4 * Audio mixer core. Combines all the active streams into a single sound
7 * Copyright (c) 2005 Nathan Keynes.
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
20 #include "aica/aica.h"
21 #include "aica/audio.h"
22 #include <glib/gmem.h>
28 extern struct audio_driver audio_null_driver;
29 extern struct audio_driver audio_osx_driver;
30 extern struct audio_driver audio_pulse_driver;
31 extern struct audio_driver audio_esd_driver;
32 extern struct audio_driver audio_alsa_driver;
34 audio_driver_t audio_driver_list[] = {
35 #ifdef HAVE_CORE_AUDIO
51 #define MS_PER_BUFFER 100
53 #define BUFFER_EMPTY 0
54 #define BUFFER_WRITING 1
58 audio_buffer_t output_buffers[NUM_BUFFERS];
61 uint32_t output_format;
63 uint32_t output_sample_size;
64 struct audio_channel channels[AUDIO_CHANNEL_COUNT];
67 audio_driver_t audio_driver = NULL;
69 #define NEXT_BUFFER() ((audio.write_buffer == NUM_BUFFERS-1) ? 0 : audio.write_buffer+1)
72 * Preserve audio channel state only - don't bother saving the buffers
74 void audio_save_state( FILE *f )
76 fwrite( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
79 int audio_load_state( FILE *f )
81 int read = fread( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
82 return (read == AUDIO_CHANNEL_COUNT ? 0 : -1 );
85 audio_driver_t get_audio_driver_by_name( const char *name )
89 return audio_driver_list[0];
91 for( i=0; audio_driver_list[i] != NULL; i++ ) {
92 if( strcasecmp( audio_driver_list[i]->name, name ) == 0 ) {
93 return audio_driver_list[i];
100 void print_audio_drivers( FILE * out )
103 fprintf( out, "Available audio drivers:\n" );
104 for( i=0; audio_driver_list[i] != NULL; i++ ) {
105 fprintf( out, " %-8s %s\n", audio_driver_list[i]->name,
106 gettext(audio_driver_list[i]->description) );
110 audio_driver_t audio_init_driver( const char *preferred_driver )
112 audio_driver_t audio_driver = get_audio_driver_by_name(preferred_driver);
113 if( audio_driver == NULL ) {
114 ERROR( "Audio driver '%s' not found, aborting.", preferred_driver );
116 } else if( audio_set_driver( audio_driver ) == FALSE ) {
118 for( i=0; audio_driver_list[i] != NULL; i++ ) {
119 if( audio_driver_list[i] != audio_driver &&
120 audio_set_driver( audio_driver_list[i] ) ) {
121 ERROR( "Failed to initialize audio driver %s, falling back to %s",
122 audio_driver->name, audio_driver_list[i]->name );
123 return audio_driver_list[i];
126 ERROR( "Unable to intialize any audio driver, aborting." );
133 * Set the output driver, sample rate and format. Also initializes the
134 * output buffers, flushing any current data and reallocating as
137 gboolean audio_set_driver( audio_driver_t driver )
139 uint32_t bytes_per_sample = 1;
140 uint32_t samples_per_buffer;
143 if( audio_driver == NULL || driver != NULL ) {
145 driver = &audio_null_driver;
146 if( driver != audio_driver ) {
147 if( !driver->init() )
149 audio_driver = driver;
153 switch( driver->sample_format & AUDIO_FMT_SAMPLE_MASK ) {
155 bytes_per_sample = 1;
157 case AUDIO_FMT_16BIT:
158 bytes_per_sample = 2;
160 case AUDIO_FMT_FLOAT:
161 bytes_per_sample = 4;
165 if( driver->sample_format & AUDIO_FMT_STEREO )
166 bytes_per_sample <<= 1;
167 if( driver->sample_rate == audio.output_rate &&
168 bytes_per_sample == audio.output_sample_size )
170 samples_per_buffer = (driver->sample_rate * MS_PER_BUFFER / 1000);
171 for( i=0; i<NUM_BUFFERS; i++ ) {
172 if( audio.output_buffers[i] != NULL )
173 free(audio.output_buffers[i]);
174 audio.output_buffers[i] = g_malloc0( sizeof(struct audio_buffer) + samples_per_buffer * bytes_per_sample );
175 audio.output_buffers[i]->length = samples_per_buffer * bytes_per_sample;
176 audio.output_buffers[i]->posn = 0;
177 audio.output_buffers[i]->status = BUFFER_EMPTY;
179 audio.output_format = driver->sample_format;
180 audio.output_rate = driver->sample_rate;
181 audio.output_sample_size = bytes_per_sample;
182 audio.write_buffer = 0;
183 audio.read_buffer = 0;
189 * Mark the current write buffer as full and prepare the next buffer for
190 * writing. Returns the next buffer to write to.
191 * If all buffers are full, returns NULL.
193 audio_buffer_t audio_next_write_buffer( )
195 audio_buffer_t result = NULL;
196 audio_buffer_t current = audio.output_buffers[audio.write_buffer];
197 current->status = BUFFER_FULL;
198 if( audio.read_buffer == audio.write_buffer &&
199 audio_driver->process_buffer( current ) ) {
200 audio_next_read_buffer();
202 int next_buffer = NEXT_BUFFER();
203 result = audio.output_buffers[next_buffer];
204 if( result->status == BUFFER_FULL )
207 audio.write_buffer = next_buffer;
208 result->status = BUFFER_WRITING;
214 * Mark the current read buffer as empty and return the next buffer for
215 * reading. If there is no next buffer yet, returns NULL.
217 audio_buffer_t audio_next_read_buffer( )
219 audio_buffer_t current = audio.output_buffers[audio.read_buffer];
220 if( current->status == BUFFER_FULL ) {
221 // Current read buffer has data, which we've just emptied
222 current->status = BUFFER_EMPTY;
225 if( audio.read_buffer == NUM_BUFFERS )
226 audio.read_buffer = 0;
228 current = audio.output_buffers[audio.read_buffer];
229 if( current->status == BUFFER_FULL ) {
240 /*************************** ADPCM ***********************************/
243 * The following section borrows heavily from ffmpeg, which is
244 * copyright (c) 2001-2003 by the fine folks at the ffmpeg project,
245 * distributed under the GPL version 2 or later.
248 #define CLAMP_TO_SHORT(value) \
251 else if (value < -32768) \
254 static const int yamaha_indexscale[] = {
255 230, 230, 230, 230, 307, 409, 512, 614,
256 230, 230, 230, 230, 307, 409, 512, 614
259 static const int yamaha_difflookup[] = {
260 1, 3, 5, 7, 9, 11, 13, 15,
261 -1, -3, -5, -7, -9, -11, -13, -15
264 static inline short adpcm_yamaha_decode_nibble( audio_channel_t c,
265 unsigned char nibble )
267 if( c->adpcm_step == 0 ) {
268 c->adpcm_predict = 0;
272 c->adpcm_predict += (c->adpcm_step * yamaha_difflookup[nibble]) >> 3;
273 CLAMP_TO_SHORT(c->adpcm_predict);
274 c->adpcm_step = (c->adpcm_step * yamaha_indexscale[nibble]) >> 8;
275 c->adpcm_step = CLAMP(c->adpcm_step, 127, 24567);
276 return c->adpcm_predict;
279 /*************************** Sample mixer *****************************/
282 * Mix a single output sample.
284 void audio_mix_samples( int num_samples )
287 int32_t result_buf[num_samples][2];
289 memset( &result_buf, 0, sizeof(result_buf) );
291 for( i=0; i < AUDIO_CHANNEL_COUNT; i++ ) {
292 audio_channel_t channel = &audio.channels[i];
293 if( channel->active ) {
295 int vol_left = (channel->vol * (32 - channel->pan)) >> 5;
296 int vol_right = (channel->vol * (channel->pan + 1)) >> 5;
297 switch( channel->sample_format ) {
298 case AUDIO_FMT_16BIT:
299 for( j=0; j<num_samples; j++ ) {
300 sample = ((int16_t *)(aica_main_ram + channel->start))[channel->posn];
301 result_buf[j][0] += sample * vol_left;
302 result_buf[j][1] += sample * vol_right;
304 channel->posn_left += channel->sample_rate;
305 while( channel->posn_left > audio.output_rate ) {
306 channel->posn_left -= audio.output_rate;
309 if( channel->posn == channel->end ) {
310 if( channel->loop ) {
311 channel->posn = channel->loop_start;
312 channel->loop = LOOP_LOOPED;
314 audio_stop_channel(i);
323 for( j=0; j<num_samples; j++ ) {
324 sample = ((int8_t *)(aica_main_ram + channel->start))[channel->posn] << 8;
325 result_buf[j][0] += sample * vol_left;
326 result_buf[j][1] += sample * vol_right;
