diff --git a/src/meta/ta_aac.c b/src/meta/ta_aac.c
index 59388f20..2f4b0f25 100644
--- a/src/meta/ta_aac.c
+++ b/src/meta/ta_aac.c
@@ -1,373 +1,373 @@
-#include "meta.h"
-#include "../coding/coding.h"
-
-/* AAC - tri-Ace (Aska engine) Audio Container */
-
-/* Xbox 360 Variants (Star Ocean 4, End of Eternity, Infinite Undiscovery) */
-VGMSTREAM * init_vgmstream_ta_aac_x360(STREAMFILE *streamFile) {
-    VGMSTREAM * vgmstream = NULL;
-    off_t start_offset;
-    int loop_flag, channel_count;
-    size_t sampleRate, numSamples, startSample, dataSize, blockSize, blockCount; // A mess
-
-    /* check extension, case insensitive */
-    /* .aac: expected, .laac/ace: for players to avoid hijacking MP4/AAC */
-    if ( !check_extensions(streamFile,"aac,laac,ace"))
-        goto fail;
-
-    if (read_32bitBE(0x00,streamFile) != 0x41414320)   /* "AAC " */
-        goto fail;
-
-    /* Ok, let's check what's behind door number 1 */
-    if (read_32bitBE(0x1000, streamFile) == 0x41534320) /* "ASC " */
-    {
-        loop_flag = read_32bitBE(0x1118, streamFile);
-
-        /*Funky Channel Count Checking */
-        if (read_32bitBE(0x1184, streamFile) == 0x7374726D)
-            channel_count = 6;
-        else if (read_32bitBE(0x1154, streamFile) == 0x7374726D)
-            channel_count = 4;
-        else
-            channel_count = read_8bit(0x1134, streamFile);
-
-        sampleRate = read_32bitBE(0x10F4, streamFile);
-        numSamples = read_32bitBE(0x10FC, streamFile);
-        startSample = read_32bitBE(0x10F8, streamFile);
-        dataSize = read_32bitBE(0x10F0, streamFile);
-        blockSize = read_32bitBE(0x1100, streamFile);
-        blockCount = read_32bitBE(0x110C, streamFile);
-    }
-    else if (read_32bitBE(0x1000, streamFile) == 0x57415645) /* "WAVE" */
-    {
-        loop_flag = read_32bitBE(0x1048, streamFile);
-
-        /*Funky Channel Count Checking */
-        if (read_32bitBE(0x10B0, streamFile) == 0x7374726D)
-            channel_count = 6;
-        else if (read_32bitBE(0x1080, streamFile) == 0x7374726D)
-            channel_count = 4;
-        else
-            channel_count = read_8bit(0x1060, streamFile);
-
-        sampleRate = read_32bitBE(0x1024, streamFile);
-        numSamples = read_32bitBE(0x102C, streamFile);
-        startSample = read_32bitBE(0x1028, streamFile);
-        dataSize = read_32bitBE(0x1020, streamFile);
-        blockSize = read_32bitBE(0x1030, streamFile);
-        blockCount = read_32bitBE(0x103C, streamFile);
-    }
-    else if (read_32bitBE(0x1000, streamFile) == 0x00000000) /* some like to be special */
-    {
-        loop_flag = read_32bitBE(0x6048, streamFile);
-
-        /*Funky Channel Count Checking */
-        if (read_32bitBE(0x60B0, streamFile) == 0x7374726D)
-            channel_count = 6;
-        else if (read_32bitBE(0x6080, streamFile) == 0x7374726D)
-            channel_count = 4;
-        else
-            channel_count = read_8bit(0x6060, streamFile);
-
-        sampleRate = read_32bitBE(0x6024, streamFile);
-        numSamples = read_32bitBE(0x602C, streamFile);
-        startSample = read_32bitBE(0x6028, streamFile);
-        dataSize = read_32bitBE(0x6020, streamFile);
-        blockSize = read_32bitBE(0x6030, streamFile);
-        blockCount = read_32bitBE(0x603C, streamFile);
-    }
-    else
-        goto fail; //cuz I don't know if there are other variants
-
-   /* build the VGMSTREAM */
