Merge pull request #121 from bnnm/awc-rws-misc

AWC, RWS, misc

fb2k/foo_filetypes.h

@@ -49,6 +49,7 @@ VGMSTREAM_DECLARE_FILE_TYPE("AST", ast);
 VGMSTREAM_DECLARE_FILE_TYPE("ATRAC3plus", at3);
 VGMSTREAM_DECLARE_FILE_TYPE("AUD", aud);
 VGMSTREAM_DECLARE_FILE_TYPE("AUS", aus);
+VGMSTREAM_DECLARE_FILE_TYPE("AWC", awc);
 
 VGMSTREAM_DECLARE_FILE_TYPE("B1S", b1s);
 VGMSTREAM_DECLARE_FILE_TYPE("BAF", baf);

readme.txt

@@ -63,7 +63,7 @@ supports Winamp plugins you may also use in_vgmstream.dll instead.
 
 Because the XMPlay MP3 decoder incorrectly tries to play some vgmstream exts,
 you need to manually fix it by going to options > plugins > input > vgmstream
-and in the "priority filetypes" put: ckd,fsb,genh,msf,p3d,rak,scd,xvag
+and in the "priority filetypes" put: awc,ckd,fsb,genh,msf,p3d,rak,scd,xvag
 
 --- foo_input_vgmstream ---
 Every should be installed automatically by the .fb2k-component bundle.

src/coding/coding.h

@@ -30,7 +30,8 @@ void decode_ms_ima(VGMSTREAM * vgmstream,VGMSTREAMCHANNEL * stream, sample * out
 void decode_otns_ima(VGMSTREAM * vgmstream, VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int channel);
 void decode_fsb_ima(VGMSTREAM * vgmstream, VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int channel);
 void decode_wwise_ima(VGMSTREAM * vgmstream, VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int channel);
-void decode_ref_ima(VGMSTREAM * vgmstream,VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do,int channel);
+void decode_ref_ima(VGMSTREAM * vgmstream, VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do,int channel);
+void decode_awc_ima(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
 size_t ms_ima_bytes_to_samples(size_t bytes, int block_align, int channels);
 size_t ima_bytes_to_samples(size_t bytes, int channels);
 
@@ -54,16 +55,16 @@ void decode_ngc_afc(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspaci
 
 /* pcm_decoder */
 void decode_pcm16LE(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
-void decode_pcm16LE_int(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
 void decode_pcm16LE_XOR_int(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
 void decode_pcm16BE(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
+void decode_pcm16_int(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int big_endian);
 void decode_pcm8(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
 void decode_pcm8_int(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
 void decode_pcm8_sb_int(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
 void decode_pcm8_unsigned_int(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
 void decode_pcm8_unsigned(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
 void decode_ulaw(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
-void decode_pcmfloat(VGMSTREAM *vgmstream, VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do);
+void decode_pcmfloat(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int big_endian);
 size_t pcm_bytes_to_samples(size_t bytes, int channels, int bits_per_sample);
 
 /* psx_decoder */

src/coding/ea_xa_decoder.c

@@ -212,23 +212,3 @@ void decode_maxis_xa(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspac
         stream->offset=0;
     }
 }
-
-
-/* EA MicroTalk 10:1 / 5:1 */
-/**
- * Rarely used but can be found in the wild: FIFA 2001 (PS2), FIFA Soccer 2002 (PS2)
- *
- * Decoding algorithm is unknown; some info found by analyzing sx.exe output:
- * - Comes in 10:1 or 5:1 compression varieties (the later's byte layout looks similar but has roughly double frame size)
- * - Also with "PCM samples" flag before each frame (later version) or without (first version)
- * - When PCM flag is 0xEE it has 16b ? + 16b num_samples + PCM samples placed right after the frame, but they
- *   are written *before* (presumably so they have something while the frame is decoded), like EALayer3.
- * - VBR ADPCM, apparently similar to Westwood VBR ADPCM: first byte seems a header with mode+count, but after it may
- *   be 8 bytes(?) of coefs/hist (unlike Westwood's), then data. Samples per frame changes with the mode used.
- *   ex. decoding pure silence (0000) takes 0x2E (10:1) or 0x48 (5:1) into 432 samples (RLE mode)
- * - Variable frame size but seems to range from 0x20 to 0x80 (in 5:1 at least)
- * - After a new SCDl block, first byte (in each channel) is a flag but various values have no effect in the output
- *   (01=first block, 00=normal block?) and should be skipped in the block parser.
- *
- */
-//void decode_ea_mt10(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do,int channel) {

src/coding/ima_decoder.c

@@ -678,6 +678,43 @@ void decode_ref_ima(VGMSTREAM * vgmstream,VGMSTREAMCHANNEL * stream, sample * ou
     stream->adpcm_step_index = step_index;
 }
 
+void decode_awc_ima(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do) {
+    int i, sample_count;
+
+    int32_t hist1 = stream->adpcm_history1_32;
+    int step_index = stream->adpcm_step_index;
+
+    //internal interleave, mono
+    int block_samples = (0x800 - 4) * 2;
+    first_sample = first_sample % block_samples;
+
+    //inverted header
+    if (first_sample == 0) {
+        off_t header_offset = stream->offset;
+
+        step_index = read_16bitLE(header_offset,stream->streamfile);
+        hist1 = read_16bitLE(header_offset+2,stream->streamfile);
+        if (step_index < 0) step_index=0;
+        if (step_index > 88) step_index=88;
+    }
+
+    for (i=first_sample,sample_count=0; i<first_sample+samples_to_do; i++,sample_count+=channelspacing) {
+        off_t byte_offset = stream->offset + 4 + i/2;
+        int nibble_shift = (i&1?4:0); //low nibble first
+
+        ms_ima_expand_nibble(stream, byte_offset,nibble_shift, &hist1, &step_index);
+        outbuf[sample_count] = (short)(hist1);
+    }
+
+    //internal interleave: increment offset on complete frame
+    if (i == block_samples) stream->offset += 0x800;
+
+    stream->adpcm_history1_32 = hist1;
+    stream->adpcm_step_index = step_index;
+}
+
+
+
 size_t ms_ima_bytes_to_samples(size_t bytes, int block_align, int channels) {
     /* MS IMA blocks have a 4 byte header per channel; 2 samples per byte (2 nibbles) */
     return (bytes / block_align) * (block_align - 4 * channels) * 2 / channels;

src/coding/mpeg_custom_utils.c

@@ -54,7 +54,6 @@ int mpeg_custom_setup_init_default(STREAMFILE *streamFile, off_t start_offset, m
             break;
 
         case MPEG_LYN:
-        case MPEG_AWC:
             goto fail; /* not fully implemented */
 
         case MPEG_STANDARD:

