cog/Frameworks/vgmstream/vgmstream/src/coding/coding_utils.c

1175 lines
51 KiB
C

#include "coding.h"
#include <math.h>
#include "../vgmstream.h"
/**
* Various utils for formats that aren't handled their own decoder or meta
*
* ffmpeg_make_riff_* utils don't depend on FFmpeg, but rather, they make headers that FFmpeg
* can use (it doesn't understand all valid RIFF headers, nor the utils make 100% correct headers).
*/
/* ******************************************** */
/* INTERNAL UTILS */
/* ******************************************** */
/**
* read num_bits (up to 25) from a bit offset.
* 25 since we read a 32 bit int, and need to adjust up to 7 bits from the byte-rounded fseek (32-7=25)
*/
static uint32_t read_bitsBE_b(int64_t bit_offset, int num_bits, STREAMFILE *streamFile) {
uint32_t num, mask;
if (num_bits > 25) return -1; //???
num = read_32bitBE(bit_offset / 8, streamFile); /* fseek rounded to 8 */
num = num << (bit_offset % 8); /* offset adjust (up to 7) */
num = num >> (32 - num_bits);
mask = 0xffffffff >> (32 - num_bits);
return num & mask;
}
/* ******************************************** */
/* FAKE RIFF HELPERS */
/* ******************************************** */
/* All helpers copy a RIFF header to buf and returns the number of bytes in buf or -1 when buf is not big enough */
int ffmpeg_make_riff_atrac3plus(uint8_t * buf, size_t buf_size, size_t sample_count, size_t data_size, int channels, int sample_rate, int block_align, int encoder_delay) {
uint16_t codec_ATRAC3plus = 0xfffe; /* wave format extensible */
size_t riff_size = 4+4+ 4 + 0x3c + 0x14 + 4+4;
if (buf_size < riff_size)
return -1;
memcpy(buf+0x00, "RIFF", 4);
put_32bitLE(buf+0x04, (int32_t)(riff_size-4-4 + data_size)); /* riff size */
memcpy(buf+0x08, "WAVE", 4);
memcpy(buf+0x0c, "fmt ", 4);
put_32bitLE(buf+0x10, 0x34);/*fmt size*/
put_16bitLE(buf+0x14, codec_ATRAC3plus);
put_16bitLE(buf+0x16, channels);
put_32bitLE(buf+0x18, sample_rate);
put_32bitLE(buf+0x1c, sample_rate*channels / sizeof(sample)); /* average bytes per second (wrong) */
put_32bitLE(buf+0x20, (int16_t)(block_align)); /* block align */
put_16bitLE(buf+0x24, 0x22); /* extra data size */
put_16bitLE(buf+0x26, 0x0800); /* samples per block */
put_32bitLE(buf+0x28, 0x0000003); /* unknown */
put_32bitBE(buf+0x2c, 0xBFAA23E9); /* GUID1 */
put_32bitBE(buf+0x30, 0x58CB7144); /* GUID2 */
put_32bitBE(buf+0x34, 0xA119FFFA); /* GUID3 */
put_32bitBE(buf+0x38, 0x01E4CE62); /* GUID4 */
put_16bitBE(buf+0x3c, 0x0010); /* unknown */
put_16bitBE(buf+0x3e, 0x0000); /* config */ //todo this varies with block size, but FFmpeg doesn't use it
put_32bitBE(buf+0x40, 0x00000000); /* empty */
put_32bitBE(buf+0x44, 0x00000000); /* empty */
memcpy(buf+0x48, "fact", 4);
put_32bitLE(buf+0x4c, 0x0c); /* fact size */
put_32bitLE(buf+0x50, sample_count);
put_32bitLE(buf+0x54, 0); /* unknown */
put_32bitLE(buf+0x58, encoder_delay);
memcpy(buf+0x5c, "data", 4);
put_32bitLE(buf+0x60, data_size); /* data size */
return riff_size;
}
int ffmpeg_make_riff_xma1(uint8_t * buf, size_t buf_size, size_t sample_count, size_t data_size, int channels, int sample_rate, int stream_mode) {
uint16_t codec_XMA1 = 0x0165;
size_t riff_size;
int streams, i;
/* stream disposition:
* 0: default (ex. 5ch = 2ch + 2ch + 1ch = 3 streams)
* 1: lineal (ex. 5ch = 1ch + 1ch + 1ch + 1ch + 1ch = 5 streams), unusual but exists
* others: not seen (ex. maybe 5ch = 2ch + 1ch + 1ch + 1ch = 4 streams) */
switch(stream_mode) {
case 0 : streams = (channels + 1) / 2; break;
case 1 : streams = channels; break;
default: return 0;
}
riff_size = 4+4+ 4 + 0x14 + 0x14*streams + 4+4;
if (buf_size < riff_size)
return -1;
memcpy(buf+0x00, "RIFF", 4);
put_32bitLE(buf+0x04, (int32_t)(riff_size-4-4 + data_size)); /* riff size */
memcpy(buf+0x08, "WAVE", 4);
memcpy(buf+0x0c, "fmt ", 4);
put_32bitLE(buf+0x10, 0xc + 0x14*streams);/*fmt size*/
put_16bitLE(buf+0x14, codec_XMA1);
put_16bitLE(buf+0x16, 16); /* bits per sample */
put_16bitLE(buf+0x18, 0x10D6); /* encoder options */
put_16bitLE(buf+0x1a, 0); /* largest stream skip (wrong, unneeded) */
put_16bitLE(buf+0x1c, streams); /* number of streams */
put_8bit (buf+0x1e, 0); /* loop count */
put_8bit (buf+0x1f, 2); /* version */
for (i = 0; i < streams; i++) {
int stream_channels;
uint32_t speakers;
off_t off = 0x20 + 0x14*i;/* stream riff offset */
if (stream_mode == 1) {
/* lineal */
stream_channels = 1;
switch(i) { /* per stream, values observed */
case 0: speakers = 0x0001; break;/* L */
case 1: speakers = 0x0002; break;/* R */
case 2: speakers = 0x0004; break;/* C */
case 3: speakers = 0x0008; break;/* LFE */
case 4: speakers = 0x0040; break;/* LB */
case 5: speakers = 0x0080; break;/* RB */
case 6: speakers = 0x0000; break;/* ? */
case 7: speakers = 0x0000; break;/* ? */
default: speakers = 0;
}
}
else {
/* with odd channels the last stream is mono */
stream_channels = channels / streams + (channels%2 != 0 && i+1 != streams ? 1 : 0);
switch(i) { /* per stream, values from xmaencode */
case 0: speakers = stream_channels == 1 ? 0x0001 : 0x0201; break;/* L R */
case 1: speakers = stream_channels == 1 ? 0x0004 : 0x0804; break;/* C LFE */
case 2: speakers = stream_channels == 1 ? 0x0040 : 0x8040; break;/* LB RB */
case 3: speakers = stream_channels == 1 ? 0x0000 : 0x0000; break;/* somehow empty (maybe should use 0x2010 LS RS) */
default: speakers = 0;
}
}
put_32bitLE(buf+off+0x00, sample_rate*stream_channels / sizeof(sample)); /* average bytes per second (wrong, unneeded) */
