Merge pull request #324 from bnnm/genh-sli-xps

genh sli xps
This commit is contained in:
Christopher Snowhill 2018-12-01 13:11:42 -08:00 committed by GitHub
commit 23a35175ed
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GPG Key ID: 4AEE18F83AFDEB23
38 changed files with 444 additions and 142 deletions

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@ -30,7 +30,7 @@
**With GCC**: use the *./Makefile* in the root folder, see inside for options. For compilation flags check the *Makefile* in each folder.
You may need to manually rebuild if you change a *.h* file (use *make clean*).
In Linux, Makefiles can be used to cross-compile with the MingW headers, but may not be updated to generate native code at the moment. It should be fixable with some effort. Autotools should build it as vgmstream-cli instead (see the Audacious section).
In Linux, Makefiles can be used to cross-compile with the MingW headers, but may not be updated to generate native code at the moment. It should be fixable with some effort. Autotools should build it as vgmstream-cli instead (see the Audacious section). Some Linux distributions like Arch Linux include pre-patched vgmstream with most libraries, you may want that instead (see: https://aur.archlinux.org/packages/vgmstream-kode54-git/).
Windows CMD example:
```

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@ -1551,9 +1551,9 @@ static const unsigned int decode5_imdct_window_int[128] = {
static const float *decode5_imdct_window = (const float *)decode5_imdct_window_int;
static void decoder5_run_imdct(stChannel *ch, int subframe) {
const static unsigned int size = HCA_SAMPLES_PER_SUBFRAME;
const static unsigned int half = HCA_SAMPLES_PER_SUBFRAME / 2;
const static unsigned int mdct_bits = HCA_MDCT_BITS;
static const unsigned int size = HCA_SAMPLES_PER_SUBFRAME;
static const unsigned int half = HCA_SAMPLES_PER_SUBFRAME / 2;
static const unsigned int mdct_bits = HCA_MDCT_BITS;
/* apply DCT-IV to dequantized spectra */

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@ -128,7 +128,7 @@ void decode_atrac9(VGMSTREAM *vgmstream, sample * outbuf, int32_t samples_to_do,
decode_fail:
/* on error just put some 0 samples */
VGM_LOG("ATRAC9: decode fail at %"PRIx64", missing %i samples\n", (off64_t)stream->offset, (samples_to_do - samples_done));
VGM_LOG("ATRAC9: decode fail at %x, missing %i samples\n", (uint32_t)stream->offset, (samples_to_do - samples_done));
memset(outbuf + samples_done * channels, 0, (samples_to_do - samples_done) * sizeof(sample) * channels);
}

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@ -157,7 +157,7 @@ void decode_celt_fsb(VGMSTREAM *vgmstream, sample * outbuf, int32_t samples_to_d
decode_fail:
/* on error just put some 0 samples */
VGM_LOG("CELT: decode fail at %"PRIx64", missing %i samples\n", (off64_t)stream->offset, (samples_to_do - samples_done));
VGM_LOG("CELT: decode fail at %x, missing %i samples\n", (uint32_t)stream->offset, (samples_to_do - samples_done));
memset(outbuf + samples_done * channels, 0, (samples_to_do - samples_done) * sizeof(sample) * channels);
}

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@ -470,17 +470,17 @@ static void ms_audio_parse_header(STREAMFILE *streamFile, int xma_version, off_t
/* 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 at 0x%"PRIx64"\n", (off64_t)offset_b/8);
VGM_LOG("MS_SAMPLES: XMA1 full packet_skip at 0x%x\n", (uint32_t)offset_b/8);
*packet_skip_count = 0x800;
}
else if (*packet_skip_count == 0xFF) { /* XMA2, 8b*/
VGM_LOG("MS_SAMPLES: XMA2 full packet_skip at 0x%"PRIx64"\n", (off64_t)offset_b/8);
VGM_LOG("MS_SAMPLES: XMA2 full packet_skip at 0x%x\n", (uint32_t)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%"PRIx64"\n", *packet_skip_count, (off64_t)offset_b/8);
"MS_SAMPLES: found big packet skip %i at 0x%x\n", *packet_skip_count, (uint32_t)offset_b/8);
}
/**
@ -636,7 +636,7 @@ static void ms_audio_get_skips(STREAMFILE *streamFile, int xma_version, off_t da
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%"PRIx64" (bit 0x%"PRIx64")\n", new_skip, (off64_t)frame_offset_b/8, (off64_t)frame_offset_b);
//;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 */
@ -651,7 +651,7 @@ static void ms_audio_get_skips(STREAMFILE *streamFile, int xma_version, off_t da
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%"PRIx64" (bit 0x%"PRIx64")\n", new_skip, (off64_t)frame_offset_b/8, (off64_t)frame_offset_b);
//;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 */

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@ -119,7 +119,7 @@ static size_t opus_io_read(STREAMFILE *streamfile, uint8_t *dest, off_t offset,
data->page_size = oggs_size + data_size;
if (data->page_size > sizeof(data->page_buffer)) { /* happens on bad reads/EOF too */
VGM_LOG("OPUS: buffer can't hold OggS at %"PRIx64"\n", (off64_t)data->physical_offset);
VGM_LOG("OPUS: buffer can't hold OggS at %x\n", (uint32_t)data->physical_offset);
data->page_size = 0;
break;
}
@ -173,7 +173,7 @@ static size_t opus_io_size(STREAMFILE *streamfile, opus_io_data* data) {
return data->logical_size;
if (data->stream_offset + data->stream_size > get_streamfile_size(streamfile)) {
VGM_LOG("OPUS: wrong streamsize %"PRIx64" + %x vs %x\n", (off64_t)data->stream_offset, data->stream_size, get_streamfile_size(streamfile));
VGM_LOG("OPUS: wrong streamsize %x + %x vs %x\n", (uint32_t)data->stream_offset, data->stream_size, get_streamfile_size(streamfile));
return 0;
}
@ -205,6 +205,12 @@ static size_t opus_io_size(STREAMFILE *streamfile, opus_io_data* data) {
default:
return 0;
}
if (data_size == 0 ) {
VGM_LOG("OPUS: data_size is 0 at %x\n", (uint32_t)physical_offset);
return 0; /* bad rip? or could 'break' and truck along */
}
oggs_size = 0x1b + (int)(data_size / 0xFF + 1); /* OggS page: base size + lacing values */
physical_offset += data_size + skip_size;
@ -501,14 +507,22 @@ fail:
return 0;
}
/************************** */
#ifdef VGM_USE_FFMPEG
static size_t opus_get_packet_samples(const uint8_t * buf, int len) {
return opus_packet_get_nb_frames(buf, len) * opus_packet_get_samples_per_frame(buf, 48000);
}
/************************** */
static size_t get_xopus_packet_size(int packet, STREAMFILE * streamfile) {
/* XOPUS has a packet size table at the beginning, get size from there.
* Maybe should copy the table during setup to avoid IO, but all XOPUS are
* quite small so it isn't very noticeable. */
return (uint16_t)read_16bitLE(0x20 + packet*0x02, streamfile);
}
#ifdef VGM_USE_FFMPEG
static size_t custom_opus_get_samples(off_t offset, size_t data_size, STREAMFILE *streamFile, opus_type_t type) {
size_t num_samples = 0;
off_t end_offset = offset + data_size;
@ -596,13 +610,6 @@ size_t ea_opus_get_encoder_delay(off_t offset, STREAMFILE *streamFile) {
}
static size_t get_xopus_packet_size(int packet, STREAMFILE * streamfile) {
/* XOPUS has a packet size table at the beginning, get size from there.