src/coding/mpeg_custom_utils_awc.c

@@ -0,0 +1,138 @@
+#include "mpeg_decoder.h"
+
+#ifdef VGM_USE_MPEG
+
+/**
+ * AWC music uses blocks (sfx doesn't), the fun part being each channel has different num_samples/frames
+ * per block, so it's unsuitable for the normal "blocked" layout and parsed here instead.
+ * Channel data is separate within the block (first all frames of ch0, then ch1, etc), padded, and sometimes
+ * the last few frames of a channel are repeated in the new block (marked with the "discard samples" field).
+ */
+
+
+static size_t get_block_header_size(STREAMFILE *streamFile, off_t offset, mpeg_codec_data *data) {
+    size_t header_size = 0;
+    int i;
+    int entries = data->config.channels;
+    int32_t (*read_32bit)(off_t,STREAMFILE*) = data->config.big_endian ? read_32bitBE : read_32bitLE;
+
+    for (i = 0; i < entries; i++) {
+        header_size += 0x18;
+        header_size += read_32bit(offset + 0x18*i + 0x04, streamFile) * 0x04; /* entries in the table */
+    }
+
+    if (header_size % 0x800) /* padded */
+        header_size +=  0x800 - (header_size % 0x800);
+
+    return header_size;
+}
+
+
+/* init config and validate */
+int mpeg_custom_setup_init_awc(STREAMFILE *streamFile, off_t start_offset, mpeg_codec_data *data, coding_t *coding_type) {
+    mpeg_frame_info info;
+    int is_music;
+
+    /* start_offset can point to a block header that always starts with 0 (music) or normal data (sfx) */
+    is_music = read_32bitBE(start_offset, streamFile) == 0x00000000;
+    if (is_music)
+        start_offset += get_block_header_size(streamFile, start_offset, data);
+
+    /* get frame info at offset */
+    if ( !mpeg_get_frame_info(streamFile, start_offset, &info))
+        goto fail;
+    switch(info.layer) {
+        case 1: *coding_type = coding_MPEG_layer1; break;
+        case 2: *coding_type = coding_MPEG_layer2; break;
+        case 3: *coding_type = coding_MPEG_layer3; break;
+        default: goto fail;
+    }
+    data->channels_per_frame = info.channels;
+    data->samples_per_frame = info.frame_samples;
+
+
+    /* extra checks */
+    if (is_music) {
+        if (data->config.chunk_size <= 0)
+            goto fail; /* needs block size */
+    }
+
+    /* no big encoder delay added (for sfx they can start in less than ~300 samples) */
+
+    return 1;
+fail:
+    return 0;
+}
+
+
+/* writes data to the buffer and moves offsets, parsing AWC blocks */
+int mpeg_custom_parse_frame_awc(VGMSTREAMCHANNEL *stream, mpeg_codec_data *data, int num_stream) {
+    mpeg_custom_stream *ms = data->streams[num_stream];
+    mpeg_frame_info info;
+    size_t current_data_size = 0, data_offset;
+    size_t file_size = get_streamfile_size(stream->streamfile);
+    int i;
+
+
+    /* blocked layout used for music */
+    if (data->config.chunk_size) {
+
+        /* block ended for this channel, move to next block start */
+        if (ms->current_size_count > 0 && ms->current_size_count == ms->current_size_target) {
+            //mpg123_open_feed(ms->m); //todo reset maybe needed?
+
+            data_offset = stream->offset - stream->channel_start_offset; /* ignoring header */
+            data_offset -= data_offset % data->config.chunk_size; /* start of current block */
+            stream->offset = stream->channel_start_offset + data_offset + data->config.chunk_size;
+
+            ms->current_size_count = 0;
+            ms->current_size_target = 0;
+        }
+
+        /* just in case, shouldn't happen */
+        if (stream->offset >= file_size) {
+            goto fail;
+        }
+
+        /* block starts for this channel, point to mpeg data */
+        if (ms->current_size_count == 0) {
+            int32_t (*read_32bit)(off_t,STREAMFILE*) = data->config.big_endian ? read_32bitBE : read_32bitLE;
+            off_t channel_offset = 0;
+
+            /* block has a header with base info per channel and table per channel (see blocked_awc.c) */
+            ms->decode_to_discard   = read_32bit(stream->offset + 0x18*num_stream + 0x08, stream->streamfile);
+            ms->current_size_target = read_32bit(stream->offset + 0x18*num_stream + 0x14, stream->streamfile);
+
+            for (i = 0; i < num_stream; i++) { /* num_stream serves as channel */
+                size_t channel_size = read_32bit(stream->offset + 0x18*i + 0x14, stream->streamfile);
+                if (channel_size % 0x10) /* 32b aligned */
+                    channel_size += 0x10 - channel_size % 0x10;
+
+                channel_offset += channel_size;
+            }
+
+            //VGM_ASSERT(ms->decode_to_discard > 0, "AWC: s%i discard of %x found at chunk %lx\n", num_stream, ms->decode_to_discard, stream->offset);
+            //;VGM_LOG("AWC: s%i off=%lx to %lx\n", num_stream, stream->offset, stream->offset + channel_offset + get_block_header_size(stream->streamfile, stream->offset, data));
+
+            stream->offset += channel_offset + get_block_header_size(stream->streamfile, stream->offset, data);
+        }
+    }
+
+
+    /* update frame */
+    if ( !mpeg_get_frame_info(stream->streamfile, stream->offset, &info) )
+        goto fail;
+    current_data_size = info.frame_size;
+
+    data->bytes_in_buffer = read_streamfile(data->buffer,stream->offset, current_data_size, stream->streamfile);
+
+    stream->offset += current_data_size;
+
+    ms->current_size_count += current_data_size;
+
+    return 1;
+fail:
+    return 0;
+}
+
+#endif

src/coding/mpeg_custom_utils_ealayer3.c

@@ -579,7 +579,7 @@ static int ealayer3_write_pcm_block(VGMSTREAMCHANNEL *stream, mpeg_codec_data *d
             if (decode_to_discard == 576)
                 decode_to_discard = data->samples_per_frame;//+ eaf->v1_pcm_number;
 
-            data->decode_to_discard += decode_to_discard;
+            ms->decode_to_discard += decode_to_discard;
         }
     }
 

src/coding/mpeg_decoder.c

@@ -140,6 +140,7 @@ mpeg_codec_data *init_mpeg_custom_codec_data(STREAMFILE *streamFile, off_t start
         case MPEG_EAL31:
         case MPEG_EAL32P:
         case MPEG_EAL32S:   ok = mpeg_custom_setup_init_ealayer3(streamFile, start_offset, data, coding_type); break;
+        case MPEG_AWC:      ok = mpeg_custom_setup_init_awc(streamFile, start_offset, data, coding_type); break;
         default:            ok = mpeg_custom_setup_init_default(streamFile, start_offset, data, coding_type); break;
     }
     if (!ok)
@@ -393,6 +394,7 @@ static void decode_mpeg_custom_stream(VGMSTREAMCHANNEL *stream, mpeg_codec_data
             case MPEG_EAL32P:
             case MPEG_EAL32S:   ok = mpeg_custom_parse_frame_ealayer3(stream, data, num_stream); break;
             case MPEG_AHX:      ok = mpeg_custom_parse_frame_ahx(stream, data); break;
+            case MPEG_AWC:      ok = mpeg_custom_parse_frame_awc(stream, data, num_stream); break;
             default:            ok = mpeg_custom_parse_frame_default(stream, data); break;
         }
         if (!ok) {
@@ -440,16 +442,16 @@ static void decode_mpeg_custom_stream(VGMSTREAMCHANNEL *stream, mpeg_codec_data
     }
     samples_filled = (bytes_done / sizeof(sample) / data->channels_per_frame);
 
-    /* for EALayer3, that discards decoded samples and writes PCM blocks instead */
+    /* discard for weird features (EALayer3 and PCM blocks, AWC and repeated frames) */
-    if (data->decode_to_discard) {
+    if (ms->decode_to_discard) {
         size_t bytes_to_discard = 0;
-        size_t decode_to_discard = data->decode_to_discard;
+        size_t decode_to_discard = ms->decode_to_discard;
         if (decode_to_discard > samples_filled)
             decode_to_discard = samples_filled;
         bytes_to_discard = sizeof(sample)*decode_to_discard*data->channels_per_frame;
 
         bytes_done -= bytes_to_discard;
-        data->decode_to_discard -= decode_to_discard;
+        ms->decode_to_discard -= decode_to_discard;
         ms->samples_used += decode_to_discard;
     }
 
@@ -522,10 +524,10 @@ void reset_mpeg(VGMSTREAM *vgmstream) {
             mpg123_feedseek(data->streams[i]->m,0,SEEK_SET,&input_offset);
             data->streams[i]->samples_filled = 0;
             data->streams[i]->samples_used = 0;
+            data->streams[i]->decode_to_discard = 0;
         }
 
         data->samples_to_discard = data->skip_samples; /* initial delay */
-        data->decode_to_discard = 0;
     }
 }
 
@@ -546,6 +548,7 @@ void seek_mpeg(VGMSTREAM *vgmstream, int32_t num_sample) {
             mpg123_feedseek(data->streams[i]->m,0,SEEK_SET,&input_offset);
             data->streams[i]->samples_filled = 0;
             data->streams[i]->samples_used = 0;
+            data->streams[i]->decode_to_discard = 0;
 
             if (vgmstream->loop_ch)
                 vgmstream->loop_ch[i].offset = vgmstream->loop_ch[i].channel_start_offset;
@@ -558,7 +561,6 @@ void seek_mpeg(VGMSTREAM *vgmstream, int32_t num_sample) {
 
     data->buffer_full = 0;
     data->buffer_used = 0;
-    data->decode_to_discard = 0;
 }
 
