360 lines
12 KiB
C
360 lines
12 KiB
C
/*
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* This file is part of Anemone3DS
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* Copyright (C) 2016-2020 Contributors in CONTRIBUTORS.md
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*
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* Additional Terms 7.b and 7.c of GPLv3 apply to this file:
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* * Requiring preservation of specified reasonable legal notices or
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* author attributions in that material or in the Appropriate Legal
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* Notices displayed by works containing it.
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* * Prohibiting misrepresentation of the origin of that material,
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* or requiring that modified versions of such material be marked in
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* reasonable ways as different from the original version.
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*/
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#include "music.h"
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#include "loading.h"
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// BCSTM Player adapted from BCSTM-Player by tobid7
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// https://github.com/NPI-D7/BCSTM-Player/blob/main/source/bcstm.hpp
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u32 read32(char *music_buf, ssize_t *cursor)
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{
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u32 ret;
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memcpy(&ret, music_buf + *cursor, 4);
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*cursor += 4;
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return ret;
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}
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u16 read16(char *music_buf, ssize_t *cursor)
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{
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u16 ret;
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memcpy(&ret, music_buf + *cursor, 2);
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*cursor += 2;
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return ret;
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}
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u8 read8(char *music_buf, ssize_t *cursor)
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{
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u8 ret;
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memcpy(&ret, music_buf + *cursor, 1);
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*cursor += 1;
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return ret;
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}
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Result update_audio_ogg(audio_ogg_s * audio)
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{
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u32 size = audio->wave_buf[audio->buf_pos].nsamples * 4 - audio->data_read;
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DEBUG("<update_audio> Audio Size: %ld\n", size);
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if (audio->wave_buf[audio->buf_pos].status == NDSP_WBUF_DONE) // only run if the current selected buffer has already finished playing
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{
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DEBUG("<update_audio> Attempting ov_read\n");
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int bitstream;
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u32 read = ov_read(&audio->vf, (char *)audio->wave_buf[audio->buf_pos].data_vaddr + audio->data_read, size, &bitstream); // read 1 vorbis packet into wave buffer
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DEBUG("<update_audio> ov_read successful\n");
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if (read <= 0) // EoF or error
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{
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ov_clear(&audio->vf);
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if (read == 0) // EoF
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{
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ov_open(fmemopen(audio->filebuf, audio->filesize, "rb"), &audio->vf, NULL, 0); // Reopen file. Don't need to reinit channel stuff since it's all the same as before
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} else // Error :(
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{
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DEBUG("<update_audio> Vorbis play error: %ld\n", read);
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ndspChnReset(0);
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return MAKERESULT(RL_FATAL, RS_INVALIDARG, RM_APPLICATION, RD_NO_DATA);
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}
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} else
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{
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audio->data_read += read;
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if (read == size) {
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audio->data_read = 0;
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ndspChnWaveBufAdd(0, &audio->wave_buf[audio->buf_pos]); // Add buffer to ndsp
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audio->buf_pos = 1 - audio->buf_pos; // switch to other buffer to load and prepare it while the current one is playing
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}
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}
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}
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return MAKERESULT(RL_SUCCESS, RS_SUCCESS, RM_APPLICATION, RD_SUCCESS);
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}
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void thread_audio_ogg(void * data) {
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audio_ogg_s * audio = (audio_ogg_s *)data;
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while(!audio->stop) {
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update_audio_ogg(audio);
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}
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ndspChnWaveBufClear(0);
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ndspChnReset(0);
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ov_clear(&audio->vf);
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free(audio->filebuf);
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linearFree((void *)audio->wave_buf[0].data_vaddr);
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linearFree((void *)audio->wave_buf[1].data_vaddr);
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}
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void play_audio_ogg(audio_ogg_s * audio) {
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audio->playing_thread = threadCreate(thread_audio_ogg, audio, 0x1000, 0x3F, 1, false);
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}
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void stop_audio_ogg(audio_ogg_s ** audio_ptr) {
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audio_ogg_s * audio = *audio_ptr;
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if(audio->playing_thread)
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{
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audio->stop = true;
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threadJoin(audio->playing_thread, U64_MAX);
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threadFree(audio->playing_thread);
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}
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free(audio);
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*audio_ptr = NULL;
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}
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int init_audio(audio_s *audio)
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{
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u32 magic = read32(audio->music_buf, &audio->cursor);
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DEBUG("Loading music, music_size: %lu, magic: 0x%08lx\n", audio->music_size, magic);
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audio->is_little_endian = read16(audio->music_buf, &audio->cursor) == 0xFEFF;
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audio->info_offset = 0;
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audio->data_offset = 0;
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if (magic != 0x4D545343) // CSTM
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{
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free(audio->music_buf);
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return -1;
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}
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audio->cursor = 0x10;
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u16 sbc = read16(audio->music_buf, &audio->cursor);
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audio->cursor += 2;
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for (u16 i = 0; i < sbc; ++i)
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{
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u16 sec = read16(audio->music_buf, &audio->cursor);
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audio->cursor += 2;
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u32 off = read32(audio->music_buf, &audio->cursor);
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audio->cursor += 4;
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DEBUG("Reading sbc: %04lx, %08lx\n", sec, off);
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if (sec == 0x4000) // Info block
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audio->info_offset = off;
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if (sec == 0x4002) // Data block
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audio->data_offset = off;
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}
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DEBUG("Info offset: 0x%08lx, data offset: 0x%08lx\n", audio->info_offset, audio->data_offset);
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if (audio->data_offset == 0 || audio->info_offset == 0)
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{
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free(audio->music_buf);
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return -2;
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}
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audio->cursor = audio->info_offset + 0x20;
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if (read8(audio->music_buf, &audio->cursor) != 2) // Encoding - 2 is DSP_ADPCM
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{
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free(audio->music_buf);
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return -3;
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}
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audio->is_looping = read8(audio->music_buf, &audio->cursor);
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audio->channel_count = read8(audio->music_buf, &audio->cursor);
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if (audio->channel_count > 2)
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{
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free(audio->music_buf);
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return -4;
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}
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DEBUG("Channel count: %d, is looping: %d\n", audio->channel_count, audio->is_looping);
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audio->cursor = audio->info_offset + 0x24;
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audio->sample_rate = read32(audio->music_buf, &audio->cursor);
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u32 _loop_pos = read32(audio->music_buf, &audio->cursor);
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u32 _loop_end = read32(audio->music_buf, &audio->cursor);
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audio->num_blocks = read32(audio->music_buf, &audio->cursor);
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audio->block_size = read32(audio->music_buf, &audio->cursor);
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audio->block_samples = read32(audio->music_buf, &audio->cursor);
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audio->cursor += 4;
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audio->last_block_samples = read32(audio->music_buf, &audio->cursor);
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audio->last_block_size = read32(audio->music_buf, &audio->cursor);
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DEBUG("sample_rate: %lu, loop_start: %lu, loop_end: %lu, num_blocks: %lu, block_size: %lu, block_samples: %lu, last_block_samples: %lu, last_block_size: %lu\n",
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audio->sample_rate, _loop_pos, _loop_end, audio->num_blocks, audio->block_size, audio->block_samples, audio->last_block_samples, audio->last_block_size);
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audio->loop_start = _loop_pos / audio->block_samples;
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audio->loop_end = (_loop_end % audio->block_samples ? audio->num_blocks : _loop_end / audio->block_samples);
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while (read32(audio->music_buf, &audio->cursor) != 0x4102); // find channel info header
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audio->cursor += read32(audio->music_buf, &audio->cursor) + audio->channel_count * 8 - 12;
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for (u8 i = 0; i < audio->channel_count; ++i)
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{
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memcpy(audio->adpcm_coefs[i], audio->music_buf + audio->cursor, sizeof(unsigned short) * 16);
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audio->cursor += sizeof(unsigned short) * 16;
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memcpy(&(audio->adpcm_data[i][0]), audio->music_buf + audio->cursor, sizeof(ndspAdpcmData));
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audio->cursor += sizeof(ndspAdpcmData);
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memcpy(&(audio->adpcm_data[i][1]), audio->music_buf + audio->cursor, sizeof(ndspAdpcmData));
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audio->cursor += sizeof(ndspAdpcmData);
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audio->cursor += 2; // skip padding
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}
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audio->cursor = audio->data_offset + 0x20;
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return 0;
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}
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int start_play(audio_s *audio) {
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audio->current_block = 0;
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for (u8 i = 0; i < audio->channel_count; ++i)
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{
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audio->channel[i] = 0;
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while (audio->channel[i] < 24 && ((audio->active_channels >> audio->channel[i]) & 1)) {
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audio->channel[i]++;
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}
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if (audio->channel[i] == 24) {
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return -1;
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}
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audio->active_channels |= 1 << audio->channel[i];
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ndspChnWaveBufClear(audio->channel[i]);
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static float mix[16];
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ndspChnSetFormat(audio->channel[i], NDSP_FORMAT_ADPCM | NDSP_3D_SURROUND_PREPROCESSED);
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ndspChnSetRate(audio->channel[i], audio->sample_rate);
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if (audio->channel_count == 1)
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{
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mix[0] = mix[1] = 0.5f;
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} else if (audio->channel_count == 2)
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{
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if (i == 0)
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{
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mix[0] = 0.8f;
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mix[1] = 0.0f;
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mix[2] = 0.2f;
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mix[3] = 0.0f;
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} else
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{
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mix[0] = 0.0f;
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mix[1] = 0.8f;
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mix[2] = 0.0f;
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mix[3] = 0.2f;
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}
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}
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ndspChnSetMix(audio->channel[i], mix);
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ndspChnSetAdpcmCoefs(audio->channel[i], audio->adpcm_coefs[i]);
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for (u8 j = 0; j < BUFFER_COUNT; ++j)
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{
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memset(&(audio->wave_buf[i][j]), 0, sizeof(ndspWaveBuf));
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audio->wave_buf[i][j].status = NDSP_WBUF_DONE;
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audio->buffer_data[i][j] = linearAlloc(audio->block_size);
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}
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}
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return 0;
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}
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void fill_buffers(audio_s *audio)
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{
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DEBUG("Filling buffers...\n");
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for (u8 bufIndex = 0; bufIndex < BUFFER_COUNT; ++bufIndex)
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{
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if (audio->wave_buf[0][bufIndex].status != NDSP_WBUF_DONE) continue;
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if (audio->channel_count == 2 && audio->wave_buf[1][bufIndex].status != NDSP_WBUF_DONE) continue;
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if (audio->is_looping && audio->current_block == audio->loop_end)
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{
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audio->current_block = audio->loop_start;
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audio->cursor = audio->data_offset + 0x20 + audio->block_size * audio->channel_count * audio->loop_start;
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}
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if (!audio->is_looping && audio->current_block == audio->loop_end)
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{
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// stop playing
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}
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for (u8 channelIndex = 0; channelIndex < audio->channel_count; ++channelIndex)
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{
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ndspWaveBuf *buf = &(audio->wave_buf[channelIndex][bufIndex]);
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memset(buf, 0, sizeof(ndspWaveBuf));
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buf->data_adpcm = audio->buffer_data[channelIndex][bufIndex];
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memcpy(buf->data_adpcm, audio->music_buf + audio->cursor, (audio->current_block == audio->num_blocks - 1) ? audio->last_block_size : audio->block_size);
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audio->cursor += (audio->current_block == audio->num_blocks - 1) ? audio->last_block_size : audio->block_size;
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DSP_FlushDataCache(buf->data_adpcm, audio->block_size);
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if (audio->current_block == 0)
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buf->adpcm_data = &(audio->adpcm_data[channelIndex][0]);
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else if (audio->current_block == audio->loop_start)
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buf->adpcm_data = &(audio->adpcm_data[channelIndex][1]);
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if (audio->current_block == audio->num_blocks - 1)
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buf->nsamples = audio->last_block_samples;
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else
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buf->nsamples = audio->block_samples;
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ndspChnWaveBufAdd(audio->channel[channelIndex], buf);
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}
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audio->current_block++;
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}
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}
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// Play a given audio struct
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void update_audio(audio_s * audio)
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{
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u32 current_time = svcGetSystemTick();
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if (current_time - audio->last_time > 1e8)
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{
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fill_buffers(audio);
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audio->last_time = current_time;
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}
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}
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void thread_audio(void * data) {
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audio_s * audio = (audio_s *)data;
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int res = init_audio(audio);
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if (res < 0)
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{
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return;
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}
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start_play(audio);
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while(!audio->stop) {
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update_audio(audio);
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}
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for (u8 i = 0; i < audio->channel_count; ++i)
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{
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ndspChnWaveBufClear(audio->channel[i]);
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ndspChnReset(audio->channel[i]);
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audio->active_channels &= ~(1 << audio->channel[i]);
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for (u8 j = 0; j < BUFFER_COUNT; ++j) {
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linearFree(audio->buffer_data[i][j]);
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}
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}
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free(audio->music_buf);
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}
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void play_audio(audio_s * audio) {
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audio->playing_thread = threadCreate(thread_audio, audio, 0x1000, 0x3F, 1, false);
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}
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void stop_audio(audio_s ** audio_ptr) {
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audio_s * audio = *audio_ptr;
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if(audio->playing_thread)
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{
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audio->stop = true;
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threadJoin(audio->playing_thread, U64_MAX);
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threadFree(audio->playing_thread);
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}
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free(audio);
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*audio_ptr = NULL;
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}
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