ref: e98a69b9db88ef00994bbc7a95add426a36adcf6
parent: 9831e451d4dffcfe05f7d8d60e3c96c8f04d7eef
author: menno <menno>
date: Mon Jul 7 17:11:18 EDT 2003
Added file playback (WIN32) with priority setting Coupling channels are now skipped Minor updates in frontend ADIF bugfix Small fixes
--- a/aacDECdrop/decode.c
+++ b/aacDECdrop/decode.c
@@ -16,8 +16,8 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: decode.c,v 1.13 2003/05/07 18:30:48 menno Exp $
-** $Id: decode.c,v 1.13 2003/05/07 18:30:48 menno Exp $
+** $Id: decode.c,v 1.14 2003/07/07 21:11:17 menno Exp $
+** $Id: decode.c,v 1.14 2003/07/07 21:11:17 menno Exp $
**/
#ifdef _WIN32
@@ -42,7 +42,7 @@
#define min(a,b) ( (a) < (b) ? (a) : (b) )
#endif
-#define MAX_CHANNELS 6 /* make this higher to support files with
+#define MAX_CHANNELS 8 /* make this higher to support files with
more channels */
/* FAAD file buffering routines */
--- a/common/mp4v2/Makefile.am
+++ b/common/mp4v2/Makefile.am
@@ -16,13 +16,18 @@
atom_drep.cpp \
atom_edts.cpp \
atom_elst.cpp \
+ atom_enca.cpp \
+ atom_encv.cpp \
atom_esds.cpp \
atom_free.cpp \
+ atom_frma.cpp \
atom_ftyp.cpp \
atom_hdlr.cpp \
atom_hinf.cpp \
atom_hmhd.cpp \
atom_hnti.cpp \
+ atom_iKMS.cpp \
+ atom_iSFM.cpp \
atom_iods.cpp \
atom_maxr.cpp \
atom_mdat.cpp \
@@ -44,7 +49,10 @@
atom_pmax.cpp \
atom_root.cpp \
atom_rtp.cpp \
+ atom_schi.cpp \
+ atom_schm.cpp \
atom_sdp.cpp \
+ atom_sinf.cpp \
atoms.h \
atom_smhd.cpp \
atom_snro.cpp \
@@ -57,7 +65,6 @@
atom_stss.cpp \
atom_stsz.cpp \
atom_stts.cpp \
- atom_tag4.cpp \
atom_tfhd.cpp \
atom_tims.cpp \
atom_tkhd.cpp \
--- a/frontend/audio.c
+++ b/frontend/audio.c
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: audio.c,v 1.12 2003/02/27 11:37:26 menno Exp $
+** $Id: audio.c,v 1.13 2003/07/07 21:11:17 menno Exp $
**/
#ifdef _WIN32
@@ -63,13 +63,17 @@
return NULL;
}
-#ifdef _WIN32
- if(infile[0] == '-')
+ if(!infile)
{+#ifdef __BORLANDC__
setmode(fileno(stdout), O_BINARY);
- }
+#elif defined _MSC_VER || defined __MINGW32__
+ _setmode(_fileno(stdout), _O_BINARY);
#endif
- aufile->sndfile = fopen(infile, "wb");
+ aufile->sndfile = stdout;
+ } else {+ aufile->sndfile = fopen(infile, "wb");
+ }
if (aufile->sndfile == NULL)
{@@ -80,6 +84,7 @@
if (aufile->fileType == OUTPUT_WAV)
{write_wav_header(aufile);
+ aufile->total_samples = 0;
}
return aufile;
@@ -108,17 +113,16 @@
return 0;
}
-void close_audio_file(audio_file *aufile)
+void close_audio_file(audio_file *aufile, char *fileName)
{- if (aufile->fileType == OUTPUT_WAV)
+ if (aufile->fileType == OUTPUT_WAV && fileName)
{fseek(aufile->sndfile, 0, SEEK_SET);
write_wav_header(aufile);
+ fclose(aufile->sndfile);
}
- fclose(aufile->sndfile);
-
if (aufile) free(aufile);
}
@@ -127,8 +131,13 @@
unsigned char header[44];
unsigned char* p = header;
unsigned int bytes = (aufile->bits_per_sample + 7) / 8;
- float data_size = (float)bytes * aufile->total_samples;
+ float data_size;
unsigned long word32;
+
+ if (aufile->total_samples)
+ data_size = aufile->total_samples * bytes;
+ else
+ data_size = (float)(long)(0x7fffffff);
*p++ = 'R'; *p++ = 'I'; *p++ = 'F'; *p++ = 'F';
--- a/frontend/audio.h
+++ b/frontend/audio.h
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: audio.h,v 1.5 2002/09/08 18:14:36 menno Exp $
+** $Id: audio.h,v 1.6 2003/07/07 21:11:17 menno Exp $
**/
#ifndef AUDIO_H_INCLUDED
@@ -45,7 +45,7 @@
audio_file *open_audio_file(char *infile, int samplerate, int channels,
int outputFormat, int fileType);
int write_audio_file(audio_file *aufile, void *sample_buffer, int samples);
-void close_audio_file(audio_file *aufile);
+void close_audio_file(audio_file *aufile, char *fileName);
static int write_wav_header(audio_file *aufile);
static int write_audio_16bit(audio_file *aufile, void *sample_buffer,
unsigned int samples);
--- a/frontend/faad.dsp
+++ b/frontend/faad.dsp
@@ -50,7 +50,7 @@
# ADD BSC32 /nologo
LINK32=xilink6.exe
# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /machine:I386
-# ADD LINK32 ws2_32.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /profile /machine:I386
+# ADD LINK32 ws2_32.lib winmm.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /profile /machine:I386
!ELSEIF "$(CFG)" == "faad - Win32 Debug"
@@ -74,7 +74,7 @@
# ADD BSC32 /nologo
LINK32=xilink6.exe
# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept
-# ADD LINK32 ws2_32.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept
+# ADD LINK32 ws2_32.lib winmm.lib kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /subsystem:console /debug /machine:I386 /pdbtype:sept
!ENDIF
@@ -97,6 +97,10 @@
SOURCE=.\main.c
# End Source File
+# Begin Source File
+
+SOURCE=.\wave_out.c
+# End Source File
# End Group
# Begin Group "Header Files"
@@ -124,6 +128,10 @@
# Begin Source File
SOURCE=..\common\mp4v2\systems.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\wave_out.h
# End Source File
# Begin Source File
--- a/frontend/main.c
+++ b/frontend/main.c
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: main.c,v 1.36 2003/07/07 09:48:31 menno Exp $
+** $Id: main.c,v 1.37 2003/07/07 21:11:17 menno Exp $
**/
#ifdef _WIN32
@@ -34,14 +34,17 @@
#include <mp4.h>
#include "audio.h"
+#ifdef _WIN32
+#include "wave_out.h"
+#endif
#ifndef min
#define min(a,b) ( (a) < (b) ? (a) : (b) )
#endif