 /* resets mpg123 decoder and its internals (with mpg123_open_feed as mpg123_feedseek won't work) */
@@ -577,10 +579,10 @@ void flush_mpeg(mpeg_codec_data * data) {
             mpg123_open_feed(data->streams[i]->m);
             data->streams[i]->samples_filled = 0;
             data->streams[i]->samples_used = 0;
+            data->streams[i]->decode_to_discard = 0;
         }
 
         data->samples_to_discard = data->skip_samples; /* initial delay */
-        data->decode_to_discard = 0;
     }
 
     data->bytes_in_buffer = 0;

src/coding/mpeg_decoder.h

@@ -19,10 +19,12 @@ int mpeg_get_frame_info(STREAMFILE *streamfile, off_t offset, mpeg_frame_info *
 
 int mpeg_custom_setup_init_default(STREAMFILE *streamFile, off_t start_offset, mpeg_codec_data *data, coding_t *coding_type);
 int mpeg_custom_setup_init_ealayer3(STREAMFILE *streamFile, off_t start_offset, mpeg_codec_data *data, coding_t *coding_type);
+int mpeg_custom_setup_init_awc(STREAMFILE *streamFile, off_t start_offset, mpeg_codec_data *data, coding_t *coding_type);
 
 int mpeg_custom_parse_frame_default(VGMSTREAMCHANNEL *stream, mpeg_codec_data *data);
 int mpeg_custom_parse_frame_ahx(VGMSTREAMCHANNEL *stream, mpeg_codec_data *data);
 int mpeg_custom_parse_frame_ealayer3(VGMSTREAMCHANNEL *stream, mpeg_codec_data *data, int num_stream);
+int mpeg_custom_parse_frame_awc(VGMSTREAMCHANNEL *stream, mpeg_codec_data *data, int num_stream);
 
 #endif/* VGM_USE_MPEG */
 

src/coding/pcm_decoder.c

@@ -68,12 +68,12 @@ void decode_pcm8_unsigned(VGMSTREAMCHANNEL * stream, sample * outbuf, int channe
     }
 }
 
-void decode_pcm16LE_int(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do) {
+void decode_pcm16_int(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int big_endian) {
-    int i;
+    int i, sample_count;
-    int32_t sample_count;
+    int16_t (*read_16bit)(off_t,STREAMFILE*) = big_endian ? read_16bitBE : read_16bitLE;
 
     for (i=first_sample,sample_count=0; i<first_sample+samples_to_do; i++,sample_count+=channelspacing) {
-        outbuf[sample_count]=read_16bitLE(stream->offset+i*2*channelspacing,stream->streamfile);
+        outbuf[sample_count]=read_16bit(stream->offset+i*2*channelspacing,stream->streamfile);
     }
 }
 
@@ -118,17 +118,17 @@ void decode_ulaw(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing,
     }
 }
 
-void decode_pcmfloat(VGMSTREAM *vgmstream, VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do) {
+void decode_pcmfloat(VGMSTREAMCHANNEL * stream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int big_endian) {
     int i, sample_count;
-    int32_t (*read_32bit)(off_t,STREAMFILE*) = vgmstream->codec_endian ? read_32bitBE : read_32bitLE;
+    int32_t (*read_32bit)(off_t,STREAMFILE*) = big_endian ? read_32bitBE : read_32bitLE;
 
     for (i=first_sample,sample_count=0; i<first_sample+samples_to_do; i++,sample_count+=channelspacing) {
         uint32_t sample_int = read_32bit(stream->offset+i*4,stream->streamfile);
-        float sample_float;
+        float* sample_float;
         int sample_pcm;
 
-        memcpy(&sample_float, &sample_int, 4); /* maybe unorthodox but simplest */
+        sample_float = (float*)&sample_int;
-        sample_pcm = floor(sample_float * 32767.f + .5f);
+        sample_pcm = floor((*sample_float) * 32767.f + .5f);
 
         outbuf[sample_count] = clamp16(sample_pcm);
     }

src/formats.c

@@ -42,6 +42,7 @@ static const char* extension_list[] = {
     "at3",
     "aud",
     "aus",
+    "awc",
 
     "b1s",
     "baf",
@@ -395,10 +396,10 @@ typedef struct {
 
 
 static const coding_info coding_info_list[] = {
-        {coding_PCM16BE,            "Big Endian 16-bit PCM"},
         {coding_PCM16LE,            "Little Endian 16-bit PCM"},
-        {coding_PCM16LE_int,        "Little Endian 16-bit PCM with 2 byte interleave"},
         {coding_PCM16LE_XOR_int,    "Little Endian 16-bit PCM with 2 byte interleave and XOR obfuscation"},
+        {coding_PCM16BE,            "Big Endian 16-bit PCM"},
+        {coding_PCM16_int,          "16-bit PCM with 2 byte interleave"},
         {coding_PCM8,               "8-bit PCM"},
         {coding_PCM8_U,             "8-bit unsigned PCM"},
         {coding_PCM8_U_int,         "8-bit unsigned PCM with 1 byte interleave"},
@@ -448,6 +449,7 @@ static const coding_info coding_info_list[] = {
         {coding_FSB_IMA,            "FSB multichannel 4-bit IMA ADPCM"},
         {coding_WWISE_IMA,          "Audiokinetic Wwise 4-bit IMA ADPCM"},
         {coding_REF_IMA,            "Reflections 4-bit IMA ADPCM"},
+        {coding_AWC_IMA,            "Rockstar AWC 4-bit IMA ADPCM"},
         {coding_WS,                 "Westwood Studios VBR ADPCM"},
         {coding_ACM,                "InterPlay ACM"},
         {coding_NWA0,               "NWA DPCM Level 0"},
@@ -532,6 +534,8 @@ static const layout_info layout_info_list[] = {
         {layout_aax,                    "AAX blocked, 18-byte interleaved"},
         {layout_scd_int,                "SCD multistream interleave"},
         {layout_ea_sns_blocked,         "Electronic Arts SNS blocked"},
+        {layout_blocked_awc,            "blocked (AWC)"},
+        {layout_blocked_vgs,            "blocked (VGS)"},
 #ifdef VGM_USE_VORBIS
         {layout_ogg_vorbis,             "Ogg"},
 #endif
@@ -716,7 +720,7 @@ static const meta_info meta_info_list[] = {
         {meta_MSVP,                 "MSVP Header"},
         {meta_NGC_SSM,              "SSM DSP Header"},
         {meta_PS2_JOE,              "Disney/Pixar JOE Header"},
-        {meta_VGS,                  "Guitar Hero Encore Rocks the 80's Header"},
+        {meta_VGS,                  "Guitar Hero VGS Header"},
         {meta_DC_DCSW_DCS,          "Evil Twin DCS file with helper"},
         {meta_WII_SMP,              "SMP DSP Header"},
         {meta_EMFF_PS2,             "Eidos Music File Format Header"},
@@ -880,6 +884,7 @@ static const meta_info meta_info_list[] = {
         {meta_STM,                  "Angel Studios/Rockstar San Diego STMA header"},
         {meta_BINK,                 "RAD Game Tools Bink header"},
         {meta_EA_SNU,               "Electronic Arts SNU header"},
+        {meta_AWC,                  "Rockstar AWC header"},
 
