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Merge pull request #869 from bnnm/mpc-aac
- Fix .mpc/mp+ total samples and encoder delay - Fix .vxn encoder delay / gapless [Asphalt 9 (Android)] - Fix EA SWVR .stream sample rate in some cases [Nascar Rumble (PS1)] - Fix .wem with fully truncated PCM [Metal Gear Solid V (PC)] - Fix some .aac encoder delay/sample issues - Remove unneeded .hca subkeys (use .acb/awb)
This commit is contained in:
commit
7cfa1f2960
@ -570,6 +570,7 @@ STREAMFILE* ffmpeg_get_streamfile(ffmpeg_codec_data* data);
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/* ffmpeg_decoder_utils.c (helper-things) */
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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);
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ffmpeg_codec_data* init_ffmpeg_atrac3_riff(STREAMFILE* sf, off_t offset, int* out_samples);
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ffmpeg_codec_data* init_ffmpeg_aac(STREAMFILE* sf, off_t offset, size_t size);
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/* ffmpeg_decoder_custom_opus.c (helper-things) */
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@ -655,6 +656,7 @@ size_t ac3_bytes_to_samples(size_t bytes, int full_block_align, int channels);
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size_t aac_get_samples(STREAMFILE* sf, off_t start_offset, size_t bytes);
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size_t mpeg_get_samples(STREAMFILE* sf, off_t start_offset, size_t bytes);
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int32_t mpeg_get_samples_clean(STREAMFILE* sf, off_t start, size_t size, size_t* p_loop_start, size_t* p_loop_end, int is_vbr);
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int mpc_get_samples(STREAMFILE* sf, off_t offset, int32_t* p_samples, int32_t* p_delay);
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/* helper to pass a wrapped, clamped, fake extension-ed, SF to another meta */
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@ -1006,6 +1006,7 @@ size_t aac_get_samples(STREAMFILE* sf, off_t start_offset, size_t bytes) {
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if (frame_size <= 0x08)
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break;
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//;VGM_LOG("AAC: %lx, %x\n", offset, frame_size);
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frames++;
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offset += frame_size;
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}
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@ -1013,6 +1014,82 @@ size_t aac_get_samples(STREAMFILE* sf, off_t start_offset, size_t bytes) {
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return frames * samples_per_frame;
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}
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/* variable-sized var reader */
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static int mpc_get_size(uint8_t* header, int header_size, int pos, int32_t* p_size) {
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uint8_t tmp;
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int32_t size = 0;
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do {
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if (pos >= header_size)
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return pos;
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tmp = header[pos];
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size = (size << 7) | (tmp & 0x7F);
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pos++;
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}
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while((tmp & 0x80));
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*p_size = size;
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return pos;
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}
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int mpc_get_samples(STREAMFILE* sf, off_t offset, int32_t* p_samples, int32_t* p_delay) {
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uint8_t header[0x20];
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int pos, size;
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int32_t samples = 0, delay = 0;
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if (read_streamfile(header, offset, sizeof(header), sf) != sizeof(header))
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goto fail;;
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if ((get_u32be(header) & 0xFFFFFF0F) == get_id32be("MP+\x07")) {
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samples = get_u32le(header + 0x04) * 1152; /* total frames */
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delay = 481; /* MPC_DECODER_SYNTH_DELAY */
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samples -= delay;
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/* in theory one header field can contain actual delay, not observed */
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}
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else if (get_u32be(header) == get_id32be("MPCK")) {
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/* V8 header is made if mini chunks (16b type, 8b size including type+size):
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* - SH: stream header
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* - RG: replay gain
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* - EI: encoder info
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* - SO: seek?
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* - ST: stream?