-#define MAX_CHANNELS 6 /* make this higher to support files with
-
+#define MAX_CHANNELS 8 /* make this higher to support files with
more channels */
+
/* FAAD file buffering routines */
typedef struct {long bytes_into_buffer;
@@ -172,43 +175,51 @@
void usage(void)
{- fprintf(stderr, "\nUsage:\n");
- fprintf(stderr, "%s [options] infile.aac\n", progName);
- fprintf(stderr, "Options:\n");
- fprintf(stderr, " -h Shows this help screen.\n");
- fprintf(stderr, " -i Shows info about the input file.\n");
- fprintf(stderr, " -o X Set output filename.\n");
- fprintf(stderr, " -f X Set output format. Valid values for X are:\n");
- fprintf(stderr, " 1: Microsoft WAV format (default).\n");
- fprintf(stderr, " 2: RAW PCM data.\n");
- fprintf(stderr, " -b X Set output sample format. Valid values for X are:\n");
- fprintf(stderr, " 1: 16 bit PCM data (default).\n");
- fprintf(stderr, " 2: 24 bit PCM data.\n");
- fprintf(stderr, " 3: 32 bit PCM data.\n");
- fprintf(stderr, " 4: 32 bit floating point data.\n");
- fprintf(stderr, " 5: 64 bit floating point data.\n");
- fprintf(stderr, " 6: 16 bit PCM data (dithered).\n");
- fprintf(stderr, " 7: 16 bit PCM data (dithered with LIGHT noise shaping).\n");
- fprintf(stderr, " 8: 16 bit PCM data (dithered with MEDIUM noise shaping).\n");
- fprintf(stderr, " 9: 16 bit PCM data (dithered with HEAVY noise shaping).\n");
- fprintf(stderr, " -s X Force the samplerate to X (for RAW files).\n");
- fprintf(stderr, " -l X Set object type. Supported object types:\n");
- fprintf(stderr, " 0: Main object type.\n");
- fprintf(stderr, " 1: LC (Low Complexity) object type.\n");
- fprintf(stderr, " 3: LTP (Long Term Prediction) object type.\n");
- fprintf(stderr, " 23: LD (Low Delay) object type.\n");
- fprintf(stderr, " -w Write output to stdio instead of a file.\n");
- fprintf(stderr, "Example:\n");
- fprintf(stderr, " faad infile.aac\n");
- fprintf(stderr, " faad infile.mp4\n");
- fprintf(stderr, " faad -o outfile.wav infile.aac\n");
- fprintf(stderr, " faad -w infile.aac > outfile.wav\n");
+ fprintf(stdout, "\nUsage:\n");
+ fprintf(stdout, "%s [options] infile.aac\n", progName);
+ fprintf(stdout, "Options:\n");
+ fprintf(stdout, " -h Shows this help screen.\n");
+ fprintf(stdout, " -i Shows info about the input file.\n");
+ fprintf(stdout, " -o X Set output filename.\n");
+ fprintf(stdout, " -f X Set output format. Valid values for X are:\n");
+ fprintf(stdout, " 1: Microsoft WAV format (default).\n");
+ fprintf(stdout, " 2: RAW PCM data.\n");
+ fprintf(stdout, " -b X Set output sample format. Valid values for X are:\n");
+ fprintf(stdout, " 1: 16 bit PCM data (default).\n");
+ fprintf(stdout, " 2: 24 bit PCM data.\n");
+ fprintf(stdout, " 3: 32 bit PCM data.\n");
+ fprintf(stdout, " 4: 32 bit floating point data.\n");
+ fprintf(stdout, " 5: 64 bit floating point data.\n");
+ fprintf(stdout, " 6: 16 bit PCM data (dithered).\n");
+ fprintf(stdout, " 7: 16 bit PCM data (dithered with LIGHT noise shaping).\n");
+ fprintf(stdout, " 8: 16 bit PCM data (dithered with MEDIUM noise shaping).\n");
+ fprintf(stdout, " 9: 16 bit PCM data (dithered with HEAVY noise shaping).\n");
+ fprintf(stdout, " -s X Force the samplerate to X (for RAW files).\n");
+ fprintf(stdout, " -l X Set object type. Supported object types:\n");
+ fprintf(stdout, " 0: Main object type.\n");
+ fprintf(stdout, " 1: LC (Low Complexity) object type.\n");
+ fprintf(stdout, " 3: LTP (Long Term Prediction) object type.\n");
+ fprintf(stdout, " 23: LD (Low Delay) object type.\n");
+ fprintf(stdout, " -w Write output to stdio instead of a file.\n");
+#ifdef _WIN32
+ fprintf(stdout, " -p Plays aac/mp4 files thru the soundcard using Windows audio.\n");
+ fprintf(stdout, " Playback Priority Options (ONE option ONLY may be used)\n");
+ fprintf(stdout, " -c X Set playback priority class, where X =\n");
+ fprintf(stdout, " 0: NORMAL priority.\n");
+ fprintf(stdout, " 1: HIGH priority (default).\n");
+ fprintf(stdout, " 2: REALTIME priority.\n");
+#endif
+ fprintf(stdout, "Example:\n");
+ fprintf(stdout, " faad infile.aac\n");
+ fprintf(stdout, " faad infile.mp4\n");
+ fprintf(stdout, " faad -o outfile.wav infile.aac\n");
+ fprintf(stdout, " faad -w infile.aac > outfile.wav\n");
return;
}
int decodeAACfile(char *aacfile, char *sndfile, int to_stdout,
int def_srate, int object_type, int outputFormat, int fileType,
- int infoOnly)
+ int infoOnly, int playback, unsigned int play_priority)
{int tagsize;
unsigned long samplerate;
@@ -336,7 +347,7 @@
fill_buffer(&b);
/* print AAC file info */
- fprintf(stderr, "AAC file info:\n");
+ fprintf(stderr, "%s file info:\n", aacfile);
switch (header_type)
{case 0:
@@ -356,8 +367,6 @@
{faacDecClose(hDecoder);
fclose(b.infile);
- if (!first_time)
- close_audio_file(aufile);
if (b.buffer)
free(b.buffer);
return 0;
@@ -391,28 +400,49 @@
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{- if (!to_stdout)
+ if(to_stdout)
+ sndfile = NULL;
+
+ if(playback == 1)
{+#ifdef _WIN32
+ if (Set_WIN_Params(INVALID_FILEDESC, samplerate, SAMPLE_SIZE,
+ frameInfo.channels, play_priority) < 0)
+ {+ fprintf(stderr, "ERROR: Can't access %s\n", "WAVE OUT");
+ if (b.buffer)
+ free(b.buffer);
+ faacDecClose(hDecoder);
+ fclose(b.infile);
+ return 0;
+ }
+#endif
