diff --git a/doc/TXTH.md b/doc/TXTH.md
index 34af042d..92eba6c1 100644
--- a/doc/TXTH.md
+++ b/doc/TXTH.md
@@ -139,10 +139,14 @@ as explained below, but often will use default values. Accepted codec strings:
 # - PCM4_U         PCM 4-bit unsigned
 #   * Variation with modified encoding
 # - OKI16          OKI ADPCM with 16-bit output (not VOX/Dialogic 12-bit)
-#   * For few PS2 games (Sweet Legacy, Hooligan)
+#   * For rare PS2 games (Sweet Legacy, Hooligan)
 # - AAC            Advanced Audio Coding (raw without .mp4)
 #   * For some 3DS games and many iOS games
 #   * Should set skip_samples (around 1024 but varies)
+# - TGC            Tiger Game.com 4-bit ADPCM
+#   * For Tiger Game.com
+# - ASF            Argonaut ASF ADPCM
+#   * For rare Argonaut games [Croc (SAT)]
 codec = (codec string)
 ```
 
@@ -679,7 +683,7 @@ sample_rate = @0x04                 #reads at 0x1204
 
 ## Examples
 
-**Colin McRae DiRT (PC) .wip.txth**
+#### Colin McRae DiRT (PC) .wip.txth
 ```
 id_value = 0x00000000   #check that value at 0x00 is really 0x00000000
 id_offset = @0x00:BE
@@ -693,7 +697,7 @@ loop_start_sample = 0
 loop_end_sample = data_size
 ```
 
-**Kim Possible: What's the Switch (PS2) .str.txth**
+#### Kim Possible: What's the Switch (PS2) .str.txth
 ```
 codec = PSX
 interleave = 0x2000
@@ -703,7 +707,7 @@ num_samples = data_size
 interleave_last = auto
 ```
 
-**Manhunt (Xbox) .rib.txth**
+#### Manhunt (Xbox) .rib.txth
 ```
 codec = XBOX
 codec_mode = 1 #interleaved XBOX
@@ -715,7 +719,7 @@ start_offset = 0x00
 num_samples = data_size
 ```
 
-**Pitfall The Lost Expedition (PC) .txth**
+#### Pitfall The Lost Expedition (PC) .txth
 ```
 codec = DVI_IMA
 interleave = 0x80
@@ -725,7 +729,7 @@ sample_rate = 44100
 num_samples = data_size
 ```
 
-**Spy Hunter (GC) .pcm.txth**
+#### Spy Hunter (GC) .pcm.txth
 ```
 codec = PCM8
 sample_rate = 32000
@@ -734,7 +738,7 @@ start_offset = 0
 num_samples = data_size
 ```
 
-**Ultimate Board Game Collection (Wii) .dsp.txth**
+#### Ultimate Board Game Collection (Wii) .dsp.txth
 ```
 codec = NGC_DSP
 interleave = 0x10000
@@ -753,7 +757,8 @@ coef_offset = 0x1c
 coef_spacing = 0x10000
 coef_endianness = BE
 ```
-**Aladdin in Nasira's Revenge (PS1) .cvs.txth**
+
+#### Aladdin in Nasira's Revenge (PS1) .cvs.txth
 ```
 codec = PSX
 interleave = 0x10
@@ -763,7 +768,7 @@ padding_size = auto-empty
 num_samples = data_size
 ```
 
-**Shikigami no Shiro - Nanayozuki Gensoukyoku (PS2) bgm.txth**
+#### Shikigami no Shiro - Nanayozuki Gensoukyoku (PS2) bgm.txth
 ```
 codec = PSX
 interleave = 0x1000
@@ -790,7 +795,7 @@ loop_end_sample   = @0x10 * channels
 data_size         = @0x08 * channels  #for bitrate
 ```
 
-**Dragon Poker (Mobile) .snd.txth**
+#### Dragon Poker (Mobile) .snd.txth
 ```
 # parse MP3 inside the .snd
 subfile_extension = mp3
@@ -805,7 +810,7 @@ loop_start_sample = 0
 loop_end_sample = data_size
 ```
 
-**Simple 2000 Series Vol. 120 - The Saigo no Nihonhei (PS2) .xag.txth**
+#### Simple 2000 Series Vol. 120 - The Saigo no Nihonhei (PS2) .xag.txth
 ```
 header_file = TSNDDRVC.IRX
 
