BCSTM Player

source/music.c

@@ -27,8 +27,31 @@
 #include "music.h"
 #include "loading.h"
 
-// Play a given audio struct
-Result update_audio(audio_s * audio) 
+u32 read32(char *music_buf, ssize_t *cursor)
+{
+    u32 ret;
+    memcpy(&ret, music_buf + *cursor, 4);
+    *cursor += 4;
+    return ret;
+}
+
+u16 read16(char *music_buf, ssize_t *cursor)
+{
+    u16 ret;
+    memcpy(&ret, music_buf + *cursor, 2);
+    *cursor += 2;
+    return ret;
+}
+
+u8 read8(char *music_buf, ssize_t *cursor)
+{
+    u8 ret;
+    memcpy(&ret, music_buf + *cursor, 1);
+    *cursor += 1;
+    return ret;
+}
+
+Result update_audio_ogg(audio_ogg_s * audio) 
 {
     u32 size = audio->wave_buf[audio->buf_pos].nsamples * 4 - audio->data_read;
     DEBUG("<update_audio> Audio Size: %ld\n", size);
@@ -64,10 +87,10 @@ Result update_audio(audio_s * audio)
     return MAKERESULT(RL_SUCCESS, RS_SUCCESS, RM_APPLICATION, RD_SUCCESS);
 }
 
-void thread_audio(void * data) {
-    audio_s * audio = (audio_s *)data;
+void thread_audio_ogg(void * data) {
+    audio_ogg_s * audio = (audio_ogg_s *)data;
     while(!audio->stop) {
-        update_audio(audio);
+        update_audio_ogg(audio);
     }
     ndspChnWaveBufClear(0);
     ndspChnReset(0);
@@ -77,6 +100,245 @@ void thread_audio(void * data) {
     linearFree((void *)audio->wave_buf[1].data_vaddr);
 }
 