* Maybe should copy the table during setup to avoid IO, but all XOPUS are
* quite small so it isn't very noticeable. */
return (uint16_t)read_16bitLE(0x20 + packet*0x02, streamfile);
}
/* ******************************************************* */
@ -616,9 +623,12 @@ static ffmpeg_codec_data * init_ffmpeg_custom_opus(STREAMFILE *streamFile, off_t
ffmpeg_data = init_ffmpeg_offset(temp_streamFile, 0x00,get_streamfile_size(temp_streamFile));
if (!ffmpeg_data) goto fail;
if (ffmpeg_data->skipSamples <= 0) {
ffmpeg_set_skip_samples(ffmpeg_data, skip);
}
/* FFmpeg + libopus: skips samples, notifies skip in codecCtx->delay (not in stream->skip_samples)
* FFmpeg + opus: *doesn't* skip, also notifies skip in codecCtx->delay, hurray (possibly fixed in recent versions)
* FFmpeg + opus is audibly buggy with some low bitrate SSB Ultimate files too */
//if (ffmpeg_data->skipSamples <= 0) {
// ffmpeg_set_skip_samples(ffmpeg_data, skip);
//}
close_streamfile(temp_streamFile);
return ffmpeg_data;

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@ -99,14 +99,14 @@ void decode_hca(hca_codec_data * data, sample * outbuf, int32_t samples_to_do) {
/* read frame */
bytes = read_streamfile(data->data_buffer, offset, blockSize, data->streamfile);
if (bytes != blockSize) {
VGM_LOG("HCA: read %x vs expected %x bytes at %"PRIx64"\n", bytes, blockSize, (off64_t)offset);
VGM_LOG("HCA: read %x vs expected %x bytes at %x\n", bytes, blockSize, (uint32_t)offset);
break;
}
/* decode frame */
status = clHCA_DecodeBlock(data->handle, (void*)(data->data_buffer), blockSize);
if (status < 0) {
VGM_LOG("HCA: decode fail at %"PRIx64", code=%i\n", (off64_t)offset, status);
VGM_LOG("HCA: decode fail at %x, code=%i\n", (uint32_t)offset, status);
break;
}

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@ -157,7 +157,7 @@ int mpeg_custom_parse_frame_default(VGMSTREAMCHANNEL *stream, mpeg_codec_data *d
}
VGM_ASSERT(data->streams_size > 1 && current_interleave != current_data_size+current_padding,
"MPEG FSB: %i streams with non-constant interleave found @ 0x%08"PRIx64"\n", data->streams_size, (off64_t)stream->offset);
"MPEG FSB: %i streams with non-constant interleave found @ 0x%08x\n", data->streams_size, (uint32_t)stream->offset);
break;
case MPEG_P3D: /* fixed interleave, not frame-aligned (ie. blocks may end/start in part of a frame) */

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@ -59,7 +59,7 @@ static size_t get_repeated_data_size(STREAMFILE *streamFile, off_t new_offset, o
}
fail:
VGM_LOG("AWC: can't find repeat size, new=0x%08"PRIx64", last=0x%08"PRIx64"\n", (off64_t)new_offset, (off64_t)last_offset);
VGM_LOG("AWC: can't find repeat size, new=0x%08x, last=0x%08x\n", (uint32_t)new_offset, (uint32_t)last_offset);
return 0; /* keep on truckin' */
}

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@ -529,7 +529,7 @@ static int ealayer3_rebuild_mpeg_frame(vgm_bitstream* is_0, ealayer3_frame_info*
if (os->b_off/8 > expected_frame_size) {
/* bit reservoir! shouldn't happen with free bitrate, otherwise it's hard to fix as needs complex buffering/calcs */
VGM_LOG("MPEG EAL3: written 0x%"PRIx64" but expected less than 0x%x at 0x%"PRIx64"\n", (off64_t)(os->b_off/8), expected_frame_size, (off64_t)os->info_offset);
VGM_LOG("MPEG EAL3: written 0x%x but expected less than 0x%x at 0x%x\n", (uint32_t)(os->b_off/8), expected_frame_size, (uint32_t)os->info_offset);
}
else {
/* fill ancillary data (should be ignored, but 0x00 seems to improve mpg123's free bitrate detection) */
@ -564,8 +564,8 @@ static int ealayer3_write_pcm_block(VGMSTREAMCHANNEL *stream, mpeg_codec_data *d
if (!eaf->pcm_size)
return 1;
VGM_ASSERT(eaf->v1_pcm_decode_discard > 576, "MPEG EAL3: big discard %i at 0x%"PRIx64"\n", eaf->v1_pcm_decode_discard, (off64_t)stream->offset);
VGM_ASSERT(eaf->v1_pcm_number > 0x100, "MPEG EAL3: big samples %i at 0x%"PRIx64"\n", eaf->v1_pcm_number, (off64_t)stream->offset);
VGM_ASSERT(eaf->v1_pcm_decode_discard > 576, "MPEG EAL3: big discard %i at 0x%x\n", eaf->v1_pcm_decode_discard, (uint32_t)stream->offset);
VGM_ASSERT(eaf->v1_pcm_number > 0x100, "MPEG EAL3: big samples %i at 0x%x\n", eaf->v1_pcm_number, (uint32_t)stream->offset);
/* read + write PCM block samples (always BE) */
for (i = 0; i < eaf->v1_pcm_number * data->channels_per_frame; i++) {

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@ -402,7 +402,7 @@ static void decode_mpeg_custom_stream(VGMSTREAMCHANNEL *stream, mpeg_codec_data
default: ok = mpeg_custom_parse_frame_default(stream, data, num_stream); break;
}
if (!ok) {
VGM_LOG("MPEG: cannot parse frame @ around %"PRIx64"\n",(off64_t)stream->offset);
VGM_LOG("MPEG: cannot parse frame @ around %x\n",(uint32_t)stream->offset);
goto decode_fail; /* mpg123 could resync but custom MPEGs wouldn't need that */
}
//;VGM_LOG("MPEG: read results: bytes_in_buffer=0x%x, new offset=%lx\n", ms->bytes_in_buffer, stream->offset);

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@ -79,7 +79,7 @@ static void mta2_block_update(VGMSTREAMCHANNEL * stream) {
}
if (block_size <= 0 || block_tracks < 0) { /* nonsense block (maybe at EOF) */
VGM_LOG("MTA2: bad block @ 0x%"PRIx64"\n", (off64_t)stream->offset);
VGM_LOG("MTA2: bad block @ 0x%x\n", (uint32_t)stream->offset);
stream->offset += 0x10;
repeat = 0;
}

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@ -109,7 +109,7 @@ static void mtaf_block_update(VGMSTREAMCHANNEL * stream) {
}
if (block_size <= 0 || block_tracks < 0) { /* nonsense block (maybe at EOF) */
VGM_LOG("MTAF: bad block @ %"PRIx64"\n", (off64_t)stream->offset);
VGM_LOG("MTAF: bad block @ %x\n", (uint32_t)stream->offset);
stream->offset += 0x10;
repeat = 0;
}
@ -146,7 +146,7 @@ void decode_mtaf(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing,
int32_t init_idx = read_16bitLE(stream->offset+4+0+c*2, stream->streamfile); /* step-L/R */
int32_t init_hist = read_16bitLE(stream->offset+4+4+c*4, stream->streamfile); /* hist-L/R: hist 16bit + empty 16bit */
VGM_ASSERT(init_idx < 0 || init_idx > 31, "MTAF: bad header idx @ 0x%"PRIx64"\n", (off64_t)stream->offset);
VGM_ASSERT(init_idx < 0 || init_idx > 31, "MTAF: bad header idx @ 0x%x\n", (uint32_t)stream->offset);
/* avoid index out of range in corrupt files */
if (init_idx < 0) {
init_idx = 0;

