodecG722_1c_48;
}
else if(codecInst.rate == 32000)
{
_codecId = kCodecG722_1c_32;
}
else if(codecInst.rate == 24000)
{
_codecId = kCodecG722_1c_24;
}
}
#endif
}
#ifdef WEBRTC_CODEC_G726
else if(STR_CASE_CMP(codecInst.plname, "G726-40") == 0)
{
_codecId = kCodecG726_40;
}
else if(STR_CASE_CMP(codecInst.plname, "G726-32") == 0)
{
_codecId = kCodecG726_24;
}
else if(STR_CASE_CMP(codecInst.plname, "G726-24") == 0)
{
_codecId = kCodecG726_32;
}
else if(STR_CASE_CMP(codecInst.plname, "G726-16") == 0)
{
_codecId = kCodecG726_16;
}
#endif
#ifdef WEBRTC_CODEC_G729
else if(STR_CASE_CMP(codecInst.plname, "G729") == 0)
{
_codecId = kCodecG729;
}
#endif
#ifdef WEBRTC_CODEC_G729_1
else if(STR_CASE_CMP(codecInst.plname, "G7291") == 0)
{
_codecId = kCodecG729_1;
}
#endif
#ifdef WEBRTC_CODEC_SPEEX
else if(STR_CASE_CMP(codecInst.plname, "speex") == 0)
{
if(codecInst.plfreq == 8000)
{
_codecId = kCodecSpeex8Khz;
}
else if(codecInst.plfreq == 16000)
{
_codecId = kCodecSpeex16Khz;
}
}
#endif
if(_codecId == kCodecNoCodec)
{
return -1;
}
memcpy(&codec_info_, &codecInst, sizeof(CodecInst));
return 0;
}
WebRtc_Word32 ModuleFileUtility::FileDurationMs(const WebRtc_Word8* fileName,
const FileFormats fileFormat,
const WebRtc_UWord32 freqInHz)
{
WEBRTC_TRACE(
kTraceModuleCall,
kTraceFile,
_id,
"ModuleFileUtility::FileDuration(%s, format= %d, frequency %d)",
fileName,
fileFormat,
freqInHz);
if(fileName == NULL)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id, "filename NULL");
}
WebRtc_Word32 time_in_ms = -1;
struct stat file_size;
if(stat(fileName,&file_size) == -1)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"failed to retrieve file size with stat!");
return -1;
}
FileWrapper* inStreamObj = FileWrapper::Create();
if(inStreamObj == NULL)
{
WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
"failed to create InStream object!");
return -1;
}
if(inStreamObj->OpenFile(fileName, true) == -1)
{
delete inStreamObj;
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"failed to open file %s!", fileName);
return -1;
}
switch (fileFormat)
{
case kFileFormatWavFile:
{
if(ReadWavHeader(*inStreamObj) == -1)
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"failed to read WAV file header!");
return -1;
}
time_in_ms = ((file_size.st_size - 44) /
(_wavFormatObj.nAvgBytesPerSec/1000));
break;
}
case kFileFormatPcm16kHzFile:
{
// 16 samples per ms. 2 bytes per sample.
WebRtc_Word32 denominator = 16*2;
time_in_ms = (file_size.st_size)/denominator;
break;
}
case kFileFormatPcm8kHzFile:
{
// 8 samples per ms. 2 bytes per sample.
WebRtc_Word32 denominator = 8*2;
time_in_ms = (file_size.st_size)/denominator;
break;
}
case kFileFormatCompressedFile:
{
WebRtc_Word32 cnt = 0;
WebRtc_Word32 read_len = 0;
WebRtc_Word8 buf[64];
do
{
read_len = inStreamObj->Read(&buf[cnt++], 1);
if(read_len != 1)
{
return -1;
}
} while ((buf[cnt-1] != '\n') && (64 > cnt));
if(cnt == 64)
{
return -1;
}
else
{
buf[cnt] = 0;
}
#ifdef WEBRTC_CODEC_GSMAMR
if(!strcmp("#!AMR\n", buf))
{
WebRtc_UWord8 dummy;
read_len = inStreamObj->Read(&dummy, 1);
if(read_len != 1)
{
return -1;
}
WebRtc_Word16 AMRMode = (dummy>>3)&0xF;
// TODO (hellner): use tables instead of hardcoding like this!
// Additionally, this calculation does not
// take octet alignment into consideration.
switch (AMRMode)
{
// Mode 0: 4.75 kbit/sec -> 95 bits per 20 ms frame.
