c97c5e5d17
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30490 a1c6a512-1295-4272-9138-f99709370657
858 lines
22 KiB
C
858 lines
22 KiB
C
// Game_Music_Emu 0.5.5. http://www.slack.net/~ant/
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#include "vgm_emu.h"
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#include "blargg_endian.h"
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#include <string.h>
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#include <math.h>
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/* Copyright (C) 2003-2006 Shay Green. This module is free software; you
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can redistribute it and/or modify it under the terms of the GNU Lesser
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General Public License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version. This
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module is distributed in the hope that it will be useful, but WITHOUT ANY
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WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
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FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more
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details. You should have received a copy of the GNU Lesser General Public
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License along with this module; if not, write to the Free Software Foundation,
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Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */
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#include "blargg_source.h"
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const char* const gme_wrong_file_type = "Wrong file type for this emulator";
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int const fm_gain = 3; // FM emulators are internally quieter to avoid 16-bit overflow
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// VGM commands (Spec v1.50)
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enum {
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cmd_gg_stereo = 0x4F,
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cmd_psg = 0x50,
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cmd_ym2413 = 0x51,
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cmd_ym2612_port0 = 0x52,
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cmd_ym2612_port1 = 0x53,
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cmd_ym2151 = 0x54,
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cmd_delay = 0x61,
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cmd_delay_735 = 0x62,
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cmd_delay_882 = 0x63,
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cmd_byte_delay = 0x64,
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cmd_end = 0x66,
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cmd_data_block = 0x67,
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cmd_short_delay = 0x70,
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cmd_pcm_delay = 0x80,
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cmd_pcm_seek = 0xE0,
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pcm_block_type = 0x00,
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ym2612_dac_port = 0x2A,
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ym2612_dac_pan_port = 0xB6
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};
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static void clear_track_vars( struct Vgm_Emu* this )
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{
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this->current_track = -1;
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track_stop( &this->track_filter );
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}
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int play_frame( struct Vgm_Emu* this, blip_time_t blip_time, int sample_count, sample_t* buf );
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static int play_frame_( void* data, blip_time_t blip_time, int sample_count, short int* buf )
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{
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return play_frame( (struct Vgm_Emu*) data, blip_time, sample_count, buf );
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}
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void Vgm_init( struct Vgm_Emu* this )
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{
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this->sample_rate = 0;
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this->mute_mask_ = 0;
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this->tempo = (int)(FP_ONE_TEMPO);
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// defaults
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this->tfilter = *track_get_setup( &this->track_filter );
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this->tfilter.max_initial = 2;
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this->tfilter.lookahead = 1;
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this->track_filter.silence_ignored_ = false;
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// Disable oversampling by default
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this->disable_oversampling = true;
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this->psg_rate = 0;
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Sms_apu_init( &this->psg );
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Synth_init( &this->pcm );
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Buffer_init( &this->stereo_buf );
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Blip_init( &this->blip_buf );
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// Init fm chips
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Ym2413_init( &this->ym2413 );
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Ym2612_init( &this->ym2612 );
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// Init resampler
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Resampler_init( &this->resampler );
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Resampler_set_callback( &this->resampler, play_frame_, this );
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// Set sound gain, a value too high
