--- /dev/null
+/* OggEnc
+ **
+ ** This program is distributed under the GNU General Public License, version 2.
+ ** A copy of this license is included with this source.
+ **
+ ** Copyright 2000-2002, Michael Smith <msmith@xiph.org>
+ ** 2010, Monty <monty@xiph.org>
+ **
+ ** AIFF/AIFC support from OggSquish, (c) 1994-1996 Monty <xiphmont@xiph.org>
+ **/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <sys/types.h>
+#include <math.h>
+
+#include "audio.h"
+#include "platform.h"
+#include "i18n.h"
+#include "resample.h"
+
+#ifdef HAVE_LIBFLAC
+#include "flac.h"
+#endif
+
+/* Macros to read header data */
+#define READ_U32_LE(buf) \
+ (((buf)[3]<<24)|((buf)[2]<<16)|((buf)[1]<<8)|((buf)[0]&0xff))
+
+#define READ_U16_LE(buf) \
+ (((buf)[1]<<8)|((buf)[0]&0xff))
+
+#define READ_U32_BE(buf) \
+ (((buf)[0]<<24)|((buf)[1]<<16)|((buf)[2]<<8)|((buf)[3]&0xff))
+
+#define READ_U16_BE(buf) \
+ (((buf)[0]<<8)|((buf)[1]&0xff))
+
+/* Define the supported formats here */
+input_format formats[] = {
+ {wav_id, 12, wav_open, wav_close, "wav", N_("WAV file reader")},
+ {aiff_id, 12, aiff_open, wav_close, "aiff", N_("AIFF/AIFC file reader")},
+#ifdef HAVE_LIBFLAC
+ {flac_id, 4, flac_open, flac_close, "flac", N_("FLAC file reader")},
+ {oggflac_id, 32, flac_open, flac_close, "ogg", N_("Ogg FLAC file reader")},
+#endif
+ {NULL, 0, NULL, NULL, NULL, NULL}
+};
+
+input_format *open_audio_file(FILE *in, oe_enc_opt *opt)
+{
+ int j=0;
+ unsigned char *buf=NULL;
+ int buf_size=0, buf_filled=0;
+ int size,ret;
+
+ while(formats[j].id_func)
+ {
+ size = formats[j].id_data_len;
+ if(size >= buf_size)
+ {
+ buf = realloc(buf, size);
+ buf_size = size;
+ }
+
+ if(size > buf_filled)
+ {
+ ret = fread(buf+buf_filled, 1, buf_size-buf_filled, in);
+ buf_filled += ret;
+
+ if(buf_filled < size)
+ { /* File truncated */
+ j++;
+ continue;
+ }
+ }
+
+ if(formats[j].id_func(buf, buf_filled))
+ {
+ /* ok, we now have something that can handle the file */
+ if(formats[j].open_func(in, opt, buf, buf_filled)) {
+ free(buf);
+ return &formats[j];
+ }
+ }
+ j++;
+ }
+
+ free(buf);
+
+ return NULL;
+}
+
+static int seek_forward(FILE *in, unsigned int length)
+{
+ if(fseek(in, length, SEEK_CUR))
+ {
+ /* Failed. Do it the hard way. */
+ unsigned char buf[1024];
+ unsigned int seek_needed = length;
+ int seeked;
+ while(seek_needed > 0)
+ {
+ seeked = fread(buf, 1, seek_needed>1024?1024:seek_needed, in);
+ if(!seeked)
+ return 0; /* Couldn't read more, can't read file */
+ else
+ seek_needed -= seeked;
+ }
+ }
+ return 1;
+}
+
+
+static int find_wav_chunk(FILE *in, char *type, unsigned int *len)
+{
+ unsigned char buf[8];
+
+ while(1)
+ {
+ if(fread(buf,1,8,in) < 8) /* Suck down a chunk specifier */
+ {
+ fprintf(stderr, _("Warning: Unexpected EOF in reading WAV header\n"));
+ return 0; /* EOF before reaching the appropriate chunk */
+ }
+
+ if(memcmp(buf, type, 4))
+ {
+ *len = READ_U32_LE(buf+4);
+ if(!seek_forward(in, *len))
+ return 0;
+
+ buf[4] = 0;
+ fprintf(stderr, _("Skipping chunk of type \"%s\", length %d\n"), buf, *len);
+ }
+ else
+ {
+ *len = READ_U32_LE(buf+4);
+ return 1;
+ }
+ }
+}
+