put_32bitLE(buf+off+0x04, sample_rate);
put_32bitLE(buf+off+0x08, 0); /* loop start */
put_32bitLE(buf+off+0x0c, 0); /* loop end */
put_8bit (buf+off+0x10, 0); /* loop subframe */
put_8bit (buf+off+0x11, stream_channels);
put_16bitLE(buf+off+0x12, speakers);
}
/* xmaencode decoding rejects XMA1 without "seek" chunk, though it doesn't seem to use it
* (needs to be have entries but can be bogus, also generates seek for even small sounds) */
memcpy(buf+riff_size-4-4, "data", 4);
put_32bitLE(buf+riff_size-4, data_size); /* data size */
return riff_size;
}
int ffmpeg_make_riff_xma2(uint8_t * buf, size_t buf_size, size_t sample_count, size_t data_size, int channels, int sample_rate, int block_count, int block_size) {
uint16_t codec_XMA2 = 0x0166;
size_t riff_size = 4+4+ 4 + 0x3c + 4+4;
size_t bytecount;
int streams;
uint32_t speakers;
/* info from xma2defs.h, xact3wb.h and audiodefs.h */
streams = (channels + 1) / 2;
switch (channels) {
case 1: speakers = 0x04; break; /* 1.0: FC */
case 2: speakers = 0x01 | 0x02; break; /* 2.0: FL FR */
case 3: speakers = 0x01 | 0x02 | 0x08; break; /* 2.1: FL FR LF */
case 4: speakers = 0x01 | 0x02 | 0x10 | 0x20; break; /* 4.0: FL FR BL BR */
case 5: speakers = 0x01 | 0x02 | 0x08 | 0x10 | 0x20; break; /* 4.1: FL FR LF BL BR */
case 6: speakers = 0x01 | 0x02 | 0x04 | 0x08 | 0x10 | 0x20; break; /* 5.1: FL FR FC LF BL BR */
case 7: speakers = 0x01 | 0x02 | 0x04 | 0x08 | 0x10 | 0x20 | 0x0100; break; /* 6.1: FL FR FC LF BL BR BC */
case 8: speakers = 0x01 | 0x02 | 0x04 | 0x08 | 0x10 | 0x20 | 0x40 | 0x80; break; /* 7.1: FL FR FC LF BL BR FLC FRC */
default: speakers = 0; break;
}
if (buf_size < riff_size)
return -1;
bytecount = sample_count * channels * sizeof(sample);
memcpy(buf+0x00, "RIFF", 4);
put_32bitLE(buf+0x04, (int32_t)(riff_size-4-4 + data_size)); /* riff size */
memcpy(buf+0x08, "WAVE", 4);
memcpy(buf+0x0c, "fmt ", 4);
put_32bitLE(buf+0x10, 0x34);/*fmt size*/
put_16bitLE(buf+0x14, codec_XMA2);
put_16bitLE(buf+0x16, channels);
put_32bitLE(buf+0x18, sample_rate);
put_32bitLE(buf+0x1c, sample_rate*channels / sizeof(sample)); /* average bytes per second (wrong, unneeded) */
put_16bitLE(buf+0x20, (int16_t)(channels*sizeof(sample))); /* block align */
put_16bitLE(buf+0x22, 16); /* bits per sample */
put_16bitLE(buf+0x24, 0x22); /* extra data size */
put_16bitLE(buf+0x26, streams); /* number of streams */
put_32bitLE(buf+0x28, speakers); /* speaker position */
put_32bitLE(buf+0x2c, bytecount); /* PCM samples */
put_32bitLE(buf+0x30, block_size); /* XMA block size (can be zero, it's for seeking only) */
/* (looping values not set, expected to be handled externally) */
put_32bitLE(buf+0x34, 0); /* play begin */
put_32bitLE(buf+0x38, 0); /* play length */
put_32bitLE(buf+0x3c, 0); /* loop begin */
put_32bitLE(buf+0x40, 0); /* loop length */
put_8bit(buf+0x44, 0); /* loop count */
put_8bit(buf+0x45, 4); /* encoder version */
put_16bitLE(buf+0x46, block_count); /* blocks count (entries in seek table, can be zero) */
memcpy(buf+0x48, "data", 4);
put_32bitLE(buf+0x4c, data_size); /* data size */
return riff_size;
}
/* Makes a XMA1/2 RIFF header for FFmpeg using a "fmt " chunk (XMAWAVEFORMAT or XMA2WAVEFORMATEX) as a base:
* Useful to preserve the stream layout */
int ffmpeg_make_riff_xma_from_fmt_chunk(uint8_t * buf, size_t buf_size, off_t fmt_offset, size_t fmt_size, size_t data_size, STREAMFILE *streamFile, int big_endian) {
size_t riff_size = 4+4+ 4 + 4+4+fmt_size + 4+4;
uint8_t chunk[0x100];
if (buf_size < riff_size || fmt_size > 0x100)
goto fail;
if (read_streamfile(chunk,fmt_offset,fmt_size, streamFile) != fmt_size)
goto fail;
if (big_endian) {
int codec = read_16bitBE(fmt_offset,streamFile);
ffmpeg_fmt_chunk_swap_endian(chunk, fmt_size, codec);
}
memcpy(buf+0x00, "RIFF", 4);
put_32bitLE(buf+0x04, (int32_t)(riff_size-4-4 + data_size)); /* riff size */
memcpy(buf+0x08, "WAVE", 4);
memcpy(buf+0x0c, "fmt ", 4);
put_32bitLE(buf+0x10, fmt_size);/*fmt size*/
memcpy(buf+0x14, chunk, fmt_size);
memcpy(buf+0x14+fmt_size, "data", 4);
put_32bitLE(buf+0x14+fmt_size+4, data_size); /* data size */
return riff_size;
fail:
return -1;
}
/* Makes a XMA2 RIFF header for FFmpeg using a "XMA2" chunk (XMA2WAVEFORMAT) as a base.
* Useful to preserve the stream layout */
int ffmpeg_make_riff_xma2_from_xma2_chunk(uint8_t * buf, size_t buf_size, off_t xma2_offset, size_t xma2_size, size_t data_size, STREAMFILE *streamFile) {
uint8_t chunk[0x100];
size_t riff_size;
riff_size = 4+4+ 4 + 4+4+xma2_size + 4+4;
if (buf_size < riff_size || xma2_size > 0x100)
goto fail;
if (read_streamfile(chunk,xma2_offset,xma2_size, streamFile) != xma2_size)
goto fail;
memcpy(buf+0x00, "RIFF", 4);
put_32bitLE(buf+0x04, (int32_t)(riff_size-4-4 + data_size)); /* riff size */
memcpy(buf+0x08, "WAVE", 4);
memcpy(buf+0x0c, "XMA2", 4);
put_32bitLE(buf+0x10, xma2_size);
memcpy(buf+0x14, chunk, xma2_size);
memcpy(buf+0x14+xma2_size, "data", 4);
put_32bitLE(buf+0x14+xma2_size+4, data_size); /* data size */
return riff_size;
fail:
return -1;
}
int ffmpeg_make_riff_xwma(uint8_t * buf, size_t buf_size, int codec, size_t data_size, int channels, int sample_rate, int avg_bps, int block_align) {
size_t riff_size = 4+4+ 4 + 0x1a + 4+4;
if (buf_size < riff_size)
return -1;
/* XWMA encoder only allows a few channel/sample rate/bitrate combinations,
* but some create identical files with fake bitrate (1ch 22050hz at
* 20/48/192kbps are all 20kbps, with the exact same codec data).