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@ -23,7 +23,7 @@ void decode_ngc_dtk(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspaci
coef_index = ((uint8_t)read_8bit(frame_offset+channel,stream->streamfile) >> 4) & 0xf;
shift_factor = ((uint8_t)read_8bit(frame_offset+channel,stream->streamfile) >> 0) & 0xf;
/* rare but happens, also repeated headers don't match (ex. Ikaruga (GC) SONG02.adp) */
VGM_ASSERT_ONCE(coef_index > 4 || shift_factor > 12, "DTK: incorrect coefs/shift at %"PRIx64"\n", (off64_t)frame_offset);
VGM_ASSERT_ONCE(coef_index > 4 || shift_factor > 12, "DTK: incorrect coefs/shift at %x\n", (uint32_t)frame_offset);
/* decode nibbles */
for (i = first_sample; i < first_sample+samples_to_do; i++) {

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@ -63,7 +63,7 @@ void decode_psx(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing,
shift_factor = ((uint8_t)read_8bit(frame_offset+0x00,stream->streamfile) >> 0) & 0xf;
flag = (uint8_t)read_8bit(frame_offset+0x01,stream->streamfile); /* only lower nibble needed */
VGM_ASSERT_ONCE(coef_index > 5 || shift_factor > 12, "PS-ADPCM: incorrect coefs/shift at %"PRIx64"\n", (off64_t)frame_offset);
VGM_ASSERT_ONCE(coef_index > 5 || shift_factor > 12, "PS-ADPCM: incorrect coefs/shift at %x\n", (uint32_t)frame_offset);
if (coef_index > 5) /* needed by inFamous (PS3) (maybe it's supposed to use more filters?) */
coef_index = 0; /* upper filters aren't used in PS1/PS2, maybe in PSP/PS3? */
if (shift_factor > 12)
@ -71,7 +71,7 @@ void decode_psx(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing,
if (is_badflags) /* some games store garbage or extra internal logic in the flags, must be ignored */
flag = 0;
VGM_ASSERT_ONCE(flag > 7,"PS-ADPCM: unknown flag at %"PRIx64"\n", (off64_t)frame_offset); /* meta should use PSX-badflags */
VGM_ASSERT_ONCE(flag > 7,"PS-ADPCM: unknown flag at %x\n", (uint32_t)frame_offset); /* meta should use PSX-badflags */
/* decode nibbles */
for (i = first_sample; i < first_sample + samples_to_do; i++) {
@ -123,7 +123,7 @@ void decode_psx_configurable(VGMSTREAMCHANNEL * stream, sample * outbuf, int cha
coef_index = ((uint8_t)read_8bit(frame_offset+0x00,stream->streamfile) >> 4) & 0xf;
shift_factor = ((uint8_t)read_8bit(frame_offset+0x00,stream->streamfile) >> 0) & 0xf;
VGM_ASSERT_ONCE(coef_index > 5 || shift_factor > 12, "PS-ADPCM: incorrect coefs/shift at %"PRIx64"\n", (off64_t)frame_offset);
VGM_ASSERT_ONCE(coef_index > 5 || shift_factor > 12, "PS-ADPCM: incorrect coefs/shift at %x\n", (uint32_t)frame_offset);
if (coef_index > 5) /* needed by Afrika (PS3) (maybe it's supposed to use more filters?) */
coef_index = 0; /* upper filters aren't used in PS1/PS2, maybe in PSP/PS3? */
if (shift_factor > 12)
@ -180,7 +180,7 @@ static int ps_find_loop_offsets_internal(STREAMFILE *streamFile, off_t start_off
uint8_t flag = (uint8_t)read_8bit(offset+0x01,streamFile) & 0x0F; /* lower nibble only (for HEVAG) */
/* theoretically possible and would use last 0x06 */
VGM_ASSERT_ONCE(loop_start_found && flag == 0x06, "PS LOOPS: multiple loop start found at %"PRIx64"\n", (off64_t)offset);
VGM_ASSERT_ONCE(loop_start_found && flag == 0x06, "PS LOOPS: multiple loop start found at %x\n", (uint32_t)offset);
if (flag == 0x06 && !loop_start_found) {
loop_start = num_samples; /* loop start before this frame */

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@ -69,7 +69,7 @@ vorbis_custom_codec_data * init_vorbis_custom(STREAMFILE *streamFile, off_t star
return data;
fail:
VGM_LOG("VORBIS: init fail at around 0x%"PRIx64"\n", (off64_t)start_offset);
VGM_LOG("VORBIS: init fail at around 0x%x\n", (uint32_t)start_offset);
free_vorbis_custom(data);
return NULL;
}
@ -144,7 +144,7 @@ void decode_vorbis_custom(VGMSTREAM * vgmstream, sample * outbuf, int32_t sample
/* parse the fake ogg packet into a logical vorbis block */
rc = vorbis_synthesis(&data->vb,&data->op);
if (rc == OV_ENOTAUDIO) {
VGM_LOG("Vorbis: not an audio packet (size=0x%x) @ %"PRIx64"\n",(size_t)data->op.bytes,(off64_t)stream->offset);
VGM_LOG("Vorbis: not an audio packet (size=0x%x) @ %x\n",(size_t)data->op.bytes,(uint32_t)stream->offset);
//VGM_LOGB(data->op.packet, (size_t)data->op.bytes,0);
continue; /* rarely happens, seems ok? */
} else if (rc != 0) goto decode_fail;
@ -162,7 +162,7 @@ void decode_vorbis_custom(VGMSTREAM * vgmstream, sample * outbuf, int32_t sample
decode_fail:
/* on error just put some 0 samples */
VGM_LOG("VORBIS: decode fail at %"PRIx64", missing %i samples\n", (off64_t)stream->offset, (samples_to_do - samples_done));
VGM_LOG("VORBIS: decode fail at %x, missing %i samples\n", (uint32_t)stream->offset, (samples_to_do - samples_done));
memset(outbuf + samples_done * channels, 0, (samples_to_do - samples_done) * channels * sizeof(sample));
}

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@ -76,7 +76,7 @@ void decode_xa(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, i
if (read_32bitBE(frame_offset+0x00,stream->streamfile) != read_32bitBE(frame_offset+0x04,stream->streamfile) ||
read_32bitBE(frame_offset+0x08,stream->streamfile) != read_32bitBE(frame_offset+0x0c,stream->streamfile)) {
VGM_LOG("bad frames at %"PRIx64"\n", (off64_t)frame_offset);
VGM_LOG("bad frames at %x\n", (uint32_t)frame_offset);
}
@ -90,7 +90,7 @@ void decode_xa(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, i
coef_index = ((uint8_t)read_8bit(sp_offset,stream->streamfile) >> 4) & 0xf;
shift_factor = ((uint8_t)read_8bit(sp_offset,stream->streamfile) >> 0) & 0xf;
VGM_ASSERT(coef_index > 4 || shift_factor > 12, "XA: incorrect coefs/shift at %"PRIx64"\n", (off64_t)sp_offset);
VGM_ASSERT(coef_index > 4 || shift_factor > 12, "XA: incorrect coefs/shift at %x\n", (uint32_t)sp_offset);
if (coef_index > 4)
coef_index = 0; /* only 4 filters are used, rest is apparently 0 */
if (shift_factor > 12)

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@ -472,6 +472,7 @@ static const char* extension_list[] = {
"xwb",
"xmd",
"xopus",
"xps",
"xwc",
"xwm",
"xwma",
@ -1046,9 +1047,9 @@ static const meta_info meta_info_list[] = {
{meta_PC_FLX, "Ultima IX .FLX header"},
{meta_MOGG, "Harmonix Music Systems MOGG Vorbis"},
{meta_OGG_VORBIS, "Ogg Vorbis"},
{meta_OGG_SLI, "Ogg Vorbis with .sli (start,length) for looping"},
{meta_OGG_SLI2, "Ogg Vorbis with .sli (from,to) for looping"},