+static int find_aiff_chunk(FILE *in, char *type, unsigned int *len)
+{
+ unsigned char buf[8];
+ int restarted = 0;
+
+ while(1)
+ {
+ if(fread(buf,1,8,in) <8)
+ {
+ if(!restarted) {
+ /* Handle out of order chunks by seeking back to the start
+ * to retry */
+ restarted = 1;
+ fseek(in, 12, SEEK_SET);
+ continue;
+ }
+ fprintf(stderr, _("Warning: Unexpected EOF in AIFF chunk\n"));
+ return 0;
+ }
+
+ *len = READ_U32_BE(buf+4);
+
+ if(memcmp(buf,type,4))
+ {
+ if((*len) & 0x1)
+ (*len)++;
+
+ if(!seek_forward(in, *len))
+ return 0;
+ }
+ else
+ return 1;
+ }
+}
+
+
+
+double read_IEEE80(unsigned char *buf)
+{
+ int s=buf[0]&0xff;
+ int e=((buf[0]&0x7f)<<8)|(buf[1]&0xff);
+ double f=((unsigned long)(buf[2]&0xff)<<24)|
+ ((buf[3]&0xff)<<16)|
+ ((buf[4]&0xff)<<8) |
+ (buf[5]&0xff);
+
+ if(e==32767)
+ {
+ if(buf[2]&0x80)
+ return HUGE_VAL; /* Really NaN, but this won't happen in reality */
+ else
+ {
+ if(s)
+ return -HUGE_VAL;
+ else
+ return HUGE_VAL;
+ }
+ }
+
+ f=ldexp(f,32);
+ f+= ((buf[6]&0xff)<<24)|
+ ((buf[7]&0xff)<<16)|
+ ((buf[8]&0xff)<<8) |
+ (buf[9]&0xff);
+
+ return ldexp(f, e-16446);
+}
+
+/* AIFF/AIFC support adapted from the old OggSQUISH application */
+int aiff_id(unsigned char *buf, int len)
+{
+ if(len<12) return 0; /* Truncated file, probably */
+
+ if(memcmp(buf, "FORM", 4))
+ return 0;
+
+ if(memcmp(buf+8, "AIF",3))
+ return 0;
+
+ if(buf[11]!='C' && buf[11]!='F')
+ return 0;
+
+ return 1;
+}
+
+static int aiff_permute_matrix[6][6] =
+{
+ {0}, /* 1.0 mono */
+ {0,1}, /* 2.0 stereo */
+ {0,2,1}, /* 3.0 channel ('wide') stereo */
+ {0,1,2,3}, /* 4.0 discrete quadraphonic (WARN) */
+ {0,2,1,3,4}, /* 5.0 surround (WARN) */
+ {0,1,2,3,4,5}, /* 5.1 surround (WARN)*/
+};
+
+
+int aiff_open(FILE *in, oe_enc_opt *opt, unsigned char *buf, int buflen)
+{
+ int aifc; /* AIFC or AIFF? */
+ unsigned int len;
+ unsigned char *buffer;
+ unsigned char buf2[8];
+ aiff_fmt format;
+ aifffile *aiff = malloc(sizeof(aifffile));
+ int i;
+
+ if(buf[11]=='C')
+ aifc=1;
+ else
+ aifc=0;
+
+ if(!find_aiff_chunk(in, "COMM", &len))
+ {
+ fprintf(stderr, _("Warning: No common chunk found in AIFF file\n"));
+ return 0; /* EOF before COMM chunk */
+ }
+
+ if(len < 18)
+ {
+ fprintf(stderr, _("Warning: Truncated common chunk in AIFF header\n"));
+ return 0; /* Weird common chunk */
+ }
+
+ buffer = alloca(len);
+
+ if(fread(buffer,1,len,in) < len)
+ {
+ fprintf(stderr, _("Warning: Unexpected EOF in reading AIFF header\n"));
+ return 0;
+ }
+
+ format.channels = READ_U16_BE(buffer);
+ format.totalframes = READ_U32_BE(buffer+2);
+ format.samplesize = READ_U16_BE(buffer+6);
+ format.rate = (int)read_IEEE80(buffer+8);
+
+ aiff->bigendian = 1;
+
+ if(aifc)
+ {
+ if(len < 22)
+ {
+ fprintf(stderr, _("Warning: AIFF-C header truncated.\n"));
+ return 0;
+ }
+
+ if(!memcmp(buffer+18, "NONE", 4))
+ {
+ aiff->bigendian = 1;
+ }
+ else if(!memcmp(buffer+18, "sowt", 4))
+ {
+ aiff->bigendian = 0;
+ }
+ else
+ {
+ fprintf(stderr, _("Warning: Can't handle compressed AIFF-C (%c%c%c%c)\n"), *(buffer+18), *(buffer+19), *(buffer+20), *(buffer+21));
+ return 0; /* Compressed. Can't handle */
+ }
+ }
+
+ if(!find_aiff_chunk(in, "SSND", &len))
+ {