+ } else {aufile = open_audio_file(sndfile, frameInfo.samplerate, frameInfo.channels,
outputFormat, fileType);
- } else {- aufile = open_audio_file("-", frameInfo.samplerate, frameInfo.channels,- outputFormat, fileType);
+
+ if (aufile == NULL)
+ {+ if (b.buffer)
+ free(b.buffer);
+ faacDecClose(hDecoder);
+ fclose(b.infile);
+ return 0;
+ }
}
- if (aufile == NULL)
- {- if (b.buffer)
- free(b.buffer);
- faacDecClose(hDecoder);
- fclose(b.infile);
- return 0;
- }
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{- write_audio_file(aufile, sample_buffer, frameInfo.samples);
+ if(playback == 1)
+#ifdef _WIN32
+ WIN_Play_Samples((short*)sample_buffer, frameInfo.channels*frameInfo.samples);
+#else
+ fprintf(stderr, "Playback on anything other than WIN32 not possible.\n");
+#endif
+ else
+ write_audio_file(aufile, sample_buffer, frameInfo.samples);
}
/* fill buffer */
@@ -428,8 +458,15 @@
fclose(b.infile);
- if (!first_time)
- close_audio_file(aufile);
+ if (playback == 1)
+ {+#ifdef _WIN32
+ WIN_Audio_close();
+#endif
+ } else {+ if (!first_time)
+ close_audio_file(aufile, sndfile);
+ }
if (b.buffer)
free(b.buffer);
@@ -479,7 +516,8 @@
};
int decodeMP4file(char *mp4file, char *sndfile, int to_stdout,
- int outputFormat, int fileType, int infoOnly)
+ int outputFormat, int fileType, int infoOnly,
+ int playback, unsigned int play_priority)
{int track;
unsigned int tracks;
@@ -526,7 +564,7 @@
char *file_info;
unsigned int i;
- fprintf(stderr, "MP4 file info:\n");
+ fprintf(stderr, "%s file info:\n", mp4file);
for (i = 0; i < tracks; i++)
{file_info = MP4Info(infile, i+1);
@@ -607,29 +645,45 @@
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{- if(!to_stdout)
+ if (to_stdout)
+ sndfile = NULL;
+
+ if(playback == 1)
{- aufile = open_audio_file(sndfile, frameInfo.samplerate, frameInfo.channels,
- outputFormat, fileType);
- } else {#ifdef _WIN32
- setmode(fileno(stdout), O_BINARY);
+ if (Set_WIN_Params (INVALID_FILEDESC, samplerate, SAMPLE_SIZE,
+ frameInfo.channels, play_priority) < 0)
+ {+ fprintf(stderr, "\nCan't access %s\n", "WAVE OUT");
+ faacDecClose(hDecoder);
+ MP4Close(infile);
+ return (0);
+ }
#endif
- aufile = open_audio_file("-", frameInfo.samplerate, frameInfo.channels,+ } else {+ aufile = open_audio_file(sndfile, frameInfo.samplerate, frameInfo.channels,
outputFormat, fileType);
+
+ if (aufile == NULL)
+ {+ faacDecClose(hDecoder);
+ MP4Close(infile);
+ return 0;
+ }
}
- if (aufile == NULL)
- {- faacDecClose(hDecoder);
- MP4Close(infile);
- return 0;
- }
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{- write_audio_file(aufile, sample_buffer, frameInfo.samples);
+ if (playback == 1)
+#ifdef _WIN32
+ WIN_Play_Samples((short*)sample_buffer, frameInfo.channels*frameInfo.samples);
+#else
+ fprintf(stderr, "Playback on anything other than WIN32 not possible.\n");
+#endif
+ else
+ write_audio_file(aufile, sample_buffer, frameInfo.samples);
}
if (frameInfo.error > 0)
@@ -644,8 +698,15 @@
MP4Close(infile);
- if (!first_time)
- close_audio_file(aufile);
+ if (playback == 1)
+ {+#ifdef _WIN32
+ WIN_Audio_close();
+#endif
+ } else {+ if (!first_time)
+ close_audio_file(aufile, sndfile);
+ }
return frameInfo.error;
}
@@ -661,6 +722,8 @@
int outputFormat = FAAD_FMT_16BIT;
int outfile_set = 0;
int showHelp = 0;
+ int playback = 0;
+ unsigned int play_priority = 1;
int mp4file = 0;
char *fnp;
char aacFileName[255];
@@ -700,11 +763,13 @@
{ "samplerate", 0, 0, 's' }, { "objecttype", 0, 0, 'l' }, { "info", 0, 0, 'i' },+ { "playback", 0, 0, 'p' },+ { "priority", 0, 0, 'c' }, { "stdio", 0, 0, 'w' }, { "help", 0, 0, 'h' }};
- c = getopt_long(argc, argv, "o:s:f:b:l:whi",
+ c = getopt_long(argc, argv, "o:s:f:b:l:whipc:",
long_options, &option_index);
if (c == -1)
@@ -768,7 +833,27 @@
}
}
break;
- case 'w':
+#ifdef _WIN32
+ case 'p':
+ playback = 1;
+ break;
+ case 'c':
+ if (optarg) {+ char dr[10];
+ if (sscanf(optarg, "%s", dr) < 1) {+ play_priority = 1; /* default */
+ } else {+ play_priority = atoi(dr);
+ if(play_priority > 2)
+ {+ fprintf(stderr, "Warning: priority class %s not recognised, using default\n", optarg);
+ play_priority = 1;
+ }
+ }
+ }
+ break;
+#endif
+ case 'w':
writeToStdio = 1;
break;
case 'i':
@@ -823,10 +908,11 @@
if (mp4file)
{result = decodeMP4file(aacFileName, audioFileName, writeToStdio,
- outputFormat, format, infoOnly);
+ outputFormat, format, infoOnly, playback, play_priority);
} else {result = decodeAACfile(aacFileName, audioFileName, writeToStdio,
- def_srate, object_type, outputFormat, format, infoOnly);
+ def_srate, object_type, outputFormat, format, infoOnly,
+ playback, play_priority);
}
--- /dev/null
+++ b/frontend/wave_out.c
@@ -1,0 +1,212 @@
+/* Set TABS = 4 */
+/*
+** FAAD - Freeware Advanced Audio Decoder
+** Copyright (C) 2002 M. Bakker
+**
+** This program is free software; you can redistribute it and/or modify
+** it under the terms of the GNU General Public License as published by
+** the Free Software Foundation; either version 2 of the License, or
+** (at your option) any later version.
+**
+** This program is distributed in the hope that it will be useful,
+** but WITHOUT ANY WARRANTY; without even the implied warranty of
+** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+** GNU General Public License for more details.