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// will cause saturation
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Sound_set_gain(this, (int)(FP_ONE_GAIN*0.7));
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// Unload
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this->voice_count = 0;
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this->voice_types = 0;
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clear_track_vars( this );
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}
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// Track info
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static byte const* skip_gd3_str( byte const* in, byte const* end )
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{
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while ( end - in >= 2 )
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{
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in += 2;
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if ( !(in [-2] | in [-1]) )
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break;
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}
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return in;
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}
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static byte const* get_gd3_str( byte const* in, byte const* end, char* field )
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{
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byte const* mid = skip_gd3_str( in, end );
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int i, len = (mid - in) / 2 - 1;
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if ( field && len > 0 )
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{
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len = min( len, (int) gme_max_field );
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field [len] = 0;
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for ( i = 0; i < len; i++ )
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field [i] = (in [i * 2 + 1] ? '?' : in [i * 2]); // TODO: convert to utf-8
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}
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return mid;
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}
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static byte const* get_gd3_pair( byte const* in, byte const* end, char* field )
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{
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return skip_gd3_str( get_gd3_str( in, end, field ), end );
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}
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static void parse_gd3( byte const* in, byte const* end, struct track_info_t* out )
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{
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in = get_gd3_pair( in, end, out->song );
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in = get_gd3_pair( in, end, out->game );
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in = get_gd3_pair( in, end, NULL ); // Skip system
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in = get_gd3_pair( in, end, out->author );
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}
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int const gd3_header_size = 12;
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static int check_gd3_header( byte const* h, int remain )
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{
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if ( remain < gd3_header_size ) return 0;
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if ( memcmp( h, "Gd3 ", 4 ) ) return 0;
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if ( get_le32( h + 4 ) >= 0x200 ) return 0;
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int gd3_size = get_le32( h + 8 );
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if ( gd3_size > remain - gd3_header_size )
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gd3_size = remain - gd3_header_size;
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return gd3_size;
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}
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static byte const* gd3_data( struct Vgm_Emu* this, int* size )
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{
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if ( size )
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*size = 0;
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int gd3_offset = get_le32( header( this )->gd3_offset ) - 0x2C;
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if ( gd3_offset < 0 )
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return 0;
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byte const* gd3 = this->file_begin + header_size + gd3_offset;
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int gd3_size = check_gd3_header( gd3, this->file_end - gd3 );
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if ( !gd3_size )
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return 0;
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if ( size )
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*size = gd3_size + gd3_header_size;
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return gd3;
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}
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static void get_vgm_length( struct header_t const* h, struct track_info_t* out )
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{
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int length = get_le32( h->track_duration ) * 10 / 441;
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if ( length > 0 )
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{
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int loop = get_le32( h->loop_duration );
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if ( loop > 0 && get_le32( h->loop_offset ) )
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{
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out->loop_length = loop * 10 / 441;
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out->intro_length = length - out->loop_length;
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}
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else
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{