+ fprintf(stderr, _("Warning: No SSND chunk found in AIFF file\n"));
+ return 0; /* No SSND chunk -> no actual audio */
+ }
+
+ if(len < 8)
+ {
+ fprintf(stderr, _("Warning: Corrupted SSND chunk in AIFF header\n"));
+ return 0;
+ }
+
+ if(fread(buf2,1,8, in) < 8)
+ {
+ fprintf(stderr, _("Warning: Unexpected EOF reading AIFF header\n"));
+ return 0;
+ }
+
+ format.offset = READ_U32_BE(buf2);
+ format.blocksize = READ_U32_BE(buf2+4);
+
+ if( format.blocksize == 0 &&
+ (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 */
+ opt->total_samples_per_channel = format.totalframes;
+
+ aiff->f = in;
+ aiff->samplesread = 0;
+ aiff->channels = format.channels;
+ aiff->samplesize = format.samplesize;
+ aiff->totalsamples = format.totalframes;
+
+ if(aiff->channels>3)
+ fprintf(stderr,"WARNING: AIFF[-C] files with greater than three channels use\n"
+ "speaker locations incompatable with Vorbis suppound definitions.\n"
+ "Not performaing channel location mapping.\n");
+
+ opt->readdata = (void *)aiff;
+
+ aiff->channel_permute = malloc(aiff->channels * sizeof(int));
+ if (aiff->channels <= 6)
+ /* Where we know the mappings, use them. */
+ memcpy(aiff->channel_permute, aiff_permute_matrix[aiff->channels-1],
+ sizeof(int) * aiff->channels);
+ else
+ /* Use a default 1-1 mapping */
+ for (i=0; i < aiff->channels; i++)
+ aiff->channel_permute[i] = i;
+
+ seek_forward(in, format.offset); /* Swallow some data */
+ return 1;
+ }
+ else
+ {
+ fprintf(stderr,
+ _("Warning: OggEnc does not support this type of AIFF/AIFC file\n"
+ " Must be 8 or 16 bit PCM.\n"));
+ return 0;
+ }
+}
+
+
+int wav_id(unsigned char *buf, int len)
+{
+ unsigned int flen;
+
+ if(len<12) return 0; /* Something screwed up */
+
+ if(memcmp(buf, "RIFF", 4))
+ return 0; /* Not wave */
+
+ flen = READ_U32_LE(buf+4); /* We don't use this */
+
+ if(memcmp(buf+8, "WAVE",4))
+ return 0; /* RIFF, but not wave */
+
+ return 1;
+}
+
+static int wav_permute_matrix[8][8] =
+{
+ {0}, /* 1.0 mono */
+ {0,1}, /* 2.0 stereo */
+ {0,2,1}, /* 3.0 channel ('wide') stereo */
+ {0,1,2,3}, /* 4.0 discrete quadraphonic */
+ {0,2,1,3,4}, /* 5.0 surround */
+ {0,2,1,4,5,3}, /* 5.1 surround */
+ {0,2,1,4,5,6,3}, /* 6.1 surround */
+ {0,2,1,6,7,4,5,3} /* 7.1 surround (classic theater 8-track) */
+};
+
+
+int wav_open(FILE *in, oe_enc_opt *opt, unsigned char *oldbuf, int buflen)
+{
+ unsigned char buf[40];
+ unsigned int len;
+ int samplesize;
+ wav_fmt format;
+ wavfile *wav = malloc(sizeof(wavfile));
+ int i;
+
+ /* Ok. At this point, we know we have a WAV file. Now we have to detect
+ * whether we support the subtype, and we have to find the actual data
+ * We don't (for the wav reader) need to use the buffer we used to id this
+ * as a wav file (oldbuf)
+ */
+
+ if(!find_wav_chunk(in, "fmt ", &len))
+ return 0; /* EOF */
+
+ if(len < 16)
+ {
+ fprintf(stderr, _("Warning: Unrecognised format chunk in WAV header\n"));
+ return 0; /* Weird format chunk */
+ }
+
+ /* A common error is to have a format chunk that is not 16, 18 or
+ * 40 bytes in size. This is incorrect, but not fatal, so we only
+ * warn about it instead of refusing to work with the file.