* Decoder needs correct bitrate to work, so it's normalized here. */
/* (may be removed once FFmpeg fixes this) */
if (codec == 0x161) { /* WMAv2 only */
int ch = channels;
int sr = sample_rate;
int br = avg_bps * 8;
/* Must be a bug in MS's encoder, as later versions of xWMAEncode remove these bitrates */
if (ch == 1) {
if (sr == 22050 && (br==48000 || br==192000))
br = 20000;
else if (sr == 32000 && (br==48000 || br==192000))
br = 20000;
else if (sr == 44100 && (br==96000 || br==192000))
br = 48000;
}
else if (ch == 2) {
if (sr == 22050 && (br==48000 || br==192000))
br = 32000;
else if (sr == 32000 && (br==192000))
br = 48000;
}
avg_bps = br / 8;
}
memcpy(buf+0x00, "RIFF", 4);
put_32bitLE(buf+0x04, (int32_t)(riff_size-4-4 + data_size)); /* riff size */
memcpy(buf+0x08, "XWMA", 4);
memcpy(buf+0x0c, "fmt ", 4);
put_32bitLE(buf+0x10, 0x12);/*fmt size*/
put_16bitLE(buf+0x14, codec);
put_16bitLE(buf+0x16, channels);
put_32bitLE(buf+0x18, sample_rate);
put_32bitLE(buf+0x1c, avg_bps); /* average bytes per second, somehow vital for XWMA */
put_16bitLE(buf+0x20, block_align); /* block align */
put_16bitLE(buf+0x22, 16); /* bits per sample */
put_16bitLE(buf+0x24, 0); /* extra size */
/* here goes the "dpds" seek table, but it's optional and not needed by FFmpeg (and also buggy) */
memcpy(buf+0x26, "data", 4);
put_32bitLE(buf+0x2a, data_size); /* data size */
return riff_size;
}
int ffmpeg_fmt_chunk_swap_endian(uint8_t * chunk, size_t chunk_size, uint16_t codec) {
int i;
/* swap from LE to BE or the other way around, doesn't matter */
switch(codec) {
case 0x165: { /* XMA1 */
put_16bitLE(chunk + 0x00, get_16bitBE(chunk + 0x00));/*FormatTag*/
put_16bitLE(chunk + 0x02, get_16bitBE(chunk + 0x02));/*BitsPerSample*/
put_16bitLE(chunk + 0x04, get_16bitBE(chunk + 0x04));/*EncodeOptions*/
put_16bitLE(chunk + 0x06, get_16bitBE(chunk + 0x06));/*LargestSkip*/
put_16bitLE(chunk + 0x08, get_16bitBE(chunk + 0x08));/*NumStreams*/
// put_8bit(chunk + 0x0a, get_8bit(chunk + 0x0a));/*LoopCount*/
// put_8bit(chunk + 0x0b, get_8bit(chunk + 0x0b));/*Version*/
for (i = 0xc; i < chunk_size; i += 0x14) { /* reverse endianness for each stream */
put_32bitLE(chunk + i + 0x00, get_32bitBE(chunk + i + 0x00));/*PsuedoBytesPerSec*/
put_32bitLE(chunk + i + 0x04, get_32bitBE(chunk + i + 0x04));/*SampleRate*/
put_32bitLE(chunk + i + 0x08, get_32bitBE(chunk + i + 0x08));/*LoopStart*/
put_32bitLE(chunk + i + 0x0c, get_32bitBE(chunk + i + 0x0c));/*LoopEnd*/
// put_8bit(chunk + i + 0x10, get_8bit(chunk + i + 0x10));/*SubframeData*/
// put_8bit(chunk + i + 0x11, get_8bit(chunk + i + 0x11));/*Channels*/
put_16bitLE(chunk + i + 0x12, get_16bitBE(chunk + i + 0x12));/*ChannelMask*/
}
break;
}
case 0x166: { /* XMA2 */
put_16bitLE(chunk + 0x00, get_16bitBE(chunk + 0x00));/*wFormatTag*/
put_16bitLE(chunk + 0x02, get_16bitBE(chunk + 0x02));/*nChannels*/
put_32bitLE(chunk + 0x04, get_32bitBE(chunk + 0x04));/*nSamplesPerSec*/
put_32bitLE(chunk + 0x08, get_32bitBE(chunk + 0x08));/*nAvgBytesPerSec*/
put_16bitLE(chunk + 0x0c, get_16bitBE(chunk + 0x0c));/*nBlockAlign*/
put_16bitLE(chunk + 0x0e, get_16bitBE(chunk + 0x0e));/*wBitsPerSample*/
put_16bitLE(chunk + 0x10, get_16bitBE(chunk + 0x10));/*cbSize*/
put_16bitLE(chunk + 0x12, get_16bitBE(chunk + 0x12));/*NumStreams*/
put_32bitLE(chunk + 0x14, get_32bitBE(chunk + 0x14));/*ChannelMask*/
put_32bitLE(chunk + 0x18, get_32bitBE(chunk + 0x18));/*SamplesEncoded*/
put_32bitLE(chunk + 0x1c, get_32bitBE(chunk + 0x1c));/*BytesPerBlock*/
put_32bitLE(chunk + 0x20, get_32bitBE(chunk + 0x20));/*PlayBegin*/
put_32bitLE(chunk + 0x24, get_32bitBE(chunk + 0x24));/*PlayLength*/
put_32bitLE(chunk + 0x28, get_32bitBE(chunk + 0x28));/*LoopBegin*/
put_32bitLE(chunk + 0x2c, get_32bitBE(chunk + 0x2c));/*LoopLength*/
/* put_8bit(chunk + 0x30, get_8bit(chunk + 0x30));*//*LoopCount*/
/* put_8bit(chunk + 0x31, get_8bit(chunk + 0x31));*//*EncoderVersion*/
put_16bitLE(chunk + 0x32, get_16bitBE(chunk + 0x32));/*BlockCount*/
break;
}
default:
goto fail;
}
return 1;
fail:
return 0;
}
/* ******************************************** */
/* XMA PARSING */
/* ******************************************** */
static void ms_audio_parse_header(STREAMFILE *streamFile, int xma_version, int64_t offset_b, int bits_frame_size, size_t *first_frame_b, size_t *packet_skip_count, size_t *header_size_b) {
if (xma_version == 1) { /* XMA1 */
//packet_sequence = read_bitsBE_b(offset_b+0, 4, streamFile); /* numbered from 0 to N */
//unknown = read_bitsBE_b(offset_b+4, 2, streamFile); /* packet_metadata? (always 2) */
*first_frame_b = read_bitsBE_b(offset_b+6, bits_frame_size, streamFile); /* offset in bits inside the packet */
*packet_skip_count = read_bitsBE_b(offset_b+21, 11, streamFile); /* packets to skip for next packet of this stream */
*header_size_b = 32;
} else if (xma_version == 2) { /* XMA2 */
//frame_count = read_bitsBE_b(offset_b+0, 6, streamFile); /* frames that begin in this packet */
*first_frame_b = read_bitsBE_b(offset_b+6, bits_frame_size, streamFile); /* offset in bits inside this packet */
//packet_metadata = read_bitsBE_b(offset_b+21, 3, streamFile); /* packet_metadata (always 1) */
*packet_skip_count = read_bitsBE_b(offset_b+24, 8, streamFile); /* packets to skip for next packet of this stream */
*header_size_b = 32;
} else { /* WMAPRO(v3) */
//packet_sequence = read_bitsBE_b(offset_b+0, 4, streamFile); /* numbered from 0 to N */
//unknown = read_bitsBE_b(offset_b+4, 2, streamFile); /* packet_metadata? (always 2) */
*first_frame_b = read_bitsBE_b(offset_b+6, bits_frame_size, streamFile); /* offset in bits inside the packet */
*packet_skip_count = 0; /* xwma has no need to skip packets since it uses real multichannel audio */
*header_size_b = 4+2+bits_frame_size; /* variable-sized header */
}
/* XMA2 packets with XMA1 RIFF (transmogrified), remove the packet metadata flag */
if (xma_version == 1 && (*packet_skip_count & 0x700) == 0x100) {
//VGM_LOG("MS_SAMPLES: XMA1 transmogrified packet header at 0x%lx\n", (off_t)offset_b/8);