{meta_OGG_SFL, "Ogg Vorbis with SFPL for looping"},
{meta_OGG_SLI, "Ogg Vorbis with .sli looping"},
{meta_OPUS_SLI, "Ogg Opus with .sli looping"},
{meta_OGG_SFL, "Ogg Vorbis with SFPL looping"},
{meta_OGG_KOVS, "Ogg Vorbis (KOVS header)"},
{meta_OGG_encrypted, "Ogg Vorbis (encrypted)"},
{meta_KMA9, "Koei Tecmo KMA9 header"},
@ -1112,7 +1113,8 @@ static const meta_info meta_info_list[] = {
{meta_VS_FFX, "Square VS header"},
{meta_NWAV, "Chunsoft NWAV header"},
{meta_XPCM, "Circus XPCM header"},
{meta_MSF_TAMASOFT, "TamaSoft MSF header"},
{meta_MSF_TAMASOFT, "Tama-Soft MSF header"},
{meta_XPS_DAT, "From Software .XPS+DAT header"},
};

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@ -40,7 +40,7 @@ void render_vgmstream_blocked(sample * buffer, int32_t sample_count, VGMSTREAM *
if (samples_this_block < 0) {
/* probably block bug or EOF, next calcs would give wrong values/segfaults/infinite loop */
VGM_LOG("layout_blocked: wrong block samples at 0x%"PRIx64"\n", (off64_t)vgmstream->current_block_offset);
VGM_LOG("layout_blocked: wrong block samples at 0x%x\n", (uint32_t)vgmstream->current_block_offset);
memset(buffer + samples_written*vgmstream->channels, 0, (sample_count - samples_written) * vgmstream->channels * sizeof(sample));
break;
}

View File

@ -48,7 +48,7 @@ void block_update_h4m(off_t block_offset, VGMSTREAM * vgmstream) {
block_skip += (audio_bytes / vgmstream->num_streams) * (vgmstream->stream_index-1);
}
VGM_ASSERT(frame_format == 1, "H4M: unknown frame_format %x at %"PRIx64"\n", frame_format, (off64_t)block_offset);
VGM_ASSERT(frame_format == 1, "H4M: unknown frame_format %x at %x\n", frame_format, (uint32_t)block_offset);
/* pass current mode to the decoder */
vgmstream->codec_config = (frame_format << 8) | (vgmstream->codec_config & 0xFF);

View File

@ -26,7 +26,7 @@ void block_update_xa(off_t block_offset, VGMSTREAM * vgmstream) {
VGM_ASSERT(block_offset + 0x930 < get_streamfile_size(streamFile) &&
(uint8_t)read_8bit(block_offset + 0x000 + 0x11,streamFile) !=
(uint8_t)read_8bit(block_offset + 0x930 + 0x11,streamFile),
"XA block: subchannel change at %"PRIx64"\n", (off64_t)block_offset);
"XA block: subchannel change at %x\n", (uint32_t)block_offset);
/* submode flag bits (typical audio value = 0x64 01100100)
* - 7 (0x80 10000000): end of file
@ -53,7 +53,7 @@ void block_update_xa(off_t block_offset, VGMSTREAM * vgmstream) {
}
else {
;VGM_ASSERT_ONCE(block_offset < get_streamfile_size(streamFile),
"XA block: non audio block found at %"PRIx64"\n", (off64_t)block_offset);
"XA block: non audio block found at %x\n", (uint32_t)block_offset);
block_samples = 0; /* not an audio sector */
}

View File

@ -1625,6 +1625,10 @@
<File
RelativePath=".\meta\xpcm.c"
>
</File>
<File
RelativePath=".\meta\xps.c"
>
</File>
<File
RelativePath=".\meta\xss.c"

View File

@ -481,6 +481,7 @@
<ClCompile Include="meta\xnb.c" />
<ClCompile Include="meta\xopus.c" />
<ClCompile Include="meta\xpcm.c" />
<ClCompile Include="meta\xps.c" />
<ClCompile Include="meta\xss.c" />
<ClCompile Include="meta\xvag.c" />
<ClCompile Include="meta\xmd.c" />

View File

@ -1465,6 +1465,9 @@
<ClCompile Include="meta\xpcm.c">
<Filter>meta\Source Files</Filter>
</ClCompile>
<ClCompile Include="meta\xps.c">
<Filter>meta\Source Files</Filter>
</ClCompile>
<ClCompile Include="meta\x360_cxs.c">
<Filter>meta\Source Files</Filter>
</ClCompile>

View File

@ -337,7 +337,7 @@ static int parse_awc_header(STREAMFILE* streamFile, awc_header* awc) {
/* If music, data is divided into blocks of block_chunk size with padding.
* Each block has a header/seek table and interleaved data for all channels */
if (awc->is_music && read_32bit(awc->stream_offset, streamFile) != 0) {
VGM_LOG("AWC: music found, but block doesn't start with seek table at %"PRIx64"\n", (off64_t)awc->stream_offset);
VGM_LOG("AWC: music found, but block doesn't start with seek table at %x\n", (uint32_t)awc->stream_offset);
goto fail;
}

View File

@ -188,7 +188,7 @@ VGMSTREAM * init_vgmstream_fsb5(STREAMFILE *streamFile) {
// /* found in some XMA2/Vorbis/FADPCM */
// break;
default:
VGM_LOG("FSB5: unknown extraflag 0x%x at %"PRIx64" + 0x04 (size 0x%x)\n", extraflag_type, (off64_t)extraflag_offset, extraflag_size);
VGM_LOG("FSB5: unknown extraflag 0x%x at %x + 0x04 (size 0x%x)\n", extraflag_type, (uint32_t)extraflag_offset, extraflag_size);
break;
}

View File

@ -52,8 +52,10 @@ typedef struct {
int loop_flag;
int32_t coef[2];
int32_t coef_splitted[2];
int32_t coef_offset;
int32_t coef_spacing;
int32_t coef_split_offset;
int32_t coef_split_spacing;
int32_t coef_type;
int32_t coef_interleave_type;
int coef_big_endian;
@ -233,16 +235,16 @@ VGMSTREAM * init_vgmstream_genh(STREAMFILE *streamFile) {
for (i=0;i<vgmstream->channels;i++) {
int16_t (*read_16bit)(off_t , STREAMFILE*) = genh.coef_big_endian ? read_16bitBE : read_16bitLE;
/* normal/split coefs bit flag */
if ((genh.coef_type & 1) == 0) { /* not set: normal coefs, all 16 interleaved into one array */
for (j=0;j<16;j++) {
vgmstream->ch[i].adpcm_coef[j] = read_16bit(genh.coef[i]+j*2,streamFile);
/* normal/split coefs */
if ((genh.coef_type & 1) == 0) { /* normal mode */
for (j = 0; j < 16; j++) {
vgmstream->ch[i].adpcm_coef[j] = read_16bit(genh.coef_offset + i*genh.coef_spacing + j*2, streamFile);
}
}
else { /* set: split coefs, 8 coefs in the main array, additional offset to 2nd array given at 0x34 for left, 0x38 for right */
for (j=0;j<8;j++) {
vgmstream->ch[i].adpcm_coef[j*2]=read_16bit(genh.coef[i]+j*2,streamFile);
vgmstream->ch[i].adpcm_coef[j*2+1]=read_16bit(genh.coef_splitted[i]+j*2,streamFile);
else { /* split coefs, 8 coefs in the main array, additional offset to 2nd array given at 0x34 for left, 0x38 for right */
for (j = 0; j < 8; j++) {
vgmstream->ch[i].adpcm_coef[j*2] = read_16bit(genh.coef_offset + i*genh.coef_spacing + j*2, streamFile);
vgmstream->ch[i].adpcm_coef[j*2+1] = read_16bit(genh.coef_split_offset + i*genh.coef_split_spacing + j*2, streamFile);
}
}
}
@ -344,10 +346,10 @@ fail:
static int parse_genh(STREAMFILE * streamFile, genh_header * genh) {
size_t header_size;
genh->channels = read_32bitLE(0x4,streamFile);
genh->channels = read_32bitLE(0x04,streamFile);
genh->interleave = read_32bitLE(0x8,streamFile);
genh->sample_rate = read_32bitLE(0xc,streamFile);