+ * Please, if you have a program that's creating format chunks of
+ * sizes other than 16 or 18 bytes in size, report a bug to the
+ * author.
+ */
+ if(len!=16 && len!=18 && len!=40)
+ fprintf(stderr,
+ _("Warning: INVALID format chunk in wav header.\n"
+ " Trying to read anyway (may not work)...\n"));
+
+ if(len>40)len=40;
+
+ if(fread(buf,1,len,in) < len)
+ {
+ fprintf(stderr, _("Warning: Unexpected EOF in reading WAV header\n"));
+ return 0;
+ }
+
+ format.format = READ_U16_LE(buf);
+ format.channels = READ_U16_LE(buf+2);
+ format.samplerate = READ_U32_LE(buf+4);
+ format.bytespersec = READ_U32_LE(buf+8);
+ format.align = READ_U16_LE(buf+12);
+ format.samplesize = READ_U16_LE(buf+14);
+
+ if(format.format == -2) /* WAVE_FORMAT_EXTENSIBLE */
+ {
+ if(len<40)
+ {
+ fprintf(stderr,"ERROR: Extended WAV format header invalid (too small)\n");
+ return 0;
+ }
+
+ format.mask = READ_U32_LE(buf+20);
+ /* warn the user if the format mask is not a supported/expected type */
+ switch(format.mask){
+ case 1539: /* 4.0 using side surround instead of back */
+ fprintf(stderr,"WARNING: WAV file uses side surround instead of rear for quadraphonic;\n"
+ "remapping side speakers to rear in encoding.\n");
+ break;
+ case 1551: /* 5.1 using side instead of rear */
+ fprintf(stderr,"WARNING: WAV file uses side surround instead of rear for 5.1;\n"
+ "remapping side speakers to rear in encoding.\n");
+ break;
+ case 319: /* 6.1 using rear instead of side */
+ fprintf(stderr,"WARNING: WAV file uses rear surround instead of side for 6.1;\n"
+ "remapping rear speakers to side in encoding.\n");
+ break;
+ case 255: /* 7.1 'Widescreen' */
+ fprintf(stderr,"WARNING: WAV file is a 7.1 'Widescreen' channel mapping;\n"
+ "remapping speakers to Vorbis 7.1 format.\n");
+ break;
+ case 0: /* default/undeclared */
+ case 1: /* mono */
+ case 3: /* stereo */
+ case 51: /* quad */
+ case 55: /* 5.0 */
+ case 63: /* 5.1 */
+ case 1807: /* 6.1 */
+ case 1599: /* 7.1 */
+ break;
+ default:
+ fprintf(stderr,"WARNING: Unknown WAV surround channel mask: %d\n"
+ "blindly mapping speakers using default SMPTE/ITU ordering.\n",
+ format.mask);
+ break;
+ }
+ format.format = READ_U16_LE(buf+24);
+ }
+
+ if(!find_wav_chunk(in, "data", &len))
+ return 0; /* EOF */
+
+ if(format.format == 1)
+ {
+ samplesize = format.samplesize/8;
+ opt->read_samples = wav_read;
+ }
+ else if(format.format == 3)
+ {
+ samplesize = 4;
+ opt->read_samples = wav_ieee_read;
+ }
+ else
+ {
+ fprintf(stderr,
+ _("ERROR: Wav file is unsupported type (must be standard PCM\n"
+ " or type 3 floating point PCM\n"));
+ return 0;
+ }
+
+ if(format.align != format.channels * samplesize) {
+ /* This is incorrect according to the spec. Warn loudly, then ignore
+ * this value.