+void play_audio_ogg(audio_ogg_s * audio) {
+    audio->playing_thread = threadCreate(thread_audio_ogg, audio, 0x1000, 0x3F, 1, false);
+}
+
+void stop_audio_ogg(audio_ogg_s ** audio_ptr) {
+    audio_ogg_s * audio = *audio_ptr;
+    if(audio->playing_thread)
+    {
+        audio->stop = true;
+        threadJoin(audio->playing_thread, U64_MAX);
+        threadFree(audio->playing_thread);
+    }
+    free(audio);
+    *audio_ptr = NULL;
+}
+
+int init_audio(audio_s *audio)
+{
+    u32 magic = read32(audio->music_buf, &audio->cursor);
+    DEBUG("Loading music, music_size: %lu, magic: 0x%08lx\n", audio->music_size, magic);
+    audio->is_little_endian = read16(audio->music_buf, &audio->cursor) == 0xFEFF;
+    audio->info_offset = 0;
+    audio->data_offset = 0;
+
+    if (magic != 0x4D545343) // CSTM
+    {
+        free(audio->music_buf);
+        return -1;
+    }
+
+    audio->cursor = 0x10;
+    u16 sbc = read16(audio->music_buf, &audio->cursor);
+    audio->cursor += 2;
+
+    for (u16 i = 0; i < sbc; ++i)
+    {
+        u16 sec = read16(audio->music_buf, &audio->cursor);
+        audio->cursor += 2;
+        u32 off = read32(audio->music_buf, &audio->cursor);
+        audio->cursor += 4;
+        DEBUG("Reading sbc: %04lx, %08lx\n", sec, off);
+        if (sec == 0x4000) // Info block
+            audio->info_offset = off;
+        if (sec == 0x4002) // Data block
+            audio->data_offset = off;
+    }
+
+    DEBUG("Info offset: 0x%08lx, data offset: 0x%08lx\n", audio->info_offset, audio->data_offset);
+    
+    if (audio->data_offset == 0 || audio->info_offset == 0)
+    {
+        free(audio->music_buf);
+        return -2;
+    }
+
+    audio->cursor = audio->info_offset + 0x20;
+
+    if (read8(audio->music_buf, &audio->cursor) != 2) // Encoding - 2 is DSP_ADPCM
+    {
+        free(audio->music_buf);
+        return -3;
+    }
+
+    audio->is_looping = read8(audio->music_buf, &audio->cursor);
+    audio->channel_count = read8(audio->music_buf, &audio->cursor);
+
+    if (audio->channel_count > 2)
+    {
+        free(audio->music_buf);
+        return -4;
+    }
+
+    DEBUG("Channel count: %d, is looping: %d\n", audio->channel_count, audio->is_looping);
+    
+    audio->cursor = audio->info_offset + 0x24;
+    audio->sample_rate = read32(audio->music_buf, &audio->cursor);
+    u32 _loop_pos = read32(audio->music_buf, &audio->cursor);
+    u32 _loop_end = read32(audio->music_buf, &audio->cursor);
+    audio->num_blocks = read32(audio->music_buf, &audio->cursor);
+    audio->block_size = read32(audio->music_buf, &audio->cursor);
+    audio->block_samples = read32(audio->music_buf, &audio->cursor);
+    audio->cursor += 4;
+    audio->last_block_samples = read32(audio->music_buf, &audio->cursor);
+    audio->last_block_size = read32(audio->music_buf, &audio->cursor);
+
+    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", 
+        audio->sample_rate, _loop_pos, _loop_end, audio->num_blocks, audio->block_size, audio->block_samples, audio->last_block_samples, audio->last_block_size);
+
+    audio->loop_start = _loop_pos / audio->block_samples;
+    audio->loop_end = (_loop_end % audio->block_samples ? audio->num_blocks : _loop_end / audio->block_samples);
+
+    while (read32(audio->music_buf, &audio->cursor) != 0x4102); // find channel info header
+    audio->cursor += read32(audio->music_buf, &audio->cursor) + audio->channel_count * 8 - 12;
+
+    for (u8 i = 0; i < audio->channel_count; ++i)
+    {
+        memcpy(audio->adpcm_coefs[i], audio->music_buf + audio->cursor, sizeof(unsigned short) * 16);
+        audio->cursor += sizeof(unsigned short) * 16;
+        memcpy(&(audio->adpcm_data[i][0]), audio->music_buf + audio->cursor, sizeof(ndspAdpcmData));
+        audio->cursor += sizeof(ndspAdpcmData);
+        memcpy(&(audio->adpcm_data[i][1]), audio->music_buf + audio->cursor, sizeof(ndspAdpcmData));
+        audio->cursor += sizeof(ndspAdpcmData);
+        audio->cursor += 2; // skip padding
+    }
+
+    audio->cursor = audio->data_offset + 0x20;
+
+    return 0;
+}
+
+int start_play(audio_s *audio) {
+    audio->current_block = 0;
+    for (u8 i = 0; i < audio->channel_count; ++i)
+    {
+        audio->channel[i] = 0;
+        while (audio->channel[i] < 24 && ((audio->active_channels >> audio->channel[i]) & 1)) {