-    vgmstream = allocate_vgmstream(channel_count,loop_flag);
-    if (!vgmstream) goto fail;
-
-    if (read_32bitBE(0x1000, streamFile) == 0x00000000)
-        start_offset = 0x7000;
-    else
-        start_offset = 0x2000;
-
-    vgmstream->sample_rate = sampleRate;
-    vgmstream->channels = channel_count;
-    vgmstream->num_samples = numSamples;
-    if (loop_flag) {
-        vgmstream->loop_start_sample = startSample;
-        vgmstream->loop_end_sample = vgmstream->num_samples;
-    }
-    vgmstream->meta_type = meta_TA_AAC_X360;
-
-#ifdef VGM_USE_FFMPEG
-    {
-        ffmpeg_codec_data *ffmpeg_data = NULL;
-        uint8_t buf[100];
-        size_t bytes, datasize, block_size, block_count;
-
-        block_count = blockCount;
-        block_size = blockSize;
-        datasize = dataSize;
-
-        bytes = ffmpeg_make_riff_xma2(buf,100, vgmstream->num_samples, datasize, vgmstream->channels, vgmstream->sample_rate, block_count, block_size);
-        ffmpeg_data = init_ffmpeg_header_offset(streamFile, buf,bytes, start_offset,datasize);
-        if ( !ffmpeg_data ) goto fail;
-        vgmstream->codec_data = ffmpeg_data;
-        vgmstream->coding_type = coding_FFmpeg;
-        vgmstream->layout_type = layout_none;
-
-        xma_fix_raw_samples(vgmstream, streamFile, start_offset, datasize, 0, 1,1);
-        if (loop_flag) { /* reapply adjusted samples */
-            vgmstream->loop_end_sample = vgmstream->num_samples;
-        }
-
-    }
-#else
-    goto fail;
-#endif
-
-    /* open the file for reading */
-    if ( !vgmstream_open_stream(vgmstream, streamFile, start_offset) )
-        goto fail;
-    return vgmstream;
-
-fail:
-    close_vgmstream(vgmstream);
-    return NULL;
-}
-
-/* PlayStation 3 Variants (Star Ocean International, Resonance of Fate) */
-VGMSTREAM * init_vgmstream_ta_aac_ps3(STREAMFILE *streamFile) {
-    VGMSTREAM * vgmstream = NULL;
-    off_t start_offset;
-    int loop_flag, channel_count;
-    uint32_t data_size, loop_start, loop_end, codec_id, asc_chunk;
-
-    /* check extension, case insensitive */
-    /* .aac: expected, .laac/ace: for players to avoid hijacking MP4/AAC */
-    if (!check_extensions(streamFile, "aac,laac,ace"))
-        goto fail;
-
-    if (read_32bitBE(0x00, streamFile) != 0x41414320)   /* "AAC " */
-        goto fail;
-
-    /* Find the ASC chunk, That's where the goodies are */
-    asc_chunk = read_32bitBE(0x40, streamFile);
-    if (read_32bitBE(asc_chunk, streamFile) != 0x41534320)   /* "ASC " */
-        goto fail;
-
-    if (read_32bitBE(asc_chunk+0x104, streamFile) != 0xFFFFFFFF)
-        loop_flag = 1;
-    else
-        loop_flag = 0;
-
-    channel_count = read_32bitBE(asc_chunk + 0xF4, streamFile);
-    codec_id = read_32bitBE(asc_chunk + 0xF0, streamFile);
-
-    /* build the VGMSTREAM */
-    vgmstream = allocate_vgmstream(channel_count, loop_flag);
-    if (!vgmstream) goto fail;
-
-    /* ASC header */
-    start_offset = asc_chunk + 0x110;
-    vgmstream->sample_rate = read_32bitBE(asc_chunk + 0xFC, streamFile);
-    vgmstream->channels = channel_count;
-    vgmstream->meta_type = meta_TA_AAC_PS3;
-    data_size = read_32bitBE(asc_chunk + 0xF8, streamFile);
-    loop_start = read_32bitBE(asc_chunk + 0x104, streamFile);
-    loop_end = read_32bitBE(asc_chunk + 0x108, streamFile);
-
-#ifdef VGM_USE_FFMPEG
-    {
-        int block_align, encoder_delay;
-
-        block_align = (codec_id == 4 ? 0x60 : (codec_id == 5 ? 0x98 : 0xC0)) * vgmstream->channels;