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* - AP: audio part start */
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if (get_u16be(header + 0x04) != 0x5348)
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goto fail;
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size = get_u8(header + 0x06);
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if (0x04 + size > sizeof(header))
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goto fail;
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if (get_u8(header + 0x0b) != 0x08)
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goto fail;
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/* SH chunk: */
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/* 0x00: CRC */
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/* 0x04: header version (8) */
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/* 0x05: samples (variable sized) */
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/* 0xNN: "beginning silence" (variable sized) */
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/* 0xNN: bitpacked channels/rate/etc */
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pos = mpc_get_size(header, sizeof(header), 0x0C, &samples);
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pos = mpc_get_size(header, sizeof(header), pos, &delay);
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samples -= delay; /* original delay, not SYNTH_DELAY */
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delay += 481; /* MPC_DECODER_SYNTH_DELAY */
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/* SYNTH_DELAY seems to be forced, but official code isn't very clear (known samples set delay to 0 but need SYNTH DELAY) */
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}
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else {
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goto fail;
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}
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if (p_samples) *p_samples = samples;
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if (p_delay) *p_delay = delay;
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return 1;
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fail:
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return 0;
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}
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/* ******************************************** */
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/* CUSTOM STREAMFILES */
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/* ******************************************** */
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@ -187,4 +187,24 @@ fail:
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return NULL;
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}
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ffmpeg_codec_data* init_ffmpeg_aac(STREAMFILE* sf, off_t offset, size_t size) {
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ffmpeg_codec_data* data = NULL;
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data = init_ffmpeg_offset(sf, offset, size);
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if (!data) goto fail;
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/* seeks to 0 eats first frame for whatever reason */
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ffmpeg_set_force_seek(data);
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/* raw AAC doesn't set this, while some decoders like FAAD remove 1024,
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* but should be handled in container as each encoder uses its own value
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* (Apple: 2112, FAAD: probably 1024, etc) */
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//ffmpeg_set_skip_samples(data, 1024);
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return data;
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fail:
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free_ffmpeg(data);
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return NULL;
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}
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#endif
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@ -4,43 +4,46 @@
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/* SWVR - from EA games, demuxed from .av/trk/mis/etc [Future Cop L.A.P.D. (PS/PC), Freekstyle (PS2/GC), EA Sports Supercross (PS)] */
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VGMSTREAM * init_vgmstream_ea_swvr(STREAMFILE *streamFile) {
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VGMSTREAM* init_vgmstream_ea_swvr(STREAMFILE* sf) {
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VGMSTREAM * vgmstream = NULL;
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off_t start_offset;
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int loop_flag = 0, channel_count, sample_rate, big_endian;
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int loop_flag = 0, channels, sample_rate, big_endian;
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coding_t coding;
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uint32_t block_id;
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int32_t (*read_32bit)(off_t,STREAMFILE*) = NULL;
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int16_t (*read_16bit)(off_t,STREAMFILE*) = NULL;
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int total_subsongs, target_subsong = streamFile->stream_index;
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int total_subsongs, target_subsong = sf->stream_index;
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/* checks */