@@ -853,15 +858,14 @@ subfile_extension = seb
 subfile_size      = ((@0x04 - @0x00) & 0xFFFFF) * 0x800
 ```
 
-
-**Zack & Wiki (Wii) .ssd.txth**
+#### Zack & Wiki (Wii) .ssd.txth
 ```
 header_file = bgm_S01.srt
 name_table = .names.txt
 
 base_offset = @0x0c:BE
 base_offset = base_offset + @0x08:BE + name_value
-base_offset = base_offset + @0x00:BE
+base_offset = base_offset + @0x00:BE - name_value
 
 codec = NGC_DSP
 channels = 2
@@ -891,7 +895,7 @@ st_s01_02c.ssd: 7*0x04
 ```
 
 
-**Zack & Wiki (Wii) st_s01_00a.txth**
+#### Zack & Wiki (Wii) st_s01_00a.txth
 ```
 #alt from above with untouched folders
 header_file = Sound/BGM/bgm_S01.srt
@@ -900,7 +904,7 @@ name_table = .names.txt
 
 base_offset = @0x0c:BE
 base_offset = base_offset + @0x08:BE + name_value
-base_offset = base_offset + @0x00:BE
+base_offset = base_offset + @0x00:BE - name_value
 
 codec = NGC_DSP
 channels = 2
@@ -929,3 +933,11 @@ coef_endianness = BE
 *snd/stream/st_s01_02c.ssd: 7*0x04
 # uses wildcards for full paths from plugins
 ```
+
+####  Croc (SAT) .asf.txth
+```
+codec = ASF
+sample_rate = 22050
+channels = 2
+num_samples = data_size
+```
diff --git a/src/coding/asf_decoder.c b/src/coding/asf_decoder.c
index 00ee61d7..db81b75b 100644
--- a/src/coding/asf_decoder.c
+++ b/src/coding/asf_decoder.c
@@ -54,3 +54,8 @@ void decode_asf(VGMSTREAMCHANNEL *stream, sample_t *outbuf, int channelspacing,
     stream->adpcm_history1_32 = hist1;
     stream->adpcm_history2_32 = hist2;
 }
+
+int32_t asf_bytes_to_samples(size_t bytes, int channels) {
+    if (channels <= 0) return 0;
+    return bytes / channels / 0x11 * 32;
+}
diff --git a/src/meta/txth.c b/src/meta/txth.c
index f97a8167..f98e6296 100644
--- a/src/meta/txth.c
+++ b/src/meta/txth.c
@@ -36,7 +36,8 @@ typedef enum {
     PCM4_U = 26,        /* 4-bit unsigned PCM (3rd and 4th gen games) */
     OKI16 = 27,         /* OKI ADPCM with 16-bit output (unlike OKI/VOX/Dialogic ADPCM's 12-bit) */
     AAC = 28,           /* Advanced Audio Coding (raw without .mp4) */
-    TGC = 29            /* Tiger Game.com 4-bit ADPCM */
+    TGC = 29,           /* Tiger Game.com 4-bit ADPCM */
+    ASF = 30,           /* Argonaut ASF 4-bit ADPCM */
 } txth_type;
 
 typedef struct {
@@ -223,6 +224,7 @@ VGMSTREAM * init_vgmstream_txth(STREAMFILE *streamFile) {
         case PCM4_U:     coding = coding_PCM4_U; break;
         case OKI16:      coding = coding_OKI16; break;
         case TGC:        coding = coding_TGC; break;
+        case ASF:        coding = coding_ASF; break;
         default:
             goto fail;
     }
@@ -334,6 +336,11 @@ VGMSTREAM * init_vgmstream_txth(STREAMFILE *streamFile) {
             vgmstream->layout_type = layout_none;
             break;
 
+        case coding_ASF:
+            vgmstream->layout_type = layout_interleave;
+            vgmstream->interleave_block_size = 0x11;
+            break;
+
         case coding_MS_IMA:
             if (!txth.interleave) goto fail; /* creates garbage */
 
@@ -450,7 +457,7 @@ VGMSTREAM * init_vgmstream_txth(STREAMFILE *streamFile) {
             }
             else {
                 /* fake header FFmpeg */
-                uint8_t buf[200];