-        encoder_delay = 1024 + 69; /* approximate, gets good loops */
-        vgmstream->num_samples = atrac3_bytes_to_samples(data_size, block_align) - encoder_delay;
-
-        vgmstream->codec_data = init_ffmpeg_atrac3_raw(streamFile, start_offset,data_size, vgmstream->num_samples,vgmstream->channels,vgmstream->sample_rate, block_align, encoder_delay);
-        if (!vgmstream->codec_data) goto fail;
-        vgmstream->coding_type = coding_FFmpeg;
-        vgmstream->layout_type = layout_none;
-
-        /* set offset samples (offset 0 jumps to sample 0 > pre-applied delay, and offset end loops after sample end > adjusted delay) */
-        vgmstream->loop_start_sample = atrac3_bytes_to_samples(loop_start, block_align); // - encoder_delay
-        vgmstream->loop_end_sample = atrac3_bytes_to_samples(loop_end, block_align) - encoder_delay;
-    }
-#endif
-
-    /* open the file for reading */
-    if (!vgmstream_open_stream(vgmstream, streamFile, start_offset))
-        goto fail;
-    return vgmstream;
-
-fail:
-    close_vgmstream(vgmstream);
-    return NULL;
-}
-
-/* Android/iOS Variants (Star Ocean Anamnesis (APK v1.9.2), Heaven x Inferno (iOS)) */
-VGMSTREAM * init_vgmstream_ta_aac_mobile_vorbis(STREAMFILE *streamFile) {
-#ifdef VGM_USE_VORBIS
-    off_t start_offset;
-    int8_t codec_id;
-
-    /* check extension, case insensitive */
-    /* .aac: expected, .laac/ace: for players to avoid hijacking MP4/AAC */
-    if (!check_extensions(streamFile, "aac,laac,ace"))
-        goto fail;
-
-    if (read_32bitLE(0x00, streamFile) != 0x41414320)   /* "AAC " */
-        goto fail;
-
-    if (read_32bitLE(0xf0, streamFile) != 0x57415645)   /* "WAVE" */
-        goto fail;
-
-    codec_id = read_8bit(0x104, streamFile);
-    if (codec_id == 0xe) /* Vorbis */
-    {
-        ogg_vorbis_meta_info_t ovmi = {0};
-        VGMSTREAM * result = NULL;
-
-        ovmi.meta_type = meta_TA_AAC_MOBILE;
-        ovmi.loop_start = read_32bitLE(0x140, streamFile);
-        ovmi.loop_end = read_32bitLE(0x144, streamFile);
-        ovmi.loop_flag = ovmi.loop_end > ovmi.loop_start;
-        ovmi.loop_end_found = ovmi.loop_flag;
-
-        start_offset = read_32bitLE(0x120, streamFile);
-        result = init_vgmstream_ogg_vorbis_callbacks(streamFile, NULL, start_offset, &ovmi);
-
-        if (result != NULL) {
-            return result;
-        }
-    }
-
-fail:
-    /* clean up anything we may have opened */
-#endif
-    return NULL;
-}
-
-/* Android/iOS Variants, before they switched to Vorbis (Star Ocean Anamnesis (Android), Heaven x Inferno (iOS)) */
-VGMSTREAM * init_vgmstream_ta_aac_mobile(STREAMFILE *streamFile) {
-    VGMSTREAM * vgmstream = NULL;
-    off_t start_offset;
-    int channel_count, loop_flag, codec;
-    size_t data_size;
-
-
-    /* check extension, case insensitive */
-    /* .aac: expected, .laac: for players to avoid hijacking MP4/AAC */
-    if (!check_extensions(streamFile, "aac,laac"))
-        goto fail;
-
-    if (read_32bitLE(0x00, streamFile) != 0x41414320)   /* "AAC " */
-        goto fail;
-
-    if (read_32bitLE(0xf0, streamFile) != 0x57415645)   /* "WAVE" */
-        goto fail;
-
-    codec = read_8bit(0x104, streamFile);
-    channel_count = read_8bit(0x105, streamFile);
-    /* 0x106: 0x01?, 0x107: 0x10? */
-    data_size = read_32bitLE(0x10c, streamFile); /* usable data only, cuts last frame */
-    start_offset = read_32bitLE(0x120, streamFile);