+ */
+ fprintf(stderr, _("Warning: WAV 'block alignment' value is incorrect, "
+ "ignoring.\n"
+ "The software that created this file is incorrect.\n"));
+ }
+
+ if(format.samplesize == samplesize*8 &&
+ (format.samplesize == 24 || format.samplesize == 16 ||
+ format.samplesize == 8 ||
+ (format.samplesize == 32 && format.format == 3)))
+ {
+ /* OK, good - we have the one supported format,
+ now we want to find the size of the file */
+ opt->rate = format.samplerate;
+ opt->channels = format.channels;
+
+ wav->f = in;
+ wav->samplesread = 0;
+ wav->bigendian = 0;
+ wav->channels = format.channels; /* This is in several places. The price
+ of trying to abstract stuff. */
+ wav->samplesize = format.samplesize;
+
+ if(len)
+ {
+ opt->total_samples_per_channel = len/(format.channels*samplesize);
+ }
+ else
+ {
+ long pos;
+ pos = ftell(in);
+ if(fseek(in, 0, SEEK_END) == -1)
+ {
+ opt->total_samples_per_channel = 0; /* Give up */
+ }
+ else
+ {
+ opt->total_samples_per_channel = (ftell(in) - pos)/
+ (format.channels*samplesize);
+ fseek(in,pos, SEEK_SET);
+ }
+ }
+ wav->totalsamples = opt->total_samples_per_channel;
+
+ opt->readdata = (void *)wav;
+
+ wav->channel_permute = malloc(wav->channels * sizeof(int));
+ if (wav->channels <= 8)
+ /* Where we know the mappings, use them. */
+ memcpy(wav->channel_permute, wav_permute_matrix[wav->channels-1],
+ sizeof(int) * wav->channels);
+ else
+ /* Use a default 1-1 mapping */
+ for (i=0; i < wav->channels; i++)
+ wav->channel_permute[i] = i;
+
+ return 1;
+ }
+ else
+ {
+ fprintf(stderr,
+ _("ERROR: Wav file is unsupported subformat (must be 8,16, or 24 bit PCM\n"
+ "or floating point PCM\n"));
+ return 0;
+ }
+}
+
+long wav_read(void *in, float **buffer, int samples)
+{
+ wavfile *f = (wavfile *)in;
+ int sampbyte = f->samplesize / 8;
+ signed char *buf = alloca(samples*sampbyte*f->channels);
+ long bytes_read = fread(buf, 1, samples*sampbyte*f->channels, f->f);
+ int i,j;
+ long realsamples;
+ int *ch_permute = f->channel_permute;
+
+ if(f->totalsamples && f->samplesread +
+ bytes_read/(sampbyte*f->channels) > f->totalsamples) {
+ bytes_read = sampbyte*f->channels*(f->totalsamples - f->samplesread);
+ }
+
+ realsamples = bytes_read/(sampbyte*f->channels);
+ f->samplesread += realsamples;
+
+ if(f->samplesize==8)
+ {
+ unsigned char *bufu = (unsigned char *)buf;
+ for(i = 0; i < realsamples; i++)
+ {
+ for(j=0; j < f->channels; j++)
+ {
+ buffer[j][i]=((int)(bufu[i*f->channels + ch_permute[j]])-128)/128.0f;
+ }
+ }
+ }
+ else if(f->samplesize==16)
+ {
+ if(!f->bigendian)
+ {
+ for(i = 0; i < realsamples; i++)
+ {
+ for(j=0; j < f->channels; j++)
+ {
+ buffer[j][i] = ((buf[i*2*f->channels + 2*ch_permute[j] + 1]<<8) |
+ (buf[i*2*f->channels + 2*ch_permute[j]] & 0xff))/32768.0f;
+ }
+ }
+ }
+ else
+ {
+ for(i = 0; i < realsamples; i++)
+ {
+ for(j=0; j < f->channels; j++)
+ {
+ buffer[j][i]=((buf[i*2*f->channels + 2*ch_permute[j]]<<8) |
+ (buf[i*2*f->channels + 2*ch_permute[j] + 1] & 0xff))/32768.0f;
+ }
+ }
+ }
+ }
+ else if(f->samplesize==24)
+ {
+ if(!f->bigendian) {
+ for(i = 0; i < realsamples; i++)