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/* .stream: common (found inside files)
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* .str: shortened, probably unnecessary */
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if (!check_extensions(streamFile,"stream,str"))
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if (!check_extensions(sf,"stream,str"))
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goto fail;
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/* Files have no actual audio headers, so we inspect the first block for known values.
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* Freekstyle uses multiblocks/subsongs (though some subsongs may be clones?) */
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/* blocks ids are in machine endianness */
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if (read_32bitBE(0x00,streamFile) == 0x52565753) { /* "RVWS" (PS1/PS2/PC) */
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if (read_u32be(0x00,sf) == get_id32be("RVWS")) { /* PS1/PS2/PC */
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big_endian = 0;
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read_32bit = read_32bitLE;
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read_16bit = read_16bitLE;
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start_offset = read_32bit(0x04, streamFile);
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start_offset = read_32bit(0x04, sf);
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}
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else if (read_32bitBE(0x00,streamFile) == 0x53575652) { /* "SWVR" (GC) */
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else if (read_u32be(0x00,sf) == get_id32be("SWVR")) { /* GC */
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big_endian = 1;
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read_32bit = read_32bitBE;
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read_16bit = read_16bitBE;
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start_offset = read_32bit(0x04, streamFile);
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start_offset = read_32bit(0x04, sf);
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}
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else if (read_32bitBE(0x00,streamFile) == 0x4D474156) { /* "MGAV", Freekstyle (PS2) raw movies */
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else if (read_u32be(0x00,sf) == get_id32be("MGAV")) { /* Freekstyle (PS2) raw movies */
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big_endian = 0;
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read_32bit = read_32bitLE;
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read_16bit = read_16bitLE;
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start_offset = 0x00;
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}
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else if (read_32bitBE(0x00,streamFile) == 0x4453504D) { /* "DSPM", Freekstyle (GC) raw movies */
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else if (read_u32be(0x00,sf) == get_id32be("DSPM")) { /* Freekstyle (GC) raw movies */
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big_endian = 1;
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read_32bit = read_32bitBE;
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read_16bit = read_16bitBE;
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@ -50,64 +53,81 @@ VGMSTREAM * init_vgmstream_ea_swvr(STREAMFILE *streamFile) {
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goto fail;
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}
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if (read_32bit(start_offset+0x00, streamFile) == 0x50414444) /* "PADD" (Freekstyle) */
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start_offset += read_32bit(start_offset+0x04, streamFile);
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else if (read_32bit(start_offset+0x10, streamFile) == 0x53484452) /* "SHDR" (Future Cop PC) */
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start_offset += read_32bit(start_offset+0x04, streamFile);
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if (read_32bit(start_offset+0x00, sf) == get_id32be("PADD")) /* Freekstyle */
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start_offset += read_32bit(start_offset+0x04, sf);
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if (read_32bit(start_offset+0x00, streamFile) == 0x46494C4C) /* "FILL" (Freekstyle) */
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start_offset += read_32bit(start_offset+0x04, streamFile);
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if (read_32bit(start_offset+0x00, sf) == get_id32be("FILL")) /* Freekstyle */
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start_offset += read_32bit(start_offset+0x04, sf);
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total_subsongs = 1;
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block_id = read_32bit(start_offset, streamFile);
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block_id = read_32bit(start_offset, sf);
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/* value after block id (usually at 0x38) is number of blocks of 0x6000 (results in file size, including FILLs) */
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/* files are basically headerless so we inspect the first block
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* Freekstyle uses multiblocks/subsongs (though some subsongs may be clones?) */
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/* intended sample rate for PSX music (verified in emus) should be 14260, but is found in ELF as pitch value
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* (ex. Nascar Rumble 0x052C in SLUS_010.68 at 0x000143BC): 0x052C * 44100 / 4096 ~= 14254.98046875hz