+**
+** You should have received a copy of the GNU General Public License
+** along with this program; if not, write to the Free Software
+** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ ********************************************************************
+
+ function: To provide playback of 16 bit PCM wave data in Win32
+ environments from decoded aac/mp4 files.
+
+ ********************************************************************/
+
+#include <string.h>
+#include <errno.h>
+#include "wave_out.h"
+
+#ifdef _WIN32
+
+#define MAXWAVESIZE 4294967040LU
+#define MAX_WAVEBLOCKS 32
+
+static CRITICAL_SECTION cs;
+static HWAVEOUT dev = NULL;
+static int ScheduledBlocks = 0;
+static int PlayedWaveHeadersCount = 0; // free index
+static WAVEHDR* PlayedWaveHeaders [MAX_WAVEBLOCKS];
+
+static int
+Box ( const char* msg )
+{+ MessageBox ( NULL, msg, " Error Message . . .", MB_OK | MB_ICONEXCLAMATION );
+ return -1;
+}
+
+
+/*
+ * This function registers already played WAVE chunks. Freeing is done by free_memory(),
+ */
+
+static void CALLBACK
+wave_callback ( HWAVE hWave, UINT uMsg, DWORD dwInstance, DWORD dwParam1, DWORD dwParam2 )
+{+ if ( uMsg == WOM_DONE ) {+ EnterCriticalSection ( &cs );
+ PlayedWaveHeaders [PlayedWaveHeadersCount++] = (WAVEHDR*) dwParam1;
+ LeaveCriticalSection ( &cs );
+ }
+}
+
+
+static void
+free_memory ( void )
+{+ WAVEHDR* wh;
+ HGLOBAL hg;
+
+ EnterCriticalSection ( &cs );
+ wh = PlayedWaveHeaders [--PlayedWaveHeadersCount];
+ ScheduledBlocks--; // decrease the number of USED blocks
+ LeaveCriticalSection ( &cs );
+
+ waveOutUnprepareHeader ( dev, wh, sizeof (WAVEHDR) );
+
+ hg = GlobalHandle ( wh -> lpData ); // Deallocate the buffer memory
+ GlobalUnlock (hg);
+ GlobalFree (hg);
+
+ hg = GlobalHandle ( wh ); // Deallocate the header memory
+ GlobalUnlock (hg);
+ GlobalFree (hg);
+}
+
+
+int
+Set_WIN_Params ( FILE_T dummyFile ,
+ long double SampleFreq,
+ unsigned int BitsPerSample,
+ unsigned int Channels,
+ unsigned int play_priority )
+{+ WAVEFORMATEX outFormat;
+ UINT deviceID = WAVE_MAPPER;
+
+ (void) dummyFile;
+
+ if ( waveOutGetNumDevs () == 0 )
+ return Box ( "No audio device present." );
+
+ outFormat.wFormatTag = WAVE_FORMAT_PCM;
+ outFormat.wBitsPerSample = BitsPerSample;
+ outFormat.nChannels = Channels;
+ outFormat.nSamplesPerSec = (unsigned long)(SampleFreq + 0.5);
+ outFormat.nBlockAlign = (outFormat.wBitsPerSample + 7) / 8 * outFormat.nChannels;
+ outFormat.nAvgBytesPerSec = outFormat.nSamplesPerSec * outFormat.nBlockAlign;
+
+ switch ( waveOutOpen ( &dev, deviceID, &outFormat, (DWORD)wave_callback, 0, CALLBACK_FUNCTION ) )
+ {+ case MMSYSERR_ALLOCATED: return Box ( "Device is already open." );
+ case MMSYSERR_BADDEVICEID: return Box ( "The specified device is out of range." );
+ case MMSYSERR_NODRIVER: return Box ( "There is no audio driver in this system." );
+ case MMSYSERR_NOMEM: return Box ( "Unable to allocate sound memory." );
+ case WAVERR_BADFORMAT: return Box ( "This audio format is not supported." );
+ case WAVERR_SYNC: return Box ( "The device is synchronous." );
+ default: return Box ( "Unknown media error." );
+ case MMSYSERR_NOERROR: break;
+ }
+
+ waveOutReset ( dev );
+ InitializeCriticalSection ( &cs );
+ switch(play_priority)
+ {+ case 0:
+ SetPriorityClass ( GetCurrentProcess (), NORMAL_PRIORITY_CLASS );
+ break;
+ case 1:
+ SetPriorityClass ( GetCurrentProcess (), HIGH_PRIORITY_CLASS );
+ break;
+ case 2:
+ SetPriorityClass ( GetCurrentProcess (), REALTIME_PRIORITY_CLASS );
+ break;
+ }
+ return 0;
+}
+
+
+int
+WIN_Play_Samples ( const void* data, size_t len )
+{+ HGLOBAL hg;
+ HGLOBAL hg2;
+ LPWAVEHDR wh;
+ void* allocptr;
+
+ do {+ while ( PlayedWaveHeadersCount > 0 ) // free used blocks ...
+ free_memory ();
+
+ if ( ScheduledBlocks < sizeof(PlayedWaveHeaders)/sizeof(*PlayedWaveHeaders) ) // wait for a free block ...
+ break;
+ Sleep (26);
+ } while (1);
+
+ if ( (hg2 = GlobalAlloc ( GMEM_MOVEABLE, len )) == NULL ) // allocate some memory for a copy of the buffer
+ return Box ( "GlobalAlloc failed." );
+
+ allocptr = GlobalLock (hg2);
+ CopyMemory ( allocptr, data, len ); // Here we can call any modification output functions we want....
+
+ if ( (hg = GlobalAlloc (GMEM_MOVEABLE | GMEM_ZEROINIT, sizeof (WAVEHDR))) == NULL ) // now make a header and WRITE IT!
+ return -1;
+
+ wh = GlobalLock (hg);
+ wh -> dwBufferLength = len;
+ wh -> lpData = allocptr;
+
+ if ( waveOutPrepareHeader ( dev, wh, sizeof (WAVEHDR)) != MMSYSERR_NOERROR ) {+ GlobalUnlock (hg);
+ GlobalFree (hg);
+ return -1;
+ }
+
+ if ( waveOutWrite ( dev, wh, sizeof (WAVEHDR)) != MMSYSERR_NOERROR ) {+ GlobalUnlock (hg);
+ GlobalFree (hg);
+ return -1;
+ }
+
+ EnterCriticalSection ( &cs );
+ ScheduledBlocks++;
+ LeaveCriticalSection ( &cs );
+
+ return len;
+}
+
+
+int
+WIN_Audio_close ( void )
+{+ if ( dev != NULL ) {+
+ while ( ScheduledBlocks > 0 ) {+ Sleep (ScheduledBlocks);
+ while ( PlayedWaveHeadersCount > 0 ) // free used blocks ...
