ref: 579fae4ab7d37423aee4d9cdf7a29f9d96c2b764
dir: /src/wav.c/
/* libSoX microsoft's WAVE sound format handler * * Copyright 1998-2006 Chris Bagwell and SoX Contributors * Copyright 1997 Graeme W. Gill, 93/5/17 * Copyright 1992 Rick Richardson * Copyright 1991 Lance Norskog And Sundry Contributors * * Info for format tags can be found at: * http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html * */ #include "sox_i.h" #include <string.h> #include <stdlib.h> #include <stdio.h> #include "ima_rw.h" #include "adpcm.h" #ifdef HAVE_LIBGSM #ifdef HAVE_GSM_GSM_H #include <gsm/gsm.h> #else #include <gsm.h> #endif #endif /* Magic length sometimes used to indicate unknown or too large size. * When detected on inputs, disable any length logic. */ #define MS_UNSPEC 0x7ffff000 #define WAVE_FORMAT_UNKNOWN 0x0000 #define WAVE_FORMAT_PCM 0x0001 #define WAVE_FORMAT_ADPCM 0x0002 #define WAVE_FORMAT_IEEE_FLOAT 0x0003 #define WAVE_FORMAT_IBM_CVSD 0x0005 #define WAVE_FORMAT_ALAW 0x0006 #define WAVE_FORMAT_MULAW 0x0007 #define WAVE_FORMAT_OKI_ADPCM 0x0010 #define WAVE_FORMAT_IMA_ADPCM 0x0011 #define WAVE_FORMAT_MEDIASPACE_ADPCM 0x0012 #define WAVE_FORMAT_SIERRA_ADPCM 0x0013 #define WAVE_FORMAT_G723_ADPCM 0x0014 #define WAVE_FORMAT_DIGISTD 0x0015 #define WAVE_FORMAT_DIGIFIX 0x0016 #define WAVE_FORMAT_YAMAHA_ADPCM 0x0020 #define WAVE_FORMAT_SONARC 0x0021 #define WAVE_FORMAT_TRUESPEECH 0x0022 #define WAVE_FORMAT_ECHOSC1 0x0023 #define WAVE_FORMAT_AUDIOFILE_AF36 0x0024 #define WAVE_FORMAT_APTX 0x0025 #define WAVE_FORMAT_AUDIOFILE_AF10 0x0026 #define WAVE_FORMAT_DOLBY_AC2 0x0030 #define WAVE_FORMAT_GSM610 0x0031 #define WAVE_FORMAT_ADPCME 0x0033 #define WAVE_FORMAT_CONTROL_RES_VQLPC 0x0034 #define WAVE_FORMAT_DIGIREAL 0x0035 #define WAVE_FORMAT_DIGIADPCM 0x0036 #define WAVE_FORMAT_CONTROL_RES_CR10 0x0037 #define WAVE_FORMAT_ROCKWELL_ADPCM 0x003b #define WAVE_FORMAT_ROCKWELL_DIGITALK 0x003c #define WAVE_FORMAT_G721_ADPCM 0x0040 #define WAVE_FORMAT_G728_CELP 0x0041 #define WAVE_FORMAT_MPEG 0x0050 #define WAVE_FORMAT_MPEGLAYER3 0x0055 #define WAVE_FORMAT_G726_ADPCM 0x0064 #define WAVE_FORMAT_G722_ADPCM 0x0065 #define WAVE_FORMAT_CREATIVE_ADPCM 0x0200 #define WAVE_FORMAT_CREATIVE_FSP8 0x0202 #define WAVE_FORMAT_CREATIVE_FSP10 0x0203 #define WAVE_FORMAT_FM_TOWNS_SND 0x0300 #define WAVE_FORMAT_OLIGSM 0x1000 #define WAVE_FORMAT_OLIADPCM 0x1001 #define WAVE_FORMAT_OLISBC 0x1003 #define WAVE_FORMAT_OLIOPR 0x1004 #define WAVE_FORMAT_EXTENSIBLE 0xfffe /* To allow padding to samplesPerBlock. Works, but currently never true. */ static const size_t pad_nsamps = sox_false; /* Private data for .wav file */ typedef struct { /* samples/channel reading: starts at total count and decremented */ /* writing: starts at 0 and counts samples written */ uint64_t numSamples; size_t dataLength; /* needed for ADPCM writing */ unsigned short formatTag; /* What type of encoding file is using */ unsigned short samplesPerBlock; unsigned short blockAlign; uint16_t bitsPerSample; /* bits per sample */ size_t dataStart; /* need to for seeking */ int ignoreSize; /* ignoreSize allows us to process 32-bit WAV files that are * greater then 2 Gb and can't be represented by the * 32-bit size field. */ /* FIXME: Have some front-end code which sets this flag. */ /* following used by *ADPCM wav files */ unsigned short nCoefs; /* ADPCM: number of coef sets */ short *lsx_ms_adpcm_i_coefs; /* ADPCM: coef sets */ void *ms_adpcm_data; /* Private data of adpcm decoder */ unsigned char *packet; /* Temporary buffer for packets */ short *samples; /* interleaved samples buffer */ short *samplePtr; /* Pointer to current sample */ short *sampleTop; /* End of samples-buffer */ unsigned short blockSamplesRemaining;/* Samples remaining per channel */ int state[16]; /* step-size info for *ADPCM writes */ #ifdef HAVE_LIBGSM /* following used by GSM 6.10 wav */ gsm gsmhandle; gsm_signal *gsmsample; int gsmindex; size_t gsmbytecount; /* counts bytes written to data block */ #endif } priv_t; struct wave_format { uint16_t tag; const char *name; sox_encoding_t encoding; int (*read_fmt)(sox_format_t *ft, uint32_t len); }; static const char *wav_format_str(unsigned tag); static int wavwritehdr(sox_format_t *, int); /****************************************************************************/ /* IMA ADPCM Support Functions Section */ /****************************************************************************/ static int wav_ima_adpcm_fmt(sox_format_t *ft, uint32_t len) { priv_t *wav = ft->priv; size_t bytesPerBlock; int err; if (wav->bitsPerSample != 4) { lsx_fail_errno(ft, SOX_EOF, "Can only handle 4-bit IMA ADPCM in wav files"); return SOX_EOF; } err = lsx_read_fields(ft, &len, "h", &wav->samplesPerBlock); if (err) return SOX_EOF; bytesPerBlock = lsx_ima_bytes_per_block(ft->signal.channels, wav->samplesPerBlock); if (bytesPerBlock != wav->blockAlign || wav->samplesPerBlock % 8 != 1) { lsx_fail_errno(ft, SOX_EOF, "format[%s]: samplesPerBlock(%d) != blockAlign(%d)", wav_format_str(wav->formatTag), wav->samplesPerBlock, wav->blockAlign); return SOX_EOF; } wav->packet = lsx_malloc(wav->blockAlign); wav->samples = lsx_malloc(ft->signal.channels * wav->samplesPerBlock * sizeof(short)); return SOX_SUCCESS; } /* * * ImaAdpcmReadBlock - Grab and decode complete block of samples * */ static unsigned short ImaAdpcmReadBlock(sox_format_t * ft) { priv_t * wav = (priv_t *) ft->priv; size_t bytesRead; int samplesThisBlock; /* Pull in the packet and check the header */ bytesRead = lsx_readbuf(ft, wav->packet, (size_t)wav->blockAlign); samplesThisBlock = wav->samplesPerBlock; if (bytesRead < wav->blockAlign) { /* If it looks like a valid header is around then try and */ /* work with partial blocks. Specs say it should be null */ /* padded but I guess this is better than trailing quiet. */ samplesThisBlock = lsx_ima_samples_in((size_t)0, (size_t)ft->signal.channels, bytesRead, (size_t) 0); if (samplesThisBlock == 0 || samplesThisBlock > wav->samplesPerBlock) { lsx_warn("Premature EOF on .wav input file"); return 0; } } wav->samplePtr = wav->samples; /* For a full block, the following should be true: */ /* wav->samplesPerBlock = blockAlign - 8byte header + 1 sample in header */ lsx_ima_block_expand_i(ft->signal.channels, wav->packet, wav->samples, samplesThisBlock); return samplesThisBlock; } /****************************************************************************/ /* MS ADPCM Support Functions Section */ /****************************************************************************/ static int wav_ms_adpcm_fmt(sox_format_t *ft, uint32_t len) { priv_t *wav = ft->priv; size_t bytesPerBlock; int i, errct = 0; int err; if (wav->bitsPerSample != 4) { lsx_fail_errno(ft, SOX_EOF, "Can only handle 4-bit MS ADPCM in wav files"); return SOX_EOF; } err = lsx_read_fields(ft, &len, "hh", &wav->samplesPerBlock, &wav->nCoefs); if (err) return SOX_EOF; bytesPerBlock = lsx_ms_adpcm_bytes_per_block(ft->signal.channels, wav->samplesPerBlock); if (bytesPerBlock != wav->blockAlign) { lsx_fail_errno(ft, SOX_EOF, "format[%s]: samplesPerBlock(%d) != blockAlign(%d)", wav_format_str(wav->formatTag), wav->samplesPerBlock, wav->blockAlign); return SOX_EOF; } if (wav->nCoefs < 7 || wav->nCoefs > 0x100) { lsx_fail_errno(ft, SOX_EOF, "ADPCM file nCoefs (%.4hx) makes no sense", wav->nCoefs); return SOX_EOF; } if (len < 4 * wav->nCoefs) { lsx_fail_errno(ft, SOX_EOF, "wave header error: cbSize too small"); return SOX_EOF; } wav->packet = lsx_malloc(wav->blockAlign); wav->samples = lsx_malloc(ft->signal.channels * wav->samplesPerBlock * sizeof(short)); /* nCoefs, lsx_ms_adpcm_i_coefs used by adpcm.c */ wav->lsx_ms_adpcm_i_coefs = lsx_malloc(wav->nCoefs * 2 * sizeof(short)); wav->ms_adpcm_data = lsx_ms_adpcm_alloc(ft->signal.channels); err = lsx_read_fields(ft, &len, "*h", 2 * wav->nCoefs, wav->lsx_ms_adpcm_i_coefs); if (err) return SOX_EOF; for (i = 0; i < 14; i++) errct += wav->lsx_ms_adpcm_i_coefs[i] != lsx_ms_adpcm_i_coef[i/2][i%2]; if (errct) lsx_warn("base lsx_ms_adpcm_i_coefs differ in %d/14 positions", errct); return SOX_SUCCESS; } /* * * AdpcmReadBlock - Grab and decode complete block of samples * */ static unsigned short AdpcmReadBlock(sox_format_t * ft) { priv_t * wav = (priv_t *) ft->priv; size_t bytesRead; int samplesThisBlock; const char *errmsg; /* Pull in the packet and check the header */ bytesRead = lsx_readbuf(ft, wav->packet, (size_t) wav->blockAlign); samplesThisBlock = wav->samplesPerBlock; if (bytesRead < wav->blockAlign) { /* If it looks like a valid header is around then try and */ /* work with partial blocks. Specs say it should be null */ /* padded but I guess this is better than trailing quiet. */ samplesThisBlock = lsx_ms_adpcm_samples_in((size_t)0, (size_t)ft->signal.channels, bytesRead, (size_t)0); if (samplesThisBlock == 0 || samplesThisBlock > wav->samplesPerBlock) { lsx_warn("Premature EOF on .wav input file"); return 0; } } errmsg = lsx_ms_adpcm_block_expand_i(wav->ms_adpcm_data, ft->signal.channels, wav->nCoefs, wav->lsx_ms_adpcm_i_coefs, wav->packet, wav->samples, samplesThisBlock); if (errmsg) lsx_warn("%s", errmsg); return samplesThisBlock; } /****************************************************************************/ /* Common ADPCM Write Function */ /****************************************************************************/ static int xxxAdpcmWriteBlock(sox_format_t * ft) { priv_t * wav = (priv_t *) ft->priv; size_t chans, ct; short *p; chans = ft->signal.channels; p = wav->samplePtr; ct = p - wav->samples; if (ct>=chans) { /* zero-fill samples if needed to complete block */ for (p = wav->samplePtr; p < wav->sampleTop; p++) *p=0; /* compress the samples to wav->packet */ if (wav->formatTag == WAVE_FORMAT_ADPCM) { lsx_ms_adpcm_block_mash_i((unsigned) chans, wav->samples, wav->samplesPerBlock, wav->state, wav->packet, wav->blockAlign); }else{ /* WAVE_FORMAT_IMA_ADPCM */ lsx_ima_block_mash_i((unsigned) chans, wav->samples, wav->samplesPerBlock, wav->state, wav->packet, 9); } /* write the compressed packet */ if (lsx_writebuf(ft, wav->packet, (size_t) wav->blockAlign) != wav->blockAlign) { lsx_fail_errno(ft,SOX_EOF,"write error"); return (SOX_EOF); } /* update lengths and samplePtr */ wav->dataLength += wav->blockAlign; if (pad_nsamps) wav->numSamples += wav->samplesPerBlock; else wav->numSamples += ct/chans; wav->samplePtr = wav->samples; } return (SOX_SUCCESS); } #ifdef HAVE_LIBGSM /****************************************************************************/ /* WAV GSM6.10 support functions */ /****************************************************************************/ static int wav_gsm_fmt(sox_format_t *ft, uint32_t len) { priv_t *wav = ft->priv; int err; err = lsx_read_fields(ft, &len, "h", &wav->samplesPerBlock); if (err) return SOX_EOF; if (wav->blockAlign != 65) { lsx_fail_errno(ft, SOX_EOF, "format[%s]: expects blockAlign(%d) = %d", wav_format_str(wav->formatTag), wav->blockAlign, 65); return SOX_EOF; } if (wav->samplesPerBlock != 320) { lsx_fail_errno(ft, SOX_EOF, "format[%s]: expects