[xiph-commits] r15310 - in trunk/vorbis-tools: . oggenc oggenc/man
msmith at svn.xiph.org
msmith at svn.xiph.org
Thu Sep 11 23:06:14 PDT 2008
Author: msmith
Date: 2008-09-11 23:06:14 -0700 (Thu, 11 Sep 2008)
New Revision: 15310
Modified:
trunk/vorbis-tools/CHANGES
trunk/vorbis-tools/oggenc/audio.c
trunk/vorbis-tools/oggenc/audio.h
trunk/vorbis-tools/oggenc/man/oggenc.1
trunk/vorbis-tools/oggenc/oggenc.c
Log:
Revert r15308: it adds network headers (!!?), incorrect documentation, c++
comments, and is full of bugs, for a feature of dubious utility.
Modified: trunk/vorbis-tools/CHANGES
===================================================================
--- trunk/vorbis-tools/CHANGES 2008-09-12 03:50:55 UTC (rev 15309)
+++ trunk/vorbis-tools/CHANGES 2008-09-12 06:06:14 UTC (rev 15310)
@@ -18,7 +18,6 @@
* oggenc: support for 32 bit Wave files (#1326)
* oggenc: --ignorelength; support for Wave files > 4 GB (#1326)
* oggenc: split help text into manageable chunks to help translators (#1385)
- * oggenc: support for linear 16 bit u-Law/.au (#1105)
* ogginfo: support for information in Kate streams (#1360)
* vcut: 64 bit fixes (#1366)
* vorbiscomment: correct memory allocation (#472)
Modified: trunk/vorbis-tools/oggenc/audio.c
===================================================================
--- trunk/vorbis-tools/oggenc/audio.c 2008-09-12 03:50:55 UTC (rev 15309)
+++ trunk/vorbis-tools/oggenc/audio.c 2008-09-12 06:06:14 UTC (rev 15310)
@@ -17,7 +17,6 @@
#include <string.h>
#include <sys/types.h>
#include <math.h>
-#include <arpa/inet.h>
#include "audio.h"
#include "platform.h"
@@ -54,7 +53,6 @@
{flac_id, 4, flac_open, flac_close, "flac", N_("FLAC file reader")},
{oggflac_id, 33, flac_open, flac_close, "ogg", N_("Ogg FLAC file reader")},
#endif
- {au_id, sizeof(auhdr),au_open,au_close, "au",N_("AU file reader") },
{NULL, 0, NULL, NULL, NULL, NULL}
};
@@ -326,7 +324,7 @@
(format.samplesize == 16 || format.samplesize == 8))
{
/* From here on, this is very similar to the wav code. Oh well. */
-
+
opt->rate = format.rate;
opt->channels = format.channels;
opt->read_samples = wav_read; /* Similar enough, so we use the same */
@@ -362,7 +360,7 @@
int wav_id(unsigned char *buf, int len)
{
unsigned int flen;
-
+
if(len<12) return 0; /* Something screwed up */
if(memcmp(buf, "RIFF", 4))
@@ -406,7 +404,7 @@
if(len < 16)
{
- fprintf(stderr, _("Warning: Unrecognised format chunk in Wave header\n"));
+ fprintf(stderr, _("Warning: Unrecognised format chunk in WAV header\n"));
return 0; /* Weird format chunk */
}
@@ -419,12 +417,12 @@
*/
if(len!=16 && len!=18 && len!=40)
fprintf(stderr,
- _("Warning: INVALID format chunk in Wave header.\n"
+ _("Warning: INVALID format chunk in wav header.\n"
" Trying to read anyway (may not work)...\n"));
if(fread(buf,1,len,in) < len)
{
- fprintf(stderr, _("Warning: Unexpected EOF in reading Wave header\n"));
+ fprintf(stderr, _("Warning: Unexpected EOF in reading WAV header\n"));
return 0;
}
@@ -457,7 +455,7 @@
else
{
fprintf(stderr,
- _("ERROR: Wave file is unsupported type (must be standard PCM\n"
+ _("ERROR: Wav file is unsupported type (must be standard PCM\n"
" or type 3 floating point PCM)\n"));
return 0;
}
@@ -825,96 +823,6 @@
free(d);
}
-
-// ------------- Basic AU support --------------
-
-// 1=OK
-// 0=KO
-int au_id(unsigned char *buf, int len)
-{
- if (len < sizeof(auhdr))
- return 0;
- if (memcmp(buf,".snd",4))
- return 0;
- return 1;
-}
-
-typedef union {
- int16_t signed16;
- uint16_t unsigned16;
-} sixteen_bit;
-
-long au_read(void *in, float **buffer, int samples)
-{
- wavfile *wav = (wavfile *)in;
- unsigned char *buf = alloca(samples*wav->channels*2);
- long bytes_read = fread(buf, 1, samples*2*wav->channels, wav->f);
- int i,j;
- long realsamples;
- uint16_t *sample;
- sixteen_bit converter;
-
- realsamples = bytes_read / (2 * wav->channels);
-
- sample = (uint16_t *)buf;
-
- for(i = 0; i < realsamples; i++)
- {
- for(j=0; j < wav->channels; j++)
- {
- converter.unsigned16 = ntohs(*sample++);
- buffer[j][i] = converter.signed16 / 32768.0f;
- }
- }
-
- return realsamples;
-}
-
-int au_open(FILE *in,
- oe_enc_opt *opt,
- unsigned char *buf, int buflen)
-{
- auhdr auHdr;
- wavfile *wav = malloc(sizeof(wavfile));
- int dataOffset;
-
- if (NULL == wav) {