-    /* 0x124: full data size */
-    loop_flag = (read_32bitLE(0x134, streamFile) > 0);
-
-
-    /* build the VGMSTREAM */
-    vgmstream = allocate_vgmstream(channel_count, loop_flag);
-    if (!vgmstream) goto fail;
-
-    vgmstream->sample_rate = read_32bitLE(0x108, streamFile);
-    vgmstream->meta_type = meta_TA_AAC_MOBILE;
-
-    switch(codec) {
-        case 0x0d:
-            if (read_32bitLE(0x144, streamFile) != 0x40) goto fail; /* frame size */
-            if (read_32bitLE(0x148, streamFile) != (0x40-0x04*channel_count)*2 / channel_count) goto fail; /* frame samples */
-            if (channel_count > 2) goto fail; /* unknown data layout */
-
-            vgmstream->coding_type = coding_ASKA;
-            vgmstream->layout_type = layout_none;
-
-            vgmstream->num_samples = aska_bytes_to_samples(data_size, channel_count);
-            vgmstream->loop_start_sample = aska_bytes_to_samples(read_32bitLE(0x130, streamFile), channel_count);
-            vgmstream->loop_end_sample = aska_bytes_to_samples(read_32bitLE(0x134, streamFile), channel_count);
-            break;
-
-        default:
-            goto fail;
-    }
-
-    if (!vgmstream_open_stream(vgmstream,streamFile,start_offset))
-        goto fail;
-    return vgmstream;
-
-fail:
-    close_vgmstream(vgmstream);
-    return NULL;
-}
-
-/* Vita variants [Judas Code (Vita)] */
-VGMSTREAM * init_vgmstream_ta_aac_vita(STREAMFILE *streamFile) {
-    VGMSTREAM * vgmstream = NULL;
-    off_t start_offset;
-    int channel_count, loop_flag;
-
-
-    /* check extension, case insensitive */
-    /* .aac: expected, .laac: for players to avoid hijacking MP4/AAC */
-    if (!check_extensions(streamFile, "aac,laac"))
-        goto fail;
-
-    if (read_32bitLE(0x00, streamFile) != 0x41414320)   /* "AAC " */
-        goto fail;
-    if (read_32bitLE(0x14, streamFile) != 0x56495441)   /* "VITA" */
-        goto fail;
-    if (read_32bitLE(0x10d0, streamFile) != 0x57415645)   /* "WAVE" */
-        goto fail;
-
-    /* there is a bunch of chunks but we simplify */
-
-    /* 0x10E4: codec 0x08? */
-    channel_count = read_8bit(0x10E5, streamFile);
-    start_offset = read_32bitLE(0x1100, streamFile);
-    loop_flag = (read_32bitLE(0x1114, streamFile) > 0);
-
-
-    /* build the VGMSTREAM */
-    vgmstream = allocate_vgmstream(channel_count, loop_flag);
-    if (!vgmstream) goto fail;
-
-    vgmstream->sample_rate = read_32bitLE(0x10e8, streamFile);
-    vgmstream->meta_type = meta_TA_AAC_VITA;
-
-#ifdef VGM_USE_ATRAC9
-    {
-        atrac9_config cfg = {0};
-
-        cfg.channels = vgmstream->channels;
-        cfg.encoder_delay = read_32bitLE(0x1124,streamFile);
-        cfg.config_data = read_32bitBE(0x1128,streamFile);
-
-        vgmstream->codec_data = init_atrac9(&cfg);
-        if (!vgmstream->codec_data) goto fail;
-        vgmstream->coding_type = coding_ATRAC9;
-        vgmstream->layout_type = layout_none;
-
-        vgmstream->num_samples = atrac9_bytes_to_samples(read_32bitLE(0x10EC, streamFile), vgmstream->codec_data);
-        vgmstream->num_samples -= cfg.encoder_delay;
-        vgmstream->loop_start_sample = atrac9_bytes_to_samples(read_32bitLE(0x1110, streamFile), vgmstream->codec_data);
-        vgmstream->loop_end_sample = atrac9_bytes_to_samples(read_32bitLE(0x1114, streamFile), vgmstream->codec_data);
-    }
-#endif
-
-    if (!vgmstream_open_stream(vgmstream,streamFile,start_offset))
-        goto fail;