samplesPerBlock(%d) = %d", wav_format_str(wav->formatTag), wav->samplesPerBlock, 320); return SOX_EOF; } return SOX_SUCCESS; } /* create the gsm object, malloc buffer for 160*2 samples */ static int wavgsminit(sox_format_t * ft) { int valueP=1; priv_t * wav = (priv_t *) ft->priv; wav->gsmbytecount=0; wav->gsmhandle=gsm_create(); if (!wav->gsmhandle) { lsx_fail_errno(ft,SOX_EOF,"cannot create GSM object"); return (SOX_EOF); } if(gsm_option(wav->gsmhandle,GSM_OPT_WAV49,&valueP) == -1){ lsx_fail_errno(ft,SOX_EOF,"error setting gsm_option for WAV49 format. Recompile gsm library with -DWAV49 option and relink sox"); return (SOX_EOF); } wav->gsmsample=lsx_malloc(sizeof(gsm_signal)*160*2); wav->gsmindex=0; return (SOX_SUCCESS); } /*destroy the gsm object and free the buffer */ static void wavgsmdestroy(sox_format_t * ft) { priv_t * wav = (priv_t *) ft->priv; gsm_destroy(wav->gsmhandle); free(wav->gsmsample); } static size_t wavgsmread(sox_format_t * ft, sox_sample_t *buf, size_t len) { priv_t * wav = (priv_t *) ft->priv; size_t done=0; int bytes; gsm_byte frame[65]; ft->sox_errno = SOX_SUCCESS; /* copy out any samples left from the last call */ while(wav->gsmindex && (wav->gsmindex<160*2) && (done < len)) buf[done++]=SOX_SIGNED_16BIT_TO_SAMPLE(wav->gsmsample[wav->gsmindex++],); /* read and decode loop, possibly leaving some samples in wav->gsmsample */ while (done < len) { wav->gsmindex=0; bytes = lsx_readbuf(ft, frame, (size_t)65); if (bytes <=0) return done; if (bytes<65) { lsx_warn("invalid wav gsm frame size: %d bytes",bytes); return done; } /* decode the long 33 byte half */ if(gsm_decode(wav->gsmhandle,frame, wav->gsmsample)<0) { lsx_fail_errno(ft,SOX_EOF,"error during gsm decode"); return 0; } /* decode the short 32 byte half */ if(gsm_decode(wav->gsmhandle,frame+33, wav->gsmsample+160)<0) { lsx_fail_errno(ft,SOX_EOF,"error during gsm decode"); return 0; } while ((wav->gsmindex <160*2) && (done < len)){ buf[done++]=SOX_SIGNED_16BIT_TO_SAMPLE(wav->gsmsample[(wav->gsmindex)++],); } } return done; } static int wavgsmflush(sox_format_t * ft) { gsm_byte frame[65]; priv_t * wav = (priv_t *) ft->priv; /* zero fill as needed */ while(wav->gsmindex<160*2) wav->gsmsample[wav->gsmindex++]=0; /*encode the even half short (32 byte) frame */ gsm_encode(wav->gsmhandle, wav->gsmsample, frame); /*encode the odd half long (33 byte) frame */ gsm_encode(wav->gsmhandle, wav->gsmsample+160, frame+32); if (lsx_writebuf(ft, frame, (size_t) 65) != 65) { lsx_fail_errno(ft,SOX_EOF,"write error"); return (SOX_EOF); } wav->gsmbytecount += 65; wav->gsmindex = 0; return (SOX_SUCCESS); } static size_t wavgsmwrite(sox_format_t * ft, const sox_sample_t *buf, size_t len) { priv_t * wav = (priv_t *) ft->priv; size_t done = 0; int rc; ft->sox_errno = SOX_SUCCESS; while (done < len) { SOX_SAMPLE_LOCALS; while ((wav->gsmindex < 160*2) && (done < len)) wav->gsmsample[(wav->gsmindex)++] = SOX_SAMPLE_TO_SIGNED_16BIT(buf[done++], ft->clips); if (wav->gsmindex < 160*2) break; rc = wavgsmflush(ft); if (rc) return 0; } return done; } static void wavgsmstopwrite(sox_format_t * ft) { priv_t * wav = (priv_t *) ft->priv; ft->sox_errno = SOX_SUCCESS; if (wav->gsmindex) wavgsmflush(ft); /* Add a pad byte if amount of written bytes is not even. */ if (wav->gsmbytecount && wav->gsmbytecount % 2){ if(lsx_writeb(ft, 0)) lsx_fail_errno(ft,SOX_EOF,"write error"); else wav->gsmbytecount += 1; } wavgsmdestroy(ft); } #endif /* HAVE_LIBGSM */ /****************************************************************************/ /* General Sox WAV file code */ /****************************************************************************/ static int wav_pcm_fmt(sox_format_t *ft, uint32_t len) { priv_t *wav = ft->priv; int bps = (wav->bitsPerSample + 7) / 8; if (bps == 1) { ft->encoding.encoding = SOX_ENCODING_UNSIGNED; } else if (bps <= 4) { ft->encoding.encoding = SOX_ENCODING_SIGN2; } else { lsx_fail_errno(ft, SOX_EFMT, "%d bytes per sample not suppored", bps); return SOX_EOF; } return SOX_SUCCESS; } static const struct wave_format wave_formats[] = { { WAVE_FORMAT_UNKNOWN, "Unknown Wave Type" }, { WAVE_FORMAT_PCM, "PCM", SOX_ENCODING_UNKNOWN, wav_pcm_fmt, }, { WAVE_FORMAT_ADPCM, "Microsoft ADPCM", SOX_ENCODING_MS_ADPCM, wav_ms_adpcm_fmt, }, { WAVE_FORMAT_IEEE_FLOAT, "IEEE Float", SOX_ENCODING_FLOAT }, { WAVE_FORMAT_IBM_CVSD, "Digispeech CVSD" }, { WAVE_FORMAT_ALAW, "CCITT A-law", SOX_ENCODING_ALAW }, { WAVE_FORMAT_MULAW, "CCITT u-law", SOX_ENCODING_ULAW }, { WAVE_FORMAT_OKI_ADPCM, "OKI ADPCM" }, { WAVE_FORMAT_IMA_ADPCM, "IMA ADPCM", SOX_ENCODING_IMA_ADPCM, wav_ima_adpcm_fmt, }, { WAVE_FORMAT_MEDIASPACE_ADPCM, "MediaSpace ADPCM" }, { WAVE_FORMAT_SIERRA_ADPCM, "Sierra ADPCM" }, { WAVE_FORMAT_G723_ADPCM, "G.723 ADPCM" }, { WAVE_FORMAT_DIGISTD, "DIGISTD" }, { WAVE_FORMAT_DIGIFIX, "DigiFix" }, { WAVE_FORMAT_YAMAHA_ADPCM, "Yamaha ADPCM" }, { WAVE_FORMAT_SONARC, "Sonarc" }, { WAVE_FORMAT_TRUESPEECH, "Truespeech" }, { WAVE_FORMAT_ECHOSC1, "ECHO SC-1", }, { WAVE_FORMAT_AUDIOFILE_AF36, "Audio File AF36" }, { WAVE_FORMAT_APTX, "aptX" }, { WAVE_FORMAT_AUDIOFILE_AF10, "Audio File AF10" }, { WAVE_FORMAT_DOLBY_AC2, "Dolby AC-2" }, { WAVE_FORMAT_GSM610, "GSM 6.10", #ifdef HAVE_LIBGSM SOX_ENCODING_GSM, wav_gsm_fmt, #endif }, { WAVE_FORMAT_ADPCME, "Antex ADPCME" }, { WAVE_FORMAT_CONTROL_RES_VQLPC, "Control Resources VQLPC" }, { WAVE_FORMAT_DIGIREAL, "DSP Solutions REAL" }, { WAVE_FORMAT_DIGIADPCM, "DSP Solutions ADPCM" }, { WAVE_FORMAT_CONTROL_RES_CR10, "Control Resources CR10" }, { WAVE_FORMAT_ROCKWELL_ADPCM, "Rockwell