*packet_skip_count &= ~0x100;
}
/* full packet skip, no new frames start in this packet (prev frames can end here)
* standardized to some value */
if (*packet_skip_count == 0x7FF) { /* XMA1, 11b */
VGM_LOG("MS_SAMPLES: XMA1 full packet_skip\n");// at %"PRIx64"\n", offset_b/8);
*packet_skip_count = 0x800;
}
else if (*packet_skip_count == 0xFF) { /* XMA2, 8b*/
VGM_LOG("MS_SAMPLES: XMA2 full packet_skip\n");// at %"PRIx64"\n", offset_b/8);
*packet_skip_count = 0x800;
}
/* unusual but not impossible, as the encoder can interleave packets in any way */
VGM_ASSERT((*packet_skip_count > 10 && *packet_skip_count < 0x800),
"MS_SAMPLES: found big packet skip %i at 0x%x\n", *packet_skip_count, (uint32_t)offset_b/8);
}
/**
* Find total and loop samples of Microsoft audio formats (WMAPRO/XMA1/XMA2) by reading frame headers.
*
* The stream is made of packets, each containing N small frames of X samples. Frames are further divided into subframes.
* XMA1/XMA2 can divided into streams for multichannel (1/2ch ... 1/2ch). From the file start, packet 1..N is owned by
* stream 1..N. Then must follow "packet_skip" value to find the stream next packet, as they are arbitrarily interleaved.
* We only need to follow the first stream, as all must contain the same number of samples.
*
* XMA1/XMA2/WMAPRO data only differs in the packet headers.
*/
static void ms_audio_get_samples(ms_sample_data * msd, STREAMFILE *streamFile, int channels_per_packet, int bytes_per_packet, int samples_per_frame, int samples_per_subframe, int bits_frame_size) {
int frames = 0, samples = 0, loop_start_frame = 0, loop_end_frame = 0;
size_t first_frame_b, packet_skip_count, header_size_b, frame_size_b;
int64_t offset_b, packet_offset_b, frame_offset_b;
size_t packet_size = bytes_per_packet;
size_t packet_size_b = packet_size * 8;
off_t offset = msd->data_offset;
off_t max_offset = msd->data_offset + msd->data_size;
off_t stream_offset_b = msd->data_offset * 8;
/* read packets */
while (offset < max_offset) {
offset_b = offset * 8; /* global offset in bits */
offset += packet_size; /* global offset in bytes */
/* packet header */
ms_audio_parse_header(streamFile, msd->xma_version, offset_b, bits_frame_size, &first_frame_b, &packet_skip_count, &header_size_b);
if (packet_skip_count > 0x7FF) {
continue; /* full skip */
}
packet_offset_b = header_size_b + first_frame_b;
/* skip packets not owned by the first stream for next time */
offset += packet_size * (packet_skip_count);
/* read packet frames */
while (packet_offset_b < packet_size_b) {
frame_offset_b = offset_b + packet_offset_b; /* in bits for aligment stuff */
/* frame loops, later adjusted with subframes (seems correct vs tests) */
if (msd->loop_flag && (offset_b + packet_offset_b) - stream_offset_b == msd->loop_start_b)
loop_start_frame = frames;
if (msd->loop_flag && (offset_b + packet_offset_b) - stream_offset_b == msd->loop_end_b)
loop_end_frame = frames;
/* frame header */
frame_size_b = read_bitsBE_b(frame_offset_b, bits_frame_size, streamFile);
frame_offset_b += bits_frame_size;
/* stop when packet padding starts (0x00 for XMA1 or 0xFF in XMA2) */
if (frame_size_b == 0 || frame_size_b == (0xffffffff >> (32 - bits_frame_size))) {
break;
}
packet_offset_b += frame_size_b; /* including header */
samples += samples_per_frame;
frames++;
/* last bit in frame = more frames flag, end packet to avoid reading garbage in some cases
* (last frame spilling to other packets also has this flag, though it's ignored here) */
if (packet_offset_b < packet_size_b && !read_bitsBE_b(offset_b + packet_offset_b - 1, 1, streamFile)) {
break;
}
}
}
/* result */
msd->num_samples = samples;
if (msd->loop_flag && loop_end_frame > loop_start_frame) {
msd->loop_start_sample = loop_start_frame * samples_per_frame + msd->loop_start_subframe * samples_per_subframe;
msd->loop_end_sample = loop_end_frame * samples_per_frame + (msd->loop_end_subframe) * samples_per_subframe;
}
/* the above can't properly read skips for WMAPro ATM, but should fixed to 1 frame anyway */
if (msd->xma_version == 0) {
msd->num_samples -= samples_per_frame; /* FFmpeg does skip this */
#if 0
msd->num_samples += (samples_per_frame / 2); /* but doesn't add extra samples */
#endif
}
}
/* simlar to the above but only gets skips */
static void ms_audio_get_skips(STREAMFILE *streamFile, int xma_version, off_t data_offset, int channels_per_packet, int bytes_per_packet, int samples_per_frame, int bits_frame_size, int *out_start_skip, int *out_end_skip) {
int start_skip = 0, end_skip = 0;
size_t first_frame_b, packet_skip_count, header_size_b, frame_size_b;
int64_t offset_b, packet_offset_b, frame_offset_b;
size_t packet_size = bytes_per_packet;
size_t packet_size_b = packet_size * 8;
int64_t offset = data_offset;
/* read packet */
{
offset_b = offset * 8; /* global offset in bits */
offset += packet_size; /* global offset in bytes */
/* packet header */
ms_audio_parse_header(streamFile, 2, offset_b, bits_frame_size, &first_frame_b, &packet_skip_count, &header_size_b);
if (packet_skip_count > 0x7FF) {
return; /* full skip */
}
packet_offset_b = header_size_b + first_frame_b;
/* read packet frames */
while (packet_offset_b < packet_size_b) {
frame_offset_b = offset_b + packet_offset_b; /* in bits for aligment stuff */
/* frame header */
frame_size_b = read_bitsBE_b(frame_offset_b, bits_frame_size, streamFile);
frame_offset_b += bits_frame_size;
/* stop when packet padding starts (0x00 for XMA1 or 0xFF in XMA2) */
if (frame_size_b == 0 || frame_size_b == (0xffffffff >> (32 - bits_frame_size))) {
break;
}
packet_offset_b += frame_size_b; /* including header */
/* find skips (info from FFmpeg) */
if (channels_per_packet && (xma_version == 1 || xma_version == 2)) {
int flag;
int len_tilehdr_size = 15; //todo incorrect but usable for XMA, fix for WMAPro (complex, see ffmpeg decode_tilehdr)
frame_offset_b += len_tilehdr_size;
/* ignore "postproc transform" */
if (channels_per_packet > 1) {