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out->length = length; // 1000 / 44100 (VGM files used 44100 as timebase)
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out->intro_length = length; // make it clear that track is no longer than length
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out->loop_length = 0;
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}
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}
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}
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static blargg_err_t track_info( struct Vgm_Emu* this, struct track_info_t* out )
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{
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memset(out, 0, sizeof out);
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get_vgm_length( header( this ), out );
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int size;
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byte const* gd3 = gd3_data( this, &size );
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if ( gd3 )
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parse_gd3( gd3 + gd3_header_size, gd3 + size, out );
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return 0;
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}
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static blargg_err_t check_vgm_header( struct header_t* h )
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{
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if ( memcmp( h->tag, "Vgm ", 4 ) )
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return gme_wrong_file_type;
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return 0;
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}
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static void set_voice( struct Vgm_Emu* this, int i, struct Blip_Buffer* c, struct Blip_Buffer* l, struct Blip_Buffer* r )
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{
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if ( i < sms_osc_count ) {
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Sms_apu_set_output( &this->psg, i, c, l, r );
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}
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}
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blargg_err_t setup_fm( struct Vgm_Emu* this );
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blargg_err_t Vgm_load_mem( struct Vgm_Emu* this, byte const* new_data, long new_size, bool parse_info )
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{
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// Unload
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this->voice_count = 0;
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clear_track_vars( this );
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// Clear info
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memset( &this->info, 0, sizeof this->info );
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assert( offsetof (struct header_t,unused2 [8]) == header_size );
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if ( new_size <= header_size )
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return gme_wrong_file_type;
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// Reset data pointers
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this->file_begin = new_data;
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this->file_end = new_data + new_size;
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struct header_t* h = (struct header_t*) new_data;
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RETURN_ERR( check_vgm_header( h ) );
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check( get_le32( h.version ) <= 0x150 );
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// If this was VGZ file gd3 parse info
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if ( parse_info ) {
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track_info( this, &this->info );
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// If file was trimmed add an
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// incomplete token to the game tag
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if ( get_le32( h->data_size ) > (unsigned) new_size ) {
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*((char *) this->file_end) = cmd_end;
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strcat(this->info.game, "(Trimmed VGZ file)" );
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}
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}
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// Get loop
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this->loop_begin = this->file_end;
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// If file was trimmed don't loop
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if ( get_le32( h->loop_offset ) && get_le32( h->data_size ) <= (unsigned) new_size )
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this->loop_begin = &new_data [get_le32( h->loop_offset ) + offsetof (struct header_t,loop_offset)];
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// PSG rate
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this->psg_rate = get_le32( h->psg_rate );
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if ( !this->psg_rate )
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this->psg_rate = 3579545;
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Blip_set_clock_rate( &this->blip_buf, this->psg_rate );
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// Disable FM
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this->fm_rate = 0;
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Ym2612_enable( &this->ym2612, false );
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Ym2413_enable( &this->ym2413, false );
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this->voice_count = sms_osc_count;
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static int const types [8] = {
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wave_type+1, wave_type+2, wave_type+3, noise_type+1,
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0, 0, 0, 0
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};
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this->voice_types = types;
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RETURN_ERR( setup_fm( this ) );
