ref: 686823647f4b0971c685865e14b5cd0422c8fdd0
parent: 89c244d6a2a0e96c998ce31510a5a114e9006008
author: Gregory Maxwell <greg@xiph.org>
date: Mon Jan 13 06:55:10 EST 2014
Add support in opusdec to write out 32-bit floating-point.
--- a/man/opusdec.1
+++ b/man/opusdec.1
@@ -17,6 +17,8 @@
] [
.B --no-dither
] [
+.B --float
+] [
.B --force-wav
] [
.B --packet-loss pct
@@ -61,6 +63,8 @@
Adjust the output volume n dB, negative values make the signal quieter.
.IP "--no-dither"
Do not dither 16-bit output
+.IP "--float"
+32-bit floating-point files instead of 16-bit files
.IP "--force-wav"
Force including a wav header on output (e.g. for non-wav extensions and stdout)
.IP "--packet-loss"
--- a/src/opusdec.c
+++ b/src/opusdec.c
@@ -278,7 +278,7 @@
}
}
-FILE *out_file_open(char *outFile, int *wav_format, int rate, int mapping_family, int *channels)
+FILE *out_file_open(char *outFile, int *wav_format, int rate, int mapping_family, int *channels, int fp)
{FILE *fout=NULL;
/*Open output file*/
@@ -429,7 +429,7 @@
}
if (*wav_format)
{- *wav_format = write_wav_header(fout, rate, mapping_family, *channels);
+ *wav_format = write_wav_header(fout, rate, mapping_family, *channels, fp);
if (*wav_format < 0)
{fprintf (stderr, "Error writing WAV header.\n");
@@ -460,6 +460,7 @@
printf (" --rate n Force decoding at sampling rate n Hz\n"); printf (" --gain n Manually adjust gain by n.nn dB (0 default)\n"); printf (" --no-dither Do not dither 16-bit output\n");+ printf (" --float 32-bit floating-point output\n"); printf (" --force-wav Force wav header on output\n"); printf (" --packet-loss n Simulate n %% random packet loss\n"); printf (" --save-range file Saves check values for every frame to a file\n");@@ -558,7 +559,7 @@
}
opus_int64 audio_write(float *pcm, int channels, int frame_size, FILE *fout, SpeexResamplerState *resampler,
- int *skip, shapestate *shapemem, int file, opus_int64 maxout)
+ int *skip, shapestate *shapemem, int file, opus_int64 maxout, int fp)
{opus_int64 sampout=0;
int i,ret,tmp_skip;
@@ -590,17 +591,20 @@
frame_size=0;
}
- /*Convert to short and save to output file*/
- if (shapemem){- shape_dither_toshort(shapemem,out,output,out_len,channels);
- }else{- for (i=0;i<(int)out_len*channels;i++)
- out[i]=(short)float2int(fmaxf(-32768,fminf(output[i]*32768.f,32767)));
+ if(!file||!fp)
+ {+ /*Convert to short and save to output file*/
+ if (shapemem){+ shape_dither_toshort(shapemem,out,output,out_len,channels);
+ }else{+ for (i=0;i<(int)out_len*channels;i++)
+ out[i]=(short)float2int(fmaxf(-32768,fminf(output[i]*32768.f,32767)));
+ }
+ if ((le_short(1)!=1)&&file){+ for (i=0;i<(int)out_len*channels;i++)
+ out[i]=le_short(out[i]);
+ }
}
- if ((le_short(1)!=1)&&file){- for (i=0;i<(int)out_len*channels;i++)
- out[i]=le_short(out[i]);
- }
if(maxout>0)
{@@ -617,7 +621,7 @@
else fprintf(stderr, "Error playing audio.\n");
}else
#endif
- ret=fwrite(out, 2*channels, out_len<maxout?out_len:maxout, fout);
+ ret=fwrite(fp?(char *)output:(char *)out, (fp?4:2)*channels, out_len<maxout?out_len:maxout, fout);
sampout+=ret;
maxout-=ret;
}
@@ -650,6 +654,7 @@
{"rate", required_argument, NULL, 0}, {"gain", required_argument, NULL, 0}, {"no-dither", no_argument, NULL, 0},+ {"float", no_argument, NULL, 0}, {"force-wav", no_argument, NULL, 0}, {"packet-loss", required_argument, NULL, 0}, {"save-range", required_argument, NULL, 0},@@ -675,6 +680,7 @@
int has_tags_packet=0;
ogg_int32_t opus_serialno;
int dither=1;
+ int fp=0;
shapestate shapemem;
SpeexResamplerState *resampler=NULL;
float gain=1;
@@ -733,6 +739,9 @@
} else if (strcmp(long_options[option_index].name,"no-dither")==0)
{dither=0;
+ } else if (strcmp(long_options[option_index].name,"float")==0)
+ {+ fp=1;
} else if (strcmp(long_options[option_index].name,"force-wav")==0)
{forcewav=1;
@@ -799,7 +808,11 @@
relevant for playback. Many audio devices sound
better at 48kHz and not resampling also saves CPU.*/
if(rate==0)rate=48000;
+ /*Playback is 16-bit only.*/
+ fp=0;
}
+ /*If the output is floating point, don't dither.*/
+ if(fp)dither=0;
/*Open input file*/
if (strcmp(inFile, "-")==0)