ADPCM" }, { WAVE_FORMAT_ROCKWELL_DIGITALK, "Rockwell DIGITALK" }, { WAVE_FORMAT_G721_ADPCM, "G.721 ADPCM" }, { WAVE_FORMAT_G728_CELP, "G.728 CELP" }, { WAVE_FORMAT_MPEG, "MPEG-1 Audio" }, { WAVE_FORMAT_MPEGLAYER3, "MPEG-1 Layer 3" }, { WAVE_FORMAT_G726_ADPCM, "G.726 ADPCM" }, { WAVE_FORMAT_G722_ADPCM, "G.722 ADPCM" }, { WAVE_FORMAT_CREATIVE_ADPCM, "Creative Labs ADPCM" }, { WAVE_FORMAT_CREATIVE_FSP8, "Creative Labs FastSpeech 8" }, { WAVE_FORMAT_CREATIVE_FSP10, "Creative Labs FastSpeech 10" }, { WAVE_FORMAT_FM_TOWNS_SND, "Fujitsu FM Towns SND" }, { WAVE_FORMAT_OLIGSM, "Olivetti GSM" }, { WAVE_FORMAT_OLIADPCM, "Olivetti ADPCM" }, { WAVE_FORMAT_OLISBC, "Olivetti CELP" }, { WAVE_FORMAT_OLIOPR, "Olivetti OPR" }, { } }; static const struct wave_format *wav_find_format(unsigned tag) { const struct wave_format *f; for (f = wave_formats; f->name; f++) if (f->tag == tag) return f; return NULL; } static int wavfail(sox_format_t *ft, int tag, const char *name) { if (name) lsx_fail_errno(ft, SOX_EHDR, "WAVE format '%s' (%04x) not supported", name, tag); else lsx_fail_errno(ft, SOX_EHDR, "Unknown WAVE format %04x", tag); return SOX_EOF; } static int wav_read_fmt(sox_format_t *ft, uint32_t len) { priv_t *wav = ft->priv; uint16_t wChannels; /* number of channels */ uint32_t dwSamplesPerSecond; /* samples per second per channel */ uint32_t dwAvgBytesPerSec; /* estimate of bytes per second needed */ uint16_t wExtSize = 0; /* extended field for non-PCM */ const struct wave_format *fmt; sox_encoding_t user_enc = ft->encoding.encoding; int err; if (len < 16) { lsx_fail_errno(ft, SOX_EHDR, "WAVE file fmt chunk is too short"); return SOX_EOF; } err = lsx_read_fields(ft, &len, "hhiihh", &wav->formatTag, &wChannels, &dwSamplesPerSecond, &dwAvgBytesPerSec, &wav->blockAlign, &wav->bitsPerSample); if (err) return SOX_EOF; /* non-PCM formats except alaw and mulaw formats have extended fmt chunk. * Check for those cases. */ if (wav->formatTag != WAVE_FORMAT_PCM && wav->formatTag != WAVE_FORMAT_ALAW && wav->formatTag != WAVE_FORMAT_MULAW && len < 2) lsx_warn("WAVE file missing extended part of fmt chunk"); if (len >= 2) { err = lsx_read_fields(ft, &len, "h", &wExtSize); if (err) return SOX_EOF; } if (wExtSize != len) { lsx_fail_errno(ft, SOX_EOF, "WAVE header error: cbSize inconsistent with fmt size"); return SOX_EOF; } if (wav->formatTag == WAVE_FORMAT_EXTENSIBLE) { uint16_t numberOfValidBits; uint32_t speakerPositionMask; uint16_t subFormatTag; if (len < 22) { lsx_fail_errno(ft, SOX_EHDR, "WAVE file fmt chunk is too short"); return SOX_EOF; } err = lsx_read_fields(ft, &len, "hih14x", &numberOfValidBits, &speakerPositionMask, &subFormatTag); if (err) return SOX_EOF; if (numberOfValidBits > wav->bitsPerSample) { lsx_fail_errno(ft, SOX_EHDR, "wValidBitsPerSample > wBitsPerSample"); return SOX_EOF; } wav->formatTag = subFormatTag; lsx_report("EXTENSIBLE"); } /* User options take precedence */ if (ft->signal.channels == 0 || ft->signal.channels == wChannels) ft->signal.channels = wChannels; else lsx_report("User options overriding channels read in .wav header"); if (ft->signal.channels == 0) { lsx_fail_errno(ft, SOX_EHDR, "Channel count is zero"); return SOX_EOF; } if (ft->signal.rate == 0 || ft->signal.rate == dwSamplesPerSecond) ft->signal.rate = dwSamplesPerSecond; else lsx_report("User options overriding rate read in .wav header"); fmt = wav_find_format(wav->formatTag); if (!fmt) return wavfail(ft, wav->formatTag, NULL); /* format handler might override */ ft->encoding.encoding = fmt->encoding; if (fmt->read_fmt) { if (fmt->read_fmt(ft, len)) return SOX_EOF; } else if (!fmt->encoding) { return wavfail(ft, wav->formatTag, fmt->name); } /* User options take precedence */ if (!ft->encoding.bits_per_sample || ft->encoding.bits_per_sample == wav->bitsPerSample) ft->encoding.bits_per_sample = wav->bitsPerSample; else lsx_warn("User options overriding size read in .wav header"); if (user_enc && user_enc != ft->encoding.encoding) { lsx_report("User options overriding encoding read in .wav header"); ft->encoding.encoding = user_enc; } return 0; } static sox_bool valid_chunk_id(const char p[4]) { int i; for (i = 0; i < 4; i++) if (p[i] < 0x20 || p[i] > 0x7f) return sox_false; return sox_true; } static int read_chunk_header(sox_format_t *ft, char tag[4], uint32_t *len) { int r; r = lsx_readbuf(ft, tag, 4); if (r < 4) return SOX_EOF; return lsx_readdw(ft, len); } /* * Do anything required before you start reading samples. * Read file header. * Find out sampling rate, * size and encoding of samples, * mono/stereo/quad. */ static int startread(sox_format_t *ft) { priv_t *wav = ft->priv; char magic[5] = { 0 }; uint32_t clen; int err; sox_bool isRF64 = sox_false; uint64_t ds64_riff_size; uint64_t ds64_data_size; uint64_t ds64_sample_count; /* wave file characteristics */ uint64_t qwRiffLength; uint64_t qwDataLength = 0; sox_bool have_fmt = sox_false; ft->sox_errno = SOX_SUCCESS; wav->ignoreSize = ft->signal.length == SOX_IGNORE_LENGTH; ft->encoding.reverse_bytes = MACHINE_IS_BIGENDIAN; if (read_chunk_header(ft, magic, &clen)) return SOX_EOF; if (!memcmp(magic, "RIFX", 4)) { lsx_debug("Found RIFX header"); ft->encoding.reverse_bytes = MACHINE_IS_LITTLEENDIAN; } else if (!memcmp(magic, "RF64", 4)) { lsx_debug("Found RF64 header"); isRF64 = sox_true; } else if (memcmp(magic, "RIFF", 4)) { lsx_fail_errno(ft, SOX_EHDR, "WAVE: RIFF header not found"); return SOX_EOF; } qwRiffLength = clen; if (lsx_readbuf(ft, magic, 4) < 4 || memcmp(magic, "WAVE", 