+ free_memory ();
+ }
+
+ waveOutReset (dev); // reset the device
+ waveOutClose (dev); // close the device
+ dev = NULL;
+ }
+
+ DeleteCriticalSection ( &cs );
+ ScheduledBlocks = 0;
+ return 0;
+}
+
+#endif
+
+/* end of wave_out.c */
--- /dev/null
+++ b/frontend/wave_out.h
@@ -1,0 +1,45 @@
+/*
+** FAAD - Freeware Advanced Audio Decoder
+** Copyright (C) 2002 M. Bakker
+**
+** This program is free software; you can redistribute it and/or modify
+** it under the terms of the GNU General Public License as published by
+** the Free Software Foundation; either version 2 of the License, or
+** (at your option) any later version.
+**
+** This program is distributed in the hope that it will be useful,
+** but WITHOUT ANY WARRANTY; without even the implied warranty of
+** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+** GNU General Public License for more details.
+**
+** You should have received a copy of the GNU General Public License
+** along with this program; if not, write to the Free Software
+** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+*/
+
+// WAVE_OUT.H - Necessary stuff for WIN_AUDIO
+
+#include <stdio.h>
+#include <windows.h>
+
+#define Cdecl __cdecl
+#define __attribute__(x)
+#define sleep(__sec) Sleep ((__sec) * 1000)
+#define inline __inline
+#define restrict
+
+//// constants /////////////////////////////////////////////////////
+
+#define CD_SAMPLE_FREQ 44.1e3
+#define SAMPLE_SIZE 16
+#define SAMPLE_SIZE_STRING ""
+#define WINAUDIO_FD ((FILE_T)-128)
+#define FILE_T FILE*
+#define INVALID_FILEDESC NULL
+
+//// procedures/functions //////////////////////////////////////////
+// wave_out.c
+int Set_WIN_Params ( FILE_T dummyFile , long double SampleFreq, unsigned int BitsPerSample, unsigned int Channels, unsigned int play_priority );
+int WIN_Play_Samples ( const void* buff, size_t len );
+int WIN_Audio_close ( void );
+
--- a/include/faad.h
+++ b/include/faad.h
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: faad.h,v 1.21 2003/04/02 20:07:45 menno Exp $
+** $Id: faad.h,v 1.22 2003/07/07 21:11:18 menno Exp $
**/
#ifndef __AACDEC_H__
@@ -47,7 +47,7 @@
#define ER_LC 17
#define ER_LTP 19
#define LD 23
-#define DRM_ER_LC 27 /* special object type for DRM */
+#define DRM_ER_LC 32 /* special object type for DRM */
/* header types */
#define RAW 0
@@ -73,8 +73,21 @@
#define ERROR_RESILIENCE_CAP (1<<4)
#define FIXED_POINT_CAP (1<<5)
-/* A decode call can eat up to FAAD_MIN_STREAMSIZE octets per decoded channel,
- so at least so much octets per channel should be available in this stream */
+/* Channel definitions */
+#define FRONT_CHANNEL_CENTER (1)
+#define FRONT_CHANNEL_LEFT (2)
+#define FRONT_CHANNEL_RIGHT (3)
+#define SIDE_CHANNEL_LEFT (4)
+#define SIDE_CHANNEL_RIGHT (5)
+#define BACK_CHANNEL_LEFT (6)
+#define BACK_CHANNEL_RIGHT (7)
+#define BACK_CHANNEL_CENTER (8)
+#define LFE_CHANNEL (9)
+#define UNKNOWN_CHANNEL (10)
+
+
+/* A decode call can eat up to FAAD_MIN_STREAMSIZE bytes per decoded channel,
+ so at least so much bytes per channel should be available in this stream */
#define FAAD_MIN_STREAMSIZE 768 /* 6144 bits/channel */
@@ -114,6 +127,9 @@
unsigned char channels;
unsigned char error;
unsigned long samplerate;
+
+ /* multichannel configuration */
+ unsigned char channel_config[64];
} faacDecFrameInfo;
char* FAADAPI faacDecGetErrorMessage(unsigned char errcode);
--- a/libfaad/Makefile.am
+++ b/libfaad/Makefile.am
@@ -8,7 +8,7 @@
rvlc.c ssr.c ssr_fb.c ssr_ipqf.c common.c
libfaad_la_INCLUDES = analysis.h bits.h cfft.h cfft_tab.h common.h \
-data.h decoder.h dither.h drc.h error.h fixed.h filtbank.h \
+decoder.h dither.h drc.h error.h fixed.h filtbank.h \
huffman.h ic_predict.h iq_table.h is.h kbd_win.h lt_predict.h mdct.h mp4.h \
ms.h output.h pns.h pulse.h rvlc.h sine_win.h ssr.h ssr_fb.h ssr_ipqf.h \
ssr_win.h specrec.h syntax.h tns.h codebook/hcb.h \
--- a/libfaad/bits.h
+++ b/libfaad/bits.h
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: bits.h,v 1.17 2003/04/01 16:34:30 menno Exp $
+** $Id: bits.h,v 1.18 2003/07/07 21:11:18 menno Exp $
**/
#ifndef __BITS_H__
@@ -162,7 +162,7 @@
return (uint8_t)faad_getbits(ld, 1 DEBUGVAR(print,var,dbg));
ld->bits_left--;
- r = (ld->bufa >> ld->bits_left) & 1;
+ r = (uint8_t)((ld->bufa >> ld->bits_left) & 1);
return r;
}
--- a/libfaad/decoder.c
+++ b/libfaad/decoder.c
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: decoder.c,v 1.57 2003/06/23 15:21:19 menno Exp $
+** $Id: decoder.c,v 1.58 2003/07/07 21:11:18 menno Exp $
**/
#include "common.h"
@@ -178,12 +178,15 @@
get_adif_header(&adif, &ld);
faad_byte_align(&ld);
- hDecoder->sf_index = adif.pce.sf_index;
- hDecoder->object_type = adif.pce.object_type;
+ hDecoder->sf_index = adif.pce[0].sf_index;
+ hDecoder->object_type = adif.pce[0].object_type;
*samplerate = sample_rates[hDecoder->sf_index];
- *channels = adif.pce.channels;
+ *channels = adif.pce[0].channels;