@@ -934,7 +947,7 @@
fprintf(stderr, "resampler error: %s\n", speex_resampler_strerror(err));
speex_resampler_skip_zeros(resampler);
}
- if(!fout)fout=out_file_open(outFile, &wav_format, rate, mapping_family, &channels);
+ if(!fout)fout=out_file_open(outFile, &wav_format, rate, mapping_family, &channels, fp);
} else if (packet_count==1)
{if (!quiet)
@@ -983,7 +996,7 @@
if(!quiet){/*Display a progress spinner while decoding.*/
static const char spinner[]="|/-\\";
- double coded_seconds = (double)audio_size/(channels*rate*sizeof(short));
+ double coded_seconds = (double)audio_size/(channels*rate*(fp?4:2));
if(coded_seconds>=last_coded_seconds+1){fprintf(stderr,"\r[%c] %02d:%02d:%02d", spinner[last_spin&3],
(int)(coded_seconds/3600),(int)(coded_seconds/60)%60,
@@ -1025,9 +1038,9 @@
the final end-trim by not letting the output sample count
get ahead of the granpos indicated value.*/
maxout=((page_granule-gran_offset)*rate/48000)-link_out;
- outsamp=audio_write(output, channels, frame_size, fout, resampler, &preskip, dither?&shapemem:0, strlen(outFile)!=0,0>maxout?0:maxout);
+ outsamp=audio_write(output, channels, frame_size, fout, resampler, &preskip, dither?&shapemem:0, strlen(outFile)!=0,0>maxout?0:maxout,fp);
link_out+=outsamp;
- audio_size+=sizeof(short)*outsamp*channels;
+ audio_size+=(fp?4:2)*outsamp*channels;
}
packet_count++;
}
@@ -1044,9 +1057,9 @@
int tmp = drain;
if (tmp > 100)
tmp = 100;
- outsamp=audio_write(zeros, channels, tmp, fout, resampler, NULL, &shapemem, strlen(outFile)!=0, ((page_granule-gran_offset)*rate/48000)-link_out);
+ outsamp=audio_write(zeros, channels, tmp, fout, resampler, NULL, &shapemem, strlen(outFile)!=0, ((page_granule-gran_offset)*rate/48000)-link_out,fp);
link_out+=outsamp;
- audio_size+=sizeof(short)*outsamp*channels;
+ audio_size+=(fp?4:2)*outsamp*channels;
drain -= tmp;
} while (drain>0);
free(zeros);
--- a/src/wav_io.c
+++ b/src/wav_io.c
@@ -70,7 +70,7 @@
return fwrite(buf,2,1,file);
}
-int write_wav_header(FILE *file, int rate, int mapping_family, int channels)
+int write_wav_header(FILE *file, int rate, int mapping_family, int channels, int fp)
{int ret;
int extensible;
@@ -79,17 +79,20 @@
proper channel meanings. */
extensible = mapping_family == 1 && 3 <= channels && channels <= 8;
+ /* >16 bit audio also requires WAVEFORMATEXTENSIBLE. */
+ extensible |= fp;
+
ret = fprintf (file, "RIFF") >= 0;
ret &= fwrite_le32 (0x7fffffff, file);
ret &= fprintf (file, "WAVEfmt ") >= 0;
ret &= fwrite_le32 (extensible ? 40 : 16, file);
- ret &= fwrite_le16 (extensible ? 0xfffe : 1, file);
+ ret &= fwrite_le16 (extensible ? 0xfffe : (fp?3:1), file);
ret &= fwrite_le16 (channels, file);
ret &= fwrite_le32 (rate, file);
- ret &= fwrite_le32 (2*channels*rate, file);
- ret &= fwrite_le16 (2*channels, file);
- ret &= fwrite_le16 (16, file);
+ ret &= fwrite_le32 ((fp?4:2)*channels*rate, file);
+ ret &= fwrite_le16 ((fp?4:2)*channels, file);
+ ret &= fwrite_le16 (fp?32:16, file);
if(extensible)
{@@ -101,6 +104,14 @@
0x80, 0x00,
0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71
};
+ static const unsigned char ksdataformat_subtype_float[16]=
+ {+ 0x03, 0x00, 0x00, 0x00,
+ 0x00, 0x00,
+ 0x10, 0x00,
+ 0x80, 0x00,
+ 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71
+ };
static const int wav_channel_masks[8] =
{1, /* 1.0 mono */
@@ -113,9 +124,14 @@
1|2|4|8|16|32|512|1024, /* 7.1 */
};
ret &= fwrite_le16 (22, file);
- ret &= fwrite_le16 (16, file);
+ ret &= fwrite_le16 (fp?32:16, file);
ret &= fwrite_le32 (wav_channel_masks[channels-1], file);
- ret &= fwrite (ksdataformat_subtype_pcm, 16, 1, file);
+ if (!fp)
+ {+ ret &= fwrite (ksdataformat_subtype_pcm, 16, 1, file);
+ } else {+ ret &= fwrite (ksdataformat_subtype_float, 16, 1, file);
+ }
}
ret &= fprintf (file, "data") >= 0;
--- a/src/wav_io.h
+++ b/src/wav_io.h
@@ -57,6 +57,6 @@
void adjust_wav_mapping(int mapping_family, int channels, unsigned char *stream_map);
-int write_wav_header(FILE *file, int rate, int mapping_family, int channels);
+int write_wav_header(FILE *file, int rate, int mapping_family, int channels, int fp);
#endif