-    return vgmstream;
-
-fail:
-    close_vgmstream(vgmstream);
-    return NULL;
-}
+#include "meta.h"
+#include "../coding/coding.h"
+
+/* AAC - tri-Ace (Aska engine) Audio Container */
+
+/* Xbox 360 Variants (Star Ocean 4, End of Eternity, Infinite Undiscovery) */
+VGMSTREAM * init_vgmstream_ta_aac_x360(STREAMFILE *streamFile) {
+    VGMSTREAM * vgmstream = NULL;
+    off_t start_offset;
+    int loop_flag, channel_count;
+    size_t sampleRate, numSamples, startSample, dataSize, blockSize, blockCount; // A mess
+
+    /* check extension, case insensitive */
+    /* .aac: expected, .laac/ace: for players to avoid hijacking MP4/AAC */
+    if ( !check_extensions(streamFile,"aac,laac,ace"))
+        goto fail;
+
+    if (read_32bitBE(0x00,streamFile) != 0x41414320)   /* "AAC " */
+        goto fail;
+
+    /* Ok, let's check what's behind door number 1 */
+    if (read_32bitBE(0x1000, streamFile) == 0x41534320) /* "ASC " */
+    {
+        loop_flag = read_32bitBE(0x1118, streamFile);
+
+        /*Funky Channel Count Checking */
+        if (read_32bitBE(0x1184, streamFile) == 0x7374726D)
+            channel_count = 6;
+        else if (read_32bitBE(0x1154, streamFile) == 0x7374726D)
+            channel_count = 4;
+        else
+            channel_count = read_8bit(0x1134, streamFile);
+
+        sampleRate = read_32bitBE(0x10F4, streamFile);
+        numSamples = read_32bitBE(0x10FC, streamFile);
+        startSample = read_32bitBE(0x10F8, streamFile);
+        dataSize = read_32bitBE(0x10F0, streamFile);
+        blockSize = read_32bitBE(0x1100, streamFile);
+        blockCount = read_32bitBE(0x110C, streamFile);
+    }
+    else if (read_32bitBE(0x1000, streamFile) == 0x57415645) /* "WAVE" */
+    {
+        loop_flag = read_32bitBE(0x1048, streamFile);
+
+        /*Funky Channel Count Checking */
+        if (read_32bitBE(0x10B0, streamFile) == 0x7374726D)
+            channel_count = 6;
+        else if (read_32bitBE(0x1080, streamFile) == 0x7374726D)
+            channel_count = 4;
+        else
+            channel_count = read_8bit(0x1060, streamFile);
+
+        sampleRate = read_32bitBE(0x1024, streamFile);
+        numSamples = read_32bitBE(0x102C, streamFile);
+        startSample = read_32bitBE(0x1028, streamFile);
+        dataSize = read_32bitBE(0x1020, streamFile);
+        blockSize = read_32bitBE(0x1030, streamFile);
+        blockCount = read_32bitBE(0x103C, streamFile);
+    }
+    else if (read_32bitBE(0x1000, streamFile) == 0x00000000) /* some like to be special */
+    {
+        loop_flag = read_32bitBE(0x6048, streamFile);
+
+        /*Funky Channel Count Checking */
+        if (read_32bitBE(0x60B0, streamFile) == 0x7374726D)
+            channel_count = 6;
+        else if (read_32bitBE(0x6080, streamFile) == 0x7374726D)
+            channel_count = 4;
+        else
+            channel_count = read_8bit(0x6060, streamFile);
+
+        sampleRate = read_32bitBE(0x6024, streamFile);
+        numSamples = read_32bitBE(0x602C, streamFile);
+        startSample = read_32bitBE(0x6028, streamFile);
+        dataSize = read_32bitBE(0x6020, streamFile);
+        blockSize = read_32bitBE(0x6030, streamFile);
+        blockCount = read_32bitBE(0x603C, streamFile);
+    }
+    else
+        goto fail; //cuz I don't know if there are other variants
+
+   /* build the VGMSTREAM */
+    vgmstream = allocate_vgmstream(channel_count,loop_flag);
+    if (!vgmstream) goto fail;
+