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* Future Cop PSX pitch looks similar (comparing vs recordings). */
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switch(block_id) {
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case 0x5641474D: /* "VAGM" */
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case 0x5641474D: /* "VAGM" (stereo music) */
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coding = coding_PSX;
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if (read_16bit(start_offset+0x1a, streamFile) == 0x0024) {
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total_subsongs = read_32bit(start_offset+0x0c, streamFile)+1;
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sample_rate = 22050;
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if (read_16bit(start_offset+0x1a, sf) == 0x0024) {
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total_subsongs = read_32bit(start_offset+0x0c, sf)+1;
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sample_rate = 22050; /* Freekstyle (PS2) */
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}
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else {
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sample_rate = 14008;
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sample_rate = 1324 * 44100 / 4096; /* ~14254 [Future Cop (PS1), Nascar Rumble (PS1), EA Sports Motocross (PS1)] */
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}
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channel_count = 2;
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channels = 2;
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break;
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case 0x56414742: /* "VAGB" */
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case 0x56414742: /* "VAGB" (mono sfx/voices)*/
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coding = coding_PSX;
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if (read_16bit(start_offset+0x1a, streamFile) == 0x6400) {
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sample_rate = 22050;
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if (read_16bit(start_offset+0x1a, sf) == 0x6400) {
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sample_rate = 22050; /* Freekstyle (PS2) */
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}
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else {
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sample_rate = 14008;
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/* approximate as pitches vary per file (ex. Nascar Rumble: engine=3779, heli=22050, commentary=11050) */
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sample_rate = 1080 * 44100 / 4096; /* ~11627 [EA Sports Motocross (PS1)] */
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}
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channel_count = 1;
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channels = 1;
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break;
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case 0x4453504D: /* "DSPM" */
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case 0x4453504D: /* "DSPM" (stereo music) */
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coding = coding_NGC_DSP;
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total_subsongs = read_32bit(start_offset+0x0c, streamFile)+1;
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sample_rate = 22050;
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channel_count = 2;
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total_subsongs = read_32bit(start_offset+0x0c, sf)+1;
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sample_rate = 22050; /* Freekstyle (GC) */
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channels = 2;
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break;
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case 0x44535042: /* "DSPB" */
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case 0x44535042: /* "DSPB" (mono voices/sfx) */
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coding = coding_NGC_DSP;
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channel_count = 1;
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sample_rate = 22050;
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channels = 1;
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sample_rate = 22050; /* Freekstyle (GC) */
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break;
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case 0x4D534943: /* "MSIC" */
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case 0x4D534943: /* "MSIC" (stereo music) */
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coding = coding_PCM8_U_int;
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channel_count = 2;
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sample_rate = 14008;
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channels = 2;
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sample_rate = 14291; /* assumed, by comparing vs PSX output [Future Cop (PC)] */
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break;
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||||
case 0x53484F43: /* "SHOC" (a generic block but hopefully has PC sounds) */
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if (read_32bit(start_offset+0x10, streamFile) == 0x53484F43) { /* SHDR */
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coding = coding_PCM8_U_int; //todo there are other codecs
|
||||
channel_count = 1;
|
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sample_rate = 14008;
|
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if (read_32bit(start_offset+0x10, sf) == get_id32be("SHDR")) { /* Future Cop (PC) */
|
||||
/* there is a mini header? after SHDR
|
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* 0x00: 5
|
||||
* 0x04: "snds"
|
||||
* 0x08: channels?
|
||||
* 0x0c: data size without blocks/padding
|
||||
* 0x10: null
|
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* 0x14: null
|
||||
* 0x18: null
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||||
* 0x1c: 1
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||||
* 0x20: 1
|
||||
* 0x24: 0x4F430000 */
|
||||
if (read_32bit(start_offset+0x18, sf) != get_id32be("snds"))
|
||||
goto fail;
|
||||
coding = coding_PCM8_U_int;
|
||||
channels = 1;
|
||||
sample_rate = 16000; /* assumed */
|
||||
}
|
||||
else {
|
||||
//todo
|
||||
/* The Lord of the Rings - The Return of the King (PC) uses IMA for sfx (in non-RVWS chunks)
|
||||
* and some unknown codec for streams (mixed with strings and cutscene(?) commands) */
|
||||
goto fail;
|
||||
}
|
||||
break;
|
||||
@ -119,25 +139,25 @@ VGMSTREAM * init_vgmstream_ea_swvr(STREAMFILE *streamFile) {
|
||||
if (target_subsong == 0) target_subsong = 1;
|
||||
if (target_subsong < 0 || target_subsong > total_subsongs || total_subsongs < 1) goto fail;
|
||||
|
||||
loop_flag = 0;//(channel_count > 1); /* some Future Cop LAPD tracks repeat but other games have fadeouts */
|
||||
loop_flag = 0;//(channels > 1); /* some Future Cop LAPD tracks repeat but other games have fadeouts */
|
||||
|
||||
|
||||
/* build the VGMSTREAM */
|
||||
vgmstream = allocate_vgmstream(channel_count,loop_flag);
|
||||
vgmstream = allocate_vgmstream(channels, loop_flag);
|
||||
if (!vgmstream) goto fail;
|
||||
|
||||
vgmstream->meta_type = meta_EA_SWVR;
|
||||
vgmstream->sample_rate = sample_rate;
|
||||
vgmstream->codec_endian = big_endian;
|
||||
vgmstream->num_streams = total_subsongs;
|
||||
vgmstream->stream_size = get_streamfile_size(streamFile) / total_subsongs; /* approx... */
|
||||
vgmstream->stream_size = get_streamfile_size(sf) / total_subsongs; /* approx... */
|
||||
|
||||
vgmstream->coding_type = coding;
|
||||
vgmstream->layout_type = layout_blocked_ea_swvr;
|
||||
/* DSP coefs are loaded per block */
|
||||
/* some files (voices etc) decode with pops but seems a mastering problem */
|
||||
|
||||
if (!vgmstream_open_stream(vgmstream,streamFile,start_offset))
|
||||
if (!vgmstream_open_stream(vgmstream,sf,start_offset))
|
||||
goto fail;
|
||||
|
||||
/* calc num_samples manually */
|
||||
@ -155,7 +175,7 @@ VGMSTREAM * init_vgmstream_ea_swvr(STREAMFILE *streamFile) {
|
||||
}
|
||||
vgmstream->num_samples += num_samples;
|
||||
}
|
||||
while (vgmstream->next_block_offset < get_streamfile_size(streamFile));
|
||||
while (vgmstream->next_block_offset < get_streamfile_size(sf));
|
||||
block_update(start_offset, vgmstream);
|
||||
}
|
||||
|
||||
|
@ -8,7 +8,7 @@ static int find_meta_loops(ffmpeg_codec_data* data, int32_t* p_loop_start, int32
|
||||
|
||||
/* parses any format supported by FFmpeg and not handled elsewhere:
|
||||
* - MP3 (.mp3, .mus): Marc Ecko's Getting Up (PC)
|
||||
* - MPC (.mpc): Moonshine Runners (PC), Asphalt 7 (PC)
|
||||
* - MPC (.mpc, mp+): Moonshine Runners (PC), Asphalt 7 (PC)
|
||||
* - FLAC (.flac): Warcraft 3 Reforged (PC), Call of Duty: Ghosts (PC)
|
||||
* - DUCK (.wav): Sonic Jam (SAT), Virtua Fighter 2 (SAT)
|
||||
* - ALAC/AAC (.caf): Chrono Trigger (iOS)
|
||||
@ -24,7 +24,7 @@ VGMSTREAM* init_vgmstream_ffmpeg(STREAMFILE* sf) {
|
||||
VGMSTREAM* vgmstream = NULL;
|
||||
ffmpeg_codec_data* data = NULL;
|
||||
int loop_flag = 0;
|
||||
int32_t loop_start = 0, loop_end = 0, num_samples = 0;
|
||||
int32_t loop_start = 0, loop_end = 0, num_samples = 0, encoder_delay = 0;
|
||||
int total_subsongs, target_subsong = sf->stream_index;
|
||||
|
||||
/* no checks */
|
||||
@ -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
|
||||
@ -76,10 +81,17 @@ VGMSTREAM* init_vgmstream_ffmpeg(STREAMFILE* sf) {
|
||||
}
|
||||
#endif
|
||||
|
||||
/* hack for MPC, that seeks/resets incorrectly due to seek table shenanigans */
|
||||
if (read_u32be(0x00, sf) == 0x4D502B07 || /* "MP+\7" (Musepack V7) */
|
||||
read_u32be(0x00, sf) == 0x4D50434B) { /* "MPCK" (Musepack V8) */
|
||||
/* hack for MPC */
|
||||
if (is_id32be(0x00, sf, "MP+\x07") || /* Musepack V7 */
|
||||
is_id32be(0x00, sf, "MP+\x17") || /* Musepack V7 with flag (seen in FFmpeg) */
|
||||
is_id32be(0x00, sf, "MPCK")) { /* Musepack V8 */
|
||||
/* FFmpeg seeks/resets incorrectly due to MPC seek table shenanigans */
|
||||
ffmpeg_set_force_seek(data);
|
||||
|
||||
/* FFmpeg gets this wrong as usual (specially V8 samples) */
|
||||
mpc_get_samples(sf, 0x00, &num_samples, &encoder_delay);
|
||||
|
||||
ffmpeg_set_skip_samples(data, encoder_delay);
|
||||
}
|
||||
|
||||
/* default but often inaccurate when calculated using bitrate (wrong for VBR) */
|
||||
@ -91,7 +103,7 @@ VGMSTREAM* init_vgmstream_ffmpeg(STREAMFILE* sf) {
|
||||
/* build VGMSTREAM */
|
||||
vgmstream = allocate_vgmstream(data->channels, loop_flag);
|
||||
if (!vgmstream) goto fail;
|
||||
|
||||
|
||||
vgmstream->sample_rate = data->sampleRate;
|
||||
vgmstream->meta_type = meta_FFMPEG;
|
||||
vgmstream->coding_type = coding_FFmpeg;
|
||||
@ -104,14 +116,13 @@ VGMSTREAM* init_vgmstream_ffmpeg(STREAMFILE* sf) {
|
||||
vgmstream->channel_layout = ffmpeg_get_channel_layout(vgmstream->codec_data);
|
||||
|
||||
return vgmstream;
|
||||
|
||||
|
||||
fail:
|
||||
free_ffmpeg(data);
|
||||
if (vgmstream) {
|
||||
vgmstream->codec_data = NULL;
|
||||
close_vgmstream(vgmstream);
|
||||
}
|
||||
|
||||
return NULL;
|
||||
}
|
||||
|
||||
|
@ -11,7 +11,7 @@ typedef struct {
|
||||
|
||||
|
||||
/**
|
||||
* List of known keys, extracted from the game files (mostly found in 2ch.net).