flag = read_bitsBE_b(frame_offset_b, 1, streamFile);
frame_offset_b += 1;
if (flag) {
flag = read_bitsBE_b(frame_offset_b, 1, streamFile);
frame_offset_b += 1;
if (flag) {
frame_offset_b += 1 + 4 * channels_per_packet*channels_per_packet; /* 4-something per double channel? */
}
}
}
/* get start/end skips to get the proper number of samples (both can be 0) */
flag = read_bitsBE_b(frame_offset_b, 1, streamFile);
frame_offset_b += 1;
if (flag) {
/* get start skip */
flag = read_bitsBE_b(frame_offset_b, 1, streamFile);
frame_offset_b += 1;
if (flag) {
int new_skip = read_bitsBE_b(frame_offset_b, 10, streamFile);
//;VGM_LOG("MS_SAMPLES: start_skip %i at 0x%x (bit 0x%x)\n", new_skip, (uint32_t)frame_offset_b/8, (uint32_t)frame_offset_b);
frame_offset_b += 10;
if (new_skip > samples_per_frame) /* from xmaencode */
new_skip = samples_per_frame;
if (start_skip==0) /* only use first skip */
start_skip = new_skip;
}
/* get end skip */
flag = read_bitsBE_b(frame_offset_b, 1, streamFile);
frame_offset_b += 1;
if (flag) {
int new_skip = read_bitsBE_b(frame_offset_b, 10, streamFile);
//;VGM_LOG("MS_SAMPLES: end_skip %i at 0x%x (bit 0x%x)\n", new_skip, (uint32_t)frame_offset_b/8, (uint32_t)frame_offset_b);
frame_offset_b += 10;
if (new_skip > samples_per_frame) /* from xmaencode */
new_skip = samples_per_frame;
end_skip = new_skip; /* always use last skip */
}
}
}
}
}
/* output results */
if (out_start_skip) *out_start_skip = start_skip;
if (out_end_skip) *out_end_skip = end_skip;
}
static int wma_get_samples_per_frame(int version, int sample_rate, uint32_t decode_flags) {
int frame_len_bits;
if (sample_rate <= 16000)
frame_len_bits = 9;
else if (sample_rate <= 22050 || (sample_rate <= 32000 && version == 1))
frame_len_bits = 10;
else if (sample_rate <= 48000 || version < 3)
frame_len_bits = 11;
else if (sample_rate <= 96000)
frame_len_bits = 12;
else
frame_len_bits = 13;
if (version == 3) {
int tmp = decode_flags & 0x6;
if (tmp == 0x2)
++frame_len_bits;
else if (tmp == 0x4)
--frame_len_bits;
else if (tmp == 0x6)
frame_len_bits -= 2;
}
return 1 << frame_len_bits;
}
static int xma_get_channels_per_stream(STREAMFILE* streamFile, off_t chunk_offset, int channels) {
int start_stream = 0;
int channels_per_stream = 0;
/* get from stream config (needed to find skips) */
if (chunk_offset) {
int format = read_16bitLE(chunk_offset,streamFile);
if (format == 0x0165 || format == 0x6501) { /* XMA1 */
channels_per_stream = read_8bit(chunk_offset + 0x0C + 0x14*start_stream + 0x11,streamFile);
} else if (format == 0x0166 || format == 0x6601) { /* new XMA2 */
channels_per_stream = channels > 1 ? 2 : 1;
} else { /* old XMA2 */
int version = read_8bit(chunk_offset,streamFile);
channels_per_stream = read_8bit(chunk_offset + 0x20 + (version==3 ? 0x00 : 0x08) + 0x4*start_stream + 0x00,streamFile);
}
}
else if (channels) {
channels_per_stream = channels == 1 ? 1 : 2; /* default for XMA without RIFF chunks, most common */
}
if (channels_per_stream > 2)
channels_per_stream = 0;
return channels_per_stream;
}
void xma_get_samples(ms_sample_data * msd, STREAMFILE *streamFile) {
const int bytes_per_packet = 2048;
const int samples_per_frame = 512;
const int samples_per_subframe = 128;
const int bits_frame_size = 15;
int channels_per_stream = xma_get_channels_per_stream(streamFile, msd->chunk_offset, msd->channels);
ms_audio_get_samples(msd, streamFile, channels_per_stream, bytes_per_packet, samples_per_frame, samples_per_subframe, bits_frame_size);
}
void wmapro_get_samples(ms_sample_data * msd, STREAMFILE *streamFile, int block_align, int sample_rate, uint32_t decode_flags) {
const int version = 3; /* WMAPRO = WMAv3 */
int bytes_per_packet = block_align;
int samples_per_frame = 0;
int samples_per_subframe = 0;
int bits_frame_size = 0;
int channels_per_stream = msd->channels;
if (!(decode_flags & 0x40)) {
VGM_LOG("MS_SAMPLES: no frame length in WMAPro\n");
msd->num_samples = 0;
return;
}
samples_per_frame = wma_get_samples_per_frame(version, sample_rate, decode_flags);
bits_frame_size = (int)floor(log(block_align) / log(2)) + 4; /* max bits needed to represent this block_align */
samples_per_subframe = 0; /* not needed as WMAPro can't use loop subframes (complex subframe lengths) */
msd->xma_version = 0; /* signal it's not XMA */
ms_audio_get_samples(msd, streamFile, channels_per_stream, bytes_per_packet, samples_per_frame, samples_per_subframe, bits_frame_size);
}
void wma_get_samples(ms_sample_data * msd, STREAMFILE *streamFile, int block_align, int sample_rate, uint32_t decode_flags) {
const int version = 2; /* WMAv1 rarely used */
int use_bit_reservoir = 0; /* last packet frame can spill into the next packet */
int samples_per_frame = 0;
int num_frames = 0;
samples_per_frame = wma_get_samples_per_frame(version, sample_rate, decode_flags);
/* assumed (ASF has a flag for this but XWMA doesn't) */
if (version == 2)
use_bit_reservoir = 1;
if (!use_bit_reservoir) {
/* 1 frame per packet */
num_frames = msd->data_size / block_align + (msd->data_size % block_align ? 1 : 0);
}
else {
/* variable frames per packet (mini-header values) */
off_t offset = msd->data_offset;
off_t max_offset = msd->data_offset + msd->data_size;
while (offset < max_offset) { /* read packets (superframes) */
int packet_frames;
uint8_t header = read_8bit(offset, streamFile); /* upper nibble: index; lower nibble: frames */
/* frames starting in this packet (ie. counts frames that spill due to bit_reservoir) */
packet_frames = (header & 0xf);
num_frames += packet_frames;
offset += block_align;
}
}
msd->num_samples = num_frames * samples_per_frame;
#if 0 //todo apply once FFmpeg decode is ok
msd->num_samples += (samples_per_frame / 2); /* last IMDCT samples */
msd->num_samples -= (samples_per_frame * 2); /* WMA default encoder delay */
#endif
}
/* XMA hell for precise looping and gapless support, fixes raw sample values from headers
* that don't count XMA's final subframe/encoder delay/encoder padding, and FFmpeg stuff.
* Configurable since different headers vary for maximum annoyance. */
void xma_fix_raw_samples_ch(VGMSTREAM *vgmstream, STREAMFILE*streamFile, off_t stream_offset, size_t stream_size, int channels_per_stream, int fix_num_samples, int fix_loop_samples) {
const int bytes_per_packet = 2048;
const int samples_per_frame = 512;
const int samples_per_subframe = 128;
const int bits_frame_size = 15;
int xma_version = 2; /* works ok even for XMA1 */
off_t first_packet = stream_offset;
off_t last_packet = stream_offset + stream_size - bytes_per_packet;
int32_t start_skip = 0, end_skip = 0;
if (stream_offset + stream_size > get_streamfile_size(streamFile)) {
VGM_LOG("XMA SKIPS: ignoring bad stream offset+size vs real size\n");
return;
}
/* find delay/padding values in the bitstream (should be safe even w/ multistreams
* as every stream repeats them). Theoretically every packet could contain skips,
* doesn't happen in practice though. */
ms_audio_get_skips(streamFile, xma_version, first_packet, channels_per_stream, bytes_per_packet, samples_per_frame, bits_frame_size, &start_skip, NULL);
ms_audio_get_skips(streamFile, xma_version, last_packet, channels_per_stream, bytes_per_packet, samples_per_frame, bits_frame_size, NULL, &end_skip);
//;VGM_LOG("XMA SKIPS: apply start=%i, end=%i\n", start_skip, end_skip);
VGM_ASSERT(start_skip < samples_per_frame, "XMA SKIPS: small start skip\n");
if (end_skip == 512) { /* most likely a read bug */
VGM_LOG("XMA SKIPS: ignoring big end_skip\n");
end_skip = 0;
}
/* apply XMA extra samples */
if (fix_num_samples) {
vgmstream->num_samples += samples_per_subframe; /* final extra IMDCT samples */
vgmstream->num_samples -= start_skip; /* first samples skipped at the beginning */
vgmstream->num_samples -= end_skip; /* last samples discarded at the end */
}
/* from xma2encode tests this is correct (probably encodes/decodes loops considering all skips), ex.-
* full loop wav to xma makes start=384 (0 + ~512 delay - 128 padding), then xma to wav has "smpl" start=0 */
if (fix_loop_samples && vgmstream->loop_flag) {
vgmstream->loop_start_sample += samples_per_subframe;
vgmstream->loop_start_sample -= start_skip;
vgmstream->loop_end_sample += samples_per_subframe;
vgmstream->loop_end_sample -= start_skip;
/* since loops are adjusted this shouldn't happen (often loop_end == num_samples after applying all) */
if (vgmstream->loop_end_sample > vgmstream->num_samples &&
vgmstream->loop_end_sample - end_skip <= vgmstream->loop_end_sample) {
VGM_LOG("XMA SAMPLES: adjusted loop end\n");
vgmstream->loop_end_sample -= end_skip;
}
}
#ifdef VGM_USE_FFMPEG
/* also fix FFmpeg, since we now know exact skips */
{
ffmpeg_codec_data* data = vgmstream->codec_data;
/* FFmpeg doesn't XMA apply encoder delay ATM so here we fix it manually.
* XMA delay is part if the bitstream and while theoretically it could be any
* value (and is honored by xmaencoder), basically it's always 512.
*
* Somehow also needs to skip 64 extra samples (looks like another FFmpeg bug
* where XMA outputs half a subframe samples late, WMAPRO isn't affected),
* which sometimes makes FFmpeg complain (=reads after end) but doesn't seem audible. */
if (data->skipSamples == 0) {
ffmpeg_set_skip_samples(data, start_skip+64);
}
}
#endif
}
void xma_fix_raw_samples_hb(VGMSTREAM *vgmstream, STREAMFILE *headerFile, STREAMFILE *bodyFile, off_t stream_offset, size_t stream_size, off_t chunk_offset, int fix_num_samples, int fix_loop_samples) {
int channels_per_stream = xma_get_channels_per_stream(headerFile, chunk_offset, vgmstream->channels);
xma_fix_raw_samples_ch(vgmstream, bodyFile, stream_offset, stream_size, channels_per_stream, fix_num_samples, fix_loop_samples);
}
void xma_fix_raw_samples(VGMSTREAM *vgmstream, STREAMFILE*streamFile, off_t stream_offset, size_t stream_size, off_t chunk_offset, int fix_num_samples, int fix_loop_samples) {
int channels_per_stream = xma_get_channels_per_stream(streamFile, chunk_offset, vgmstream->channels);
xma_fix_raw_samples_ch(vgmstream, streamFile, stream_offset, stream_size, channels_per_stream, fix_num_samples, fix_loop_samples);
}
/* ******************************************** */
/* HEADER PARSING */
/* ******************************************** */
/* Read values from a XMA1 RIFF "fmt" chunk (XMAWAVEFORMAT), starting from an offset *after* chunk type+size.