+    if (read_32bitBE(0x1000, streamFile) == 0x00000000)
+        start_offset = 0x7000;
+    else
+        start_offset = 0x2000;
+
+    vgmstream->sample_rate = sampleRate;
+    vgmstream->channels = channel_count;
+    vgmstream->num_samples = numSamples;
+    if (loop_flag) {
+        vgmstream->loop_start_sample = startSample;
+        vgmstream->loop_end_sample = vgmstream->num_samples;
+    }
+    vgmstream->meta_type = meta_TA_AAC_X360;
+
+#ifdef VGM_USE_FFMPEG
+    {
+        ffmpeg_codec_data *ffmpeg_data = NULL;
+        uint8_t buf[100];
+        size_t bytes, datasize, block_size, block_count;
+
+        block_count = blockCount;
+        block_size = blockSize;
+        datasize = dataSize;
+
+        bytes = ffmpeg_make_riff_xma2(buf,100, vgmstream->num_samples, datasize, vgmstream->channels, vgmstream->sample_rate, block_count, block_size);
+        ffmpeg_data = init_ffmpeg_header_offset(streamFile, buf,bytes, start_offset,datasize);
+        if ( !ffmpeg_data ) goto fail;
+        vgmstream->codec_data = ffmpeg_data;
+        vgmstream->coding_type = coding_FFmpeg;
+        vgmstream->layout_type = layout_none;
+
+        xma_fix_raw_samples(vgmstream, streamFile, start_offset, datasize, 0, 1,1);
+        if (loop_flag) { /* reapply adjusted samples */
+            vgmstream->loop_end_sample = vgmstream->num_samples;
+        }
+
+    }
+#else
+    goto fail;
+#endif
+
+    /* open the file for reading */
+    if ( !vgmstream_open_stream(vgmstream, streamFile, start_offset) )
+        goto fail;
+    return vgmstream;
+
+fail:
+    close_vgmstream(vgmstream);
+    return NULL;
+}
+
+/* PlayStation 3 Variants (Star Ocean International, Resonance of Fate) */
+VGMSTREAM * init_vgmstream_ta_aac_ps3(STREAMFILE *streamFile) {
+    VGMSTREAM * vgmstream = NULL;
+    off_t start_offset;
+    int loop_flag, channel_count;
+    uint32_t data_size, loop_start, loop_end, codec_id, asc_chunk;
+
+    /* check extension, case insensitive */
+    /* .aac: expected, .laac/ace: for players to avoid hijacking MP4/AAC */
+    if (!check_extensions(streamFile, "aac,laac,ace"))
+        goto fail;
+
+    if (read_32bitBE(0x00, streamFile) != 0x41414320)   /* "AAC " */
+        goto fail;
+
+    /* Find the ASC chunk, That's where the goodies are */
+    asc_chunk = read_32bitBE(0x40, streamFile);
+    if (read_32bitBE(asc_chunk, streamFile) != 0x41534320)   /* "ASC " */
+        goto fail;
+
+    if (read_32bitBE(asc_chunk+0x104, streamFile) != 0xFFFFFFFF)
+        loop_flag = 1;
+    else
+        loop_flag = 0;
+
+    channel_count = read_32bitBE(asc_chunk + 0xF4, streamFile);
+    codec_id = read_32bitBE(asc_chunk + 0xF0, streamFile);
+
+    /* build the VGMSTREAM */
+    vgmstream = allocate_vgmstream(channel_count, loop_flag);
+    if (!vgmstream) goto fail;
+
+    /* ASC header */
+    start_offset = asc_chunk + 0x110;
+    vgmstream->sample_rate = read_32bitBE(asc_chunk + 0xFC, streamFile);
+    vgmstream->channels = channel_count;
+    vgmstream->meta_type = meta_TA_AAC_PS3;
+    data_size = read_32bitBE(asc_chunk + 0xF8, streamFile);
+    loop_start = read_32bitBE(asc_chunk + 0x104, streamFile);
+    loop_end = read_32bitBE(asc_chunk + 0x108, streamFile);
+
+#ifdef VGM_USE_FFMPEG
+    {
+        int block_align, encoder_delay;
+
+        block_align = (codec_id == 4 ? 0x60 : (codec_id == 5 ? 0x98 : 0xC0)) * vgmstream->channels;
+        encoder_delay = 1024 + 69; /* approximate, gets good loops */