 #ifdef VGM_USE_VORBIS
         {meta_OGG_VORBIS,           "Ogg Vorbis"},

src/layout/blocked.c

@@ -143,6 +143,12 @@ void render_vgmstream_blocked(sample * buffer, int32_t sample_count, VGMSTREAM *
                     break;
                 case layout_ea_sns_blocked:
                     ea_sns_block_update(vgmstream->next_block_offset,vgmstream);
+                    break;
+                case layout_blocked_awc:
+                    block_update_awc(vgmstream->next_block_offset,vgmstream);
+                    break;
+                case layout_blocked_vgs:
+                    block_update_vgs(vgmstream->next_block_offset,vgmstream);
                     break;
 				default:
                     break;

src/layout/blocked_awc.c

@@ -0,0 +1,60 @@
+#include "layout.h"
+#include "../coding/coding.h"
+#include "../layout/layout.h"
+#include "../vgmstream.h"
+
+
+static size_t get_block_header_size(STREAMFILE *streamFile, off_t offset, int channels, int big_endian);
+
+/* AWC music chunks  */
+void block_update_awc(off_t block_offset, VGMSTREAM * vgmstream) {
+    STREAMFILE* streamFile = vgmstream->ch[0].streamfile;
+    int32_t (*read_32bit)(off_t,STREAMFILE*) = vgmstream->codec_endian ? read_32bitBE : read_32bitLE;
+    size_t header_size, entries, block_size, block_samples;
+    int i;
+
+    /* assumed only AWC_IMA enters here, MPEG/XMA2 need special parsing as blocked layout is too limited */
+
+    entries = read_32bit(block_offset + 0x18*0 + 0x04, streamFile); /* assumed same for all channels */
+    block_samples = entries * (0x800-4)*2;
+    block_size = vgmstream->full_block_size;
+
+    vgmstream->current_block_offset = block_offset;
+    vgmstream->next_block_offset = block_offset + block_size;
+    vgmstream->current_block_samples = block_samples;
+
+    /* starts with a header block */
+    /* for each channel
+     *   0x00: start entry within channel (ie. entries * ch)
+     *   0x04: entries
+     *   0x08: samples to discard in the beginning of this block (MPEG only?)
+     *   0x0c: samples in channel (for MPEG/XMA2 can vary between channels)
+     *   0x10: MPEG only: close to number of frames but varies a bit?
+     *   0x14: MPEG only: channel usable data size (not counting padding)
+     * for each channel
+     *   32b * entries = global samples per frame in each block (for MPEG probably per full frame)
+     */
+
+    header_size = get_block_header_size(streamFile, block_offset, vgmstream->channels, vgmstream->codec_endian);
+    for (i = 0; i < vgmstream->channels; i++) {
+        vgmstream->ch[i].offset = block_offset + header_size + 0x800*entries*i;
+    }
+
+}
+
+static size_t get_block_header_size(STREAMFILE *streamFile, off_t offset, int channels, int big_endian) {
+    size_t header_size = 0;
+    int i;
+    int entries = channels;
+    int32_t (*read_32bit)(off_t,STREAMFILE*) = big_endian ? read_32bitBE : read_32bitLE;
+
+    for (i = 0; i < entries; i++) {
+        header_size += 0x18;
+        header_size += read_32bit(offset + 0x18*i + 0x04, streamFile) * 0x04; /* entries in the table */
+    }
+
+    if (header_size % 0x800) /* padded */
+        header_size +=  0x800 - (header_size % 0x800);
+
+    return header_size;
+}

src/layout/blocked_vgs.c

@@ -0,0 +1,32 @@
+#include "layout.h"
+#include "../coding/coding.h"
+#include "../layout/layout.h"
+#include "../vgmstream.h"
+
+
+/* VGS multistream frames */
+void block_update_vgs(off_t block_offset, VGMSTREAM * vgmstream) {
+    STREAMFILE* streamFile = vgmstream->ch[0].streamfile;
+    size_t file_size = get_streamfile_size(vgmstream->ch[0].streamfile);
+    int i;
+    size_t channel_size = 0x10;
+
+
+    /* set offsets */
+    for (i = 0; i < vgmstream->channels; i++) {
+        vgmstream->ch[i].offset = block_offset + channel_size*i;
+    }
+
+    vgmstream->current_block_size = channel_size;
+    vgmstream->current_block_offset = block_offset;
+    vgmstream->next_block_offset = block_offset + channel_size*vgmstream->channels;
+
+    /* skip unhandled tracks: flag can be 0x0n per track, of 0x8x for last frame */
+    while (vgmstream->next_block_offset < file_size) {
+        if ((read_8bit(vgmstream->next_block_offset + 0x01, streamFile) & 0x0F) == 0x00)
+                break;
+
+        vgmstream->next_block_offset += channel_size;
+    }
+
+}

src/layout/ea_block.c

@@ -211,7 +211,7 @@ void eacs_block_update(off_t block_offset, VGMSTREAM * vgmstream) {
         } else {
 
             for (i=0;i<vgmstream->channels;i++) {
-                if(vgmstream->coding_type==coding_PCM16LE_int)
+                if(vgmstream->coding_type==coding_PCM16_int)
                     vgmstream->ch[i].offset = block_offset+(i*2);
                 else
                     vgmstream->ch[i].offset = block_offset+i;

src/layout/layout.h

@@ -65,6 +65,9 @@ void hwas_block_update(off_t block_offset, VGMSTREAM * vgmstream);
 
 void ea_sns_block_update(off_t block_offset, VGMSTREAM * vgmstream);
 
+void block_update_awc(off_t block_offset, VGMSTREAM * vgmstream);
+void block_update_vgs(off_t block_offset, VGMSTREAM * vgmstream);
+
 /* other layouts */
 void render_vgmstream_interleave(sample * buffer, int32_t sample_count, VGMSTREAM * vgmstream);
 

src/libvgmstream.vcproj

@@ -1398,6 +1398,10 @@
                     RelativePath=".\coding\mpeg_custom_utils_ahx.c"
                     >
                 </File>
+                <File
+                    RelativePath=".\coding\mpeg_custom_utils_awc.c"
+                    >
+                </File>
                 <File
                     RelativePath=".\coding\mpeg_custom_utils_ealayer3.c"
                     >
@@ -1530,6 +1534,14 @@
 					RelativePath=".\layout\blocked.c"
 					>
 				</File>
+                <File
+                    RelativePath=".\layout\blocked_awc.c"
+                    >
+                </File>
+                <File
+                    RelativePath=".\layout\blocked_vgs.c"
+                    >
+                </File>
 				<File
 					RelativePath=".\layout\caf_blocked.c"
 					>

src/libvgmstream.vcxproj

@@ -427,6 +427,7 @@
     <ClCompile Include="coding\mc3_decoder.c" />
     <ClCompile Include="coding\mpeg_custom_utils.c" />
     <ClCompile Include="coding\mpeg_custom_utils_ahx.c" />
+    <ClCompile Include="coding\mpeg_custom_utils_awc.c" />
     <ClCompile Include="coding\mpeg_custom_utils_ealayer3.c" />
     <ClCompile Include="coding\mpeg_decoder.c" />
     <ClCompile Include="coding\msadpcm_decoder.c" />
@@ -453,6 +454,8 @@
     <ClCompile Include="layout\ast_blocked.c" />
     <ClCompile Include="layout\bdsp_blocked.c" />
     <ClCompile Include="layout\blocked.c" />
+    <ClCompile Include="layout\blocked_awc.c" />
+    <ClCompile Include="layout\blocked_vgs.c" />
     <ClCompile Include="layout\caf_blocked.c" />
     <ClCompile Include="layout\de2_blocked.c" />
     <ClCompile Include="layout\ea_block.c" />

src/libvgmstream.vcxproj.filters

@@ -817,6 +817,9 @@
     <ClCompile Include="coding\mpeg_custom_utils_ahx.c">
       <Filter>coding\Source Files</Filter>
     </ClCompile>
+    <ClCompile Include="coding\mpeg_custom_utils_awc.c">
+      <Filter>coding\Source Files</Filter>
+    </ClCompile>
     <ClCompile Include="coding\mpeg_custom_utils_ealayer3.c">
       <Filter>coding\Source Files</Filter>
     </ClCompile>
@@ -895,6 +898,12 @@
     <ClCompile Include="layout\blocked.c">
       <Filter>layout\Source Files</Filter>
     </ClCompile>
+    <ClCompile Include="layout\blocked_awc.c">
+      <Filter>layout\Source Files</Filter>
+    </ClCompile>
+    <ClCompile Include="layout\blocked_vgs.c">
+      <Filter>layout\Source Files</Filter>
+    </ClCompile>
     <ClCompile Include="layout\caf_blocked.c">
       <Filter>layout\Source Files</Filter>
     </ClCompile>