* Useful as custom X360 headers commonly have it lurking inside. */
void xma1_parse_fmt_chunk(STREAMFILE *streamFile, off_t chunk_offset, int * channels, int * sample_rate, int * loop_flag, int32_t * loop_start_b, int32_t * loop_end_b, int32_t * loop_subframe, int be) {
int16_t (*read_16bit)(off_t,STREAMFILE*) = be ? read_16bitBE : read_16bitLE;
int32_t (*read_32bit)(off_t,STREAMFILE*) = be ? read_32bitBE : read_32bitLE;
int i, num_streams, total_channels = 0;
if (read_16bit(chunk_offset+0x00,streamFile) != 0x165)
return;
num_streams = read_16bit(chunk_offset+0x08,streamFile);
if(loop_flag) *loop_flag = (uint8_t)read_8bit(chunk_offset+0xA,streamFile) > 0;
/* sample rate and loop bit offsets are defined per stream, but the first is enough */
if(sample_rate) *sample_rate = read_32bit(chunk_offset+0x10,streamFile);
if(loop_start_b) *loop_start_b = read_32bit(chunk_offset+0x14,streamFile);
if(loop_end_b) *loop_end_b = read_32bit(chunk_offset+0x18,streamFile);
if(loop_subframe) *loop_subframe = (uint8_t)read_8bit(chunk_offset+0x1C,streamFile);
/* channels is the sum of all streams */
for (i = 0; i < num_streams; i++) {
total_channels += read_8bit(chunk_offset+0x0C+0x14*i+0x11,streamFile);
}
if(channels) *channels = total_channels;
}
/* Read values from a 'new' XMA2 RIFF "fmt" chunk (XMA2WAVEFORMATEX), starting from an offset *after* chunk type+size.
* Useful as custom X360 headers commonly have it lurking inside. Only parses the extra data (before is a normal WAVEFORMATEX). */
void xma2_parse_fmt_chunk_extra(STREAMFILE *streamFile, off_t chunk_offset, int * out_loop_flag, int32_t * out_num_samples, int32_t * out_loop_start_sample, int32_t * out_loop_end_sample, int be) {
int16_t (*read_16bit)(off_t,STREAMFILE*) = be ? read_16bitBE : read_16bitLE;
int32_t (*read_32bit)(off_t,STREAMFILE*) = be ? read_32bitBE : read_32bitLE;
int num_samples, loop_start_sample, loop_end_sample, loop_flag;
if (read_16bit(chunk_offset+0x00,streamFile) != 0x166)
return;
if (read_16bit(chunk_offset+0x10,streamFile) < 0x22)
return; /* expected extra data size */
num_samples = read_32bit(chunk_offset+0x18,streamFile);
loop_start_sample = read_32bit(chunk_offset+0x28,streamFile);
loop_end_sample = loop_start_sample + read_32bit(chunk_offset+0x2C,streamFile);
loop_flag = (uint8_t)read_8bit(chunk_offset+0x30,streamFile) != 0;
/* may need loop end +1, though some header doesn't need it (ex.- Sonic and Sega All Stars Racing .str) */
/* flag rarely set, use loop_end as marker */
if (!loop_flag) {
loop_flag = loop_end_sample > 0;
/* some XMA incorrectly do full loops for every song/jingle [Shadows of the Damned (X360)] */
if ((loop_start_sample + 128 - 512) == 0 && (loop_end_sample + 128 - 512) + 256 >= (num_samples + 128 - 512)) {
VGM_LOG("XMA2 PARSE: disabling full loop\n");
loop_flag = 0;
}
}
/* samples are "raw" values, must be fixed externally (see xma_fix_raw_samples) */
if(out_num_samples) *out_num_samples = num_samples;
if(out_loop_start_sample) *out_loop_start_sample = loop_start_sample;
if(out_loop_end_sample) *out_loop_end_sample = loop_end_sample;
if(out_loop_flag) *out_loop_flag = loop_flag;
/* play_begin+end = pcm_samples in original sample rate (not usable as file may be resampled) */
/* int32_t play_begin_sample = read_32bit(xma->chunk_offset+0x20,streamFile); */
/* int32_t play_end_sample = play_begin_sample + read_32bit(xma->chunk_offset+0x24,streamFile); */
}
/* Read values from an 'old' XMA2 RIFF "XMA2" chunk (XMA2WAVEFORMAT), starting from an offset *after* chunk type+size.
* Useful as custom X360 headers commonly have it lurking inside. */
void xma2_parse_xma2_chunk(STREAMFILE *streamFile, off_t chunk_offset, int * out_channels, int * out_sample_rate, int * out_loop_flag, int32_t * out_num_samples, int32_t * out_loop_start_sample, int32_t * out_loop_end_sample) {
int32_t (*read_32bit)(off_t,STREAMFILE*) = read_32bitBE; /* XMA2WAVEFORMAT is always big endian */
int i, xma2_chunk_version, num_streams;
int channels, sample_rate, loop_flag, num_samples, loop_start_sample, loop_end_sample;
off_t offset;
xma2_chunk_version = read_8bit(chunk_offset+0x00,streamFile);
num_streams = read_8bit(chunk_offset+0x01,streamFile);
loop_start_sample = read_32bit(chunk_offset+0x04,streamFile);
loop_end_sample = read_32bit(chunk_offset+0x08,streamFile);
loop_flag = (uint8_t)read_8bit(chunk_offset+0x03,streamFile) > 0 || loop_end_sample; /* rarely not set, encoder default */
sample_rate = read_32bit(chunk_offset+0x0c,streamFile);
/* may need loop end +1 */
offset = xma2_chunk_version == 3 ? 0x14 : 0x1C;
num_samples = read_32bit(chunk_offset+offset+0x00,streamFile);
/* pcm_samples in original sample rate (not usable as file may be resampled) */
/* pcm_samples = read_32bitBE(chunk_offset+offset+0x04,streamFile)*/
offset = xma2_chunk_version == 3 ? 0x20 : 0x28;
channels = 0; /* channels is the sum of all streams */
for (i = 0; i < num_streams; i++) {
channels += read_8bit(chunk_offset+offset+i*0x04,streamFile);
}
/* samples are "raw" values, must be fixed externally (see xma_fix_raw_samples) */
if(out_channels) *out_channels = channels;
if(out_sample_rate) *out_sample_rate = sample_rate;
if(out_num_samples) *out_num_samples = num_samples;
if(out_loop_start_sample) *out_loop_start_sample = loop_start_sample;
if(out_loop_end_sample) *out_loop_end_sample = loop_end_sample;
if(out_loop_flag) *out_loop_flag = loop_flag;
}
/* ******************************************** */
/* OTHER STUFF */
/* ******************************************** */
size_t atrac3_bytes_to_samples(size_t bytes, int full_block_align) {
if (full_block_align <= 0) return 0;
/* ATRAC3 expects full block align since as is can mix joint stereo with mono blocks;
* so (full_block_align / channels) DOESN'T give the size of a single channel (uncommon in ATRAC3 though) */
return (bytes / full_block_align) * 1024;
}
size_t atrac3plus_bytes_to_samples(size_t bytes, int full_block_align) {
if (full_block_align <= 0) return 0;
/* ATRAC3plus expects full block align since as is can mix joint stereo with mono blocks;
* so (full_block_align / channels) DOESN'T give the size of a single channel (common in ATRAC3plus) */
return (bytes / full_block_align) * 2048;
}
size_t ac3_bytes_to_samples(size_t bytes, int full_block_align, int channels) {
if (full_block_align <= 0) return 0;
return (bytes / full_block_align) * 256 * channels;
}
size_t aac_get_samples(STREAMFILE *streamFile, off_t start_offset, size_t bytes) {
const int samples_per_frame = 1024; /* theoretically 960 exists in .MP4 so may need a flag */
int frames = 0;
off_t offset = start_offset;
off_t max_offset = start_offset + bytes;
if (!streamFile)
return 0;
if (max_offset > get_streamfile_size(streamFile))
max_offset = get_streamfile_size(streamFile);
/* AAC sometimes comes with an "ADIF" header right before data but probably not in games,
* while standard raw frame headers are called "ADTS" and are similar to MPEG's:
* (see https://wiki.multimedia.cx/index.php/ADTS) */
/* AAC uses VBR so must read all frames */
while (offset < max_offset) {
uint16_t frame_sync = read_u16be(offset+0x00, streamFile);
uint32_t frame_size = read_u32be(offset+0x02, streamFile);
frame_sync = (frame_sync >> 4) & 0x0FFF; /* 12b */
frame_size = (frame_size >> 5) & 0x1FFF; /* 13b */
if (frame_sync != 0xFFF)
break;
if (frame_size <= 0x08)
break;
frames++;
offset += frame_size;
}
return frames * samples_per_frame;
}
/* ******************************************** */
/* BITSTREAM */
/* ******************************************** */
/* Read bits (max 32) from buf and update the bit offset. Vorbis packs values in LSB order and byte by byte.
* (ex. from 2 bytes 00100111 00000001 we can could read 4b=0111 and 6b=010010, 6b=remainder (second value is split into the 2nd byte) */
static int r_bits_vorbis(vgm_bitstream * ib, int num_bits, uint32_t * value) {
off_t off, pos;
int i, bit_buf, bit_val;
if (num_bits == 0) return 1;
if (num_bits > 32 || num_bits < 0 || ib->b_off + num_bits > ib->bufsize*8) goto fail;
*value = 0; /* set all bits to 0 */
off = ib->b_off / 8; /* byte offset */
pos = ib->b_off % 8; /* bit sub-offset */
for (i = 0; i < num_bits; i++) {
bit_buf = (1U << pos) & 0xFF; /* bit check for buf */
bit_val = (1U << i); /* bit to set in value */
if (ib->buf[off] & bit_buf) /* is bit in buf set? */
*value |= bit_val; /* set bit */
pos++; /* new byte starts */
if (pos%8 == 0) {
pos = 0;
off++;
}
}
ib->b_off += num_bits;
return 1;
fail:
return 0;
}
/* Write bits (max 32) to buf and update the bit offset. Vorbis packs values in LSB order and byte by byte.
* (ex. writing 1101011010 from b_off 2 we get 01101011 00001101 (value split, and 11 in the first byte skipped)*/
static int w_bits_vorbis(vgm_bitstream * ob, int num_bits, uint32_t value) {
off_t off, pos;
int i, bit_val, bit_buf;
if (num_bits == 0) return 1;
if (num_bits > 32 || num_bits < 0 || ob->b_off + num_bits > ob->bufsize*8) goto fail;
off = ob->b_off / 8; /* byte offset */
pos = ob->b_off % 8; /* bit sub-offset */
for (i = 0; i < num_bits; i++) {
bit_val = (1U << i); /* bit check for value */
bit_buf = (1U << pos) & 0xFF; /* bit to set in buf */
if (value & bit_val) /* is bit in val set? */
ob->buf[off] |= bit_buf; /* set bit */
else
ob->buf[off] &= ~bit_buf; /* unset bit */
pos++; /* new byte starts */
if (pos%8 == 0) {
pos = 0;
off++;
}
}
ob->b_off += num_bits;
return 1;
fail:
return 0;
}
/* Read bits (max 32) from buf and update the bit offset. Order is BE (MSF). */
static int r_bits_msf(vgm_bitstream * ib, int num_bits, uint32_t * value) {
off_t off, pos;
int i, bit_buf, bit_val;
if (num_bits == 0) return 1;
if (num_bits > 32 || num_bits < 0 || ib->b_off + num_bits > ib->bufsize*8) goto fail;
*value = 0; /* set all bits to 0 */
off = ib->b_off / 8; /* byte offset */
pos = ib->b_off % 8; /* bit sub-offset */
for (i = 0; i < num_bits; i++) {
bit_buf = (1U << (8-1-pos)) & 0xFF; /* bit check for buf */
bit_val = (1U << (num_bits-1-i)); /* bit to set in value */
if (ib->buf[off] & bit_buf) /* is bit in buf set? */
*value |= bit_val; /* set bit */
pos++;
if (pos%8 == 0) { /* new byte starts */
pos = 0;
off++;
}
}
ib->b_off += num_bits;
return 1;
fail:
return 0;
}
/* Write bits (max 32) to buf and update the bit offset. Order is BE (MSF). */
static int w_bits_msf(vgm_bitstream * ob, int num_bits, uint32_t value) {
off_t off, pos;
int i, bit_val, bit_buf;
if (num_bits == 0) return 1;
if (num_bits > 32 || num_bits < 0 || ob->b_off + num_bits > ob->bufsize*8) goto fail;
off = ob->b_off / 8; /* byte offset */
pos = ob->b_off % 8; /* bit sub-offset */
for (i = 0; i < num_bits; i++) {
bit_val = (1U << (num_bits-1-i)); /* bit check for value */
bit_buf = (1U << (8-1-pos)) & 0xFF; /* bit to set in buf */
if (value & bit_val) /* is bit in val set? */
ob->buf[off] |= bit_buf; /* set bit */
else
ob->buf[off] &= ~bit_buf; /* unset bit */
pos++;
if (pos%8 == 0) { /* new byte starts */
pos = 0;
off++;
}
}
ob->b_off += num_bits;
return 1;
fail:
return 0;
}
int r_bits(vgm_bitstream * ib, int num_bits, uint32_t * value) {
if (ib->mode == BITSTREAM_VORBIS)
return r_bits_vorbis(ib,num_bits,value);
else
return r_bits_msf(ib,num_bits,value);
}
int w_bits(vgm_bitstream * ob, int num_bits, uint32_t value) {
if (ob->mode == BITSTREAM_VORBIS)
return w_bits_vorbis(ob,num_bits,value);
else
return w_bits_msf(ob,num_bits,value);
}
/* ******************************************** */
/* CUSTOM STREAMFILES */
/* ******************************************** */
STREAMFILE* setup_subfile_streamfile(STREAMFILE *sf, off_t subfile_offset, size_t subfile_size, const char* extension) {
STREAMFILE *new_sf = NULL;
new_sf = open_wrap_streamfile(sf);
new_sf = open_clamp_streamfile_f(new_sf, subfile_offset, subfile_size);
if (extension) {
new_sf = open_fakename_streamfile_f(new_sf, NULL, extension);
}
return new_sf;
}