+        vgmstream->num_samples = atrac3_bytes_to_samples(data_size, block_align) - encoder_delay;
+
+        vgmstream->codec_data = init_ffmpeg_atrac3_raw(streamFile, start_offset,data_size, vgmstream->num_samples,vgmstream->channels,vgmstream->sample_rate, block_align, encoder_delay);
+        if (!vgmstream->codec_data) goto fail;
+        vgmstream->coding_type = coding_FFmpeg;
+        vgmstream->layout_type = layout_none;
+
+        /* set offset samples (offset 0 jumps to sample 0 > pre-applied delay, and offset end loops after sample end > adjusted delay) */
+        vgmstream->loop_start_sample = atrac3_bytes_to_samples(loop_start, block_align); // - encoder_delay
+        vgmstream->loop_end_sample = atrac3_bytes_to_samples(loop_end, block_align) - encoder_delay;
+    }
+#endif
+
+    /* open the file for reading */
+    if (!vgmstream_open_stream(vgmstream, streamFile, start_offset))
+        goto fail;
+    return vgmstream;
+
+fail:
+    close_vgmstream(vgmstream);
+    return NULL;
+}
+
+/* Android/iOS Variants (Star Ocean Anamnesis (APK v1.9.2), Heaven x Inferno (iOS)) */
+VGMSTREAM * init_vgmstream_ta_aac_mobile_vorbis(STREAMFILE *streamFile) {
+#ifdef VGM_USE_VORBIS
+    off_t start_offset;
+    int8_t codec_id;
+
+    /* check extension, case insensitive */
+    /* .aac: expected, .laac/ace: for players to avoid hijacking MP4/AAC */
+    if (!check_extensions(streamFile, "aac,laac,ace"))
+        goto fail;
+
+    if (read_32bitLE(0x00, streamFile) != 0x41414320)   /* "AAC " */
+        goto fail;
+
+    if (read_32bitLE(0xf0, streamFile) != 0x57415645)   /* "WAVE" */
+        goto fail;
+
+    codec_id = read_8bit(0x104, streamFile);
+    if (codec_id == 0xe) /* Vorbis */
+    {
+        ogg_vorbis_meta_info_t ovmi = {0};
+        VGMSTREAM * result = NULL;
+
+        ovmi.meta_type = meta_TA_AAC_MOBILE;
+        ovmi.loop_start = read_32bitLE(0x140, streamFile);
+        ovmi.loop_end = read_32bitLE(0x144, streamFile);
+        ovmi.loop_flag = ovmi.loop_end > ovmi.loop_start;
+        ovmi.loop_end_found = ovmi.loop_flag;
+
+        start_offset = read_32bitLE(0x120, streamFile);
+        result = init_vgmstream_ogg_vorbis_callbacks(streamFile, NULL, start_offset, &ovmi);
+
+        if (result != NULL) {
+            return result;
+        }
+    }
+
+fail:
+    /* clean up anything we may have opened */
+#endif
+    return NULL;
+}
+
+/* Android/iOS Variants, before they switched to Vorbis (Star Ocean Anamnesis (Android), Heaven x Inferno (iOS)) */
+VGMSTREAM * init_vgmstream_ta_aac_mobile(STREAMFILE *streamFile) {
+    VGMSTREAM * vgmstream = NULL;
+    off_t start_offset;
+    int channel_count, loop_flag, codec;
+    size_t data_size;
+
+
+    /* check extension, case insensitive */
+    /* .aac: expected, .laac: for players to avoid hijacking MP4/AAC */
+    if (!check_extensions(streamFile, "aac,laac"))
+        goto fail;
+
+    if (read_32bitLE(0x00, streamFile) != 0x41414320)   /* "AAC " */
+        goto fail;
+
+    if (read_32bitLE(0xf0, streamFile) != 0x57415645)   /* "WAVE" */
+        goto fail;
+
+    codec = read_8bit(0x104, streamFile);
+    channel_count = read_8bit(0x105, streamFile);
+    /* 0x106: 0x01?, 0x107: 0x10? */
+    data_size = read_32bitLE(0x10c, streamFile); /* usable data only, cuts last frame */
+    start_offset = read_32bitLE(0x120, streamFile);
+    /* 0x124: full data size */
+    loop_flag = (read_32bitLE(0x134, streamFile) > 0);