// 20 ms = 95 bits ->
// file size in bytes * 8 / 95 is the number of
// 20 ms frames in the file ->
// time_in_ms = file size * 8 / 95 * 20
case 0:
time_in_ms = ((file_size.st_size)*160)/95;
break;
// Mode 1: 5.15 kbit/sec -> 103 bits per 20 ms frame.
case 1:
time_in_ms = ((file_size.st_size)*160)/103;
break;
// Mode 2: 5.90 kbit/sec -> 118 bits per 20 ms frame.
case 2:
time_in_ms = ((file_size.st_size)*160)/118;
break;
// Mode 3: 6.70 kbit/sec -> 134 bits per 20 ms frame.
case 3:
time_in_ms = ((file_size.st_size)*160)/134;
break;
// Mode 4: 7.40 kbit/sec -> 148 bits per 20 ms frame.
case 4:
time_in_ms = ((file_size.st_size)*160)/148;
break;
// Mode 5: 7.95 bit/sec -> 159 bits per 20 ms frame.
case 5:
time_in_ms = ((file_size.st_size)*160)/159;
break;
// Mode 6: 10.2 bit/sec -> 204 bits per 20 ms frame.
case 6:
time_in_ms = ((file_size.st_size)*160)/204;
break;
// Mode 7: 12.2 bit/sec -> 244 bits per 20 ms frame.
case 7:
time_in_ms = ((file_size.st_size)*160)/244;
break;
// Mode 8: SID Mode -> 39 bits per 20 ms frame.
case 8:
time_in_ms = ((file_size.st_size)*160)/39;
break;
default:
break;
}
}
#endif
#ifdef WEBRTC_CODEC_GSMAMRWB
if(!strcmp("#!AMRWB\n", buf))
{
WebRtc_UWord8 dummy;
read_len = inStreamObj->Read(&dummy, 1);
if(read_len != 1)
{
return -1;
}
// TODO (hellner): use tables instead of hardcoding like this!
WebRtc_Word16 AMRWBMode = (dummy>>3)&0xF;
switch(AMRWBMode)
{
// Mode 0: 6.6 kbit/sec -> 132 bits per 20 ms frame.
case 0:
time_in_ms = ((file_size.st_size)*160)/132;
break;
// Mode 1: 8.85 kbit/sec -> 177 bits per 20 ms frame.
case 1:
time_in_ms = ((file_size.st_size)*160)/177;
break;
// Mode 2: 12.65 kbit/sec -> 253 bits per 20 ms frame.
case 2:
time_in_ms = ((file_size.st_size)*160)/253;
break;
// Mode 3: 14.25 kbit/sec -> 285 bits per 20 ms frame.
case 3:
time_in_ms = ((file_size.st_size)*160)/285;
break;
// Mode 4: 15.85 kbit/sec -> 317 bits per 20 ms frame.
case 4:
time_in_ms = ((file_size.st_size)*160)/317;
break;
// Mode 5: 18.25 kbit/sec -> 365 bits per 20 ms frame.
case 5:
time_in_ms = ((file_size.st_size)*160)/365;
break;
// Mode 6: 19.85 kbit/sec -> 397 bits per 20 ms frame.
case 6:
time_in_ms = ((file_size.st_size)*160)/397;
break;
// Mode 7: 23.05 kbit/sec -> 461 bits per 20 ms frame.
case 7:
time_in_ms = ((file_size.st_size)*160)/461;
break;
// Mode 8: 23.85 kbit/sec -> 477 bits per 20 ms frame.
case 8:
time_in_ms = ((file_size.st_size)*160)/477;
break;
default:
delete inStreamObj;
return -1;
}
}
#endif
#ifdef WEBRTC_CODEC_ILBC
if(!strcmp("#!iLBC20\n", buf))
{
// 20 ms is 304 bits
time_in_ms = ((file_size.st_size)*160)/304;
break;
}
if(!strcmp("#!iLBC30\n", buf))
{
// 30 ms takes 400 bits.
// file size in bytes * 8 / 400 is the number of
// 30 ms frames in the file ->
// time_in_ms = file size * 8 / 400 * 30
time_in_ms = ((file_size.st_size)*240)/400;
break;
}
#endif
}
case kFileFormatPreencodedFile:
{
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"cannot determine duration of Pre-Encoded file!");
break;
}
default:
WEBRTC_TRACE(kTraceError, kTraceFile, _id,
"unsupported file format %d!", fileFormat);
break;
}
inStreamObj->CloseFile();
delete inStreamObj;
return time_in_ms;
}
WebRtc_UWord32 ModuleFileUtility::PlayoutPositionMs()
{
WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
"ModuleFileUtility::PlayoutPosition()");
if(_reading)
{
return _playoutPositionMs;
}
else
{
return 0;
}
}
} // namespace webrtc