ffmpeg: AAC cleanup and doc

src/coding/coding.h

@@ -572,7 +572,7 @@ STREAMFILE* ffmpeg_get_streamfile(ffmpeg_codec_data* data);
 /* ffmpeg_decoder_utils.c (helper-things) */
 ffmpeg_codec_data* init_ffmpeg_atrac3_raw(STREAMFILE* sf, off_t offset, size_t data_size, int sample_count, int channels, int sample_rate, int block_align, int encoder_delay);
 ffmpeg_codec_data* init_ffmpeg_atrac3_riff(STREAMFILE* sf, off_t offset, int* out_samples);
-ffmpeg_codec_data* init_ffmpeg_aac(STREAMFILE* sf, off_t offset, size_t size);
+ffmpeg_codec_data* init_ffmpeg_aac(STREAMFILE* sf, off_t offset, size_t size, int skip_samples);
 
 
 /* ffmpeg_decoder_custom_opus.c (helper-things) */

src/coding/ffmpeg_decoder_custom_opus.c

@@ -764,10 +764,8 @@ static ffmpeg_codec_data* init_ffmpeg_custom_opus_config(STREAMFILE* sf, off_t s
 
     /* FFmpeg + libopus: skips samples, notifies skip in codecCtx->delay (not in stream->skip_samples)
      * FFmpeg + opus: *doesn't* skip, also notifies skip in codecCtx->delay, hurray (possibly fixed in recent versions)
-     * FFmpeg + opus is audibly buggy with some low bitrate SSB Ultimate files too */
+     * FFmpeg + opus is audibly buggy with some low bitrate SSB Ultimate files */
-    //if (ffmpeg_data->skipSamples <= 0) {
+    //ffmpeg_set_skip_samples(ffmpeg_data, skip);
-    //    ffmpeg_set_skip_samples(ffmpeg_data, skip);
-    //}
 
     close_streamfile(temp_sf);
     return ffmpeg_data;

src/coding/ffmpeg_decoder_utils.c

@@ -187,7 +187,7 @@ fail:
     return NULL;
 }
 
-ffmpeg_codec_data* init_ffmpeg_aac(STREAMFILE* sf, off_t offset, size_t size) {
+ffmpeg_codec_data* init_ffmpeg_aac(STREAMFILE* sf, off_t offset, size_t size, int skip_samples) {
     ffmpeg_codec_data* data = NULL;
 
     data = init_ffmpeg_offset(sf, offset, size);
@@ -199,7 +199,7 @@ ffmpeg_codec_data* init_ffmpeg_aac(STREAMFILE* sf, off_t offset, size_t size) {
     /* raw AAC doesn't set this, while some decoders like FAAD remove 1024,
      * but should be handled in container as each encoder uses its own value
      * (Apple: 2112, FAAD: probably 1024, etc) */
-    //ffmpeg_set_skip_samples(data, 1024);
+    ffmpeg_set_skip_samples(data, skip_samples);
 
     return data;
 fail:

src/meta/ffmpeg.c

@@ -82,6 +82,9 @@ VGMSTREAM* init_vgmstream_ffmpeg(STREAMFILE* sf) {
      *  .mus: Marc Ecko's Getting Up (PC) */
     if (!num_samples && check_extensions(sf, "mp3,lmp3,mus")) {
         num_samples = mpeg_get_samples(sf, 0x00, get_streamfile_size(sf));
+
+        /* this seems correct thankfully */
+        //ffmpeg_set_skip_samples(data, encoder_delay);
     }
 #endif
 

src/meta/naac.c

@@ -6,7 +6,7 @@
 VGMSTREAM* init_vgmstream_naac(STREAMFILE* sf) {
     VGMSTREAM* vgmstream = NULL;
     off_t start_offset;
-    int loop_flag, channels;
+    int loop_flag, channels, skip_samples;
     size_t data_size;
 