- fprintf(stderr, _("Out of memory opening AU driver\n"));
- }
-
- rewind(in);
- fread(&auHdr,1,sizeof(auHdr),in);
-
- if (SND_FORMAT_LINEAR_16 != ntohl(auHdr.formatCode)) {
- fprintf(stderr,_("At this moment, only linear 16 bit .au files are supported\n"));
- return 0;
- }
- dataOffset = ntohl(auHdr.dataOffset);
- opt->rate = ntohl(auHdr.samplingRate);
- opt->channels = ntohl(auHdr.numberChannels);
- opt->read_samples = au_read;
- opt->total_samples_per_channel = ntohl(auHdr.dataBytes) / (2 * opt->channels);
- opt->readdata = wav;
-
- wav->channels = opt->channels;
- wav->samplesize = 4;
- wav->totalsamples = opt->total_samples_per_channel;
- wav->samplesread = 0;
- wav->f = in;
-
- // ignore: short bigendian;
-
- fseek(in,dataOffset,SEEK_SET);
- return 1;
-}
-
-void au_close(void *info)
-{
- free(info);
-}
-/* End Basic AU Support */
-
-
typedef struct {
audio_read_func real_reader;
void *real_readdata;
@@ -938,10 +846,10 @@
downmix *d = calloc(1, sizeof(downmix));
if(opt->channels != 2) {
- fprintf(stderr, _("Internal error! Please report this bug.\n"));
+ fprintf(stderr, "Internal error! Please report this bug.\n");
return;
}
-
+
d->bufs = malloc(2 * sizeof(float *));
d->bufs[0] = malloc(4096 * sizeof(float));
d->bufs[1] = malloc(4096 * sizeof(float));
@@ -966,3 +874,4 @@
free(d->bufs);
free(d);
}
+
Modified: trunk/vorbis-tools/oggenc/audio.h
===================================================================
--- trunk/vorbis-tools/oggenc/audio.h 2008-09-12 03:50:55 UTC (rev 15309)
+++ trunk/vorbis-tools/oggenc/audio.h 2008-09-12 06:06:14 UTC (rev 15310)
@@ -5,8 +5,6 @@
#include "encode.h"
#include <stdio.h>
-#include <arpa/inet.h>
-
int setup_resample(oe_enc_opt *opt);
void clear_resample(oe_enc_opt *opt);
void setup_downmix(oe_enc_opt *opt);
@@ -69,22 +67,5 @@
long wav_ieee_read(void *, float **buffer, int samples);
long raw_read_stereo(void *, float **buffer, int samples);
-/* AU code */
-int au_open(FILE *in, oe_enc_opt *opt, unsigned char *buf, int buflen);
-int au_id(unsigned char *buf, int len);
-void au_close(void *);
-
-typedef struct {
- char magic[4];
- uint32_t dataOffset;
- uint32_t dataBytes;
- uint32_t formatCode;
- uint32_t samplingRate;
- uint32_t numberChannels;
- char optional[4];
-} auhdr;
-
-#define SND_FORMAT_LINEAR_16 3
-/* End AU code */
-
#endif /* __AUDIO_H */
+
Modified: trunk/vorbis-tools/oggenc/man/oggenc.1
===================================================================
--- trunk/vorbis-tools/oggenc/man/oggenc.1 2008-09-12 03:50:55 UTC (rev 15309)
+++ trunk/vorbis-tools/oggenc/man/oggenc.1 2008-09-12 06:06:14 UTC (rev 15310)
@@ -82,8 +82,8 @@
.SH DESCRIPTION
.B oggenc
-reads audio data in either raw, Wave, AIFF or 16 bit u-Law (.au) format and
-encodes it into an Ogg Vorbis stream.
+reads audio data in either raw, Wave, or AIFF format and encodes it into an
+Ogg Vorbis stream.
.B oggenc
may also read audio data from FLAC and Ogg FLAC files depending upon compile-time options. If the input file "-" is specified, audio data is
read from
Modified: trunk/vorbis-tools/oggenc/oggenc.c
===================================================================
--- trunk/vorbis-tools/oggenc/oggenc.c 2008-09-12 03:50:55 UTC (rev 15309)
+++ trunk/vorbis-tools/oggenc/oggenc.c 2008-09-12 06:06:14 UTC (rev 15310)
@@ -494,11 +494,11 @@
"\n"));
fprintf(stdout, _(
"INPUT FILES:\n"
- " OggEnc input files must currently be 24, 16, or 8 bit PCM Wave, 16 bit u-Law (.au), AIFF, or AIFF/C\n"
- " files, 32 bit IEEE floating point Wave, and optionally FLAC or Ogg FLAC. Files\n"
+ " OggEnc input files must currently be 24, 16, or 8 bit PCM WAV, AIFF, or AIFF/C\n"
+ " files, 32 bit IEEE floating point WAV, and optionally FLAC or Ogg FLAC. Files\n"
" may be mono or stereo (or more channels) and any sample rate.\n"
" Alternatively, the --raw option may be used to use a raw PCM data file, which\n"
- " must be 16 bit stereo little-endian PCM ('headerless wave'), unless additional\n"
+ " must be 16 bit stereo little-endian PCM ('headerless wav'), unless additional\n"
" parameters for raw mode are specified.\n"
" You can specify taking the file from stdin by using - as the input filename.\n"
" In this mode, output is to stdout unless an output filename is specified\n"
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