4)) { lsx_fail_errno(ft, SOX_EHDR, "WAVE header not found"); return SOX_EOF; } while (!read_chunk_header(ft, magic, &clen)) { uint32_t len = clen; off_t cstart = lsx_tell(ft); off_t pos; if (!valid_chunk_id(magic)) { lsx_fail_errno(ft, SOX_EHDR, "invalid chunk ID found"); return SOX_EOF; } lsx_debug("Found chunk '%s', size %u", magic, clen); if (!memcmp(magic, "ds64", 4)) { if (!isRF64) lsx_warn("ds64 chunk in non-RF64 file"); if (clen < 28) { lsx_fail_errno(ft, SOX_EHDR, "ds64 chunk too small"); return SOX_EOF; } if (clen == 32) { lsx_warn("ds64 chunk size invalid, attempting workaround"); clen = 28; } err = lsx_read_fields(ft, &len, "qqq", &ds64_riff_size, &ds64_data_size, &ds64_sample_count); if (err) return SOX_EOF; goto next; } if (!memcmp(magic, "fmt ", 4)) { err = wav_read_fmt(ft, clen); if (err) return err; have_fmt = sox_true; goto next; } if (!memcmp(magic, "fact", 4)) { uint32_t val; err = lsx_read_fields(ft, &len, "i", &val); if (err) return SOX_EOF; wav->numSamples = val; goto next; } if (!memcmp(magic, "data", 4)) { if (isRF64 && clen == UINT32_MAX) clen = ds64_data_size; qwDataLength = clen; wav->dataStart = lsx_tell(ft); if (qwDataLength == UINT32_MAX || qwDataLength == MS_UNSPEC) break; if (!ft->seekable) break; goto next; } next: pos = lsx_tell(ft); clen += clen & 1; if (pos > cstart + clen) { lsx_fail_errno(ft, SOX_EHDR, "malformed chunk %s", magic); return SOX_EOF; } err = lsx_seeki(ft, cstart + clen - pos, SEEK_CUR); if (err) return SOX_EOF; } if (isRF64) { if (wav->numSamples == UINT32_MAX) wav->numSamples = ds64_sample_count; if (qwRiffLength == UINT32_MAX) qwRiffLength = ds64_riff_size; } if (!have_fmt) { lsx_fail_errno(ft, SOX_EOF, "fmt chunk not found"); return SOX_EOF; } if (!wav->dataStart) { lsx_fail_errno(ft, SOX_EOF, "data chunk not found"); return SOX_EOF; } if (ft->seekable) lsx_seeki(ft, wav->dataStart, SEEK_SET); /* some files wrongly report total samples across all channels */ if (wav->numSamples * wav->blockAlign == qwDataLength * ft->signal.channels) wav->numSamples /= ft->signal.channels; if ((qwDataLength == UINT32_MAX && !wav->numSamples) || qwDataLength == MS_UNSPEC) { lsx_warn("WAV data length is magic value or UINT32_MAX, ignoring"); wav->ignoreSize = 1; } switch (wav->formatTag) { case WAVE_FORMAT_ADPCM: wav->numSamples = lsx_ms_adpcm_samples_in(qwDataLength, ft->signal.channels, wav->blockAlign, wav->samplesPerBlock); wav->blockSamplesRemaining = 0; /* Samples left in buffer */ break; case WAVE_FORMAT_IMA_ADPCM: /* Compute easiest part of number of samples. For every block, there are samplesPerBlock samples to read. */ wav->numSamples = lsx_ima_samples_in(qwDataLength, ft->signal.channels, wav->blockAlign, wav->samplesPerBlock); wav->blockSamplesRemaining = 0; /* Samples left in buffer */ lsx_ima_init_table(); break; #ifdef HAVE_LIBGSM case WAVE_FORMAT_GSM610: wav->numSamples = qwDataLength / wav->blockAlign * wav->samplesPerBlock; wavgsminit(ft); break; #endif } if (!wav->numSamples) wav->numSamples = div_bits(qwDataLength, ft->encoding.bits_per_sample) / ft->signal.channels; if (wav->ignoreSize) ft->signal.length = SOX_UNSPEC; else ft->signal.length = wav->numSamples * ft->signal.channels; return lsx_rawstartread(ft); } /* * Read up to len samples from file. * Convert to signed longs. * Place in buf[]. * Return number of samples read. */ static size_t read_samples(sox_format_t *ft, sox_sample_t *buf, size_t len) { priv_t *wav = ft->priv; size_t done; ft->sox_errno = SOX_SUCCESS; if (!wav->ignoreSize) len = min(len, wav->numSamples * ft->signal.channels); /* If file is in ADPCM encoding then read in multiple blocks else */ /* read as much as possible and return quickly. */ switch (ft->encoding.encoding) { case SOX_ENCODING_IMA_ADPCM: case SOX_ENCODING_MS_ADPCM: done = 0; while (done < len) { /* Still want data? */ short *p, *top; size_t ct; /* See if need to read more from disk */ if (wav->blockSamplesRemaining == 0) { if (wav->formatTag == WAVE_FORMAT_IMA_ADPCM) wav->blockSamplesRemaining = ImaAdpcmReadBlock(ft); else wav->blockSamplesRemaining = AdpcmReadBlock(ft); if (wav->blockSamplesRemaining == 0) { /* Don't try to read any more samples */ wav->numSamples = 0; return done; } wav->samplePtr = wav->samples; } /* Copy interleaved data into buf, converting to sox_sample_t */ ct = len - done; if (ct > wav->blockSamplesRemaining * ft->signal.channels) ct = wav->blockSamplesRemaining * ft->signal.channels; done += ct; wav->blockSamplesRemaining -= ct / ft->signal.channels; p = wav->samplePtr; top = p + ct; /* Output is already signed */ while (p < top) *buf++ = SOX_SIGNED_16BIT_TO_SAMPLE(*p++,); wav->samplePtr = p; } /* "done" for ADPCM equals total data processed and not * total samples procesed. The only way to take care of that * is to return here and not fall thru. */ wav->numSamples -= done / ft->signal.channels; return done; #ifdef HAVE_LIBGSM case SOX_ENCODING_GSM: done = wavgsmread(ft, buf, len); break; #endif default: /* assume PCM or float encoding */ done = lsx_rawread(ft, buf, len); break; } if (done == 0 && wav->numSamples && !wav->ignoreSize) lsx_warn("Premature EOF on .wav input file"); /* Only return buffers that contain a totally playable * amount of audio. */ done -= done % ft->signal.channels; if (done / ft->signal.channels > wav->numSamples) wav->numSamples = 0; else wav->numSamples -= done / ft->signal.channels; return done; } /* * Do anything required when you stop reading samples. * Don't close input file! */ static int stopread(sox_format_t * ft) { priv_t * wav = (priv_t *) ft->priv; ft->sox_errno = SOX_SUCCESS; free(wav->packet); free(wav->samples); free(wav->lsx_ms_adpcm_i_coefs); free(wav->ms_adpcm_data); switch (ft->encoding.encoding) { #ifdef HAVE_LIBGSM case SOX_ENCODING_GSM: wavgsmdestroy(ft); break; #endif case SOX_ENCODING_IMA_ADPCM: case SOX_ENCODING_MS_ADPCM: break; default: break; } return SOX_SUCCESS; } static int startwrite(sox_format_t * ft) { priv_t * wav = (priv_t *) ft->priv; int rc; ft->sox_errno = SOX_SUCCESS; if (ft->encoding.encoding != SOX_ENCODING_MS_ADPCM && ft->encoding.encoding != SOX_ENCODING_IMA_ADPCM && ft->encoding.encoding != SOX_ENCODING_GSM) { rc = lsx_rawstartwrite(ft); if (rc) return rc; } wav->numSamples = 0; wav->dataLength = 0; if (!ft->signal.length && !ft->seekable) lsx_warn("Length in output .wav header will be wrong since can't seek to fix it"); rc = wavwritehdr(ft, 0); /* also calculates various wav->* info */ if (rc != 0) return rc; wav->packet = NULL; wav->samples = NULL; wav->lsx_ms_adpcm_i_coefs = NULL; switch (wav->formatTag) { size_t ch, sbsize; case WAVE_FORMAT_IMA_ADPCM: lsx_ima_init_table(); /* intentional case fallthru! */ case WAVE_FORMAT_ADPCM: /* #channels already range-checked for overflow in wavwritehdr() */ for (ch=0; ch<ft->signal.channels; ch++) wav->state[ch] = 0; sbsize = ft->signal.channels * wav->samplesPerBlock; wav->packet = lsx_malloc((size_t)wav->blockAlign); wav->samples = lsx_malloc(sbsize*sizeof(short)); wav->sampleTop = wav->samples + sbsize; wav->samplePtr = wav->samples; break; #ifdef HAVE_LIBGSM case WAVE_FORMAT_GSM610: return wavgsminit(ft); #endif default: break; } return SOX_SUCCESS; } /* wavwritehdr: write .wav headers as follows: bytes variable description 0 - 3 'RIFF'/'RIFX' Little/Big-endian 4 - 7 wRiffLength length of file minus the 8 byte riff header 8 - 11 'WAVE' 12 - 15 'fmt ' 16 - 19 wFmtSize length of format chunk minus 8 byte header 20 - 21 wFormatTag identifies PCM, ULAW etc 22 - 23 wChannels 24 - 27 dwSamplesPerSecond samples per second per channel 28 - 31 dwAvgBytesPerSec non-trivial for compressed formats 32 - 33 wBlockAlign basic block size 34 - 35 wBitsPerSample non-trivial for compressed formats PCM formats then go straight to the data chunk: 36 - 39 'data' 40 - 43 dwDataLength length of data chunk minus 8 byte header 44 - (dwDataLength + 43) the data (+ a padding byte if dwDataLength is odd) non-PCM formats must write an extended format chunk and a fact chunk: ULAW, ALAW formats: 36 - 37 wExtSize = 0 the length of the format extension 38 - 41 'fact' 42 - 45 dwFactSize = 4 length of the fact chunk minus 8 byte header 46 - 49 dwSamplesWritten actual number of samples written out 50 - 53 'data' 54 - 57 dwDataLength length of data chunk minus 8 byte header 58 - (dwDataLength + 57) the data (+ a padding byte if dwDataLength is odd) GSM6.10 format: 36 - 37 wExtSize = 2 the length in bytes of the format-dependent extension 38 - 39 320 number of samples per block 40 - 43 'fact' 44 - 47 dwFactSize = 4 length of the fact chunk minus 8 byte header 48 - 51 dwSamplesWritten actual number of samples written out 52 - 55 'data' 56 - 59 dwDataLength length of data chunk minus 8 byte header 60 - (dwDataLength + 59) the data (including a padding byte, if necessary, so dwDataLength is always even) note that header contains (up to) 3 separate ways of describing the length of the file, all derived here from the number of (input) samples wav->numSamples in a way that is non-trivial for the blocked and padded compressed formats: wRiffLength - (riff header) the length of the file, minus 8 dwSamplesWritten - (fact header) the number of samples written (after padding to a complete block eg for GSM) dwDataLength - (data chunk header) the number of (valid) data bytes written */ static int wavwritehdr(sox_format_t * ft, int second_header) { priv_t * wav = (priv_t *) ft->priv; /* variables written to wav file header */ /* RIFF header */ uint64_t wRiffLength ; /* length of file after 8 byte riff header */ /* fmt chunk */ uint16_t wFmtSize = 16; /* size field of the fmt chunk */ uint16_t wFormatTag = 0; /* data format */ uint16_t wChannels; /* number of channels */ uint32_t dwSamplesPerSecond; /* samples per second per channel*/ uint32_t dwAvgBytesPerSec=0; /* estimate of bytes per second needed */ uint32_t wBlockAlign=0; /* byte alignment of a basic sample block */ uint16_t wBitsPerSample=0; /* bits per sample */ /* fmt chunk extension (not PCM) */ uint16_t wExtSize=0; /* extra bytes in the format extension */ uint16_t wSamplesPerBlock; /* samples per channel per block */ /* wSamplesPerBlock and other things may go into format extension */ /* fact chunk (not PCM) */ uint32_t dwFactSize=4; /* length of the fact chunk */ uint64_t dwSamplesWritten=0; /* windows doesnt seem to use this*/ /* data chunk */ uint64_t dwDataLength; /* length of sound data in bytes */ /* end of variables written to header */ /* internal variables, intermediate values etc */ int bytespersample; /* (uncompressed) bytes per sample (per channel) */ uint64_t blocksWritten = 0; sox_bool isExtensible = sox_false; /* WAVE_FORMAT_EXTENSIBLE? */ if (ft->signal.channels > UINT16_MAX) { lsx_fail_errno(ft, SOX_EOF, "Too many channels (%u)", ft->signal.channels); return SOX_EOF; } dwSamplesPerSecond = ft->signal.rate; wChannels = ft->signal.channels; wBitsPerSample = ft->encoding.bits_per_sample; wSamplesPerBlock = 1; /* common default for PCM data */ switch (ft->encoding.encoding) { case SOX_ENCODING_UNSIGNED: case SOX_ENCODING_SIGN2: wFormatTag = WAVE_FORMAT_PCM; bytespersample = (wBitsPerSample + 7)/8; wBlockAlign = wChannels * bytespersample; break; case SOX_ENCODING_FLOAT: wFormatTag = WAVE_FORMAT_IEEE_FLOAT; bytespersample = (wBitsPerSample + 7)/8; wBlockAlign = wChannels * bytespersample; break; case SOX_ENCODING_ALAW: wFormatTag = WAVE_FORMAT_ALAW; wBlockAlign = wChannels; break; case SOX_ENCODING_ULAW: wFormatTag = WAVE_FORMAT_MULAW; wBlockAlign = wChannels; break; case SOX_ENCODING_IMA_ADPCM: if (wChannels>16) { lsx_fail_errno(ft,SOX_EOF,"Channels(%d) must be <= 16",wChannels); return SOX_EOF; } wFormatTag = WAVE_FORMAT_IMA_ADPCM; wBlockAlign = wChannels * 256; /* reasonable default */ wBitsPerSample = 4; wExtSize = 2; wSamplesPerBlock = lsx_ima_samples_in((size_t) 0, (size_t) wChannels, (size_t) wBlockAlign, (size_t) 0); break; case SOX_ENCODING_MS_ADPCM: if (wChannels>16) { lsx_fail_errno(ft,SOX_EOF,"Channels(%d) must be <= 16",wChannels); return SOX_EOF; } wFormatTag = WAVE_FORMAT_ADPCM; wBlockAlign = ft->signal.rate / 11008; wBlockAlign = max(wBlockAlign, 1) * wChannels * 256; wBitsPerSample = 4; wExtSize = 4+4*7; /* Ext fmt data length */ wSamplesPerBlock = lsx_ms_adpcm_samples_in((size_t) 0, (size_t) wChannels, (size_t) wBlockAlign, (size_t) 0); break; #ifdef HAVE_LIBGSM case SOX_ENCODING_GSM: if (wChannels!=1) { lsx_report("Overriding GSM audio from %d channel to 1",wChannels); if (!second_header) ft->signal.length /= max(1, ft->signal.channels); wChannels = ft->signal.channels = 1; } wFormatTag = WAVE_FORMAT_GSM610; /* dwAvgBytesPerSec = 1625*(dwSamplesPerSecond/8000.)+0.5; */ wBlockAlign=65; wBitsPerSample=0; /* not representable as int */ wExtSize=2; /* length of format extension */ wSamplesPerBlock = 320; break; #endif default: break; } if (wBlockAlign > UINT16_MAX) { lsx_fail_errno(ft, SOX_EOF, "Too many channels (%u)", ft->signal.channels); return SOX_EOF; } wav->formatTag = wFormatTag; wav->blockAlign = wBlockAlign; wav->samplesPerBlock = wSamplesPerBlock; /* When creating header, use length hint given by input file. If no * hint then write default value. Also, use default value even * on header update if more then 32-bit length needs to be written. */ dwSamplesWritten = second_header ? wav->numSamples : ft->signal.length / wChannels; blocksWritten = (dwSamplesWritten + wSamplesPerBlock - 1) / wSamplesPerBlock; dwDataLength = blocksWritten * wBlockAlign; if (wFormatTag == WAVE_FORMAT_GSM610) dwDataLength = (dwDataLength+1) & ~1u; /* round up to even */ if (wFormatTag == WAVE_FORMAT_PCM && (wBitsPerSample > 16 || wChannels > 2) && strcmp(ft->filetype, "wavpcm")) { isExtensible = sox_true; wFmtSize += 2 + 22; } else if (wFormatTag != WAVE_FORMAT_PCM) wFmtSize += 2+wExtSize; /* plus ExtData */ wRiffLength = 4 + (8+wFmtSize) + (8+dwDataLength+dwDataLength%2); if (isExtensible || wFormatTag != WAVE_FORMAT_PCM) /* PCM omits the "fact" chunk */ wRiffLength += (8+dwFactSize); if (dwSamplesWritten > UINT32_MAX) dwSamplesWritten = UINT32_MAX; if (dwDataLength > UINT32_MAX) dwDataLength = UINT32_MAX; if (!second_header && !ft->signal.length) dwDataLength = UINT32_MAX; if (wRiffLength > UINT32_MAX) wRiffLength = UINT32_MAX; /* dwAvgBytesPerSec <-- this is BEFORE compression, isn't it? guess not. */ dwAvgBytesPerSec = (double)wBlockAlign*ft->signal.rate / (double)wSamplesPerBlock + 0.5; /* figured out header info, so write it */ /* If user specified opposite swap than we think, assume they are * asking to write a RIFX file. */ if (ft->encoding.reverse_bytes == MACHINE_IS_LITTLEENDIAN) { if (!second_header) lsx_report("Requested to swap bytes so writing RIFX header"); lsx_writes(ft, "RIFX"); } else lsx_writes(ft, "RIFF"); lsx_writedw(ft, wRiffLength); lsx_writes(ft, "WAVE"); lsx_writes(ft, "fmt "); lsx_writedw(ft, wFmtSize); lsx_writew(ft, isExtensible ? WAVE_FORMAT_EXTENSIBLE : wFormatTag); lsx_writew(ft, wChannels); lsx_writedw(ft, dwSamplesPerSecond); lsx_writedw(ft, dwAvgBytesPerSec); lsx_writew(ft, wBlockAlign); lsx_writew(ft, wBitsPerSample); /* end info common to all fmts */ if (isExtensible) { uint32_t dwChannelMask=0; /* unassigned speaker mapping by default */ static unsigned char const guids[][14] = { "\x00\x00\x00\x00\x10\x00\x80\x00\x00\xAA\x00\x38\x9B\x71", /* wav */ "\x00\x00\x21\x07\xd3\x11\x86\x44\xc8\xc1\xca\x00\x00\x00"}; /* amb */ /* if not amb, assume most likely channel masks from number of channels; not * ideal solution, but will make files playable in many/most situations */ if (strcmp(ft->filetype, "amb")) { if (wChannels == 1) dwChannelMask = 0x4; /* 1 channel (mono) = FC */ else if (wChannels == 2) dwChannelMask = 0x3; /* 2 channels (stereo) = FL, FR */ else if (wChannels == 4) dwChannelMask = 0x33; /* 4 channels (quad) = FL, FR, BL, BR */ else if (wChannels == 6) dwChannelMask = 0x3F; /* 6 channels (5.1) = FL, FR, FC, LF, BL, BR */ else if (wChannels == 8) dwChannelMask = 0x63F; /* 8 channels (7.1) = FL, FR, FC, LF, BL, BR, SL, SR */ } lsx_writew(ft, 22); lsx_writew(ft, wBitsPerSample); /* No padding in container */ lsx_writedw(ft, dwChannelMask); /* Speaker mapping is something reasonable */ lsx_writew(ft, wFormatTag); lsx_writebuf(ft, guids[!strcmp(ft->filetype, "amb")], (size_t)14); } else /* if not PCM, we need to write out wExtSize even if wExtSize=0 */ if (wFormatTag != WAVE_FORMAT_PCM) lsx_writew(ft,wExtSize); switch (wFormatTag) { int i; case WAVE_FORMAT_IMA_ADPCM: lsx_writew(ft, wSamplesPerBlock); break; case WAVE_FORMAT_ADPCM: lsx_writew(ft, wSamplesPerBlock); lsx_writew(ft, 7); /* nCoefs */ for (i=0; i<7; i++) { lsx_writew(ft, (uint16_t)(lsx_ms_adpcm_i_coef[i][0])); lsx_writew(ft, (uint16_t)(lsx_ms_adpcm_i_coef[i][1])); } break; case WAVE_FORMAT_GSM610: lsx_writew(ft, wSamplesPerBlock); break; default: break; } /* if not PCM, write the 'fact' chunk */ if (isExtensible || wFormatTag != WAVE_FORMAT_PCM){ lsx_writes(ft, "fact"); lsx_writedw(ft,dwFactSize); lsx_writedw(ft,dwSamplesWritten); } lsx_writes(ft, "data"); lsx_writedw(ft, dwDataLength); /* data chunk size */ if (!second_header) { lsx_debug("Writing Wave file: %s format, %d channel%s, %d samp/sec", wav_format_str(wFormatTag), wChannels, wChannels == 1 ? "" : "s", dwSamplesPerSecond); lsx_debug(" %d byte/sec, %d block align, %d bits/samp", dwAvgBytesPerSec, wBlockAlign, wBitsPerSample); } else { if (wRiffLength == UINT32_MAX || dwDataLength == UINT32_MAX || dwSamplesWritten == UINT32_MAX) lsx_warn("File too large, writing truncated values in header"); lsx_debug("Finished writing Wave file, %"PRIu64" data bytes %"PRIu64" samples", dwDataLength, wav->numSamples); #ifdef HAVE_LIBGSM if (wFormatTag == WAVE_FORMAT_GSM610){ lsx_debug("GSM6.10 format: %"PRIu64" blocks %"PRIu64" padded samples %"PRIu64" padded data bytes", blocksWritten, dwSamplesWritten, dwDataLength); if (wav->gsmbytecount != dwDataLength) lsx_warn("help ! internal inconsistency - data_written %"PRIu64" gsmbytecount %zu", dwDataLength, wav->gsmbytecount); } #endif } return SOX_SUCCESS; } static size_t write_samples(sox_format_t * ft, const sox_sample_t *buf, size_t len) { priv_t * wav = (priv_t *) ft->priv; ptrdiff_t total_len = len; ft->sox_errno = SOX_SUCCESS; switch (wav->formatTag) { case WAVE_FORMAT_IMA_ADPCM: case WAVE_FORMAT_ADPCM: while (len>0) { short *p = wav->samplePtr; short *top = wav->sampleTop; if (top>p+len) top = p+len; len -= top-p; /* update residual len */ while (p < top) *p++ = (*buf++) >> 16; wav->samplePtr = p; if (p == wav->sampleTop) xxxAdpcmWriteBlock(ft); } return total_len - len; break; #ifdef HAVE_LIBGSM case WAVE_FORMAT_GSM610: len = wavgsmwrite(ft, buf, len); wav->numSamples += (len/ft->signal.channels); return len; break; #endif default: len = lsx_rawwrite(ft, buf, len); wav->numSamples += (len/ft->signal.channels); return len; } } static int stopwrite(sox_format_t * ft) { priv_t * wav = (priv_t *) ft->priv; ft->sox_errno = SOX_SUCCESS; /* Call this to flush out any remaining data. */ switch (wav->formatTag) { case WAVE_FORMAT_IMA_ADPCM: case WAVE_FORMAT_ADPCM: xxxAdpcmWriteBlock(ft); break; #ifdef HAVE_LIBGSM case WAVE_FORMAT_GSM610: wavgsmstopwrite(ft); break; #endif } /* Add a pad byte if the number of data bytes is odd. See wavwritehdr() above for the calculation. */ if (wav->formatTag != WAVE_FORMAT_GSM610) lsx_padbytes(ft, (size_t)((wav->numSamples + wav->samplesPerBlock - 1)/wav->samplesPerBlock*wav->blockAlign) % 2); free(wav->packet); free(wav->samples); free(wav->lsx_ms_adpcm_i_coefs); /* All samples are already written out. */ /* If file header needs fixing up, for example it needs the */ /* the number of samples in a field, seek back and write them here. */ if (ft->signal.length && wav->numSamples <= 0xffffffff && wav->numSamples == ft->signal.length) return SOX_SUCCESS; if (!ft->seekable) return SOX_EOF; if (lsx_seeki(ft, (off_t)0, SEEK_SET) != 0) { lsx_fail_errno(ft,SOX_EOF,"Can't rewind output file to rewrite .wav header."); return SOX_EOF; } return (wavwritehdr(ft, 1)); } /* * Return a string corresponding to the wave format type. */ static const char *wav_format_str(unsigned tag) { const struct wave_format *f = wav_find_format(tag); return f ? f->name : "unknown"; } static int seek(sox_format_t * ft, uint64_t offset) { priv_t * wav = (priv_t *) ft->priv; if (ft->encoding.bits_per_sample & 7) lsx_fail_errno(ft, SOX_ENOTSUP, "seeking not supported with this encoding"); else if (wav->formatTag == WAVE_FORMAT_GSM610) { int alignment; size_t gsmoff; /* rounding bytes to blockAlign so that we * don't have to decode partial block. */ gsmoff = offset * wav->blockAlign / wav->samplesPerBlock + wav->blockAlign * ft->signal.channels / 2; gsmoff -= gsmoff % (wav->blockAlign * ft->signal.channels); ft->sox_errno = lsx_seeki(ft, (off_t)(gsmoff + wav->dataStart), SEEK_SET); if (ft->sox_errno == SOX_SUCCESS) { /* offset is in samples */ uint64_t new_offset = offset; alignment = offset % wav->samplesPerBlock; if (alignment != 0) new_offset += (wav->samplesPerBlock - alignment); wav->numSamples = ft->signal.length - (new_offset / ft->signal.channels); } } else { double wide_sample = offset - (offset % ft->signal.channels); double to_d = wide_sample * ft->encoding.bits_per_sample / 8; off_t to = to_d; ft->sox_errno = (to != to_d)? SOX_EOF : lsx_seeki(ft, (off_t)wav->dataStart + (off_t)to, SEEK_SET); if (ft->sox_errno == SOX_SUCCESS) wav->numSamples -= (size_t)wide_sample / ft->signal.channels; } return ft->sox_errno; } LSX_FORMAT_HANDLER(wav) { static char const * const names[] = {"wav", "wavpcm", "amb", NULL}; static unsigned const write_encodings[] = { SOX_ENCODING_SIGN2, 16, 24, 32, 0, SOX_ENCODING_UNSIGNED, 8, 0, SOX_ENCODING_ULAW, 8, 0, SOX_ENCODING_ALAW, 8, 0, #ifdef HAVE_LIBGSM SOX_ENCODING_GSM, 0, #endif SOX_ENCODING_MS_ADPCM, 4, 0, SOX_ENCODING_IMA_ADPCM, 4, 0, SOX_ENCODING_FLOAT, 32, 64, 0, 0}; static sox_format_handler_t const handler = {SOX_LIB_VERSION_CODE, "Microsoft audio format", names, SOX_FILE_LIT_END, startread, read_samples, stopread, startwrite, write_samples, stopwrite, seek, write_encodings, NULL, sizeof(priv_t) }; return &handler; }