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Fix some .aac encoder delay/sample issues

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This commit is contained in:
bnnm 2021-06-01 00:28:04 +02:00
parent c5fc5799fb
commit 89ccb06747
7 changed files with 41 additions and 9 deletions
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1
src/coding/coding.h
View File

@ -570,6 +570,7 @@ 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_decoder_custom_opus.c (helper-things) */

1
src/coding/coding_utils.c
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@ -1006,6 +1006,7 @@ size_t aac_get_samples(STREAMFILE* sf, off_t start_offset, size_t bytes) {
if (frame_size <= 0x08)
break;
//;VGM_LOG("AAC: %lx, %x\n", offset, frame_size);
frames++;
offset += frame_size;
}

20
src/coding/ffmpeg_decoder_utils.c
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@ -187,4 +187,24 @@ fail:
return NULL;
}
ffmpeg_codec_data* init_ffmpeg_aac(STREAMFILE* sf, off_t offset, size_t size) {
ffmpeg_codec_data* data = NULL;
data = init_ffmpeg_offset(sf, offset, size);
if (!data) goto fail;
/* seeks to 0 eats first frame for whatever reason */
ffmpeg_set_force_seek(data);
/* 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);
return data;
fail:
free_ffmpeg(data);
return NULL;
}
#endif

5
src/meta/ffmpeg.c
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@ -66,6 +66,11 @@ VGMSTREAM* init_vgmstream_ffmpeg(STREAMFILE* sf) {
/* hack for AAC files (will return 0 samples if not an actual file) */
if (!num_samples && check_extensions(sf, "aac,laac")) {
num_samples = aac_get_samples(sf, 0x00, get_streamfile_size(sf));
if (num_samples > 0) {
/* FFmpeg seeks to 0 eats first frame for whatever reason */
ffmpeg_set_force_seek(data);
}
}
#ifdef VGM_USE_MPEG

5
src/meta/naac.c
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@ -42,15 +42,16 @@ VGMSTREAM* init_vgmstream_naac(STREAMFILE* sf) {
#ifdef VGM_USE_FFMPEG
{
vgmstream->codec_data = init_ffmpeg_offset(sf, start_offset, data_size);
vgmstream->codec_data = init_ffmpeg_aac(sf, start_offset, data_size);
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 -= 1024; /* may end with 1024 of silence? */
vgmstream->num_samples -= 1024;
vgmstream->loop_end_sample -= 1024;
/* for some reason last frame is ignored/bugged in various decoders (gives EOF errors) */
}
#else
goto fail;

8
src/meta/strm_abylight.c
View File

@ -42,14 +42,14 @@ VGMSTREAM* init_vgmstream_strm_abylight(STREAMFILE* sf) {
#ifdef VGM_USE_FFMPEG
{
vgmstream->codec_data = init_ffmpeg_offset(sf, start_offset, data_size);
vgmstream->codec_data = init_ffmpeg_aac(sf, start_offset, data_size);
if (!vgmstream->codec_data) goto fail;
vgmstream->coding_type = coding_FFmpeg;
vgmstream->layout_type = layout_none;
/* apparently none, or maybe ~600 */
//ffmpeg_set_skip_samples(ffmpeg_data, 1024);
//vgmstream->num_samples -= 1024;
/* assumed, maybe a bit more */
ffmpeg_set_skip_samples(vgmstream->codec_data, 1024);
vgmstream->num_samples -= 1024;
}
#else
goto fail;

10
src/meta/txth.c
View File

@ -468,14 +468,18 @@ VGMSTREAM* init_vgmstream_txth(STREAMFILE* sf) {
case coding_FFmpeg: {
ffmpeg_codec_data *ffmpeg_data = NULL;
if (txth.codec == FFMPEG || txth.codec == AC3 || txth.codec == AAC) {
if (txth.codec == FFMPEG || txth.codec == AC3) {
/* default FFmpeg */
ffmpeg_data = init_ffmpeg_offset(txth.sf_body, txth.start_offset,txth.data_size);
if ( !ffmpeg_data ) goto fail;
ffmpeg_data = init_ffmpeg_offset(txth.sf_body, txth.start_offset, txth.data_size);
if (!ffmpeg_data) goto fail;
if (vgmstream->num_samples == 0)
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);
if (!ffmpeg_data) goto fail;
}
else {
/* fake header FFmpeg */
uint8_t buf[0x100];