src/meta/awc.c

@@ -0,0 +1,276 @@
+#include "meta.h"
+#include "../coding/coding.h"
+#include "../layout/layout.h"
+
+typedef struct {
+    int big_endian;
+    int is_encrypted;
+    int is_music;
+
+    int total_streams;
+
+    int channel_count;
+    int sample_rate;
+    int codec;
+    int num_samples;
+
+    int block_chunk;
+
+    off_t stream_offset;
+    off_t stream_size;
+
+} awc_header;
+
+static int parse_awc_header(STREAMFILE* streamFile, awc_header* awc);
+
+
+/* AWC - from RAGE (Rockstar Advanced Game Engine) audio (Red Dead Redemption, Max Payne 3, GTA5) */
+VGMSTREAM * init_vgmstream_awc(STREAMFILE *streamFile) {
+    VGMSTREAM * vgmstream = NULL;
+    awc_header awc;
+
+    /* check extension */
+    if (!check_extensions(streamFile,"awc"))
+        goto fail;
+
+    /* check header */
+    if (!parse_awc_header(streamFile, &awc))
+        goto fail;
+
+    if (awc.is_encrypted)
+        goto fail;
+
+
+    /* build the VGMSTREAM */
+    vgmstream = allocate_vgmstream(awc.channel_count, 0);
+    if (!vgmstream) goto fail;
+
+    vgmstream->sample_rate = awc.sample_rate;
+    vgmstream->num_samples = awc.num_samples;
+    vgmstream->num_streams = awc.total_streams;
+    vgmstream->meta_type = meta_AWC;
+
+
+    switch(awc.codec) {
+        case 0x01:      /* PCM (PC/PS3) [sfx, rarely] */
+            if (awc.is_music) goto fail; /* blocked_awc needs to be prepared */
+            vgmstream->coding_type = awc.big_endian ? coding_PCM16BE : coding_PCM16LE;
+            vgmstream->layout_type = layout_interleave;
+            vgmstream->interleave_block_size = 0x02;
+            break;
+
+        case 0x04:      /* IMA (PC) */
+            vgmstream->coding_type = coding_AWC_IMA;
+            vgmstream->layout_type = awc.is_music ? layout_blocked_awc : layout_none;
+            vgmstream->full_block_size = awc.block_chunk;
+            vgmstream->codec_endian = awc.big_endian;
+            break;
+
+#ifdef VGM_USE_MPEG
+        case 0x07: {    /* MPEG (PS3) */
+            mpeg_custom_config cfg;
+            memset(&cfg, 0, sizeof(mpeg_custom_config));
+
+            cfg.chunk_size = awc.block_chunk;
+            cfg.big_endian = awc.big_endian;
+
+            vgmstream->codec_data = init_mpeg_custom_codec_data(streamFile, awc.stream_offset, &vgmstream->coding_type, vgmstream->channels, MPEG_AWC, &cfg);
+            if (!vgmstream->codec_data) goto fail;
+            vgmstream->layout_type = layout_none;
+
+            break;
+        }
+#endif
+
+        case 0x05:      /* XMA2 (X360) */
+        default:
+            VGM_LOG("AWC: unknown codec 0x%02x\n", awc.codec);
+            goto fail;
+    }
+
+
+    /* open files; channel offsets are updated below */
+    if (!vgmstream_open_stream(vgmstream,streamFile,awc.stream_offset))
+        goto fail;
+
+    if (vgmstream->layout_type == layout_blocked_awc)
+        block_update_awc(awc.stream_offset, vgmstream);
+
+    return vgmstream;
+
+fail:
+    close_vgmstream(vgmstream);
+    return NULL;
+}
+
+
+/* Parse Rockstar's AWC header (much info from LibertyV: https://github.com/koolkdev/libertyv).
+ * Made of entries for N streams, each with a number of tags pointing to chunks (header, data, events, etc). */
+static int parse_awc_header(STREAMFILE* streamFile, awc_header* awc) {
+    int64_t (*read_64bit)(off_t,STREAMFILE*) = NULL;
+    int32_t (*read_32bit)(off_t,STREAMFILE*) = NULL;
+    int16_t (*read_16bit)(off_t,STREAMFILE*) = NULL;
+    int i, ch, entries;
+    uint32_t flags, info_header, tag_count = 0, tags_skip = 0;
+    off_t off;
+    int target_stream = streamFile->stream_index;
+
+    memset(awc,0,sizeof(awc_header));
+
+
+    /* check header */
+    if (read_32bitBE(0x00,streamFile) != 0x41444154 &&  /* "ADAT" (LE) */
+        read_32bitBE(0x00,streamFile) != 0x54414441)    /* "TADA" (BE) */
+        goto fail;
+
+    awc->big_endian = read_32bitBE(0x00,streamFile) == 0x54414441;
+    if (awc->big_endian) {
+        read_64bit = read_64bitBE;
+        read_32bit = read_32bitBE;
+        read_16bit = read_16bitBE;
+    } else {
+        read_64bit = read_64bitLE;
+        read_32bit = read_32bitLE;
+        read_16bit = read_16bitLE;
+    }
+
+
+    flags = read_32bit(0x04,streamFile);
+    entries = read_32bit(0x08,streamFile);
+    //header_size = read_32bit(0x0c,streamFile); /* after to stream id/tags, not including chunks */
+
+    off = 0x10;
+
+    if ((flags & 0xFF00FFFF) != 0xFF000001 || (flags & 0x00F00000)) {
+        VGM_LOG("AWC: unknown flags 0x%08x\n", flags);
+        goto fail;
+    }
+
+    if (flags & 0x00010000) /* some kind of mini offset table */
+        off += 0x2 * entries;
+    //if (flags % 0x00020000) /* seems to indicate chunks are not ordered (ie. header may go after data) */
+    //  ...
+    //if (flags % 0x00040000) /* music/multichannel flag? (GTA5, not seen in RDR) */
+    //  awc->is_music = 1;
+    if (flags & 0x00080000) /* encrypted data chunk (most of GTA5 PC) */
+        awc->is_encrypted = 1;
+
+
+    /* Music when the first id is 0 (base/fake entry with info for all channels), sfx pack otherwise.
+     * sfx = N single streams, music = N-1 interleaved mono channels (even for MP3/XMA).
+     * Music seems layered (N-1/2 stereo pairs), maybe set with events? */
+    awc->is_music = (read_32bit(off + 0x00,streamFile) & 0x1FFFFFFF) == 0x00000000;
+    if (awc->is_music) { /* all streams except id 0 is a channel */
+        awc->total_streams = 1;
+        target_stream = 1; /* we only need id 0, though channels may have its own tags/chunks */
+    }
+    else { /* each stream is a single sound */
+        awc->total_streams = entries;
+        if (target_stream == 0) target_stream = 1;
+        if (target_stream < 0 || target_stream > awc->total_streams || awc->total_streams < 1) goto fail;
+    }
+
+
+    /* get stream base info */
+    for (i = 0; i < entries; i++) {
+        info_header = read_32bit(off + 0x04*i, streamFile);
+        tag_count   = (info_header >> 29) & 0x7; /* 3b */
+        //id        = (info_header >>  0) & 0x1FFFFFFF; /* 29b */
+        if (target_stream-1 == i)
+            break;
+        tags_skip += tag_count; /* tags to skip to reach target's tags, in the next header */
+    }
+    off += 0x04*entries;
+    off += 0x08*tags_skip;
+
+    /* get stream tags */
+    for (i = 0; i < tag_count; i++) {
+        uint64_t tag_header, tag, size, offset;
+
+        tag_header = (uint64_t)read_64bit(off + 0x08*i,streamFile);
+        tag    = (tag_header >> 56) & 0xFF; /* 8b */
+        size   = (tag_header >> 28) & 0x0FFFFFFF; /* 28b */
+        offset = (tag_header >>  0) & 0x0FFFFFFF; /* 28b */
+
+        /* Tags are apparently part of a hash derived from a word ("data", "format", etc).
+         * If music + 1ch, the header and data chunks can repeat for no reason (sometimes not even pointed). */
+        switch(tag) {
+            case 0x55: /* data */
+                awc->stream_offset = offset;
+                awc->stream_size = size;
+                break;
+
+            case 0x48: /* music header */
+                if (!awc->is_music) {
+                    VGM_LOG("AWC: music header found in sfx\n");
+                    goto fail;
+                }
+
+                /* 0x00(32): unknown (some count?) */
+                awc->block_chunk = read_32bit(offset + 0x04,streamFile);
+                awc->channel_count = read_32bit(offset + 0x08,streamFile);
+
+                if (awc->channel_count != entries - 1) { /* not counting id-0 */
+                    VGM_LOG("AWC: number of music channels doesn't match entries\n");
+                    goto fail;
+                }
+
+                for (ch = 0; ch < awc->channel_count; ch++) {
+                    int num_samples, sample_rate, codec;
+                    /* 0x00(32): stream id (not always in the header entries order) */
+                    /* 0x08(16): headroom?, 0x0d(8): round size?, 0x0e(16): unknown (zero?) */
+                    num_samples = read_32bit(offset + 0x0c + 0x10*ch + 0x04,streamFile);
+                    sample_rate = (uint16_t)read_16bit(offset + 0x0c + 0x10*ch + 0x0a,streamFile);
+                    codec = read_8bit(offset + 0x0c + 0x10*ch + 0x0c, streamFile);
+
+                    /* validate as all channels should repeat this */
+                    if ((awc->num_samples && awc->num_samples != num_samples) ||
+                        (awc->sample_rate && awc->sample_rate != sample_rate) ||
+                        (awc->codec && awc->codec != codec)) {
+                        VGM_LOG("AWC: found header diffs between channels\n"); /* can rarely happen in stereo pairs */
+                        goto fail;
+                    }
+
+                    awc->num_samples = num_samples;
+                    awc->sample_rate = sample_rate;
+                    awc->codec = codec;
+                }
+                break;
+
+            case 0xFA: /* sfx header */
+                if (awc->is_music) {
+                    VGM_LOG("AWC: sfx header found in music\n");
+                    goto fail;
+                }
+                /*  0x04(32): -1?, 0x0a(16x4): unknown x4, 0x12: null? */
+                awc->num_samples = read_32bit(offset + 0x00,streamFile);
+                awc->sample_rate = (uint16_t)read_16bit(offset + 0x08,streamFile);
+                awc->codec = read_8bit(offset + 0x13, streamFile);
+                awc->channel_count = 1;
+                break;
+
+            case 0xA3: /* block-to-sample table (32b x number of blocks w/ num_samples at the start of each block) */
+            case 0xBD: /* events (32bx4): type_hash, params_hash, timestamp_ms, flags */
+            default: /* 0x5C=animation/RSC?,  0x68=midi?, 0x36/0x2B/0x5A/0xD9=? */
+                //VGM_LOG("AWC: ignoring unknown tag 0x%02x\n", tag);
+                break;
+        }
+    }
+
+    if (!awc->stream_offset) {
+        VGM_LOG("AWC: stream offset not found\n");
+        goto fail;
+    }
+
+    /* 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\n");
+        goto fail;
+    }
+
+
+    return 1;
+fail:
+    return 0;
+}