genh->interleave = read_32bitLE(0x08,streamFile);
genh->sample_rate = read_32bitLE(0x0c,streamFile);
genh->loop_start_sample = read_32bitLE(0x10,streamFile);
genh->loop_end_sample = read_32bitLE(0x14,streamFile);
@ -359,37 +361,54 @@ static int parse_genh(STREAMFILE * streamFile, genh_header * genh) {
genh->start_offset = 0x800;
header_size = 0x800;
}
/* check for audio data start past header end */
if (header_size > genh->start_offset) goto fail;
genh->coef[0] = read_32bitLE(0x24,streamFile);
genh->coef[1] = read_32bitLE(0x28,streamFile);
genh->coef_interleave_type = read_32bitLE(0x2C,streamFile);
if (header_size > genh->start_offset) /* audio data start past header end */
goto fail;
if (header_size < 0x24) /* absolute minimum for GENH */
goto fail;
/* DSP coefficients */
if (header_size >= 0x30) {
genh->coef_offset = read_32bitLE(0x24,streamFile);
if (genh->channels == 2) /* old meaning, "coef right offset" */
genh->coef_spacing = read_32bitLE(0x28,streamFile) - genh->coef_offset;
else if (genh->channels > 2) /* new meaning, "coef spacing" */
genh->coef_spacing = read_32bitLE(0x28,streamFile);
genh->coef_interleave_type = read_32bitLE(0x2C,streamFile);
}
/* DSP coefficient variants */
/* bit 0 flag - split coefs (2 arrays) */
/* bit 1 flag - little endian coefs (for some 3DS) */
genh->coef_type = read_32bitLE(0x30,streamFile);
genh->coef_big_endian = ((genh->coef_type & 2) == 0);
if (header_size >= 0x34) {
/* bit 0 flag - split coefs (2 arrays) */
/* bit 1 flag - little endian coefs (for some 3DS) */
genh->coef_type = read_32bitLE(0x30,streamFile);
genh->coef_big_endian = ((genh->coef_type & 2) == 0);
}
/* when using split coefficients, 2nd array is at: */
genh->coef_splitted[0] = read_32bitLE(0x34,streamFile);
genh->coef_splitted[1] = read_32bitLE(0x38,streamFile);
/* DSP split coefficients' 2nd array */
if (header_size >= 0x3c) {
genh->coef_split_offset = read_32bitLE(0x34,streamFile);
if (genh->channels == 2) /* old meaning, "coef right offset" */
genh->coef_split_spacing = read_32bitLE(0x38,streamFile) - genh->coef_split_offset;
else if (genh->channels > 2) /* new meaning, "coef spacing" */
genh->coef_split_spacing = read_32bitLE(0x38,streamFile);
}
/* extended fields */
if (header_size >= 0x54) {
genh->num_samples = read_32bitLE(0x40,streamFile);
genh->skip_samples = read_32bitLE(0x44,streamFile); /* for FFmpeg based codecs */
genh->skip_samples_mode = read_8bit(0x48,streamFile); /* 0=autodetect, 1=force manual value @ 0x44 */
genh->codec_mode = read_8bit(0x4b,streamFile);
if ((genh->codec == ATRAC3 || genh->codec == ATRAC3PLUS) && genh->codec_mode==0)
genh->codec_mode = read_8bit(0x49,streamFile);
if ((genh->codec == XMA1 || genh->codec == XMA2) && genh->codec_mode==0)
genh->codec_mode = read_8bit(0x4a,streamFile);
genh->data_size = read_32bitLE(0x50,streamFile);
}
/* other fields */
genh->num_samples = read_32bitLE(0x40,streamFile);
genh->skip_samples = read_32bitLE(0x44,streamFile); /* for FFmpeg based codecs */
genh->skip_samples_mode = read_8bit(0x48,streamFile); /* 0=autodetect, 1=force manual value @ 0x44 */
genh->codec_mode = read_8bit(0x4b,streamFile);
if ((genh->codec == ATRAC3 || genh->codec == ATRAC3PLUS) && genh->codec_mode==0)
genh->codec_mode = read_8bit(0x49,streamFile);
if ((genh->codec == XMA1 || genh->codec == XMA2) && genh->codec_mode==0)
genh->codec_mode = read_8bit(0x4a,streamFile);
genh->data_size = read_32bitLE(0x50,streamFile);
if (genh->data_size == 0)
genh->data_size = get_streamfile_size(streamFile) - genh->start_offset;
genh->num_samples = genh->num_samples > 0 ? genh->num_samples : genh->loop_end_sample;
genh->loop_flag = genh->loop_start_sample != -1;

View File

@ -810,4 +810,7 @@ VGMSTREAM * init_vgmstream_xpcm(STREAMFILE * streamFile);
VGMSTREAM * init_vgmstream_msf_tamasoft(STREAMFILE * streamFile);
VGMSTREAM * init_vgmstream_xps_dat(STREAMFILE * streamFile);
VGMSTREAM * init_vgmstream_xps(STREAMFILE * streamFile);
#endif /*_META_H*/

View File

@ -15,11 +15,12 @@ VGMSTREAM * init_vgmstream_pos(STREAMFILE *streamFile) {
streamData = open_streamfile_by_ext(streamFile, "wav");
if (streamData) {
vgmstream = init_vgmstream_riff(streamData);
vgmstream = init_vgmstream_riff(streamData);
if (!vgmstream) goto fail;
vgmstream->meta_type = meta_RIFF_WAVE_POS;
}
else {
#ifdef VGM_USE_VORBIS
/* hack for Ogg with external loops */
streamData = open_streamfile_by_ext(streamFile, "ogg");
if (streamData) {
@ -29,6 +30,9 @@ VGMSTREAM * init_vgmstream_pos(STREAMFILE *streamFile) {
else {
goto fail;
}
#else
goto fail;
#endif
}
close_streamfile(streamData);

View File

@ -2,7 +2,7 @@
#include <ctype.h>
/* .sli - loop points associated with a similarly named .ogg [Fate/Stay Night (PC), World End Economica (PC)]*/
/* .sli+ogg/opus - KiriKiri engine / WaveLoopManager loop points loader [Fate/Stay Night (PC), World End Economica (PC)] */
VGMSTREAM * init_vgmstream_sli_ogg(STREAMFILE *streamFile) {
VGMSTREAM * vgmstream = NULL;
STREAMFILE * streamData = NULL;
@ -14,23 +14,42 @@ VGMSTREAM * init_vgmstream_sli_ogg(STREAMFILE *streamFile) {
goto fail;
{
/* try with file.ogg.sli=header and file.ogg=data */
/* try with file.ogg/opus.sli=header and file.ogg/opus=data */
char basename[PATH_LIMIT];
get_streamfile_basename(streamFile,basename,PATH_LIMIT);
streamData = open_streamfile_by_filename(streamFile, basename);
if (!streamData) goto fail;
if (!check_extensions(streamData, "ogg"))
goto fail;
}
#ifdef VGM_USE_VORBIS
/* let the real initer do the parsing */
vgmstream = init_vgmstream_ogg_vorbis(streamData);
if (!vgmstream) goto fail;
if (check_extensions(streamData, "ogg")) { /* Fate/Stay Night (PC) */
#ifdef VGM_USE_VORBIS
vgmstream = init_vgmstream_ogg_vorbis(streamData);
if (!vgmstream) goto fail;
vgmstream->meta_type = meta_OGG_SLI;
#else
goto fail;
#endif
}
else if (check_extensions(streamData, "opus")) { /* Sabbat of the Witch (PC) */
#ifdef VGM_USE_FFMPEG
vgmstream = init_vgmstream_ffmpeg(streamData);
if (!vgmstream) goto fail;
/* FFmpeg's Opus encoder delay is borked but no need to fix:
* somehow sli+opus use 0 in the OpusHead (to simplify looping?) */
vgmstream->meta_type = meta_OPUS_SLI;
#else
goto fail;
#endif
}
else {
goto fail;
}
/* find loop text */
{
@ -77,13 +96,11 @@ VGMSTREAM * init_vgmstream_sli_ogg(STREAMFILE *streamFile) {
}
}
if (loop_start != -1 && loop_length != -1) {