+ memcpy(&(hDecoder->pce), &(adif.pce[0]), sizeof(program_config));
+ hDecoder->pce_set = 1;
+
bits = bit2byte(faad_get_processed_bits(&ld));
/* Check if an ADTS header is present */
@@ -403,8 +406,8 @@
#ifdef LTP_DEC
uint16_t *ltp_lag = hDecoder->ltp_lag;
#endif
+ program_config *pce = &(hDecoder->pce);
- program_config pce;
element *syntax_elements[MAX_SYNTAX_ELEMENTS];
element **elements;
int16_t *spec_data[MAX_CHANNELS];
@@ -447,7 +450,7 @@
/* decode the complete bitstream */
elements = raw_data_block(hDecoder, hInfo, ld, syntax_elements,
- spec_data, spec_coef, &pce, drc);
+ spec_data, spec_coef, pce, drc);
ch_ele = hDecoder->fr_ch_ele;
channels = hDecoder->fr_channels;
@@ -497,13 +500,10 @@
ic_stream *ics;
/* find the syntax element to which this channel belongs */
- for (i = 0; i < ch_ele; i++)
- {- if (syntax_elements[i]->channel == ch)
- ics = &(syntax_elements[i]->ics1);
- else if (syntax_elements[i]->paired_channel == ch)
- ics = &(syntax_elements[i]->ics2);
- }
+ if (syntax_elements[hDecoder->channel_element[ch]]->channel == ch)
+ ics = &(syntax_elements[hDecoder->channel_element[ch]]->ics1);
+ else if (syntax_elements[hDecoder->channel_element[ch]]->paired_channel == ch)
+ ics = &(syntax_elements[hDecoder->channel_element[ch]]->ics2);
/* inverse quantization */
inverse_quantization(spec_coef[ch], spec_data[ch], frame_len);
@@ -518,18 +518,6 @@
/* deinterleave short block grouping */
if (ics->window_sequence == EIGHT_SHORT_SEQUENCE)
quant_to_spec(ics, spec_coef[ch], frame_len);
-
-#if 0
- {- int rr;
-
- for (rr = 0; rr < frame_len; rr++)
- {-// if ((spec_coef[ch][rr] >> REAL_BITS) > (1<<12))
- printf(">>> %d\n", (spec_coef[ch][rr] >> REAL_BITS));- }
- }
-#endif
}
/* Because for ms, is and pns both channels spectral coefficients are needed
@@ -543,23 +531,20 @@
ltp_info *ltp;
/* find the syntax element to which this channel belongs */
- for (i = 0; i < ch_ele; i++)
+ if (syntax_elements[hDecoder->channel_element[ch]]->channel == ch)
{- if (syntax_elements[i]->channel == ch)
- {- ics = &(syntax_elements[i]->ics1);
- icsr = &(syntax_elements[i]->ics2);
- ltp = &(ics->ltp);
- pch = syntax_elements[i]->paired_channel;
- right_channel = 0;
- } else if (syntax_elements[i]->paired_channel == ch) {- ics = &(syntax_elements[i]->ics2);
- if (syntax_elements[i]->common_window)
- ltp = &(ics->ltp2);
- else
- ltp = &(ics->ltp);
- right_channel = 1;
- }
+ ics = &(syntax_elements[hDecoder->channel_element[ch]]->ics1);
+ icsr = &(syntax_elements[hDecoder->channel_element[ch]]->ics2);
+ ltp = &(ics->ltp);
+ pch = syntax_elements[hDecoder->channel_element[ch]]->paired_channel;
+ right_channel = 0;
+ } else if (syntax_elements[hDecoder->channel_element[ch]]->paired_channel == ch) {+ ics = &(syntax_elements[hDecoder->channel_element[ch]]->ics2);
+ if (syntax_elements[hDecoder->channel_element[ch]]->common_window)
+ ltp = &(ics->ltp2);
+ else
+ ltp = &(ics->ltp);
+ right_channel = 1;
}
/* pns decoding */
--- a/libfaad/decoder.h
+++ b/libfaad/decoder.h
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: decoder.h,v 1.22 2003/04/02 20:07:46 menno Exp $
+** $Id: decoder.h,v 1.23 2003/07/07 21:11:18 menno Exp $
**/
#ifndef __DECODER_H__
@@ -64,6 +64,18 @@
#define LD_DEC_CAP (1<<3)
#define ERROR_RESILIENCE_CAP (1<<4)
#define FIXED_POINT_CAP (1<<5)
+
+/* channel definitions */
+#define FRONT_CHANNEL_CENTER (1)
+#define FRONT_CHANNEL_LEFT (2)
+#define FRONT_CHANNEL_RIGHT (3)
+#define SIDE_CHANNEL_LEFT (4)
+#define SIDE_CHANNEL_RIGHT (5)
+#define BACK_CHANNEL_LEFT (6)
+#define BACK_CHANNEL_RIGHT (7)
+#define BACK_CHANNEL_CENTER (8)
+#define LFE_CHANNEL (9)
+#define UNKNOWN_CHANNEL (10)
int8_t* FAADAPI faacDecGetErrorMessage(uint8_t errcode);
--- a/libfaad/output.c
+++ b/libfaad/output.c
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: output.c,v 1.16 2003/04/26 13:25:13 menno Exp $
+** $Id: output.c,v 1.17 2003/07/07 21:11:18 menno Exp $
**/
#include "common.h"
@@ -56,11 +56,11 @@
double *double_sample_buffer = (double*)sample_buffer;
/* Copy output to a standard PCM buffer */
- switch (format)
+ for (ch = 0; ch < channels; ch++)
{- case FAAD_FMT_16BIT:
- for (ch = 0; ch < channels; ch++)
+ switch (format)
{+ case FAAD_FMT_16BIT:
for(i = 0; i < frame_len; i++)
{int32_t tmp;
@@ -69,11 +69,8 @@
ftemp = input[ch][i] + 0xff8000;
ftol(ftemp, short_sample_buffer[(i*channels)+ch]);
}
- }
- break;
- case FAAD_FMT_16BIT_DITHER:
- for (ch = 0; ch < channels; ch++)
- {+ break;
+ case FAAD_FMT_16BIT_DITHER:
for(i = 0; i < frame_len; i++, j++)
{double Sum = input[ch][i] * 65535.f;
@@ -87,13 +84,10 @@
val = -(1<<15);
short_sample_buffer[(i*channels)+ch] = (int16_t)val;
}
- }
- break;
- case FAAD_FMT_16BIT_L_SHAPE:
- case FAAD_FMT_16BIT_M_SHAPE:
- case FAAD_FMT_16BIT_H_SHAPE:
- for (ch = 0; ch < channels; ch++)
- {+ break;
+ case FAAD_FMT_16BIT_L_SHAPE:
+ case FAAD_FMT_16BIT_M_SHAPE:
+ case FAAD_FMT_16BIT_H_SHAPE:
for(i = 0; i < frame_len; i++, j++)