+                uint8_t buf[0x100];
                 int32_t bytes;
 
                 if (txth.codec == ATRAC3) {
@@ -465,14 +472,14 @@ VGMSTREAM * init_vgmstream_txth(STREAMFILE *streamFile) {
                 else if (txth.codec == ATRAC3PLUS) {
                     int block_size = txth.interleave;
 
-                    bytes = ffmpeg_make_riff_atrac3plus(buf, 200, vgmstream->num_samples, txth.data_size, vgmstream->channels, vgmstream->sample_rate, block_size, txth.skip_samples);
+                    bytes = ffmpeg_make_riff_atrac3plus(buf, sizeof(buf), vgmstream->num_samples, txth.data_size, vgmstream->channels, vgmstream->sample_rate, block_size, txth.skip_samples);
                     ffmpeg_data = init_ffmpeg_header_offset(txth.streamBody, buf,bytes, txth.start_offset,txth.data_size);
                     if ( !ffmpeg_data ) goto fail;
                 }
                 else if (txth.codec == XMA1) {
                     int xma_stream_mode = txth.codec_mode == 1 ? 1 : 0;
 
-                    bytes = ffmpeg_make_riff_xma1(buf, 100, vgmstream->num_samples, txth.data_size, vgmstream->channels, vgmstream->sample_rate, xma_stream_mode);
+                    bytes = ffmpeg_make_riff_xma1(buf, sizeof(buf), vgmstream->num_samples, txth.data_size, vgmstream->channels, vgmstream->sample_rate, xma_stream_mode);
                     ffmpeg_data = init_ffmpeg_header_offset(txth.streamBody, buf,bytes, txth.start_offset,txth.data_size);
                     if ( !ffmpeg_data ) goto fail;
                 }
@@ -482,7 +489,7 @@ VGMSTREAM * init_vgmstream_txth(STREAMFILE *streamFile) {
                     block_size = txth.interleave ? txth.interleave : 2048;
                     block_count = txth.data_size / block_size;
 
-                    bytes = ffmpeg_make_riff_xma2(buf, 200, vgmstream->num_samples, txth.data_size, vgmstream->channels, vgmstream->sample_rate, block_count, block_size);
+                    bytes = ffmpeg_make_riff_xma2(buf, sizeof(buf), vgmstream->num_samples, txth.data_size, vgmstream->channels, vgmstream->sample_rate, block_count, block_size);
                     ffmpeg_data = init_ffmpeg_header_offset(txth.streamBody, buf,bytes, txth.start_offset,txth.data_size);
                     if ( !ffmpeg_data ) goto fail;
                 }
@@ -848,6 +855,7 @@ static int parse_keyval(STREAMFILE * streamFile_, txth_header * txth, const char
         else if (is_string(val,"AAC"))          txth->codec = AAC;
         else if (is_string(val,"TGC"))          txth->codec = TGC;
         else if (is_string(val,"GCOM_ADPCM"))   txth->codec = TGC;
+        else if (is_string(val,"ASF"))          txth->codec = ASF;
         else goto fail;
 
         /* set common interleaves to simplify usage
@@ -1730,6 +1738,8 @@ static int get_bytes_to_samples(txth_header * txth, uint32_t bytes) {
 #endif
         case AC3:
             return ac3_bytes_to_samples(bytes, txth->interleave, txth->channels);
+        case ASF:
+            return asf_bytes_to_samples(bytes, txth->channels);
 
         /* XMA bytes-to-samples is done at the end as the value meanings are a bit different */
         case XMA1: