#include "meta.h" #include "../coding/coding.h" #include "../util.h" /* MSF - Sony's PS3 SDK format (MultiStream File) */ VGMSTREAM * init_vgmstream_ps3_msf(STREAMFILE *streamFile) { VGMSTREAM * vgmstream = NULL; off_t start_offset, header_offset = 0; uint32_t data_size, loop_start = 0, loop_end = 0; uint32_t id, codec_id, flags; int loop_flag = 0, channel_count; /* check extension, case insensitive */ if (!check_extensions(streamFile,"msf,at3")) goto fail; /* .at3: Silent Hill HD Collection */ /* "WMSF" variation with a mini header over the MSFC header, same extension */ if (read_32bitBE(0x00,streamFile) == 0x574D5346) { header_offset = 0x10; } start_offset = header_offset+0x40; /* MSF header is always 0x40 */ /* check header "MSF" + version-char * usually "MSF\0\1", "MSF\0\2", "MSF0"(\3\0), "MSF5"(\3\5), "MSFC"(\4\3) (last/common version) */ id = read_32bitBE(header_offset+0x00,streamFile); if ((id & 0xffffff00) != 0x4D534600) goto fail; codec_id = read_32bitBE(header_offset+0x04,streamFile); channel_count = read_32bitBE(header_offset+0x08,streamFile); data_size = read_32bitBE(header_offset+0x0C,streamFile); /* without header */ if (data_size == 0xFFFFFFFF) /* unneeded? */ data_size = get_streamfile_size(streamFile) - start_offset; /* byte flags, not in MSFv1 or v2 * 0x01/02/04/08: loop marker 0/1/2/3 * 0x10: resample options (force 44/48khz) * 0x20: VBR MP3 * 0x40: joint stereo MP3 (apparently interleaved stereo for other formats) * 0x80+: (none/reserved) */ flags = read_32bitBE(header_offset+0x14,streamFile); /* sometimes loop_start/end is set with flag 0x10, but from tests it only loops if 0x01/02 is set * 0x10 often goes with 0x01 but not always (Castlevania HoD); Malicious PS3 uses flag 0x2 instead */ loop_flag = flags != 0xffffffff && ((flags & 0x01) || (flags & 0x02)); /* loop markers (marker N @ 0x18 + N*(4+4), but in practice only marker 0 is used) */ if (loop_flag) { loop_start = read_32bitBE(header_offset+0x18,streamFile); loop_end = read_32bitBE(header_offset+0x1C,streamFile); /* loop duration */ loop_end = loop_start + loop_end; /* usually equals data_size but not always */ if (loop_end > data_size)/* not seen */ loop_end = data_size; } /* build the VGMSTREAM */ vgmstream = allocate_vgmstream(channel_count,loop_flag); if (!vgmstream) goto fail; /* Sample rate hack for strange MSFv1 files that don't have a specified frequency */ vgmstream->sample_rate = read_32bitBE(header_offset+0x10,streamFile); if (vgmstream->sample_rate == 0x00000000) /* PS ADPCM only? */ vgmstream->sample_rate = 48000; vgmstream->meta_type = meta_PS3_MSF; switch (codec_id) { case 0x0: /* PCM (Big Endian) */ case 0x1: { /* PCM (Little Endian) */ vgmstream->coding_type = codec_id==0 ? coding_PCM16BE : coding_PCM16LE; vgmstream->layout_type = channel_count == 1 ? layout_none : layout_interleave; vgmstream->interleave_block_size = 2; vgmstream->num_samples = data_size/2/channel_count; if (loop_flag){ vgmstream->loop_start_sample = loop_start/2/channel_count; vgmstream->loop_end_sample = loop_end/2/channel_count; } break; } case 0x2: { /* PCM 32 (Float) */ goto fail; //probably unused/spec only } case 0x3: { /* PS ADPCM */ vgmstream->coding_type = coding_PSX; vgmstream->layout_type = channel_count == 1 ? layout_none : layout_interleave; vgmstream->interleave_block_size = 0x10; vgmstream->num_samples = data_size*28/16/channel_count; if (loop_flag) { vgmstream->loop_start_sample = loop_start*28/16/channel_count; vgmstream->loop_end_sample = loop_end*28/16/channel_count; } break; } #ifdef VGM_USE_FFMPEG case 0x4: /* ATRAC3 low (66 kbps, frame size 96, Joint Stereo) */ case 0x5: /* ATRAC3 mid (105 kbps, frame size 152) */ case 0x6: { /* ATRAC3 high (132 kbps, frame size 192) */ ffmpeg_codec_data *ffmpeg_data = NULL; uint8_t buf[100]; int32_t bytes, block_size, encoder_delay, joint_stereo; block_size = (codec_id==4 ? 