{double Sum = input[ch][i] * 65535.f;
@@ -107,11 +101,8 @@
val = -(1<<15);
short_sample_buffer[(i*channels)+ch] = (int16_t)val;
}
- }
- break;
- case FAAD_FMT_24BIT:
- for (ch = 0; ch < channels; ch++)
- {+ break;
+ case FAAD_FMT_24BIT:
for(i = 0; i < frame_len; i++)
{if (input[ch][i] > (1<<15)-1)
@@ -120,11 +111,8 @@
input[ch][i] = -(1<<15);
int_sample_buffer[(i*channels)+ch] = ROUND(input[ch][i]*(1<<8));
}
- }
- break;
- case FAAD_FMT_32BIT:
- for (ch = 0; ch < channels; ch++)
- {+ break;
+ case FAAD_FMT_32BIT:
for(i = 0; i < frame_len; i++)
{if (input[ch][i] > (1<<15)-1)
@@ -133,38 +121,20 @@
input[ch][i] = -(1<<15);
int_sample_buffer[(i*channels)+ch] = ROUND32(input[ch][i]*(1<<16));
}
- }
- break;
- case FAAD_FMT_FLOAT:
- for (ch = 0; ch < channels; ch++)
- {+ break;
+ case FAAD_FMT_FLOAT:
for(i = 0; i < frame_len; i++)
{-/*
- if (input[ch][i] > (1<<15)-1)
- input[ch][i] = (1<<15)-1;
- else if (input[ch][i] < -(1<<15))
- input[ch][i] = -(1<<15);
-*/
float_sample_buffer[(i*channels)+ch] = input[ch][i]*FLOAT_SCALE;
}
- }
- break;
- case FAAD_FMT_DOUBLE:
- for (ch = 0; ch < channels; ch++)
- {+ break;
+ case FAAD_FMT_DOUBLE:
for(i = 0; i < frame_len; i++)
{-/*
- if (input[ch][i] > (1<<15)-1)
- input[ch][i] = (1<<15)-1;
- else if (input[ch][i] < -(1<<15))
- input[ch][i] = -(1<<15);
-*/
double_sample_buffer[(i*channels)+ch] = (double)input[ch][i]*FLOAT_SCALE;
}
+ break;
}
- break;
}
return sample_buffer;
--- a/libfaad/structs.h
+++ b/libfaad/structs.h
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: structs.h,v 1.5 2003/02/09 20:42:49 menno Exp $
+** $Id: structs.h,v 1.6 2003/07/07 21:11:18 menno Exp $
**/
#ifndef __STRUCTS_H__
@@ -155,7 +155,8 @@
uint8_t num_program_config_elements;
uint32_t adif_buffer_fullness;
- program_config pce;
+ /* maximum of 16 PCEs */
+ program_config pce[16];
} adif_header;
typedef struct
@@ -314,6 +315,9 @@
uint8_t channels;
uint8_t error;
uint32_t samplerate;
+
+ /* multichannel configuration */
+ uint8_t channel_config[64];
} faacDecFrameInfo;
typedef struct
@@ -364,6 +368,11 @@
real_t *pow2_table;
#endif
#endif
+
+ /* Program Config Element */
+ uint8_t pce_set;
+ program_config pce;
+ uint8_t channel_element[MAX_CHANNELS];
/* Configuration data */
faacDecConfiguration config;
--- a/libfaad/syntax.c
+++ b/libfaad/syntax.c
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: syntax.c,v 1.45 2003/06/23 15:21:20 menno Exp $
+** $Id: syntax.c,v 1.46 2003/07/07 21:11:18 menno Exp $
**/
/*
@@ -65,8 +65,10 @@
program_config_element(&pce, ld);
mp4ASC->channelsConfiguration = pce.channels;
+ /*
if (pce.num_valid_cc_elements)
return -3;
+ */
}
#ifdef ERROR_RESILIENCE
@@ -154,12 +156,16 @@
if ((pce->front_element_is_cpe[i] = faad_get1bit(ld
DEBUGVAR(1,26,"program_config_element(): front_element_is_cpe"))) & 1)
{+// printf("CPE ");pce->channels += 2;
} else {+// printf("SCE ");pce->channels++;
}
pce->front_element_tag_select[i] = (uint8_t)faad_getbits(ld, 4
DEBUGVAR(1,27,"program_config_element(): front_element_tag_select"));
+
+// printf("FRONT %d: %d\n", i, pce->front_element_tag_select[i]);}
for (i = 0; i < pce->num_side_channel_elements; i++)
@@ -167,12 +173,16 @@
if ((pce->side_element_is_cpe[i] = faad_get1bit(ld
DEBUGVAR(1,28,"program_config_element(): side_element_is_cpe"))) & 1)
{+// printf("CPE ");pce->channels += 2;
} else {+// printf("SCE ");pce->channels++;
}
pce->side_element_tag_select[i] = (uint8_t)faad_getbits(ld, 4
DEBUGVAR(1,29,"program_config_element(): side_element_tag_select"));
+
+// printf("SIDE %d: %d\n", i, pce->side_element_tag_select[i]);}
for (i = 0; i < pce->num_back_channel_elements; i++)
@@ -180,12 +190,16 @@
if ((pce->back_element_is_cpe[i] = faad_get1bit(ld
DEBUGVAR(1,30,"program_config_element(): back_element_is_cpe"))) & 1)
{+// printf("CPE ");pce->channels += 2;
} else {+// printf("SCE ");pce->channels++;
}
pce->back_element_tag_select[i] = (uint8_t)faad_getbits(ld, 4
DEBUGVAR(1,31,"program_config_element(): back_element_tag_select"));
+
+// printf("BACK %d: %d\n", i, pce->back_element_tag_select[i]);}
for (i = 0; i < pce->num_lfe_channel_elements; i++)
@@ -193,17 +207,18 @@
pce->channels++;
pce->lfe_element_tag_select[i] = (uint8_t)faad_getbits(ld, 4
DEBUGVAR(1,32,"program_config_element(): lfe_element_tag_select"));
+
+// printf("LFE %d: %d\n", i, pce->lfe_element_tag_select[i]);}
for (i = 0; i < pce->num_assoc_data_elements; i++)
+ {pce->assoc_data_element_tag_select[i] = (uint8_t)faad_getbits(ld, 4
- DEBUGVAR(1,33,"program_config_element(): assoc_data_element_tag_select"));
+ DEBUGVAR(1,33,"program_config_element(): assoc_data_element_tag_select"));
+ }
for (i = 0; i < pce->num_valid_cc_elements; i++)
{- /* have to count these as channels too?? (1 or 2) */
- pce->channels += 2;
-
pce->cc_element_is_ind_sw[i] = faad_get1bit(ld
DEBUGVAR(1,34,"program_config_element(): cc_element_is_ind_sw"));
pce->valid_cc_element_tag_select[i] = (uint8_t)faad_getbits(ld, 4