328 channel->posn_left += channel->sample_rate;
329 while( channel->posn_left > audio.output_rate ) {
330 channel->posn_left -= audio.output_rate;
333 if( channel->posn == channel->end ) {
334 if( channel->loop ) {
335 channel->posn = channel->loop_start;
336 channel->loop = LOOP_LOOPED;
338 audio_stop_channel(i);
346 case AUDIO_FMT_ADPCM:
347 for( j=0; j<num_samples; j++ ) {
348 sample = (int16_t)channel->adpcm_predict;
349 result_buf[j][0] += sample * vol_left;
350 result_buf[j][1] += sample * vol_right;
351 channel->posn_left += channel->sample_rate;
352 while( channel->posn_left > audio.output_rate ) {
353 channel->posn_left -= audio.output_rate;
355 if( channel->posn == channel->end ) {
356 if( channel->loop ) {
357 channel->posn = channel->loop_start;
358 channel->loop = LOOP_LOOPED;
359 channel->adpcm_predict = 0;
360 channel->adpcm_step = 0;
362 audio_stop_channel(i);
367 uint8_t data = ((uint8_t *)(aica_main_ram + channel->start))[channel->posn>>1];
368 if( channel->posn&1 ) {
369 adpcm_yamaha_decode_nibble( channel, (data >> 4) & 0x0F );
371 adpcm_yamaha_decode_nibble( channel, data & 0x0F );
382 /* Down-render to the final output format */
383 audio_buffer_t buf = audio.output_buffers[audio.write_buffer];
384 if( buf->status == BUFFER_FULL ) {
385 buf = audio_next_write_buffer();
386 if( buf == NULL ) { // no available space
391 switch( audio.output_format & AUDIO_FMT_SAMPLE_MASK ) {
392 case AUDIO_FMT_FLOAT: {
393 float scale = 1.0/SHRT_MAX;
394 float *data = (float *)&buf->data[buf->posn];
395 for( j=0; j<num_samples; j++ ) {
396 *data++ = scale * (result_buf[j][0] >> 6);
397 *data++ = scale * (result_buf[j][1] >> 6);
399 if( buf->posn == buf->length ) {
400 buf = audio_next_write_buffer();
404 data = (float *)&buf->data[0];
409 case AUDIO_FMT_16BIT: {
410 int16_t *data = (int16_t *)&buf->data[buf->posn];
411 for( j=0; j < num_samples; j++ ) {
412 *data++ = (int16_t)(result_buf[j][0] >> 6);
413 *data++ = (int16_t)(result_buf[j][1] >> 6);
415 if( buf->posn == buf->length ) {
416 buf = audio_next_write_buffer();
418 // All buffers are full
421 data = (int16_t *)&buf->data[0];
426 case AUDIO_FMT_8BIT: {
427 int8_t *data = (int8_t *)&buf->data[buf->posn];
428 for( j=0; j < num_samples; j++ ) {
429 *data++ = (int8_t)(result_buf[j][0] >> 16);
430 *data++ = (int8_t)(result_buf[j][1] >> 16);
432 if( buf->posn == buf->length ) {
433 buf = audio_next_write_buffer();
435 // All buffers are full
438 buf = audio.output_buffers[audio.write_buffer];
439 data = (int8_t *)&buf->data[0];
447 /********************** Internal AICA calls ***************************/
449 audio_channel_t audio_get_channel( int channel )
451 return &audio.channels[channel];
454 void audio_start_stop_channel( int channel, gboolean start )
456 if( audio.channels[channel].active ) {
458 audio_stop_channel(channel);
461 audio_start_channel(channel);
465 void audio_stop_channel( int channel )
467 audio.channels[channel].active = FALSE;
471 void audio_start_channel( int channel )
473 audio.channels[channel].posn = 0;
474 audio.channels[channel].posn_left = 0;
475 audio.channels[channel].active = TRUE;
476 if( audio.channels[channel].sample_format == AUDIO_FMT_ADPCM ) {
477 audio.channels[channel].adpcm_step = 0;
478 audio.channels[channel].adpcm_predict = 0;
479 uint8_t data = ((uint8_t *)(aica_main_ram + audio.channels[channel].start))[0];
480 adpcm_yamaha_decode_nibble( &audio.channels[channel], data & 0x0F );
.