|
||||
* List of known keys, extracted from the game files (several found in 2ch.net, others from data analisys).
|
||||
* CRI's tools expect an unsigned 64 bit number string, but keys are commonly found online in hex form.
|
||||
* Keys only use 56 bits though, so the upper 8 bits can be ignored.
|
||||
*
|
||||
@ -401,6 +401,9 @@ static const hcakey_info hcakey_list[] = {
|
||||
/* Mushoku Tensei: Game ni Nattemo Honki Dasu (Android) */
|
||||
{12281329554827291428u}, // AA700C292CFCAB24
|
||||
|
||||
/* Dragalia Lost (iOS/Android) */
|
||||
{2967411924141}, // 000002B2E7889CAD
|
||||
|
||||
/* D4DJ Groovy Mix (Android) [base files] */
|
||||
{393410674916959300}, // 0575ACECA945A444
|
||||
/* D4DJ Groovy Mix (Android) [music_* files, per-song later mixed with subkey] */
|
||||
@ -627,9 +630,6 @@ static const hcakey_info hcakey_list[] = {
|
||||
{0xb96786621e27daf3},
|
||||
{0x1111d6c10e509824},
|
||||
|
||||
/* Dragalia Lost (iOS/Android) */
|
||||
{2967411924141, subkeys_dgl, sizeof(subkeys_dgl) / sizeof(subkeys_dgl[0]) }, // 000002B2E7889CAD
|
||||
|
||||
};
|
||||
|
||||
#endif/*_HCA_KEYS_H_*/
|
||||
|
File diff suppressed because it is too large
Load Diff
@ -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;
|
||||
|
@ -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;
|
||||
|
@ -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];
|
||||
|
@ -1,10 +1,11 @@
|
||||
#include "meta.h"
|
||||
#include "../coding/coding.h"
|
||||
|
||||
|
||||
/* VXN - from Gameloft mobile games */
|
||||
VGMSTREAM* init_vgmstream_vxn(STREAMFILE* sf) {
|
||||
VGMSTREAM* vgmstream = NULL;
|
||||
int loop_flag = 0, channel_count, codec, sample_rate, block_align, bits, num_samples;
|
||||
int loop_flag = 0, channels, codec, sample_rate, block_align, bits, num_samples, encoder_delay;
|
||||
off_t start_offset, stream_offset, chunk_offset, first_offset = 0x00;
|
||||
size_t stream_size;
|
||||
int total_subsongs, target_subsong = sf->stream_index;
|
||||
@ -13,22 +14,32 @@ VGMSTREAM* init_vgmstream_vxn(STREAMFILE* sf) {
|
||||
if (!check_extensions(sf,"vxn"))
|
||||
goto fail;
|
||||
|
||||
if (read_u32be(0x00,sf) != 0x566F784E) /* "VoxN" */
|
||||
if (!is_id32be(0x00,sf, "VoxN"))
|
||||
goto fail;
|
||||
/* 0x04: chunk size */
|
||||
/* 0x08: ASCII version? ("0.0.1") */
|
||||
if (read_u32le(0x10,sf) != get_streamfile_size(sf))
|
||||
goto fail;
|
||||
/* 0x14: first MPC data start */
|
||||
|
||||
/* header is RIFF-like with many custom chunks */
|
||||
if (!find_chunk_le(sf, 0x41666D74,first_offset,0, &chunk_offset,NULL)) /* "Afmt" */
|
||||
goto fail;
|
||||
codec = read_u16le(chunk_offset+0x00, sf);
|
||||
channel_count = read_u16le(chunk_offset+0x02, sf);
|
||||
channels = read_u16le(chunk_offset+0x02, sf);
|
||||
sample_rate = read_u32le(chunk_offset+0x04, sf);
|
||||
block_align = read_u16le(chunk_offset+0x08, sf);
|
||||
bits = read_16bitLE(chunk_offset+0x0a, sf);
|
||||
bits = read_s16le(chunk_offset+0x0a, sf);
|
||||
|
||||
/* files are divided into segment subsongs, often a leadout and loop in that order
|
||||
* (the "Plst" and "Rule" chunks may have order info) */
|
||||
/* files are divided into segment subsongs, often a leadout and loop in that order,
|
||||
* but may also have +15 [Asphalt 9]. Various chunks seem to define how they are used:
|
||||
* - Rule: optional, not related to number of playlists
|
||||
* - Plst: sets N playlists + some flags?