src/meta/ea_old.c

@@ -61,7 +61,7 @@ VGMSTREAM * init_vgmstream_eacs(STREAMFILE *streamFile) {
 		vgmstream->sample_rate = ea_header->dwSampleRate;
 		
 		if(ea_header->bCompression==0) {
-			vgmstream->coding_type = coding_PCM16LE_int;
+			vgmstream->coding_type = coding_PCM16_int;
 			if(ea_header->bBits==1)
 				vgmstream->coding_type = coding_PCM8_int;
 		}

src/meta/his.c

@@ -68,7 +68,7 @@ VGMSTREAM * init_vgmstream_his(STREAMFILE *streamFile) {
         vgmstream->coding_type = coding_PCM16LE;
         if (channel_count == 2)
         {
-            vgmstream->coding_type = coding_PCM16LE_int;
+            vgmstream->coding_type = coding_PCM16_int;
             vgmstream->interleave_block_size = 2;
         }
     }

src/meta/meta.h

@@ -684,4 +684,6 @@ VGMSTREAM * init_vgmstream_stm(STREAMFILE * streamFile);
 
 VGMSTREAM * init_vgmstream_ea_snu(STREAMFILE * streamFile);
 
+VGMSTREAM * init_vgmstream_awc(STREAMFILE * streamFile);
+
 #endif /*_META_H*/

src/meta/pc_sob.c

@@ -53,7 +53,7 @@ VGMSTREAM * init_vgmstream_sab(STREAMFILE *streamFile) {
 	vgmstream->current_block_offset=8+32*numSounds;
     vgmstream->channels = channel_count;
 	vgmstream->sample_rate = read_32bitLE(0x20,streamFile);
-    vgmstream->coding_type = coding_PCM16LE_int;
+    vgmstream->coding_type = coding_PCM16_int;
     vgmstream->num_samples = (int32_t)((get_streamfile_size(streamFile)-vgmstream->current_block_offset)/2/channel_count);
 	if(loop_flag)
 	{

src/meta/ps2_vag.c

@@ -63,7 +63,11 @@ VGMSTREAM * init_vgmstream_ps2_vag(STREAMFILE *streamFile) {
             break;
         case 'p': /* "VAGp" (extended) [most common, ex Ratchet & Clank] */
 
-            if ((version <= 0x00000004) && (datasize < filesize / 2)) { /* two VAGp in the same file */
+            if (read_32bitBE(0x6000,streamFile) == 0x56414770) { /* "VAGp" */
+                channel_count = 2; /* The Simpsons Wrestling PSX interleave */
+                loop_flag = 0;
+            }
+            else if ((version <= 0x00000004) && (datasize < filesize / 2)) { /* two VAGp in the same file */
                 if (is_swag)
                     loop_flag = vag_find_loop_offsets(streamFile, 0x30, &loopStart, &loopEnd);
                 else
@@ -129,7 +133,15 @@ VGMSTREAM * init_vgmstream_ps2_vag(STREAMFILE *streamFile) {
         case 'p': // VAGp
             interleave=0x10;
 
-            if ((version == 0x00000004) && (datasize < filesize / 2)) {
+            if (read_32bitBE(0x6000,streamFile) == 0x56414770) { /* "VAGp" */
+                 interleave = 0x6000; /* The Simpsons Wrestling PSX interleave, includes header */
+                 vgmstream->layout_type = layout_interleave;
+                 vgmstream->meta_type = meta_PS2_VAGs;
+
+                 vgmstream->num_samples = datasize / 16 * 28;
+                 start_offset = 0x30;
+            }
+            else if ((version == 0x00000004) && (datasize < filesize / 2)) {
                 vgmstream->channels=2;
                 vgmstream->layout_type=layout_interleave;
                 vgmstream->meta_type=meta_PS2_VAGs;

src/meta/rws.c

@@ -160,9 +160,10 @@ VGMSTREAM * init_vgmstream_rws(STREAMFILE *streamFile) {
 
     switch(codec) {
         case 0xD01BD217:    /* PCM X360 (D01BD217 35874EED B9D9B8E8 6EA9B995) */
-            /* The Legend of Spyro (X360) */
+            /* ex. The Legend of Spyro (X360) */
-            vgmstream->coding_type = coding_PCM16BE;
+            vgmstream->coding_type = coding_PCM16_int;
-            //vgmstream->interleave_block_size = block_size / 2; //0x2; //todo 2ch PCM not working correctly (interleaved PCM not ok?)
+            vgmstream->codec_endian = 1; /* big */
+            vgmstream->interleave_block_size = 0x02; /* only used to setup channels */
 
             vgmstream->num_samples = pcm_bytes_to_samples(stream_size, channel_count, 16);
             break;

src/meta/sqex_scd.c

@@ -225,7 +225,7 @@ VGMSTREAM * init_vgmstream_sqex_scd(STREAMFILE *streamFile) {
     switch (codec_id) {
         case 0x1:
             /* PCM */
-            vgmstream->coding_type = coding_PCM16LE_int;
+            vgmstream->coding_type = coding_PCM16_int;
             vgmstream->layout_type = layout_none;
             vgmstream->num_samples = stream_size / 2 / channel_count;
 

src/meta/vgs.c

@@ -1,100 +1,76 @@
 #include "meta.h"
 #include "../util.h"
+#include "../coding/coding.h"
+#include "../layout/layout.h"
 
-/* VGS (from Guitar Hero Encore - Rocks the 80s) */
+/* VGS  - from Guitar Hero Encore - Rocks the 80s, Guitar Hero II PS2 */
 VGMSTREAM * init_vgmstream_vgs(STREAMFILE *streamFile) {
     VGMSTREAM * vgmstream = NULL;
-    char filename[PATH_LIMIT];
     off_t start_offset;
+    size_t channel_size = 0, stream_data_size, stream_frame_count;
+    int channel_count = 0, loop_flag = 0, sample_rate = 0, stream_sample_rate;
+    int i;
 
-    int loop_flag;
-	int channel_flag;
-	int channel_flag_offset;
-	int channel_count;
 
     /* check extension, case insensitive */
-    streamFile->get_name(streamFile,filename,sizeof(filename));
+    if (!check_extensions(streamFile,"vgs"))
-    if (strcasecmp("vgs",filename_extension(filename))) goto fail;
+        goto fail;
 
     /* check header */
     if (read_32bitBE(0x00,streamFile) != 0x56675321) /* "VgS!" */
         goto fail;
+    /* 0x04: version? */
 
-    loop_flag = 0;
+    /* contains N streams, which can have one less frame, or half frame and sample rate */
-    channel_flag_offset = get_streamfile_size(streamFile)-0x10;
+    for (i = 0; i < 8; i++) {
-	channel_flag = read_32bitBE(channel_flag_offset,streamFile);
+        stream_sample_rate = read_32bitLE(0x08 + 0x08*i + 0x00,streamFile);
-	
+        stream_frame_count = read_32bitLE(0x08 + 0x08*i + 0x04,streamFile);
-	/* Only seen files up to 5 channels, but just
+        stream_data_size = stream_frame_count*0x10;
-	to be sure we will look up to 8 chanels */
+
-	switch (channel_flag) {
+        if (stream_sample_rate == 0)
-		case 0x00800000:
+            break;
-			channel_count = 1;
+
-			break;
+        if (!sample_rate || !channel_size) {
-		case 0x00810000:
+            sample_rate = stream_sample_rate;
-			channel_count = 2;
+            channel_size = stream_data_size;
-			break;
+        }
-		case 0x00820000:
+
-			channel_count = 3;
+        /* some streams end 1 frame early */
-			break;
+        if (channel_size - 0x10 == stream_data_size) {
-		case 0x00830000:
+            channel_size -= 0x10;
-			channel_count = 4;
+        }
-			break;
+
-		case 0x00840000:
+        /* Guitar Hero II sometimes uses half sample rate for last stream */
-			channel_count = 5;
+        if (sample_rate != stream_sample_rate) {
-			break;
+            VGM_LOG("VGS: ignoring stream %i\n", i);
-		case 0x00850000:
+            //total_streams++; // todo handle substreams
-			channel_count = 6;
+            break;
-			break;
+        }
-		case 0x00860000:
+
-			channel_count = 7;
+        channel_count++;
-			break;
+    }
-		case 0x00870000:
-			channel_count = 8;
-		break;
-			default:
-				goto fail;
-		}
 
     
 	/* build the VGMSTREAM */
     vgmstream = allocate_vgmstream(channel_count,loop_flag);
     if (!vgmstream) goto fail;
 
-	/* fill in the vital statistics */
     start_offset = 0x80;
-	vgmstream->channels = channel_count;
+    vgmstream->sample_rate = sample_rate;
-    vgmstream->sample_rate = read_32bitLE(0x08,streamFile);
+    vgmstream->num_samples = ps_bytes_to_samples(channel_size*channel_count, channel_count);
-    vgmstream->coding_type = coding_PSX_badflags;
-    vgmstream->num_samples = (read_32bitLE(0x0C,streamFile)*channel_count*0x10)*28/16/channel_count;
-    if (loop_flag) {
-        vgmstream->loop_start_sample = 0;
-        vgmstream->loop_end_sample = (read_32bitLE(0x0C,streamFile)*channel_count*0x10)*28/16/channel_count;
-    }
 
-    vgmstream->layout_type = layout_interleave;
+    vgmstream->coding_type = coding_PSX_badflags; /* flag = stream/channel number */
-    vgmstream->interleave_block_size = 0x10;
+    vgmstream->layout_type = layout_blocked_vgs;
     vgmstream->meta_type = meta_VGS;
 
-    /* open the file for reading */
-    {
-        int i;
-        STREAMFILE * file;
-        file = streamFile->open(streamFile,filename,STREAMFILE_DEFAULT_BUFFER_SIZE);
-        if (!file) goto fail;
-        for (i=0;i<channel_count;i++) {
-            vgmstream->ch[i].streamfile = file;
 
-            vgmstream->ch[i].channel_start_offset=
+    /* open files; channel offsets are updated below */
-                vgmstream->ch[i].offset=start_offset+
+    if (!vgmstream_open_stream(vgmstream,streamFile,start_offset))
-                vgmstream->interleave_block_size*i;
+        goto fail;
-
+    block_update_vgs(start_offset, vgmstream);
-        }
-    }
 
     return vgmstream;
-
-    /* clean up anything we may have opened */
 fail:
-    if (vgmstream) close_vgmstream(vgmstream);
+    close_vgmstream(vgmstream);
     return NULL;
 }

src/streamfile.h

@@ -134,6 +134,18 @@ static inline int32_t read_32bitBE(off_t offset, STREAMFILE * streamfile) {
     if (read_streamfile(buf,offset,4,streamfile)!=4) return -1;
     return get_32bitBE(buf);
 }
+static inline int64_t read_64bitLE(off_t offset, STREAMFILE * streamfile) {
+    uint8_t buf[8];
+
+    if (read_streamfile(buf,offset,8,streamfile)!=8) return -1;
+    return get_64bitLE(buf);
+}
+static inline int64_t read_64bitBE(off_t offset, STREAMFILE * streamfile) {
+    uint8_t buf[8];
+
+    if (read_streamfile(buf,offset,8,streamfile)!=8) return -1;
+    return get_64bitBE(buf);
+}
 
 static inline int8_t read_8bit(off_t offset, STREAMFILE * streamfile) {
     uint8_t buf[1];

src/util.h

@@ -25,6 +25,14 @@ static inline int32_t get_32bitLE(uint8_t * p) {
     return (p[0]) | (p[1]<<8) | (p[2]<<16) | (p[3]<<24);
 }
 
+static inline int64_t get_64bitBE(uint8_t * p) {
+    return (uint64_t)(((uint64_t)p[0]<<56) | ((uint64_t)p[1]<<48) | ((uint64_t)p[2]<<40) | ((uint64_t)p[3]<<32) | ((uint64_t)p[4]<<24) | ((uint64_t)p[5]<<16) | ((uint64_t)p[6]<<8) | ((uint64_t)p[7]));
+}
+
+static inline int64_t get_64bitLE(uint8_t * p) {
+    return (uint64_t)(((uint64_t)p[0]) | ((uint64_t)p[1]<<8) | ((uint64_t)p[2]<<16) | ((uint64_t)p[3]<<24) | ((uint64_t)p[4]<<32) | ((uint64_t)p[5]<<40) | ((uint64_t)p[6]<<48) | ((uint64_t)p[7]<<56));
+}
+
 void put_8bit(uint8_t * buf, int8_t i);
 
 void put_16bitLE(uint8_t * buf, int16_t i);

src/vgmstream.c

@@ -47,6 +47,7 @@ VGMSTREAM * (*init_vgmstream_fcns[])(STREAMFILE *streamFile) = {
     init_vgmstream_ps2_rxws,
     init_vgmstream_ps2_rxw,
     init_vgmstream_ps2_int,
+    init_vgmstream_ngc_dsp_stm,
     init_vgmstream_ps2_exst,
     init_vgmstream_ps2_svag,
     init_vgmstream_ps2_mib,
@@ -367,6 +368,7 @@ VGMSTREAM * (*init_vgmstream_fcns[])(STREAMFILE *streamFile) = {
     init_vgmstream_sk_aud,
     init_vgmstream_stm,
     init_vgmstream_ea_snu,
+    init_vgmstream_awc,
 
