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https://github.com/vgmstream/vgmstream.git
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EAAC: Properly calculate stream size for bitrate
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8a00c7cd66
commit
519659fd3e
@ -5,29 +5,43 @@
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/* EA SNS/SPS blocks */
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void block_update_ea_sns(off_t block_offset, VGMSTREAM * vgmstream) {
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STREAMFILE* streamFile = vgmstream->ch[0].streamfile;
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uint32_t block_size, block_samples;
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uint32_t block_id, block_size, block_samples;
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size_t file_size = get_streamfile_size(streamFile);
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off_t channel_start;
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size_t channel_interleave;
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int i;
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/* always BE */
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block_size = read_32bitBE(block_offset + 0x00,streamFile);
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block_samples = read_32bitBE(block_offset + 0x04,streamFile);
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/* EOF */
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if (block_size == 0 || block_offset >= file_size) {
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vgmstream->current_block_offset = file_size;
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vgmstream->next_block_offset = file_size + 0x04;
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vgmstream->current_block_samples = vgmstream->num_samples;
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/* EOF reads: signal we have nothing and let the layout fail */
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if (block_offset >= get_streamfile_size(streamFile)) {
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vgmstream->current_block_offset = block_offset;
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vgmstream->next_block_offset = block_offset;
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vgmstream->current_block_samples = -1;
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return;
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}
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/* always BE */
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block_size = read_32bitBE(block_offset + 0x00,streamFile);
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/* At 0x00(1): block flag
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* - in SNS: 0x00=normal block, 0x80=last block (not mandatory)
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* - in SPS: 0x48=header, 0x44=normal block, 0x45=last block (empty) */
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block_id = (block_size & 0x00FFFFFF) >> 24;
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block_size &= 0x00FFFFFF;
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if (block_id == 0x00 || block_id == 0x80 || block_id == 0x44) {
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block_samples = read_32bitBE(block_offset + 0x04, streamFile);
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} else {
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block_samples = 0;
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}
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vgmstream->current_block_offset = block_offset;
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vgmstream->next_block_offset = block_offset + block_size;
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vgmstream->current_block_samples = block_samples;
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/* no need to setup offsets (plus could read over filesize near EOF) */
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if (block_samples == 0)
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return;
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switch (vgmstream->coding_type) {
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case coding_NGC_DSP:
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/* 0x04: unknown (0x00/02), 0x08: some size?, 0x34: null? */
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@ -54,8 +68,4 @@ void block_update_ea_sns(off_t block_offset, VGMSTREAM * vgmstream) {
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vgmstream->ch[i].channel_start_offset = vgmstream->ch[i].offset;
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}
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}
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vgmstream->current_block_offset = block_offset;
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vgmstream->next_block_offset = block_offset + block_size;
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vgmstream->current_block_samples = block_samples;
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}
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@ -342,7 +342,7 @@ VGMSTREAM * init_vgmstream_ea_sbr(STREAMFILE *streamFile) {
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for (i = 0; i < num_metas; i++) {
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entry_offset = metas_offset + 0x06 * i;
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meta_type = read_16bitBE(entry_offset, streamFile);
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meta_type = read_16bitBE(entry_offset + 0x00, streamFile);
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data_offset = read_32bitBE(entry_offset + 0x02, streamFile);
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type_desc = read_32bitBE(types_offset + 0x06 * meta_type, streamFile);
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@ -497,7 +497,7 @@ VGMSTREAM * init_vgmstream_ea_mpf_mus_eaac(STREAMFILE *streamFile) {
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uint32_t num_sounds;
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uint8_t version, sub_version, block_id;
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off_t table_offset, entry_offset, snr_offset, sns_offset;
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size_t /*snr_size,*/ sns_size;
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/* size_t snr_size sns_size; */
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int32_t(*read_32bit)(off_t, STREAMFILE*);
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STREAMFILE *musFile = NULL;
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VGMSTREAM *vgmstream = NULL;
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@ -547,8 +547,10 @@ VGMSTREAM * init_vgmstream_ea_mpf_mus_eaac(STREAMFILE *streamFile) {
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entry_offset = table_offset + (target_stream - 1) * 0x1c;
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snr_offset = read_32bit(entry_offset + 0x08, musFile) * 0x10;
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sns_offset = read_32bit(entry_offset + 0x0c, musFile) * 0x80;
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//snr_size = read_32bit(entry_offset + 0x10, musFile);
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/*
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snr_size = read_32bit(entry_offset + 0x10, musFile);
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sns_size = read_32bit(entry_offset + 0x14, musFile);
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*/
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block_id = read_8bit(sns_offset, musFile);
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if (block_id != EAAC_BLOCKID0_DATA && block_id != EAAC_BLOCKID0_END)
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@ -559,7 +561,6 @@ VGMSTREAM * init_vgmstream_ea_mpf_mus_eaac(STREAMFILE *streamFile) {
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goto fail;
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vgmstream->num_streams = num_sounds;
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vgmstream->stream_size = sns_size;
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close_streamfile(musFile);
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return vgmstream;
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@ -764,7 +765,7 @@ typedef struct {
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static segmented_layout_data* build_segmented_eaaudiocore_looping(STREAMFILE *streamData, eaac_header *eaac);
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static layered_layout_data* build_layered_eaaudiocore_eaxma(STREAMFILE *streamFile, eaac_header *eaac);
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static size_t calculate_eaac_size(VGMSTREAM *vgmstream, STREAMFILE *streamFile, eaac_header *eaac, off_t start_offset);
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/* EA newest header from RwAudioCore (RenderWare?) / EAAudioCore library (still generated by sx.exe).