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// do after FM in case output buffer is changed
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// setup buffer
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this->clock_rate_ = this->psg_rate;
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Buffer_clock_rate( &this->stereo_buf, this->psg_rate );
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RETURN_ERR( Buffer_set_channel_count( &this->stereo_buf, this->voice_count, this->voice_types ) );
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this->buf_changed_count = Buffer_channels_changed_count( &this->stereo_buf );
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// Post load
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Sound_set_tempo( this, this->tempo );
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Sound_mute_voices( this, this->mute_mask_ );
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// so we can start playback
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this->current_track = 0;
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return 0;
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}
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void update_fm_rates( struct Vgm_Emu* this, int* ym2413_rate, int* ym2612_rate );
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static blargg_err_t init_fm( struct Vgm_Emu* this, int* rate )
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{
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int ym2612_rate = get_le32( header( this )->ym2612_rate );
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int ym2413_rate = get_le32( header( this )->ym2413_rate );
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if ( ym2413_rate && get_le32( header( this )->version ) < 0x110 )
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update_fm_rates( this, &ym2413_rate, &ym2612_rate );
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if ( ym2612_rate )
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{
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if ( !*rate )
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*rate = ym2612_rate / 144;
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RETURN_ERR( Ym2612_set_rate( &this->ym2612, *rate, ym2612_rate ) );
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Ym2612_enable( &this->ym2612, true );
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}
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else if ( ym2413_rate )
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{
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if ( !*rate )
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*rate = ym2413_rate / 72;
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int result = Ym2413_set_rate( &this->ym2413, *rate, ym2413_rate );
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if ( result == 2 )
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return "YM2413 FM sound not supported";
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CHECK_ALLOC( !result );
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Ym2413_enable( &this->ym2413, true );
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}
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this->fm_rate = *rate;
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return 0;
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}
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blargg_err_t setup_fm( struct Vgm_Emu* this )
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{
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int fm_rate = 0;
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if ( !this->disable_oversampling )
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fm_rate = (this->sample_rate * 3) / 2; // oversample factor = 1.5
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RETURN_ERR( init_fm( this, &fm_rate ) );
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if ( uses_fm( this ) )
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{
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this->voice_count = 8;
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RETURN_ERR( Resampler_setup( &this->resampler, fm_rate, fm_gain, this->sample_rate, this->gain ) );
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RETURN_ERR( Resampler_reset( &this->resampler, Blip_length( &this->blip_buf ) * this->sample_rate / 1000 ) );
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Sms_apu_volume( &this->psg, ((this->gain/5)-(this->gain*5)/1000) * fm_gain );
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}
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else
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{
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Sms_apu_volume( &this->psg, this->gain );
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}
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return 0;
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}
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// Emulation
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blip_time_t run( struct Vgm_Emu* this, vgm_time_t end_time );
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static blip_time_t run_psg( struct Vgm_Emu* this, int msec )
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{
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blip_time_t t = run( this, msec * this->vgm_rate / 1000 );
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Sms_apu_end_frame( &this->psg, t );
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return t;
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}
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static void check_end( struct Vgm_Emu* this )
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{
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if( this->pos >= this->file_end )
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track_set_end( &this->track_filter );
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}
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static blargg_err_t run_clocks( struct Vgm_Emu* this, blip_time_t* time_io, int msec )
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{
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check_end( this );