+ {
+ for(j=0; j < f->channels; j++)
+ {
+ buffer[j][i] = ((buf[i*3*f->channels + 3*ch_permute[j] + 2] << 16) |
+ (((unsigned char *)buf)[i*3*f->channels + 3*ch_permute[j] + 1] << 8) |
+ (((unsigned char *)buf)[i*3*f->channels + 3*ch_permute[j]] & 0xff))
+ / 8388608.0f;
+
+ }
+ }
+ }
+ else {
+ fprintf(stderr, _("Big endian 24 bit PCM data is not currently "
+ "supported, aborting.\n"));
+ return 0;
+ }
+ }
+ else {
+ fprintf(stderr, _("Internal error: attempt to read unsupported "
+ "bitdepth %d\n"), f->samplesize);
+ return 0;
+ }
+
+ return realsamples;
+}
+
+long wav_ieee_read(void *in, float **buffer, int samples)
+{
+ wavfile *f = (wavfile *)in;
+ float *buf = alloca(samples*4*f->channels); /* de-interleave buffer */
+ long bytes_read = fread(buf,1,samples*4*f->channels, f->f);
+ int i,j;
+ long realsamples;
+
+
+ if(f->totalsamples && f->samplesread +
+ bytes_read/(4*f->channels) > f->totalsamples)
+ bytes_read = 4*f->channels*(f->totalsamples - f->samplesread);
+ realsamples = bytes_read/(4*f->channels);
+ f->samplesread += realsamples;
+
+ for(i=0; i < realsamples; i++)
+ for(j=0; j < f->channels; j++)
+ buffer[j][i] = buf[i*f->channels + f->channel_permute[j]];
+
+ return realsamples;
+}
+
+
+void wav_close(void *info)
+{
+ wavfile *f = (wavfile *)info;
+ free(f->channel_permute);
+
+ free(f);
+}
+
+int raw_open(FILE *in, oe_enc_opt *opt, unsigned char *buf, int buflen)
+{
+ wav_fmt format; /* fake wave header ;) */
+ wavfile *wav = malloc(sizeof(wavfile));
+ int i;
+
+ /* construct fake wav header ;) */
+ format.format = 2;
+ format.channels = opt->channels;
+ format.samplerate = opt->rate;
+ format.samplesize = opt->samplesize;
+ format.bytespersec = opt->channels * opt->rate * opt->samplesize / 8;
+ format.align = format.bytespersec;
+ wav->f = in;
+ wav->samplesread = 0;
+ wav->bigendian = opt->endianness;
+ wav->channels = format.channels;
+ wav->samplesize = opt->samplesize;
+ wav->totalsamples = 0;
+ wav->channel_permute = malloc(wav->channels * sizeof(int));
+ for (i=0; i < wav->channels; i++)
+ wav->channel_permute[i] = i;
+
+ opt->read_samples = wav_read;
+ opt->readdata = (void *)wav;
+ opt->total_samples_per_channel = 0; /* raw mode, don't bother */
+ return 1;
+}
+
+typedef struct {
+ res_state resampler;
+ audio_read_func real_reader;
+ void *real_readdata;
+ float **bufs;
+ int channels;
+ int bufsize;
+ int done;
+} resampler;
+
+static long read_resampled(void *d, float **buffer, int samples)
+{
+ resampler *rs = d;
+ long in_samples;
+ int out_samples;
+
+ in_samples = res_push_max_input(&rs->resampler, samples);
+ if(in_samples > rs->bufsize)
+ in_samples = rs->bufsize;
+
+ in_samples = rs->real_reader(rs->real_readdata, rs->bufs, in_samples);
+
+ if(in_samples <= 0) {
+ if(!rs->done) {
+ rs->done = 1;
+ out_samples = res_drain(&rs->resampler, buffer);
+ return out_samples;
+ }
+ return 0;
+ }
+
+ out_samples = res_push(&rs->resampler, buffer, (float const **)rs->bufs, in_samples);
+
+ if(out_samples <= 0) {
+ fprintf(stderr, _("BUG: Got zero samples from resampler: your file will be truncated. Please report this.\n"));