|
||||
* - Stat: N entries (1 per playlist), start? + name (null-terminated, size 0x0C) + sample values?
|
||||
* - Trsn: optional, transitions for playlists?
|
||||
* - Grps: N entries (1 per playlist), 0x18 per entry, unknown
|
||||
* - Gprs: playlist segments, 0x20 per entry, playlist id + segment + null + segment? + flags (0/1/-1)
|
||||
*/
|
||||
if (!find_chunk_le(sf, 0x5365676D,first_offset,0, &chunk_offset,NULL)) /* "Segm" */
|
||||
goto fail;
|
||||
total_subsongs = read_u32le(chunk_offset+0x00, sf);
|
||||
@ -45,7 +56,7 @@ VGMSTREAM* init_vgmstream_vxn(STREAMFILE* sf) {
|
||||
|
||||
|
||||
/* build the VGMSTREAM */
|
||||
vgmstream = allocate_vgmstream(channel_count,loop_flag);
|
||||
vgmstream = allocate_vgmstream(channels, loop_flag);
|
||||
if (!vgmstream) goto fail;
|
||||
|
||||
vgmstream->sample_rate = sample_rate;
|
||||
@ -85,16 +96,24 @@ VGMSTREAM* init_vgmstream_vxn(STREAMFILE* sf) {
|
||||
break;
|
||||
|
||||
#ifdef VGM_USE_FFMPEG
|
||||
case 0x0800: /* Musepack (ex. Asphalt Xtreme) */
|
||||
case 0x0800: /* Musepack (ex. Asphalt Xtreme, Asphalt 9) */
|
||||
if (bits != -1) goto fail;
|
||||
|
||||
vgmstream->codec_data = init_ffmpeg_offset(sf, start_offset,stream_size);
|
||||
vgmstream->codec_data = init_ffmpeg_offset(sf, start_offset, stream_size);
|
||||
if (!vgmstream->codec_data) goto fail;
|
||||
vgmstream->coding_type = coding_FFmpeg;
|
||||
vgmstream->layout_type = layout_none;
|
||||
|
||||
/* unlike standard .mpc, .vxn has no seek table so no need to fix */
|
||||
//ffmpeg_set_force_seek(vgmstream->codec_data);
|
||||
|
||||
encoder_delay = 0;
|
||||
mpc_get_samples(sf, start_offset, NULL, &encoder_delay);
|
||||
/* num_samples matches MPC samples */
|
||||
|
||||
/* FFmpeg doesn't set encoder delay */
|
||||
ffmpeg_set_skip_samples(vgmstream->codec_data, encoder_delay);
|
||||
|
||||
break;
|
||||
#endif
|
||||
|
||||
|
@ -109,6 +109,10 @@ VGMSTREAM* init_vgmstream_wwise(STREAMFILE* sf) {
|
||||
}
|
||||
|
||||
vgmstream->num_samples = pcm_bytes_to_samples(ww.data_size, ww.channels, ww.bits_per_sample);
|
||||
|
||||
/* truncated .bnk RIFFs that only have header and no data is possible [Metal Gear Solid V (PC)] */
|
||||
if (ww.truncated && !vgmstream->num_samples)
|
||||
vgmstream->num_samples = 1; /* force something to avoid broken subsongs */
|
||||
break;
|
||||
|
||||
case IMA: /* common */
|
||||
|
Loading…
x
Reference in New Issue
Block a user