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*time_io = run_psg( this, msec );
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return 0;
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}
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blargg_err_t play_( void *emu, int count, sample_t out [] )
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{
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struct Vgm_Emu* this = (struct Vgm_Emu*) emu;
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if ( !uses_fm( this ) ) {
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int remain = count;
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while ( remain )
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{
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Buffer_disable_immediate_removal( &this->stereo_buf );
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remain -= Buffer_read_samples( &this->stereo_buf, &out [count - remain], remain );
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if ( remain )
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{
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if ( this->buf_changed_count != Buffer_channels_changed_count( &this->stereo_buf ) )
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{
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this->buf_changed_count = Buffer_channels_changed_count( &this->stereo_buf );
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// Remute voices
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Sound_mute_voices( this, this->mute_mask_ );
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}
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int msec = Buffer_length( &this->stereo_buf );
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blip_time_t clocks_emulated = msec * this->clock_rate_ / 1000 - 100;
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RETURN_ERR( run_clocks( this, &clocks_emulated, msec ) );
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assert( clocks_emulated );
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Buffer_end_frame( &this->stereo_buf, clocks_emulated );
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}
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}
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return 0;
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}
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Resampler_play( &this->resampler, count, out, &this->blip_buf );
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return 0;
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}
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// Vgm_Emu_impl
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static inline int command_len( int command )
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{
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static byte const lens [0x10] = {
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// 0 1 2 3 4 5 6 7 8 9 A B C D E F
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1,1,1,2,2,3,1,1,1,1,3,3,4,4,5,5
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};
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int len = lens [command >> 4];
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check( len != 1 );
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return len;
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}
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static inline fm_time_t to_fm_time( struct Vgm_Emu* this, vgm_time_t t )
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{
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return (t * this->fm_time_factor + this->fm_time_offset) >> fm_time_bits;
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}
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static inline blip_time_t to_psg_time( struct Vgm_Emu* this, vgm_time_t t )
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{
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return (t * this->blip_time_factor) >> blip_time_bits;
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}
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static void write_pcm( struct Vgm_Emu* this, vgm_time_t vgm_time, int amp )
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{
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check( amp >= 0 );
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blip_time_t blip_time = to_psg_time( this, vgm_time );
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int old = this->dac_amp;
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int delta = amp - old;
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this->dac_amp = amp;
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Blip_set_modified( &this->blip_buf );
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if ( old >= 0 ) // first write is ignored, to avoid click
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Synth_offset_inline( &this->pcm, blip_time, delta, &this->blip_buf );
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else
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this->dac_amp |= this->dac_disabled;
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}
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blip_time_t run( struct Vgm_Emu* this, vgm_time_t end_time )
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{
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vgm_time_t vgm_time = this->vgm_time;
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byte const* pos = this->pos;
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/* if ( pos > this->file_end )
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{
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warning( "Stream lacked end event" );
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} */
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while ( vgm_time < end_time && pos < this->file_end )
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{
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// TODO: be sure there are enough bytes left in stream for particular command
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// so we don't read past end
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switch ( *pos++ )
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{
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case cmd_end:
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pos = this->loop_begin; // if not looped, loop_begin == data_end
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break;
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case cmd_delay_735:
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vgm_time += 735;
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break;
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case cmd_delay_882:
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vgm_time += 882;
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break;
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case cmd_gg_stereo:
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Sms_apu_write_ggstereo( &this->psg, to_psg_time( this, vgm_time ), *pos++ );
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break;
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case cmd_psg:
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Sms_apu_write_data( &this->psg, to_psg_time( this, vgm_time ), *pos++ );
|
|
break;
|
|
|
|
case cmd_delay:
|
|
vgm_time += pos [1] * 0x100 + pos [0];
|
|
pos += 2;
|
|
break;
|
|
|
|
case cmd_byte_delay:
|
|
vgm_time += *pos++;
|
|
break;
|
|
|
|
case cmd_ym2413:
|
|
if ( Ym2413_run_until( &this->ym2413, to_fm_time( this, vgm_time ) ) )
|
|
Ym2413_write( &this->ym2413, pos [0], pos [1] );
|
|
pos += 2;
|
|
break;
|
|
|
|
case cmd_ym2612_port0:
|
|
if ( pos [0] == ym2612_dac_port )
|
|
{
|
|
write_pcm( this, vgm_time, pos [1] );
|
|
}
|
|
else if ( Ym2612_run_until( &this->ym2612, to_fm_time( this, vgm_time ) ) )
|
|
{
|
|
if ( pos [0] == 0x2B )
|
|
{
|
|
this->dac_disabled = (pos [1] >> 7 & 1) - 1;
|
|
this->dac_amp |= this->dac_disabled;
|
|
}
|
|
Ym2612_write0( &this->ym2612, pos [0], pos [1] );
|
|
}
|
|
pos += 2;
|
|
break;
|
|
|
|
case cmd_ym2612_port1:
|
|
if ( Ym2612_run_until( &this->ym2612, to_fm_time( this, vgm_time ) ) )
|
|
Ym2612_write1( &this->ym2612, pos [0], pos [1] );
|
|
pos += 2;
|
|
break;
|
|
|
|
case cmd_data_block: {
|
|
check( *pos == cmd_end );
|
|
int type = pos [1];
|
|
int size = get_le32( pos + 2 );
|
|
pos += 6;
|
|
if ( type == pcm_block_type )
|
|
this->pcm_data = pos;
|
|
pos += size;
|
|
break;
|
|
}
|
|
|
|
case cmd_pcm_seek:
|
|
this->pcm_pos = this->pcm_data + pos [3] * 0x1000000 + pos [2] * 0x10000 +
|
|
pos [1] * 0x100 + pos [0];
|
|
pos += 4;
|
|
break;
|
|
|
|
default: {
|
|
int cmd = pos [-1];
|
|
switch ( cmd & 0xF0 )
|
|
{
|
|
case cmd_pcm_delay:
|
|
write_pcm( this, vgm_time, *this->pcm_pos++ );
|
|
vgm_time += cmd & 0x0F;
|
|
break;
|
|
|
|
case cmd_short_delay:
|
|
vgm_time += (cmd & 0x0F) + 1;
|
|
break;
|
|
|
|
case 0x50:
|
|
pos += 2;
|
|
break;
|
|
|
|
default:
|
|
pos += command_len( cmd ) - 1;
|
|
/* warning( "Unknown stream event" ); */
|
|
}
|
|
}
|
|
}
|
|
}
|
|
vgm_time -= end_time;
|
|
this->pos = pos;
|
|
this->vgm_time = vgm_time;
|
|
|
|
return to_psg_time( this, end_time );
|
|
}
|
|
|
|
int play_frame( struct Vgm_Emu* this, blip_time_t blip_time, int sample_count, blip_sample_t out [] )
|
|
{
|
|
check_end( this);
|
|
|
|
// to do: timing is working mostly by luck
|
|
int min_pairs = (unsigned) sample_count / 2;
|
|
int vgm_time = (min_pairs << fm_time_bits) / this->fm_time_factor - 1;
|
|
assert( to_fm_time( this, vgm_time ) <= min_pairs );
|
|
int pairs;
|
|
while ( (pairs = to_fm_time( this, vgm_time )) < min_pairs )
|
|
vgm_time++;
|
|
//debug_printf( "pairs: %d, min_pairs: %d\n", pairs, min_pairs );
|
|
|
|
if ( Ym2612_enabled( &this->ym2612 ) )
|
|
{
|
|
Ym2612_begin_frame( &this->ym2612, out );
|
|
memset( out, 0, pairs * stereo * sizeof *out );
|
|
}
|
|
else if ( Ym2413_enabled( &this->ym2413 ) )
|
|
{
|
|
Ym2413_begin_frame( &this->ym2413, out );
|
|
}
|
|
|
|
run( this, vgm_time );
|
|
Ym2612_run_until( &this->ym2612, pairs );
|
|
Ym2413_run_until( &this->ym2413, pairs );
|
|
|
|
this->fm_time_offset = (vgm_time * this->fm_time_factor + this->fm_time_offset) - (pairs << fm_time_bits);
|
|
|
|
Sms_apu_end_frame( &this->psg, blip_time );
|
|
|
|
return pairs * stereo;
|
|
}
|
|
|
|
// Update pre-1.10 header FM rates by scanning commands
|
|
void update_fm_rates( struct Vgm_Emu* this, int* ym2413_rate, int* ym2612_rate )
|
|
{
|
|
byte const* p = this->file_begin + 0x40;
|
|
while ( p < this->file_end )
|
|
{
|
|
switch ( *p )
|
|
{
|
|
case cmd_end:
|
|
return;
|
|
|
|
case cmd_psg:
|
|
case cmd_byte_delay:
|
|
p += 2;
|
|
break;
|
|
|
|
case cmd_delay:
|
|
p += 3;
|
|
break;
|
|
|
|
case cmd_data_block:
|
|
p += 7 + get_le32( p + 3 );
|
|
break;
|
|
|
|
case cmd_ym2413:
|
|
*ym2612_rate = 0;
|
|
return;
|
|
|
|
case cmd_ym2612_port0:
|
|
case cmd_ym2612_port1:
|
|
*ym2612_rate = *ym2413_rate;
|
|
*ym2413_rate = 0;
|
|
return;
|
|
|
|
case cmd_ym2151:
|
|
*ym2413_rate = 0;
|
|
*ym2612_rate = 0;
|
|
return;
|
|
|
|
default:
|
|
p += command_len( *p );
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
// Music Emu
|
|
|
|
blargg_err_t Vgm_set_sample_rate( struct Vgm_Emu* this, int rate )
|
|
{
|
|
require( !this->sample_rate ); // sample rate can't be changed once set
|
|
RETURN_ERR( Blip_set_sample_rate( &this->blip_buf, rate, 1000 / 30 ) );
|
|
RETURN_ERR( Buffer_set_sample_rate( &this->stereo_buf, rate, 1000 / 20 ) );
|
|
|
|
// Set bass frequency
|
|
Buffer_bass_freq( &this->stereo_buf, 80 );
|
|
|
|
this->sample_rate = rate;
|
|
RETURN_ERR( track_init( &this->track_filter, this ) );
|
|
this->tfilter.max_silence = 6 * stereo * this->sample_rate;
|
|
return 0;
|
|
}
|
|
|
|
void Sound_mute_voice( struct Vgm_Emu* this, int index, bool mute )
|
|
{
|
|
require( (unsigned) index < (unsigned) this->voice_count );
|
|
int bit = 1 << index;
|
|
int mask = this->mute_mask_ | bit;
|
|
if ( !mute )
|
|
mask ^= bit;
|
|
Sound_mute_voices( this, mask );
|
|
}
|
|
|
|
void Sound_mute_voices( struct Vgm_Emu* this, int mask )
|
|
{
|
|
require( this->sample_rate ); // sample rate must be set first
|
|
this->mute_mask_ = mask;
|
|
|
|
int i;
|
|
for ( i = this->voice_count; i--; )
|
|
{
|
|
if ( mask & (1 << i) )
|
|
{
|
|
set_voice( this, i, 0, 0, 0 );
|
|
}
|
|
else
|
|
{
|
|
struct channel_t ch = Buffer_channel( &this->stereo_buf, i );
|
|
assert( (ch.center && ch.left && ch.right) ||
|
|
(!ch.center && !ch.left && !ch.right) ); // all or nothing
|
|
set_voice( this, i, ch.center, ch.left, ch.right );
|
|
}
|
|
}
|
|
|
|
// TODO: what was this for?
|
|
//core.pcm.output( &core.blip_buf );
|
|
|
|
// TODO: silence PCM if FM isn't used?
|
|
if ( uses_fm( this ) )
|
|
{
|
|
for ( i = sms_osc_count; --i >= 0; )
|
|
Sms_apu_set_output( &this->psg, i, ( mask & 0x80 ) ? 0 : &this->blip_buf, NULL, NULL );
|
|
if ( Ym2612_enabled( &this->ym2612 ) )
|
|
{
|
|
Synth_volume( &this->pcm, (mask & 0x40) ? 0 : (int)((long long)(0.1115*FP_ONE_VOLUME) / 256 * fm_gain * this->gain / FP_ONE_VOLUME) );
|
|
Ym2612_mute_voices( &this->ym2612, mask );
|
|
}
|
|
|
|
if ( Ym2413_enabled( &this->ym2413 ) )
|
|
{
|
|
int m = mask & 0x3F;
|
|
if ( mask & 0x20 )
|
|
m |= 0x01E0; // channels 5-8
|
|
if ( mask & 0x40 )
|
|
m |= 0x3E00;
|
|
Ym2413_mute_voices( &this->ym2413, m );
|
|
}
|
|
}
|
|
}
|
|
|
|
void Sound_set_tempo( struct Vgm_Emu* this, int t )
|
|
{
|
|
require( this->sample_rate ); // sample rate must be set first
|
|
int const min = (int)(FP_ONE_TEMPO*0.02);
|
|
int const max = (int)(FP_ONE_TEMPO*4.00);
|
|
if ( t < min ) t = min;
|
|
if ( t > max ) t = max;
|
|
this->tempo = t;
|
|
|
|
if ( this->file_begin )
|
|
{
|
|
this->vgm_rate = (long) ((44100LL * t) / FP_ONE_TEMPO);
|
|
this->blip_time_factor = (int) (((1LL << blip_time_bits) * Blip_clock_rate( &this->blip_buf )) / this->vgm_rate);
|
|
//debug_printf( "blip_time_factor: %ld\n", blip_time_factor );
|
|
//debug_printf( "vgm_rate: %ld\n", vgm_rate );
|
|
// TODO: remove? calculates vgm_rate more accurately (above differs at most by one Hz only)
|
|
//blip_time_factor = (long) floor( double (1L << blip_time_bits) * psg_rate / 44100 / t + 0.5 );
|
|
//vgm_rate = (long) floor( double (1L << blip_time_bits) * psg_rate / blip_time_factor + 0.5 );
|
|
|
|
this->fm_time_factor = 2 + (int) ((this->fm_rate * (1LL << fm_time_bits)) / this->vgm_rate);
|
|
}
|
|
}
|
|
|
|
blargg_err_t Vgm_start_track( struct Vgm_Emu* this )
|
|
{
|
|
clear_track_vars( this );
|
|
|
|
Sms_apu_reset( &this->psg, get_le16( header( this )->noise_feedback ), header( this )->noise_width );
|
|
|
|
this->dac_disabled = -1;
|
|
this->pos = this->file_begin + header_size;
|
|
this->pcm_data = this->pos;
|
|
this->pcm_pos = this->pos;
|
|
this->dac_amp = -1;
|
|
this->vgm_time = 0;
|
|
if ( get_le32( header( this )->version ) >= 0x150 )
|
|
{
|
|
int data_offset = get_le32( header( this )->data_offset );
|
|
check( data_offset );
|
|
if ( data_offset )
|
|
this->pos += data_offset + offsetof (struct header_t,data_offset) - 0x40;
|
|
}
|
|
|
|
if ( uses_fm( this ) )
|
|
{
|
|
if ( Ym2413_enabled( &this->ym2413 ) )
|
|
Ym2413_reset( &this->ym2413 );
|
|
|
|
if ( Ym2612_enabled( &this->ym2612 ) )
