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Add/use atrac3_bytes_to_samples

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This commit is contained in:
bnnm 2017-04-07 21:18:07 +02:00
parent 3619b14f8e
commit b7ffd11fca
5 changed files with 25 additions and 10 deletions
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3
src/coding/coding.h
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@ -207,6 +207,9 @@ typedef struct {
void xma_get_samples(xma_sample_data * msd, STREAMFILE *streamFile);
void wmapro_get_samples(xma_sample_data * msd, STREAMFILE *streamFile, int block_align, int sample_rate, uint32_t decode_flags);
size_t atrac3_bytes_to_samples(size_t bytes, int full_block_align);
size_t atrac3plus_bytes_to_samples(size_t bytes, int full_block_align);
#endif
#endif /*_CODING_H*/

12
src/coding/ffmpeg_decoder_utils.c
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@ -631,4 +631,16 @@ void wmapro_get_samples(xma_sample_data * msd, STREAMFILE *streamFile, int block
}
size_t atrac3_bytes_to_samples(size_t bytes, int full_block_align) {
/* ATRAC3 expects full block align since as is can mix joint stereo with mono blocks;
* so (full_block_align / channels) DOESN'T give the size of a single channel (uncommon in ATRAC3 though) */
return (bytes / full_block_align) * 1024;
}
size_t atrac3plus_bytes_to_samples(size_t bytes, int full_block_align) {
/* ATRAC3plus expects full block align since as is can mix joint stereo with mono blocks;
* so (full_block_align / channels) DOESN'T give the size of a single channel (common in ATRAC3plus) */
return (bytes / full_block_align) * 2048;
}
#endif

6
src/meta/gsp_gsb.c
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@ -82,9 +82,9 @@ VGMSTREAM * init_vgmstream_gsp_gsb(STREAMFILE *streamFile) {
uint8_t buf[100];
int32_t bytes, block_size, encoder_delay, joint_stereo, max_samples;
block_size = 0x98 * vgmstream->channels;
joint_stereo = 0;
max_samples = (datasize / block_size) * 1024;
block_size = 0x98 * vgmstream->channels;
joint_stereo = 0;
max_samples = atrac3_bytes_to_samples(datasize, block_size);;
encoder_delay = max_samples - vgmstream->num_samples; /* todo guessed */
vgmstream->num_samples += encoder_delay;

12
src/meta/ps3_msf.c
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@ -104,12 +104,12 @@ VGMSTREAM * init_vgmstream_ps3_msf(STREAMFILE *streamFile) {
case 0x6: { /* ATRAC3 high (132 kbps, frame size 192) */
ffmpeg_codec_data *ffmpeg_data = NULL;
uint8_t buf[100];
int32_t bytes, samples_size = 1024, block_size, encoder_delay, joint_stereo, max_samples;
int32_t bytes, block_size, encoder_delay, joint_stereo, max_samples;
block_size = (codec_id==4 ? 0x60 : (codec_id==5 ? 0x98 : 0xC0)) * vgmstream->channels;
block_size = (codec_id==4 ? 0x60 : (codec_id==5 ? 0x98 : 0xC0)) * vgmstream->channels;
encoder_delay = 0x0; //todo MSF encoder delay (around 440-450*2)
max_samples = (data_size / block_size) * samples_size;
joint_stereo = codec_id==4; /* interleaved joint stereo (ch must be even) */
max_samples = atrac3_bytes_to_samples(data_size, block_size);
joint_stereo = codec_id==4; /* interleaved joint stereo (ch must be even) */
if (vgmstream->sample_rate==0xFFFFFFFF) /* some MSFv1 (Digi World SP) */
vgmstream->sample_rate = 44100;//voice tracks seems to use 44khz, not sure about other tracks
@ -126,8 +126,8 @@ VGMSTREAM * init_vgmstream_ps3_msf(STREAMFILE *streamFile) {
vgmstream->num_samples = max_samples;
if (loop_flag) {
vgmstream->loop_start_sample = (loop_start / block_size) * samples_size;
vgmstream->loop_end_sample = (loop_end / block_size) * samples_size;
vgmstream->loop_start_sample = atrac3_bytes_to_samples(loop_start, block_size);
vgmstream->loop_end_sample = atrac3_bytes_to_samples(loop_end, block_size);
}
break;

2
src/meta/xwb.c
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@ -226,7 +226,7 @@ VGMSTREAM * init_vgmstream_xwb(STREAMFILE *streamFile) {
/* num samples uses a modified entry_info format (maybe skip samples + samples? sfx use the standard format)
* ignore for now and just calc max samples */ //todo
xwb.num_samples = xwb.stream_size / (xwb.block_align * xwb.channels) * 1024;
xwb.num_samples = atrac3_bytes_to_samples(xwb.stream_size, xwb.block_align * xwb.channels);
}