+ }
+
+ return out_samples;
+}
+
+int setup_resample(oe_enc_opt *opt) {
+ resampler *rs = calloc(1, sizeof(resampler));
+ int c;
+
+ rs->bufsize = 4096; /* Shrug */
+ rs->real_reader = opt->read_samples;
+ rs->real_readdata = opt->readdata;
+ rs->bufs = malloc(sizeof(float *) * opt->channels);
+ rs->channels = opt->channels;
+ rs->done = 0;
+ if(res_init(&rs->resampler, rs->channels, opt->resamplefreq, opt->rate, RES_END))
+ {
+ fprintf(stderr, _("Couldn't initialise resampler\n"));
+ return -1;
+ }
+
+ for(c=0; c < opt->channels; c++)
+ rs->bufs[c] = malloc(sizeof(float) * rs->bufsize);
+
+ opt->read_samples = read_resampled;
+ opt->readdata = rs;
+ if(opt->total_samples_per_channel)
+ opt->total_samples_per_channel = (int)((float)opt->total_samples_per_channel *
+ ((float)opt->resamplefreq/(float)opt->rate));
+ opt->rate = opt->resamplefreq;
+
+ return 0;
+}
+
+void clear_resample(oe_enc_opt *opt) {
+ resampler *rs = opt->readdata;
+ int i;
+
+ opt->read_samples = rs->real_reader;
+ opt->readdata = rs->real_readdata;
+ res_clear(&rs->resampler);
+
+ for(i = 0; i < rs->channels; i++)
+ free(rs->bufs[i]);
+
+ free(rs->bufs);
+
+ free(rs);
+}
+
+typedef struct {
+ audio_read_func real_reader;
+ void *real_readdata;
+ int channels;
+ float scale_factor;
+} scaler;
+
+static long read_scaler(void *data, float **buffer, int samples) {
+ scaler *d = data;
+ long in_samples = d->real_reader(d->real_readdata, buffer, samples);
+ int i,j;
+
+ for(i=0; i < d->channels; i++) {
+ for(j=0; j < in_samples; j++) {
+ buffer[i][j] *= d->scale_factor;
+ }
+ }
+
+ return in_samples;
+}
+
+
+void setup_scaler(oe_enc_opt *opt, float scale) {
+ scaler *d = calloc(1, sizeof(scaler));
+
+ d->real_reader = opt->read_samples;
+ d->real_readdata = opt->readdata;
+
+ opt->read_samples = read_scaler;
+ opt->readdata = d;
+ d->channels = opt->channels;
+ d->scale_factor = scale;
+}
+
+void clear_scaler(oe_enc_opt *opt) {
+ scaler *d = opt->readdata;
+
+ opt->read_samples = d->real_reader;
+ opt->readdata = d->real_readdata;
+
+ free(d);
+}
+
+typedef struct {
+ audio_read_func real_reader;
+ void *real_readdata;
+ float **bufs;
+} downmix;
+
+static long read_downmix(void *data, float **buffer, int samples)
+{
+ downmix *d = data;
+ long in_samples = d->real_reader(d->real_readdata, d->bufs, samples);
+ int i;
+
+ for(i=0; i < in_samples; i++) {
+ buffer[0][i] = (d->bufs[0][i] + d->bufs[1][i])*0.5f;
+ }
+
+ return in_samples;
+}
+
+void setup_downmix(oe_enc_opt *opt) {
+ downmix *d = calloc(1, sizeof(downmix));
+
+ if(opt->channels != 2) {
+ 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));
+ d->real_reader = opt->read_samples;
+
+ d->real_readdata = opt->readdata;
+
+ opt->read_samples = read_downmix;
+ opt->readdata = d;
+
+ opt->channels = 1;
+}
+void clear_downmix(oe_enc_opt *opt) {
+ downmix *d = opt->readdata;
+
+ opt->read_samples = d->real_reader;
+ opt->readdata = d->real_readdata;
+ opt->channels = 2; /* other things in cleanup rely on this */
+
+ free(d->bufs[0]);
+ free(d->bufs[1]);
+ free(d->bufs);
+ free(d);
+}
+
projects / 16.git / blobdiff