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* Audio "assets" come in separate RAM headers (.SNR/SPH) and raw blocked streams (.SNS/SPS),
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@ -831,7 +832,7 @@ static VGMSTREAM * init_vgmstream_eaaudiocore_header(STREAMFILE * streamHead, ST
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eaac.loop_offset = eaac.stream_offset + eaac.loop_offset;
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}
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else {
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/* RAM assets only one block in case in case of full loops */
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/* RAM assets have only one block in case of full loops */
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eaac.loop_offset = eaac.stream_offset; /* implicit */
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}
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@ -1023,9 +1024,13 @@ static VGMSTREAM * init_vgmstream_eaaudiocore_header(STREAMFILE * streamHead, ST
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goto fail;
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}
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if (!vgmstream_open_stream(vgmstream, temp_streamFile ? temp_streamFile : streamData, start_offset))
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goto fail;
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if (eaac.loop_start == 0) {
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vgmstream->stream_size = calculate_eaac_size(vgmstream, temp_streamFile ? temp_streamFile : streamData, &eaac, start_offset);
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}
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close_streamfile(temp_streamFile);
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return vgmstream;
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@ -1043,6 +1048,44 @@ static size_t get_snr_size(STREAMFILE *streamFile, off_t offset) {
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}
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}
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static size_t calculate_eaac_size(VGMSTREAM *vgmstream, STREAMFILE *streamFile, eaac_header *eaac, off_t start_offset) {
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uint32_t total_samples;
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size_t stream_size, file_size;
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switch (eaac->codec) {
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case EAAC_CODEC_EAXMA:
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case EAAC_CODEC_EALAYER3_V1:
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case EAAC_CODEC_EALAYER3_V2_PCM:
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case EAAC_CODEC_EALAYER3_V2_SPIKE:
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case EAAC_CODEC_EATRAX:
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case EAAC_CODEC_EAMP3:
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case EAAC_CODEC_EAOPUS:
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stream_size = get_streamfile_size(streamFile);
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break;
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default:
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stream_size = 0;
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total_samples = 0;
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file_size = get_streamfile_size(streamFile);
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vgmstream->next_block_offset = start_offset;
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while (vgmstream->next_block_offset < file_size && total_samples != vgmstream->num_samples) {
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block_update_ea_sns(vgmstream->next_block_offset, vgmstream);
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if (vgmstream->current_block_samples == 0)
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continue;
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/* stream size is almost never provided in bank files so we have to calc it manually */
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stream_size += vgmstream->next_block_offset - vgmstream->ch[0].offset;
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total_samples += vgmstream->current_block_samples;
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}
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/* reset once we're done */
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block_update(start_offset, vgmstream);
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break;
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}
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return stream_size;
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}
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/* Actual looping uses 2 block sections, separated by a block end flag *and* padded.
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*
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@ -1108,6 +1151,8 @@ static segmented_layout_data* build_segmented_eaaudiocore_looping(STREAMFILE *st
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if (!vgmstream_open_stream(data->segments[i],temp_streamFile[i],0x00))
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goto fail;
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data->segments[i]->stream_size = calculate_eaac_size(data->segments[i], temp_streamFile[i], eaac, 0x00);
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}
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if (!setup_layout_segmented(data))
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