 
@@ -22,6 +22,7 @@ VGMSTREAM* init_vgmstream_naac(STREAMFILE* sf) {
     start_offset = 0x1000;
     loop_flag = (read_s32le(0x18,sf) != 0);
     channels = read_s32le(0x08,sf);
+    skip_samples = 1024; /* raw AAC doesn't set this */
 
     /* build the VGMSTREAM */
     vgmstream = allocate_vgmstream(channels, loop_flag);
@@ -42,15 +43,14 @@ VGMSTREAM* init_vgmstream_naac(STREAMFILE* sf) {
 
 #ifdef VGM_USE_FFMPEG
     {
-        vgmstream->codec_data = init_ffmpeg_aac(sf, start_offset, data_size);
+        vgmstream->codec_data = init_ffmpeg_aac(sf, start_offset, data_size, skip_samples);
         if (!vgmstream->codec_data) goto fail;
         vgmstream->coding_type = coding_FFmpeg;
         vgmstream->layout_type = layout_none;
 
         /* observed default, some files start without silence though seems correct when loop_start=0 */
-        ffmpeg_set_skip_samples(vgmstream->codec_data, 1024); /* raw AAC doesn't set this */
+        vgmstream->num_samples -= skip_samples;
-        vgmstream->num_samples -= 1024;
+        vgmstream->loop_end_sample -= skip_samples;
-        vgmstream->loop_end_sample -= 1024;
         /* for some reason last frame is ignored/bugged in various decoders (gives EOF errors) */
     }
 #else

src/meta/strm_abylight.c

@@ -4,9 +4,9 @@
 
 /* .STRM - from Abylight 3DS games [Cursed Castilla (3DS)] */
 VGMSTREAM* init_vgmstream_strm_abylight(STREAMFILE* sf) {
-    VGMSTREAM * vgmstream = NULL;
+    VGMSTREAM* vgmstream = NULL;
     off_t start_offset;
-    int loop_flag, channel_count, sample_rate;
+    int loop_flag, channel_count, sample_rate, skip_samples;
     size_t data_size;
 
 
@@ -22,6 +22,7 @@ VGMSTREAM* init_vgmstream_strm_abylight(STREAMFILE* sf) {
     loop_flag = 0;
     channel_count = 2; /* there are various possible fields but all files are stereo */
     sample_rate = read_32bitLE(0x08,sf);
+    skip_samples = 1024; /* assumed, maybe a bit more */
 
     start_offset = 0x1e;
     data_size = read_32bitLE(0x10,sf);
@@ -42,14 +43,12 @@ VGMSTREAM* init_vgmstream_strm_abylight(STREAMFILE* sf) {
 
 #ifdef VGM_USE_FFMPEG
     {
-        vgmstream->codec_data = init_ffmpeg_aac(sf, start_offset, data_size);
+        vgmstream->codec_data = init_ffmpeg_aac(sf, start_offset, data_size, skip_samples);
         if (!vgmstream->codec_data) goto fail;
         vgmstream->coding_type = coding_FFmpeg;
         vgmstream->layout_type = layout_none;
 
-        /* assumed, maybe a bit more */
+        vgmstream->num_samples -= skip_samples;
-        ffmpeg_set_skip_samples(vgmstream->codec_data, 1024);
-        vgmstream->num_samples -= 1024;
     }
 #else
     goto fail;

src/meta/txth.c

@@ -486,7 +486,7 @@ VGMSTREAM* init_vgmstream_txth(STREAMFILE* sf) {
                     vgmstream->num_samples = ffmpeg_data->totalSamples; /* sometimes works */
             }
             else if (txth.codec == AAC) {
-                ffmpeg_data = init_ffmpeg_aac(txth.sf_body, txth.start_offset, txth.data_size);
+                ffmpeg_data = init_ffmpeg_aac(txth.sf_body, txth.start_offset, txth.data_size, 0);
                 if (!ffmpeg_data) goto fail;
             }
             else {