if (loop_start != -1 && loop_length != -1) { /* v1 */
vgmstream_force_loop(vgmstream,1,loop_start, loop_start+loop_length);
vgmstream->meta_type = meta_OGG_SLI;
}
else if (loop_from != -1 && loop_to != -1) {
else if (loop_from != -1 && loop_to != -1) { /* v2 */
vgmstream_force_loop(vgmstream,1,loop_to, loop_from);
vgmstream->meta_type = meta_OGG_SLI2;
}
else {
goto fail; /* if there's no loop points the .sli wasn't valid */

View File

@ -303,17 +303,17 @@ VGMSTREAM * init_vgmstream_txth(STREAMFILE *streamFile) {
int16_t (*read_16bit)(off_t , STREAMFILE*) = txth.coef_big_endian ? read_16bitBE : read_16bitLE;
/* normal/split coefs */
if (txth.coef_mode == 0) {
for (j=0;j<16;j++) {
vgmstream->ch[i].adpcm_coef[j] = read_16bit(txth.coef_offset + i*txth.coef_spacing + j*2,txth.streamHead);
if (txth.coef_mode == 0) { /* normal mode */
for (j = 0; j < 16; j++) {
vgmstream->ch[i].adpcm_coef[j] = read_16bit(txth.coef_offset + i*txth.coef_spacing + j*2, txth.streamHead);
}
}
else {
else { /* split coefs */
goto fail; //IDK what is this
/*
for (j=0;j<8;j++) {
vgmstream->ch[i].adpcm_coef[j*2]=read_16bit(coef[i]+j*2,txth.streamHead);
vgmstream->ch[i].adpcm_coef[j*2+1]=read_16bit(coef_splitted[i]+j*2,txth.streamHead);
for (j = 0; j < 8; j++) {
vgmstream->ch[i].adpcm_coef[j*2] = read_16bit(genh.coef_offset + i*genh.coef_spacing + j*2, txth.streamHead);
vgmstream->ch[i].adpcm_coef[j*2+1] = read_16bit(genh.coef_split_offset + i*genh.coef_split_spacing + j*2, txth.streamHead);
}
*/
}
@ -508,6 +508,7 @@ static int parse_txth(txth_header * txth) {
bytes_read = get_streamfile_text_line(TXT_LINE_MAX,line, txt_offset,txth->streamText, &line_done);
if (!line_done) goto fail;
//;VGM_LOG("TXTH: line=%s\n",line);
txt_offset += bytes_read;
@ -759,6 +760,11 @@ static int parse_keyval(STREAMFILE * streamFile_, txth_header * txth, const char
txth->streambody_opened = 1;
}
/* use body as header when opening a .txth directly to simplify things */
if (txth->streamfile_is_txth && !txth->streamhead_opened) {
txth->streamHead = txth->streamBody;
}
txth->data_size = !txth->streamBody ? 0 :
get_streamfile_size(txth->streamBody) - txth->start_offset; /* re-evaluate */
}
@ -841,13 +847,14 @@ static int parse_num(STREAMFILE * streamFile, txth_header * txth, const char * v
else goto fail;
}
if (value_mul)
/* operators, but only if current value wasn't set to 0 right before */
if (value_mul && txth->value_mul)
*out_value = (*out_value) * value_mul;
if (value_div)
if (value_div && txth->value_div)
*out_value = (*out_value) / value_div;
if (value_add)
if (value_add && txth->value_add)
*out_value = (*out_value) + value_add;
if (value_sub)
if (value_sub && txth->value_sub)
*out_value = (*out_value) - value_sub;
//;VGM_LOG("TXTH: val=%s, read %u (0x%x)\n", val, *out_value, *out_value);

View File

@ -391,7 +391,7 @@ static int parse_pk_header(ubi_bao_header * bao, STREAMFILE *streamFile) {
goto fail;
}
;VGM_LOG("BAO stream: id=%x, offset=%"PRIx64", size=%x, res=%s\n", bao->stream_id, (off64_t)bao->stream_offset, bao->stream_size, (bao->is_external ? bao->resource_name : "internal"));
;VGM_LOG("BAO stream: id=%x, offset=%x, size=%x, res=%s\n", bao->stream_id, (uint32_t)bao->stream_offset, bao->stream_size, (bao->is_external ? bao->resource_name : "internal"));
return 1;
fail:
@ -434,7 +434,7 @@ static int parse_bao(ubi_bao_header * bao, STREAMFILE *streamFile, off_t offset)
case 0x70000000: bao->types_count[7]++; break; /* project info? (sometimes special id 0x7fffffff) */
case 0x80000000: bao->types_count[8]++; break; /* unknown (some id/info?) */
default:
VGM_LOG("UBI BAO: unknown type %x at %"PRIx64" + %x\n", descriptor_type, (off64_t)offset, 0x20);
VGM_LOG("UBI BAO: unknown type %x at %x + %x\n", descriptor_type, (uint32_t)offset, 0x20);
goto fail;
}
@ -453,7 +453,7 @@ static int parse_bao(ubi_bao_header * bao, STREAMFILE *streamFile, off_t offset)
case 0x00000007: bao->subtypes_count[7]++; break; /* related to other header BAOs? */
case 0x00000008: bao->subtypes_count[8]++; break; /* ? (almost empty with some unknown value) */
default:
VGM_LOG("UBI BAO: unknown subtype %x at %"PRIx64" + %x\n", descriptor_subtype, (off64_t)offset, header_size+0x04);
VGM_LOG("UBI BAO: unknown subtype %x at %x + %x\n", descriptor_subtype, (uint32_t)offset, header_size+0x04);
goto fail;
}
//;VGM_ASSERT(descriptor_subtype != 0x01, "UBI BAO: subtype %x at %lx (%lx)\n", descriptor_subtype, offset, offset+header_size+0x04);
@ -477,7 +477,7 @@ static int parse_bao(ubi_bao_header * bao, STREAMFILE *streamFile, off_t offset)
* - channels, ?, sample rate, average bit rate?, samples, full stream_size?, codec, etc
* - subtable entries, subtable size (may contain offsets/ids, cues, etc)
* - extra data per codec (ex. XMA header in some versions) */
;VGM_LOG("BAO header at %"PRIx64"\n", (off64_t)offset);
;VGM_LOG("BAO header at %x\n", (uint32_t)offset);
bao->version = bao_version;
switch(bao->version) {
@ -503,7 +503,7 @@ static int parse_bao(ubi_bao_header * bao, STREAMFILE *streamFile, off_t offset)
case 0x03: bao->codec = UBI_ADPCM; break;
case 0x05: bao->codec = RAW_XMA1; break;
case 0x09: bao->codec = RAW_DSP; break;
default: VGM_LOG("UBI BAO: unknown codec at %"PRIx64"\n", (off64_t)offset); goto fail;
default: VGM_LOG("UBI BAO: unknown codec at %x\n", (uint32_t)offset); goto fail;
}
bao->prefetch_size = read_32bit(offset + header_size + 0x74, streamFile);
@ -535,11 +535,11 @@ static int parse_bao(ubi_bao_header * bao, STREAMFILE *streamFile, off_t offset)
switch(bao->header_codec) {
case 0x06: bao->codec = RAW_PSX; break;
case 0x07: bao->codec = FMT_AT3; break;
default: VGM_LOG("UBI BAO: unknown codec at %"PRIx64"\n", (off64_t)offset); goto fail;
default: VGM_LOG("UBI BAO: unknown codec at %x\n", (uint32_t)offset); goto fail;
}
if (read_32bit(offset+header_size+0x20, streamFile) & 0x10) {
VGM_LOG("UBI BAO: possible full loop at %"PRIx64"\n", (off64_t)offset);
VGM_LOG("UBI BAO: possible full loop at %x\n", (uint32_t)offset);
/* RIFFs may have "smpl" and this flag, even when data shouldn't loop... */
}
@ -564,7 +564,7 @@ static int parse_bao(ubi_bao_header * bao, STREAMFILE *streamFile, off_t offset)
case 0x02: bao->codec = UBI_ADPCM; break;
case 0x03: bao->codec = FMT_OGG; break;
case 0x04: bao->codec = RAW_XMA2; break;
default: VGM_LOG("UBI BAO: unknown codec at %"PRIx64"\n", (off64_t)offset); goto fail;
default: VGM_LOG("UBI BAO: unknown codec at %x\n", (uint32_t)offset); goto fail;
}
bao->prefetch_size = read_32bit(offset+header_size+0x84, streamFile);
@ -598,7 +598,7 @@ static int parse_bao(ubi_bao_header * bao, STREAMFILE *streamFile, off_t offset)
case 0x04: bao->codec = RAW_XMA2; break;
case 0x05: bao->codec = RAW_PSX; break;
case 0x06: bao->codec = RAW_AT3; break;
default: VGM_LOG("UBI BAO: unknown codec at %"PRIx64"\n", (off64_t)offset); goto fail;
default: VGM_LOG("UBI BAO: unknown codec at %x\n", (uint32_t)offset); goto fail;
}
bao->prefetch_size = read_32bit(offset + header_size + 0x78, streamFile);
@ -615,7 +615,7 @@ static int parse_bao(ubi_bao_header * bao, STREAMFILE *streamFile, off_t offset)
case 0x00280306: /* Far Cry 3: Blood Dragon (X360)-file */
case 0x00290106: /* Splinter Cell Blacklist? */
default: /* others possibly using BAO: Avatar X360/PS3/PC, Just Dance, Watch_Dogs, Far Cry Primal, Far Cry 4 */
VGM_LOG("UBI BAO: unknown BAO version at %"PRIx64"\n", (off64_t)offset);
VGM_LOG("UBI BAO: unknown BAO version at %x\n", (uint32_t)offset);
goto fail;
}

View File

@ -29,9 +29,6 @@ VGMSTREAM * init_vgmstream_ue4opus(STREAMFILE *streamFile) {
start_offset = 0x11;
data_size = get_streamfile_size(streamFile) - start_offset;
/* usually uses 60ms for music (delay of 360 samples) */
skip = ue4_opus_get_encoder_delay(start_offset, streamFile);
/* build the VGMSTREAM */
vgmstream = allocate_vgmstream(channel_count, loop_flag);
@ -39,10 +36,13 @@ VGMSTREAM * init_vgmstream_ue4opus(STREAMFILE *streamFile) {
vgmstream->meta_type = meta_UE4OPUS;
vgmstream->sample_rate = sample_rate;
vgmstream->num_samples = num_samples - skip;
#ifdef VGM_USE_FFMPEG
{
/* usually uses 60ms for music (delay of 360 samples) */
skip = ue4_opus_get_encoder_delay(start_offset, streamFile);
vgmstream->num_samples = num_samples - skip;
vgmstream->codec_data = init_ffmpeg_ue4_opus(streamFile, start_offset,data_size, vgmstream->channels, skip, vgmstream->sample_rate);
if (!vgmstream->codec_data) goto fail;
vgmstream->coding_type = coding_FFmpeg;

223
src/meta/xps.c Normal file
View File

@ -0,0 +1,223 @@
#include "meta.h"
#include "../coding/coding.h"
static void read_xps_name(VGMSTREAM *vgmstream, STREAMFILE *streamFile, int file_id);
/* .XPS+DAT - From Software games streams [Metal Wolf Chaos (Xbox), Otogi (Xbox)] */
VGMSTREAM * init_vgmstream_xps_dat(STREAMFILE *streamFile) {
VGMSTREAM * vgmstream = NULL;
STREAMFILE * streamData = NULL;
off_t start_offset, header_offset;
size_t stream_size;
int loop_flag, channel_count, sample_rate, codec, loop_start_sample, loop_end_sample, file_id;
int total_subsongs, target_subsong = streamFile->stream_index;
/* checks */
if (!check_extensions(streamFile, "xps"))
goto fail;
if (read_32bitLE(0x00,streamFile) != get_streamfile_size(streamFile))
goto fail;
if (read_32bitBE(0x0c,streamFile) != 0x64696666) /* "diff" */
goto fail;
/* handle .xps+dat (bank .xps are done below) */
streamData = open_streamfile_by_ext(streamFile, "dat");
if (!streamData) goto fail;
/* 0x00: approximate file size */
total_subsongs = read_32bitLE(0x04,streamData);
if (target_subsong == 0) target_subsong = 1;
if (target_subsong < 0 || target_subsong > total_subsongs || total_subsongs < 1) goto fail;
header_offset = 0x20 + 0x94*(target_subsong-1); /* could start at 0x0c too */
file_id = read_32bitLE(header_offset+0x00,streamData);
start_offset = read_32bitLE(header_offset+0x04,streamData);
stream_size = read_32bitLE(header_offset+0x08,streamData);
/* 0x0c: loop start offset? */
/* 0x10: loop end offset? */
/* 0x14: always null? */
codec = read_16bitLE(header_offset+0x18,streamData);
channel_count = read_16bitLE(header_offset+0x1a,streamData);
sample_rate = read_32bitLE(header_offset+0x1c,streamData);
/* 0x20: average bitrate */
/* 0x24: block size, bps */
loop_flag = read_32bitLE(header_offset+0x5c,streamData);
loop_start_sample = read_32bitLE(header_offset+0x6c,streamData);
loop_end_sample = read_32bitLE(header_offset+0x70,streamData) + 1; /* a "smpl" chunk basically */
/* build the VGMSTREAM */
vgmstream = allocate_vgmstream(channel_count, loop_flag);
if (!vgmstream) goto fail;
vgmstream->sample_rate = sample_rate;
vgmstream->meta_type = meta_XPS_DAT;
vgmstream->loop_start_sample = loop_start_sample;
vgmstream->loop_end_sample = loop_end_sample;
vgmstream->num_streams = total_subsongs;
vgmstream->stream_size = stream_size;
switch(codec) {
case 0x01:
vgmstream->coding_type = coding_PCM16LE;
vgmstream->layout_type = layout_interleave;
vgmstream->interleave_block_size = 0x02;
vgmstream->num_samples = pcm_bytes_to_samples(stream_size, channel_count, 16);
break;
case 0x69:
vgmstream->coding_type = coding_XBOX_IMA;
vgmstream->layout_type = layout_none;
vgmstream->num_samples = xbox_ima_bytes_to_samples(stream_size, channel_count);
break;
default:
goto fail;
}
read_xps_name(vgmstream, streamFile, file_id);
if (!vgmstream_open_stream(vgmstream,streamData,start_offset))
goto fail;
close_streamfile(streamData);
return vgmstream;
fail:
close_streamfile(streamData);
close_vgmstream(vgmstream);
return NULL;
}
static void read_xps_name(VGMSTREAM *vgmstream, STREAMFILE *streamFile, int file_id) {
int i, entries;
int name_id = -1, udss_name_id;
off_t entry_offset = 0x10;
/* main section + stream sections (usually same number but not always) */
entries = read_32bitLE(0x04,streamFile);
/* "sid\0" entries: find name_id of file_id */
for (i = 0; i < entries; i++) {
off_t entry_base = entry_offset;
size_t entry_size = read_32bitLE(entry_base+0x00,streamFile);
uint32_t entry_id = read_32bitBE(entry_base+0x04,streamFile);
size_t entry_pad = read_32bitLE(entry_base+0x08,streamFile);
/* 0x0c: always null, rest: entry (format varies) */
entry_offset += entry_size + entry_pad + 0x10;
/* sound info entry */
if (entry_id == 0x73696400) { /* "sid\0" */
int entry_file_id = read_32bitLE(entry_base+0x10,streamFile);
int entry_name_id = read_32bitLE(entry_base+0x14,streamFile);
if (entry_file_id == file_id && name_id == -1) {
name_id = entry_name_id;
}
continue;
}
/* sound stream entry, otherwise no good */
if (entry_id != 0x75647373) { /* "udss" */
goto fail;
}
udss_name_id = read_32bitLE(entry_base+0x10,streamFile);
if (udss_name_id == name_id) {
off_t name_offset = entry_base + 0x10 + 0x08;
size_t name_size = entry_size - 0x08; /* includes null */
read_string(vgmstream->stream_name,name_size, name_offset,streamFile);
return;
}
}
fail:
return;
}
/* .XPS - From Software games banks [Metal Wolf Chaos (Xbox), Otogi (Xbox)] */
VGMSTREAM * init_vgmstream_xps(STREAMFILE *streamFile) {
VGMSTREAM * vgmstream = NULL;
STREAMFILE * streamData = NULL;
int i, entries;
off_t entry_offset = 0x10;
int total_subsongs, target_subsong = streamFile->stream_index;
/* checks */
if (!check_extensions(streamFile, "xps"))
goto fail;
if (read_32bitLE(0x00,streamFile) != get_streamfile_size(streamFile))
goto fail;
if (read_32bitBE(0x0c,streamFile) != 0x64696666) /* "diff" */
goto fail;
/* handle .xps alone (stream .xps+data are done above) */
streamData = open_streamfile_by_ext(streamFile, "dat");
if (streamData) goto fail;
/* main section + bank sections (usually same number but not always) */
entries = read_32bitLE(0x04,streamFile);
total_subsongs = 0;
if (target_subsong == 0) target_subsong = 1;
if (target_subsong < 0 /*|| target_subsong > total_subsongs || total_subsongs < 1*/) goto fail;
/* parse entries: skip (there is probably a stream/bank flag here) */
for (i = 0; i < entries; i++) {
off_t entry_base = entry_offset;
size_t entry_size = read_32bitLE(entry_base+0x00,streamFile);
uint32_t entry_id = read_32bitBE(entry_base+0x04,streamFile);
size_t entry_pad = read_32bitLE(entry_base+0x08,streamFile);
/* 0x0c: always null, rest: entry (format varies) */
entry_offset += entry_size + entry_pad + 0x10;
/* sound info entry */
if (entry_id == 0x73696400) { /* "sid\0" */
/* keep looking for sound banks */
continue;
}
/* sound bank entry, otherwise no good */
if (entry_id != 0x75647362) { /* "udsb" */
goto fail;
}
total_subsongs++;
/* open internal RIFF */
if (target_subsong == total_subsongs && vgmstream == NULL) {
STREAMFILE* temp_streamFile;
off_t subsong_offset = entry_base+0x18;
size_t subsong_size = read_32bitLE(entry_base+0x14,streamFile);
temp_streamFile = setup_subfile_streamfile(streamFile, subsong_offset,subsong_size, "wav");
if (!temp_streamFile) goto fail;
vgmstream = init_vgmstream_riff(temp_streamFile);
close_streamfile(temp_streamFile);
if (!vgmstream) goto fail;
}
}
/* subsong not found */
if (!vgmstream)
goto fail;
vgmstream->num_streams = total_subsongs;
return vgmstream;
fail:
close_streamfile(streamData);
close_vgmstream(vgmstream);
return NULL;
}

View File

@ -53,7 +53,7 @@ static size_t read_stdio(STDIOSTREAMFILE *streamfile,uint8_t * dest, off_t offse
/* ignore requests at EOF */
if (offset >= streamfile->filesize) {
//offset = streamfile->filesize; /* seems fseek doesn't clamp offset */
VGM_ASSERT_ONCE(offset > streamfile->filesize, "STDIO: reading over filesize 0x%x @ 0x%"PRIx64" + 0x%x\n", streamfile->filesize, (off64_t)offset, length);
VGM_ASSERT_ONCE(offset > streamfile->filesize, "STDIO: reading over filesize 0x%x @ 0x%x + 0x%x\n", streamfile->filesize, (uint32_t)offset, length);
break;
}
@ -254,7 +254,7 @@ static size_t buffer_read(BUFFER_STREAMFILE *streamfile, uint8_t * dest, off_t o
/* ignore requests at EOF */
if (offset >= streamfile->filesize) {
//offset = streamfile->filesize; /* seems fseek doesn't clamp offset */
VGM_ASSERT_ONCE(offset > streamfile->filesize, "BUFFER: reading over filesize 0x%x @ 0x%"PRIx64" + 0x%x\n", streamfile->filesize, (off64_t)offset, length);
VGM_ASSERT_ONCE(offset > streamfile->filesize, "BUFFER: reading over filesize 0x%x @ 0x%x + 0x%x\n", streamfile->filesize, (uint32_t)offset, length);
break;
}

View File

@ -450,6 +450,8 @@ VGMSTREAM * (*init_vgmstream_functions[])(STREAMFILE *streamFile) = {
init_vgmstream_nwav,
init_vgmstream_xpcm,
init_vgmstream_msf_tamasoft,
init_vgmstream_xps_dat,
init_vgmstream_xps,
/* lowest priority metas (should go after all metas, and TXTH should go before raw formats) */
init_vgmstream_txth, /* proper parsers should supersede TXTH, once added */

View File

@ -641,7 +641,7 @@ typedef enum {
meta_MOGG, /* Harmonix Music Systems MOGG Vorbis */
meta_OGG_VORBIS, /* Ogg Vorbis */
meta_OGG_SLI, /* Ogg Vorbis file w/ companion .sli for looping */
meta_OGG_SLI2, /* Ogg Vorbis file w/ different styled .sli for looping */
meta_OPUS_SLI, /* Ogg Opus file w/ companion .sli for looping */
meta_OGG_SFL, /* Ogg Vorbis file w/ .sfl (RIFF SFPL) for looping */
meta_OGG_KOVS, /* Ogg Vorbis with header and encryption (Koei Tecmo Games) */
meta_OGG_encrypted, /* Ogg Vorbis with encryption */
@ -706,6 +706,7 @@ typedef enum {
meta_NWAV,
meta_XPCM,
meta_MSF_TAMASOFT,
meta_XPS_DAT,
} meta_t;

View File

@ -1436,6 +1436,7 @@ __declspec(dllexport) In_Module * winampGetInModule2() {
#define WINAMP_TAGS_ENTRY_SIZE 2048
typedef struct {
int loaded;
in_char filename[PATH_LIMIT]; /* tags are loaded for this file */
int tag_count;
@ -1457,8 +1458,9 @@ static void load_tagfile_info(in_char* filename) {
char *path;
if (settings.tagfile_disable) {
last_tags.tag_count = 0; /* maybe helps if setting changes during play */
if (settings.tagfile_disable) { /* reset values if setting changes during play */
last_tags.loaded = 0;
last_tags.tag_count = 0;
return;
}
@ -1469,6 +1471,8 @@ static void load_tagfile_info(in_char* filename) {
return; /* not changed, tags still apply */
}
last_tags.loaded = 0;
/* tags are now for this filename, find tagfile path */
wa_ichar_to_char(filename_utf8, PATH_LIMIT, filename_clean);
strcpy(tagfile_path_utf8,filename_utf8);
@ -1517,6 +1521,7 @@ static void load_tagfile_info(in_char* filename) {
}
close_streamfile(tagFile);
last_tags.loaded = 1;
}
}
@ -1529,16 +1534,17 @@ static int winampGetExtendedFileInfo_common(in_char* filename, char *metadata, c
int i, tag_found;
int max_len;
/* load list current tags, if necessary */
load_tagfile_info(filename);
if (!last_tags.loaded) /* tagfile not found, fail so default get_title takes over */
goto fail;
/* always called (value in ms), must return ok so other tags get called */
if (strcasecmp(metadata, "length") == 0) {
strcpy(ret, "0");//todo should export but shows GetFileInfo's ms if not provided
return 1;
}
/* load list current tags, if necessary */
load_tagfile_info(filename);
#if 0
/* special case to fill WA5's unified dialog */
if (strcasecmp(metadata, "formatinformation") == 0) {