@@ -256,6 +271,11 @@
hInfo->error = single_lfe_channel_element(hDecoder, ele,
ld, spec_data[channels]);
+ if (id_syn_ele == ID_SCE)
+ hDecoder->channel_element[channels] = hDecoder->fr_ch_ele;
+ else /* LFE */
+ hDecoder->channel_element[channels] = hDecoder->fr_ch_ele;
+
hDecoder->fr_channels++;
hDecoder->fr_ch_ele++;
@@ -295,6 +315,9 @@
hInfo->error = channel_pair_element(hDecoder, ele,
ld, spec_data[channels], spec_data[channels+1]);
+ hDecoder->channel_element[channels] = hDecoder->fr_ch_ele;
+ hDecoder->channel_element[channels+1] = hDecoder->fr_ch_ele;
+
hDecoder->fr_channels += 2;
hDecoder->fr_ch_ele++;
@@ -334,9 +357,11 @@
if (hInfo->error > 0)
return elements;
break;
- case ID_CCE: /* not implemented yet */
- hInfo->error = 6;
- return elements;
+ case ID_CCE: /* not implemented yet, but skip the bits */
+ hInfo->error = coupling_channel_element(hDecoder, ld);
+ if (hInfo->error > 0)
+ return elements;
+ break;
case ID_DSE:
data_stream_element(ld);
break;
@@ -343,6 +368,7 @@
case ID_PCE:
if ((hInfo->error = program_config_element(pce, ld)) > 0)
return elements;
+ hDecoder->pce_set = 1;
break;
case ID_FIL:
if ((hInfo->error = fill_element(ld, drc)) > 0)
@@ -775,6 +801,97 @@
return 0;
}
+/* Table 4.4.8: Currently just for skipping the bits... */
+static uint8_t coupling_channel_element(faacDecHandle hDecoder, bitfile *ld)
+{+ uint8_t c, result = 0;
+ uint8_t ind_sw_cce_flag = 0;
+ uint8_t num_gain_element_lists = 0;
+ uint8_t num_coupled_elements = 0;
+
+ element el_empty;
+ ic_stream ics_empty;
+ int16_t sh_data[1024];
+
+ memset(&el_empty, 0, sizeof(element));
+ memset(&ics_empty, 0, sizeof(ic_stream));
+
+ faad_getbits(ld, LEN_TAG
+ DEBUGVAR(1,900,"coupling_channel_element(): element_instance_tag"));
+
+ ind_sw_cce_flag = faad_get1bit(ld
+ DEBUGVAR(1,901,"coupling_channel_element(): ind_sw_cce_flag"));
+ num_coupled_elements = faad_getbits(ld, 3
+ DEBUGVAR(1,902,"coupling_channel_element(): num_coupled_elements"));
+
+ for (c = 0; c < num_coupled_elements + 1; c++)
+ {+ uint8_t cc_target_is_cpe, cc_target_tag_select;
+
+ num_gain_element_lists++;
+
+ cc_target_is_cpe = faad_get1bit(ld
+ DEBUGVAR(1,903,"coupling_channel_element(): cc_target_is_cpe"));
+ cc_target_tag_select = faad_getbits(ld, 4
+ DEBUGVAR(1,904,"coupling_channel_element(): cc_target_tag_select"));
+
+ if (cc_target_is_cpe)
+ {+ uint8_t cc_l = faad_get1bit(ld
+ DEBUGVAR(1,905,"coupling_channel_element(): cc_l"));
+ uint8_t cc_r = faad_get1bit(ld
+ DEBUGVAR(1,906,"coupling_channel_element(): cc_r"));
+
+ if (cc_l && cc_r)
+ num_gain_element_lists++;
+ }
+ }
+
+ faad_get1bit(ld
+ DEBUGVAR(1,907,"coupling_channel_element(): cc_domain"));
+ faad_get1bit(ld
+ DEBUGVAR(1,908,"coupling_channel_element(): gain_element_sign"));
+ faad_getbits(ld, 2
+ DEBUGVAR(1,909,"coupling_channel_element(): gain_element_scale"));
+
+ if ((result = individual_channel_stream(hDecoder, &el_empty, ld, &ics_empty,
+ 0, sh_data)) > 0)
+ {+ return result;
+ }
+
+ for (c = 1; c < num_gain_element_lists; c++)
+ {+ uint8_t cge;
+
+ if (ind_sw_cce_flag)
+ {+ cge = 1;
+ } else {+ cge = faad_get1bit(ld
+ DEBUGVAR(1,910,"coupling_channel_element(): common_gain_element_present"));
+ }
+
+ if (cge)
+ {+ huffman_scale_factor(ld);
+ } else {+ uint8_t g, sfb;
+
+ for (g = 0; g < ics_empty.num_window_groups; g++)
+ {+ for (sfb = 0; sfb < ics_empty.max_sfb; sfb++)
+ {+ if (ics_empty.sfb_cb[g][sfb] != ZERO_HCB)
+ huffman_scale_factor(ld);
+ }
+ }
+ }
+ }
+
+ return 0;
+}
+
/* Table 4.4.10 */
static uint16_t data_stream_element(bitfile *ld)
{@@ -1624,17 +1741,17 @@
adif->num_program_config_elements = (uint8_t)faad_getbits(ld, 4
DEBUGVAR(1,116,"get_adif_header(): num_program_config_elements"));
- if(adif->bitstream_type == 0)
- {- adif->adif_buffer_fullness = faad_getbits(ld, 20
- DEBUGVAR(1,117,"get_adif_header(): adif_buffer_fullness"));
- } else {- adif->adif_buffer_fullness = 0;
- }
-
for (i = 0; i < adif->num_program_config_elements + 1; i++)
{- program_config_element(&adif->pce, ld);
+ if(adif->bitstream_type == 0)
+ {+ adif->adif_buffer_fullness = faad_getbits(ld, 20
+ DEBUGVAR(1,117,"get_adif_header(): adif_buffer_fullness"));
+ } else {+ adif->adif_buffer_fullness = 0;
+ }
+
+ program_config_element(&adif->pce[i], ld);
}
}
--- a/libfaad/syntax.h
+++ b/libfaad/syntax.h
@@ -16,7 +16,7 @@
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
-** $Id: syntax.h,v 1.28 2003/06/23 15:21:20 menno Exp $
+** $Id: syntax.h,v 1.30 2003/07/08 13:45:45 menno Exp $
**/
#ifndef __SYNTAX_H__
@@ -99,6 +99,7 @@
static uint8_t channel_pair_element(faacDecHandle hDecoder, element *cpe,
bitfile *ld, int16_t *spec_data1,
int16_t *spec_data2);
+static uint8_t coupling_channel_element(faacDecHandle hDecoder, bitfile *ld);
static uint16_t data_stream_element(bitfile *ld);
static uint8_t program_config_element(program_config *pce, bitfile *ld);
static uint8_t fill_element(bitfile *ld, drc_info *drc);