0x60 : (codec_id==5 ? 0x98 : 0xC0)) * vgmstream->channels; joint_stereo = (codec_id==4); /* interleaved joint stereo (ch must be even) */ /* MSF skip samples: from tests with MSEnc and real files (ex. TTT2 eddy.msf v43, v01 demos) seems like 1162 is consistent. * Atelier Rorona bt_normal01 needs it to properly skip the beginning garbage but usually doesn't matter. * (note that encoder may add a fade-in with looping/resampling enabled but should be skipped) */ encoder_delay = 1162; vgmstream->num_samples = atrac3_bytes_to_samples(data_size, block_size) - encoder_delay; 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 bytes = ffmpeg_make_riff_atrac3(buf, 100, vgmstream->num_samples, data_size, vgmstream->channels, vgmstream->sample_rate, block_size, joint_stereo, encoder_delay); if (bytes <= 0) goto fail; ffmpeg_data = init_ffmpeg_header_offset(streamFile, buf,bytes, start_offset,data_size); if (!ffmpeg_data) goto fail; vgmstream->codec_data = ffmpeg_data; vgmstream->coding_type = coding_FFmpeg; vgmstream->layout_type = layout_none; /* manually set skip_samples if FFmpeg didn't do it */ if (ffmpeg_data->skipSamples <= 0) { ffmpeg_set_skip_samples(ffmpeg_data, encoder_delay); } /* MSF loop/sample values are offsets so trickier to adjust the skip_samples but this seems correct */ if (loop_flag) { vgmstream->loop_start_sample = atrac3_bytes_to_samples(loop_start, block_size) /* - encoder_delay*/; vgmstream->loop_end_sample = atrac3_bytes_to_samples(loop_end, block_size) - encoder_delay; } break; } #endif #ifdef VGM_USE_FFMPEG case 0x7: { /* MPEG (LAME MP3 of any quality) */ /* delegate to FFMpeg, it can parse MSF files */ ffmpeg_codec_data *ffmpeg_data = init_ffmpeg_offset(streamFile, header_offset, streamFile->get_size(streamFile) ); if ( !ffmpeg_data ) goto fail; vgmstream->codec_data = ffmpeg_data; vgmstream->coding_type = coding_FFmpeg; vgmstream->layout_type = layout_none; /* vgmstream->num_samples = ffmpeg_data->totalSamples; */ /* duration may not be set/inaccurate */ vgmstream->num_samples = (int64_t)data_size * ffmpeg_data->sampleRate * 8 / ffmpeg_data->bitrate; if (loop_flag) { //todo properly apply encoder delay, which seems to vary between 1152 (1f), 528, 576 or 528+576 int frame_size = ffmpeg_data->frameSize; vgmstream->loop_start_sample = (int64_t)loop_start * ffmpeg_data->sampleRate * 8 / ffmpeg_data->bitrate; vgmstream->loop_start_sample -= vgmstream->loop_start_sample==frame_size ? frame_size : vgmstream->loop_start_sample % frame_size; vgmstream->loop_end_sample = (int64_t)loop_end * ffmpeg_data->sampleRate * 8 / ffmpeg_data->bitrate; vgmstream->loop_end_sample -= vgmstream->loop_end_sample==frame_size ? frame_size : vgmstream->loop_end_sample % frame_size; } break; } #endif #if defined(VGM_USE_MPEG) && !defined(VGM_USE_FFMPEG) case 0x7: { /* MPEG (LAME MP3 of any quality) */ int frame_size = 576; /* todo incorrect looping calcs */ mpeg_codec_data *mpeg_data = NULL; coding_t ct; mpeg_data = init_mpeg_codec_data(streamFile, start_offset, &ct, vgmstream->channels); if (!mpeg_data) goto fail; vgmstream->codec_data = mpeg_data; vgmstream->coding_type = ct; vgmstream->layout_type = layout_mpeg; vgmstream->num_samples = mpeg_bytes_to_samples(data_size, mpeg_data); vgmstream->num_samples -= vgmstream->num_samples % frame_size; if (loop_flag) { vgmstream->loop_start_sample = mpeg_bytes_to_samples(loop_start, mpeg_data); vgmstream->loop_start_sample -= vgmstream->loop_start_sample % frame_size; vgmstream->loop_end_sample = mpeg_bytes_to_samples(loop_end, mpeg_data); vgmstream->loop_end_sample -= vgmstream->loop_end_sample % frame_size; } vgmstream->interleave_block_size = 0; break; } #endif default: /* 8+: not defined */ goto fail; } /* open the file for reading */ if (!vgmstream_open_stream(vgmstream,streamFile,start_offset)) goto fail; return vgmstream; fail: close_vgmstream(vgmstream); return NULL; }