     init_vgmstream_txth,  /* should go at the end (lower priority) */
 #ifdef VGM_USE_FFMPEG
@@ -933,6 +935,8 @@ void render_vgmstream(sample * buffer, int32_t sample_count, VGMSTREAM * vgmstre
         case layout_rws_blocked:
         case layout_hwas_blocked:
         case layout_ea_sns_blocked:
+        case layout_blocked_awc:
+        case layout_blocked_vgs:
             render_vgmstream_blocked(buffer,sample_count,vgmstream);
             break;
         case layout_interleave_byte:
@@ -968,9 +972,9 @@ int get_vgmstream_samples_per_frame(VGMSTREAM * vgmstream) {
         case coding_NGC_DSP:
             return 14;
         case coding_PCM16LE:
-        case coding_PCM16LE_int:
         case coding_PCM16LE_XOR_int:
         case coding_PCM16BE:
+        case coding_PCM16_int:
         case coding_PCM8:
         case coding_PCM8_U:
         case coding_PCM8_int:
@@ -1052,6 +1056,8 @@ int get_vgmstream_samples_per_frame(VGMSTREAM * vgmstream) {
         case coding_WWISE_IMA:
         case coding_REF_IMA:
             return (vgmstream->interleave_block_size-4*vgmstream->channels)*2/vgmstream->channels;
+        case coding_AWC_IMA:
+            return (0x800-4)*2;
         case coding_RAD_IMA_mono:
             return 32;
         case coding_NDS_PROCYON:
@@ -1124,9 +1130,9 @@ int get_vgmstream_frame_size(VGMSTREAM * vgmstream) {
         case coding_NGC_DSP:
             return 8;
         case coding_PCM16LE:
-        case coding_PCM16LE_int:
         case coding_PCM16LE_XOR_int:
         case coding_PCM16BE:
+        case coding_PCM16_int:
             return 2;
         case coding_PCM8:
         case coding_PCM8_U:
@@ -1155,6 +1161,8 @@ int get_vgmstream_frame_size(VGMSTREAM * vgmstream) {
         case coding_WWISE_IMA:
         case coding_REF_IMA:
             return vgmstream->interleave_block_size;
+        case coding_AWC_IMA:
+            return 0x800;
         case coding_RAD_IMA_mono:
             return 0x14;
         case coding_NGC_DTK:
@@ -1306,13 +1314,6 @@ void decode_vgmstream(VGMSTREAM * vgmstream, int samples_written, int samples_to
                         samples_to_do);
             }
             break;
-        case coding_PCM16LE_int:
-            for (chan=0;chan<vgmstream->channels;chan++) {
-                decode_pcm16LE_int(&vgmstream->ch[chan],buffer+samples_written*vgmstream->channels+chan,
-                        vgmstream->channels,vgmstream->samples_into_block,
-                        samples_to_do);
-            }
-            break;
         case coding_PCM16LE_XOR_int:
             for (chan=0;chan<vgmstream->channels;chan++) {
                 decode_pcm16LE_XOR_int(&vgmstream->ch[chan],buffer+samples_written*vgmstream->channels+chan,
@@ -1327,6 +1328,14 @@ void decode_vgmstream(VGMSTREAM * vgmstream, int samples_written, int samples_to
                         samples_to_do);
             }
             break;
+        case coding_PCM16_int:
+            for (chan=0;chan<vgmstream->channels;chan++) {
+                decode_pcm16_int(&vgmstream->ch[chan],buffer+samples_written*vgmstream->channels+chan,
+                        vgmstream->channels,vgmstream->samples_into_block,
+                        samples_to_do,
+                        vgmstream->codec_endian);
+            }
+            break;
         case coding_PCM8:
             for (chan=0;chan<vgmstream->channels;chan++) {
                 decode_pcm8(&vgmstream->ch[chan],buffer+samples_written*vgmstream->channels+chan,
@@ -1371,9 +1380,10 @@ void decode_vgmstream(VGMSTREAM * vgmstream, int samples_written, int samples_to
             break;
         case coding_PCMFLOAT:
             for (chan=0;chan<vgmstream->channels;chan++) {
-                decode_pcmfloat(vgmstream, &vgmstream->ch[chan],buffer+samples_written*vgmstream->channels+chan,
+                decode_pcmfloat(&vgmstream->ch[chan],buffer+samples_written*vgmstream->channels+chan,
                         vgmstream->channels,vgmstream->samples_into_block,
-                        samples_to_do);
+                        samples_to_do,
+                        vgmstream->codec_endian);
             }
             break;
 
@@ -1650,6 +1660,13 @@ void decode_vgmstream(VGMSTREAM * vgmstream, int samples_written, int samples_to
                         samples_to_do,chan);
             }
             break;
+        case coding_AWC_IMA:
+            for (chan=0;chan<vgmstream->channels;chan++) {
+                decode_awc_ima(&vgmstream->ch[chan],buffer+samples_written*vgmstream->channels+chan,
+                        vgmstream->channels,vgmstream->samples_into_block,
+                        samples_to_do);
+            }
+            break;
 
         case coding_WS:
             for (chan=0;chan<vgmstream->channels;chan++) {

src/vgmstream.h

@@ -74,9 +74,9 @@ enum { STREAM_NAME_SIZE = 255 }; /* reasonable max */
 typedef enum {
     /* PCM */
     coding_PCM16LE,         /* little endian 16-bit PCM */
-    coding_PCM16LE_int,		/* little endian 16-bit PCM with sample-level interleave */
     coding_PCM16LE_XOR_int, /* little endian 16-bit PCM with sample-level xor */
     coding_PCM16BE,         /* big endian 16-bit PCM */
+    coding_PCM16_int,       /* 16-bit PCM with sample-level interleave */
 
     coding_PCM8,            /* 8-bit PCM */
     coding_PCM8_int,        /* 8-Bit PCM with sample-level interleave */
@@ -132,6 +132,7 @@ typedef enum {
     coding_FSB_IMA,         /* FMOD's FSB multichannel IMA ADPCM */
     coding_WWISE_IMA,       /* Audiokinetic Wwise IMA ADPCM */
     coding_REF_IMA,         /* Reflections IMA ADPCM */
+    coding_AWC_IMA,         /* Rockstar AWC IMA ADPCM */
 
     coding_MSADPCM,         /* Microsoft ADPCM */
     coding_WS,              /* Westwood Studios VBR ADPCM */
@@ -236,6 +237,8 @@ typedef enum {
     layout_rws_blocked,
     layout_hwas_blocked,
     layout_ea_sns_blocked,  /* newest Electronic Arts blocks, found in SNS/SNU/SPS/etc formats */
+    layout_blocked_awc,     /* Rockstar AWC */
+    layout_blocked_vgs,     /* Guitar Hero II */
 
     /* otherwise odd */
     layout_acm,             /* libacm layout */
@@ -621,6 +624,7 @@ typedef enum {
     meta_STM,               /* Angel Studios/Rockstar San Diego Games */
     meta_BINK,              /* RAD Game Tools BINK audio/video */
     meta_EA_SNU,            /* Electronic Arts SNU (Dead Space) */
+    meta_AWC,               /* Rockstar AWC (GTA5, RDR) */
 
 #ifdef VGM_USE_VORBIS
     meta_OGG_VORBIS,        /* Ogg Vorbis */
@@ -874,7 +878,7 @@ typedef enum {
     MPEG_EAL32P,            /* EALayer3 v2 "P" (PCM?), custom frames with v2 header */
     MPEG_EAL32S,            /* EALayer3 v2 "S" (Spike?), custom frames with v2 header */
     MPEG_LYN,               /* N streams of fixed interleave */
-    MPEG_AWC                /* N streams in absolute offsets (consecutive) */
+    MPEG_AWC                /* N streams in block layout (music) or absolute offsets (sfx) */
 } mpeg_custom_t;
 
 /* config for the above modes */
@@ -884,6 +888,7 @@ typedef struct {
     int chunk_size; /* size of a data portion */
     int interleave; /* size of stream interleave */
     int encryption; /* encryption mode */
+    int big_endian;
     /* for AHX */
     int cri_type;
     uint16_t cri_key1;
@@ -899,6 +904,11 @@ typedef struct {
     size_t output_buffer_size;
     size_t samples_filled; /* data in the buffer (in samples) */
     size_t samples_used; /* data extracted from the buffer */
+
+    size_t current_size_count; /* data read (if the parser needs to know) */
+    size_t current_size_target; /* max data, until something happens */
+    size_t decode_to_discard;  /* discard from this stream only (for EALayer3 or AWC) */
+
 } mpeg_custom_stream;
 
 typedef struct {
@@ -924,7 +934,6 @@ typedef struct {
 
     size_t skip_samples; /* base encoder delay */
     size_t samples_to_discard; /* for custom mpeg looping */
-    size_t decode_to_discard;  /* for EALayer3, that discards decoded samples and writes PCM blocks in their place */
 
 } mpeg_codec_data;
 #endif