+
+
+    /* build the VGMSTREAM */
+    vgmstream = allocate_vgmstream(channel_count, loop_flag);
+    if (!vgmstream) goto fail;
+
+    vgmstream->sample_rate = read_32bitLE(0x108, streamFile);
+    vgmstream->meta_type = meta_TA_AAC_MOBILE;
+
+    switch(codec) {
+        case 0x0d:
+            if (read_32bitLE(0x144, streamFile) != 0x40) goto fail; /* frame size */
+            if (read_32bitLE(0x148, streamFile) != (0x40-0x04*channel_count)*2 / channel_count) goto fail; /* frame samples */
+            if (channel_count > 2) goto fail; /* unknown data layout */
+
+            vgmstream->coding_type = coding_ASKA;
+            vgmstream->layout_type = layout_none;
+
+            vgmstream->num_samples = aska_bytes_to_samples(data_size, channel_count);
+            vgmstream->loop_start_sample = aska_bytes_to_samples(read_32bitLE(0x130, streamFile), channel_count);
+            vgmstream->loop_end_sample = aska_bytes_to_samples(read_32bitLE(0x134, streamFile), channel_count);
+            break;
+
+        default:
+            goto fail;
+    }
+
+    if (!vgmstream_open_stream(vgmstream,streamFile,start_offset))
+        goto fail;
+    return vgmstream;
+
+fail:
+    close_vgmstream(vgmstream);
+    return NULL;
+}
+
+/* Vita variants [Judas Code (Vita)] */
+VGMSTREAM * init_vgmstream_ta_aac_vita(STREAMFILE *streamFile) {
+    VGMSTREAM * vgmstream = NULL;
+    off_t start_offset;
+    int channel_count, loop_flag;
+
+
+    /* check extension, case insensitive */
+    /* .aac: expected, .laac: for players to avoid hijacking MP4/AAC */
+    if (!check_extensions(streamFile, "aac,laac"))
+        goto fail;
+
+    if (read_32bitLE(0x00, streamFile) != 0x41414320)   /* "AAC " */
+        goto fail;
+    if (read_32bitLE(0x14, streamFile) != 0x56495441)   /* "VITA" */
+        goto fail;
+    if (read_32bitLE(0x10d0, streamFile) != 0x57415645)   /* "WAVE" */
+        goto fail;
+
+    /* there is a bunch of chunks but we simplify */
+
+    /* 0x10E4: codec 0x08? */
+    channel_count = read_8bit(0x10E5, streamFile);
+    start_offset = read_32bitLE(0x1100, streamFile);
+    loop_flag = (read_32bitLE(0x1114, streamFile) > 0);
+
+
+    /* build the VGMSTREAM */
+    vgmstream = allocate_vgmstream(channel_count, loop_flag);
+    if (!vgmstream) goto fail;
+
+    vgmstream->sample_rate = read_32bitLE(0x10e8, streamFile);
+    vgmstream->meta_type = meta_TA_AAC_VITA;
+
+#ifdef VGM_USE_ATRAC9
+    {
+        atrac9_config cfg = {0};
+
+        cfg.channels = vgmstream->channels;
+        cfg.encoder_delay = read_32bitLE(0x1124,streamFile);
+        cfg.config_data = read_32bitBE(0x1128,streamFile);
+
+        vgmstream->codec_data = init_atrac9(&cfg);
+        if (!vgmstream->codec_data) goto fail;
+        vgmstream->coding_type = coding_ATRAC9;
+        vgmstream->layout_type = layout_none;
+
+        vgmstream->num_samples = atrac9_bytes_to_samples(read_32bitLE(0x10EC, streamFile), vgmstream->codec_data);
+        vgmstream->num_samples -= cfg.encoder_delay;
+        vgmstream->loop_start_sample = atrac9_bytes_to_samples(read_32bitLE(0x1110, streamFile), vgmstream->codec_data);
+        vgmstream->loop_end_sample = atrac9_bytes_to_samples(read_32bitLE(0x1114, streamFile), vgmstream->codec_data);
+    }
+#endif
+
+    if (!vgmstream_open_stream(vgmstream,streamFile,start_offset))
+        goto fail;
+    return vgmstream;
+
+fail:
+    close_vgmstream(vgmstream);
+    return NULL;
+}