|
|
Ym2612_reset( &this->ym2612 );
|
|
|
|
Blip_clear( &this->blip_buf );
|
|
Resampler_clear( &this->resampler );
|
|
}
|
|
|
|
this->fm_time_offset = 0;
|
|
|
|
Buffer_clear( &this->stereo_buf );
|
|
|
|
// convert filter times to samples
|
|
struct setup_t s = this->tfilter;
|
|
s.max_initial *= this->sample_rate * stereo;
|
|
#ifdef GME_DISABLE_SILENCE_LOOKAHEAD
|
|
s.lookahead = 1;
|
|
#endif
|
|
track_setup( &this->track_filter, &s );
|
|
|
|
return track_start( &this->track_filter );
|
|
}
|
|
|
|
// Tell/Seek
|
|
|
|
static int msec_to_samples( int msec, int sample_rate )
|
|
{
|
|
int sec = msec / 1000;
|
|
msec -= sec * 1000;
|
|
return (sec * sample_rate + msec * sample_rate / 1000) * stereo;
|
|
}
|
|
|
|
int Track_tell( struct Vgm_Emu* this )
|
|
{
|
|
int rate = this->sample_rate * stereo;
|
|
int sec = track_sample_count( &this->track_filter ) / rate;
|
|
return sec * 1000 + (track_sample_count( &this->track_filter ) - sec * rate) * 1000 / rate;
|
|
}
|
|
|
|
blargg_err_t Track_seek( struct Vgm_Emu* this, int msec )
|
|
{
|
|
int time = msec_to_samples( msec, this->sample_rate );
|
|
if ( time < track_sample_count( &this->track_filter ) )
|
|
RETURN_ERR( Vgm_start_track( this ) );
|
|
return Track_skip( this, time - track_sample_count( &this->track_filter ) );
|
|
}
|
|
|
|
blargg_err_t Track_skip( struct Vgm_Emu* this, int count )
|
|
{
|
|
require( this->current_track >= 0 ); // start_track() must have been called already
|
|
return track_skip( &this->track_filter, count );
|
|
}
|
|
|
|
blargg_err_t skip_( void* emu, int count )
|
|
{
|
|
struct Vgm_Emu* this = (struct Vgm_Emu*) emu;
|
|
|
|
// for long skip, mute sound
|
|
const int threshold = 32768;
|
|
if ( count > threshold )
|
|
{
|
|
int saved_mute = this->mute_mask_;
|
|
Sound_mute_voices( this, ~0 );
|
|
|
|
int n = count - threshold/2;
|
|
n &= ~(2048-1); // round to multiple of 2048
|
|
count -= n;
|
|
RETURN_ERR( skippy_( &this->track_filter, n ) );
|
|
|
|
Sound_mute_voices( this, saved_mute );
|
|
}
|
|
|
|
return skippy_( &this->track_filter, count );
|
|
}
|
|
|
|
// Fading
|
|
|
|
void Track_set_fade( struct Vgm_Emu* this, int start_msec, int length_msec )
|
|
{
|
|
track_set_fade( &this->track_filter, msec_to_samples( start_msec, this->sample_rate ),
|
|
length_msec * this->sample_rate / (1000 / stereo) );
|
|
}
|
|
|
|
blargg_err_t Vgm_play( struct Vgm_Emu* this, int out_count, sample_t* out )
|
|
{
|
|
require( this->current_track >= 0 );
|
|
require( out_count % stereo == 0 );
|
|
return track_play( &this->track_filter, out_count, out );
|
|
}
|