+            audio->channel[i]++;
+        }
+        if (audio->channel[i] == 24) {
+            return -1;
+        }
+        audio->active_channels |= 1 << audio->channel[i];
+        ndspChnWaveBufClear(audio->channel[i]);
+        
+        static float mix[16];
+        ndspChnSetFormat(audio->channel[i], NDSP_FORMAT_ADPCM | NDSP_3D_SURROUND_PREPROCESSED);
+        ndspChnSetRate(audio->channel[i], audio->sample_rate);
+        
+        if (audio->channel_count == 1)
+        {
+            mix[0] = mix[1] = 0.5f;
+        } else if (audio->channel_count == 2)
+        {
+            if (i == 0)
+            {
+                mix[0] = 0.8f;
+                mix[1] = 0.0f;
+                mix[2] = 0.2f;
+                mix[3] = 0.0f;
+            } else
+            {
+                mix[0] = 0.0f;
+                mix[1] = 0.8f;
+                mix[2] = 0.0f;
+                mix[3] = 0.2f;
+            }
+        }
+        ndspChnSetMix(audio->channel[i], mix);
+        ndspChnSetAdpcmCoefs(audio->channel[i], audio->adpcm_coefs[i]);
+
+        for (u8 j = 0; j < BUFFER_COUNT; ++j)
+        {
+            memset(&(audio->wave_buf[i][j]), 0, sizeof(ndspWaveBuf));
+            audio->wave_buf[i][j].status = NDSP_WBUF_DONE;
+            audio->buffer_data[i][j] = linearAlloc(audio->block_size);
+        }
+    }
+
+    return 0;
+}
+
+void fill_buffers(audio_s *audio)
+{
+    DEBUG("Filling buffers...\n");
+    for (u8 bufIndex = 0; bufIndex < BUFFER_COUNT; ++bufIndex)
+    {
+        if (audio->wave_buf[0][bufIndex].status != NDSP_WBUF_DONE) continue;
+        if (audio->channel_count == 2 && audio->wave_buf[1][bufIndex].status != NDSP_WBUF_DONE) continue;
+
+        if (audio->is_looping && audio->current_block == audio->loop_end)
+        {
+            audio->current_block = audio->loop_start;
+            audio->cursor = audio->data_offset + 0x20 + audio->block_size * audio->channel_count * audio->loop_start;
+        }
+
+        if (!audio->is_looping && audio->current_block == audio->loop_end)
+        {
+            // stop playing
+        }
+
+        for (u8 channelIndex = 0; channelIndex < audio->channel_count; ++channelIndex)
+        {
+            ndspWaveBuf *buf = &(audio->wave_buf[channelIndex][bufIndex]);
+            memset(buf, 0, sizeof(ndspWaveBuf));
+            buf->data_adpcm = audio->buffer_data[channelIndex][bufIndex];
+            memcpy(buf->data_adpcm, audio->music_buf + audio->cursor, (audio->current_block == audio->num_blocks - 1) ? audio->last_block_size : audio->block_size);
+            audio->cursor += (audio->current_block == audio->num_blocks - 1) ? audio->last_block_size : audio->block_size;
+            DSP_FlushDataCache(buf->data_adpcm, audio->block_size);
+
+            if (audio->current_block == 0)
+                buf->adpcm_data = &(audio->adpcm_data[channelIndex][0]);
+            else if (audio->current_block == audio->loop_start)
+                buf->adpcm_data = &(audio->adpcm_data[channelIndex][1]);
+
+            if (audio->current_block == audio->num_blocks - 1)
+                buf->nsamples = audio->last_block_samples;
+            else
+                buf->nsamples = audio->block_samples;
+
+            ndspChnWaveBufAdd(audio->channel[channelIndex], buf);
+        }
+        audio->current_block++;
+    }
+}
+
+// Play a given audio struct
+void update_audio(audio_s * audio) 
+{
+    u32 current_time = svcGetSystemTick();
+    if (current_time - audio->last_time > 1e8)
+    {
+        fill_buffers(audio);
+        audio->last_time = current_time;
+    }
+}
+
+void thread_audio(void * data) {
+    audio_s * audio = (audio_s *)data;
+    int res = init_audio(audio);
+    if (res < 0)
+    {
+        return;
+    }
+    start_play(audio);
+    while(!audio->stop) {
+        update_audio(audio);
+    }
+    for (u8 i = 0; i < audio->channel_count; ++i)
+    {
+        ndspChnWaveBufClear(audio->channel[i]);
+        ndspChnReset(audio->channel[i]);
+        audio->active_channels &= ~(1 << audio->channel[i]);
+        for (u8 j = 0; j < BUFFER_COUNT; ++j) {
+            linearFree(audio->buffer_data[i][j]);
+        }
+    }
+    free(audio->music_buf);
+}
+
 void play_audio(audio_s * audio) {
     audio->playing_thread = threadCreate(thread_audio, audio, 0x1000, 0x3F, 1, false);
 }