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https://github.com/vgmstream/vgmstream.git
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take the first channel found
git-svn-id: https://vgmstream.svn.sourceforge.net/svnroot/vgmstream@860 51a99a44-fe44-0410-b1ba-c3e57ba2b86b
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@ -20,6 +20,8 @@ VGMSTREAM * init_vgmstream_cdxa(STREAMFILE *streamFile) {
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char filename[260];
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char filename[260];
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int channel_count;
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int channel_count;
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int headerless=0;
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int* xa_channel=0;
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uint8_t bCoding;
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uint8_t bCoding;
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off_t start_offset;
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off_t start_offset;
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@ -33,41 +35,54 @@ VGMSTREAM * init_vgmstream_cdxa(STREAMFILE *streamFile) {
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if (!((read_32bitBE(0x00,streamFile) == 0x52494646) &&
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if (!((read_32bitBE(0x00,streamFile) == 0x52494646) &&
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(read_32bitBE(0x08,streamFile) == 0x43445841) &&
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(read_32bitBE(0x08,streamFile) == 0x43445841) &&
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(read_32bitBE(0x0C,streamFile) == 0x666D7420)))
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(read_32bitBE(0x0C,streamFile) == 0x666D7420)))
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goto fail;
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headerless=1;
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/* First init to have the correct info of the channel */
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/* First init to have the correct info of the channel */
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start_offset=init_xa_channel(0,streamFile);
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if (!headerless) {
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start_offset=init_xa_channel(&xa_channel,streamFile);
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/* No sound ? */
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/* No sound ? */
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if(start_offset==0)
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if(start_offset==0)
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goto fail;
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goto fail;
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bCoding = read_8bit(start_offset-5,streamFile);
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bCoding = read_8bit(start_offset-5,streamFile);
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switch (AUDIO_CODING_GET_STEREO(bCoding)) {
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switch (AUDIO_CODING_GET_STEREO(bCoding)) {
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case 0: channel_count = 1; break;
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case 0: channel_count = 1; break;
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case 1: channel_count = 2; break;
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case 1: channel_count = 2; break;
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default: channel_count = 0; break;
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default: channel_count = 0; break;
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}
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/* build the VGMSTREAM */
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vgmstream = allocate_vgmstream(channel_count,0);
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if (!vgmstream) goto fail;
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/* fill in the vital statistics */
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vgmstream->channels = channel_count;
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vgmstream->xa_channel = xa_channel;
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switch (AUDIO_CODING_GET_FREQ(bCoding)) {
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case 0: vgmstream->sample_rate = 37800; break;
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case 1: vgmstream->sample_rate = 18900; break;
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default: vgmstream->sample_rate = 0; break;
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}
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/* Check for Compression Scheme */
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vgmstream->num_samples = (int32_t)((((get_streamfile_size(streamFile) - 0x3C)/2352)*0x1F80)/(2*channel_count));
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} else
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{
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channel_count=2;
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vgmstream = allocate_vgmstream(2,0);
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if (!vgmstream) goto fail;
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vgmstream->xa_headerless=1;
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vgmstream->sample_rate=44100;
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vgmstream->channels=2;
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vgmstream->num_samples = (int32_t)(((get_streamfile_size(streamFile)/ 0x80)*0xE0)/2);
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start_offset=0;
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}
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}
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/* build the VGMSTREAM */
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vgmstream = allocate_vgmstream(channel_count,0);
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if (!vgmstream) goto fail;
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/* fill in the vital statistics */
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vgmstream->channels = channel_count;
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vgmstream->xa_channel = 0;
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switch (AUDIO_CODING_GET_FREQ(bCoding)) {
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case 0: vgmstream->sample_rate = 37800; break;
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case 1: vgmstream->sample_rate = 18900; break;
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default: vgmstream->sample_rate = 0; break;
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}
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/* Check for Compression Scheme */
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vgmstream->coding_type = coding_XA;
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vgmstream->coding_type = coding_XA;
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vgmstream->num_samples = (int32_t)((((get_streamfile_size(streamFile) - 0x3C)/2352)*0x1F80)/(2*channel_count));
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vgmstream->layout_type = layout_xa_blocked;
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vgmstream->layout_type = layout_xa_blocked;
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vgmstream->meta_type = meta_PSX_XA;
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vgmstream->meta_type = meta_PSX_XA;
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@ -93,7 +108,7 @@ fail:
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return NULL;
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return NULL;
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}
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}
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off_t init_xa_channel(int channel,STREAMFILE* streamFile) {
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off_t init_xa_channel(int* channel,STREAMFILE* streamFile) {
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off_t block_offset=0x44;
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off_t block_offset=0x44;
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size_t filelength=get_streamfile_size(streamFile);
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size_t filelength=get_streamfile_size(streamFile);
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@ -109,9 +124,10 @@ begin:
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currentChannel=read_8bit(block_offset-7,streamFile);
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currentChannel=read_8bit(block_offset-7,streamFile);
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subAudio=read_8bit(block_offset-6,streamFile);
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subAudio=read_8bit(block_offset-6,streamFile);
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if (!((currentChannel==channel) && (subAudio==0x64))) {
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*channel=currentChannel;
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block_offset+=2352;
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//if (!((currentChannel==channel) && (subAudio==0x64))) {
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goto begin;
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// block_offset+=2352;
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}
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// goto begin;
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//}
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return block_offset;
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return block_offset;
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}
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}
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