dm: sandbox: Update sound to use two buffers

At present we use a single buffer for sound which means we cannot be
playing one sound while queueing up the next. This wouldn't matter except
that a long sound (more than a second) has to be created as a single
buffer, thus using a lot of memory. To better mimic what real sound
drivers do, add support for double buffering in sandbox.

Signed-off-by: Simon Glass <sjg@chromium.org>
This commit is contained in:
Simon Glass 2018-12-10 10:37:35 -07:00
parent e96fa6c911
commit e625b68b04

View File

@ -13,6 +13,21 @@ enum {
SAMPLE_RATE = 22050,
};
/**
* struct buf_info - a data buffer holding audio data
*
* @pos: Current position playing in audio buffer
* @size: Size of data in audio buffer (0=empty)
* @alloced: Allocated size of audio buffer (max size it can hold)
* @data: Audio data
*/
struct buf_info {
uint pos;
uint size;
uint alloced;
uint8_t *data;
};
static struct sdl_info {
SDL_Surface *screen;
int width;
@ -20,12 +35,11 @@ static struct sdl_info {
int depth;
int pitch;
uint frequency;
uint audio_pos;
uint audio_size;
uint sample_rate;
uint8_t *audio_data;
bool audio_active;
bool inited;
int cur_buf;
struct buf_info buf[2];
} sdl;
static void sandbox_sdl_poll_events(void)
@ -243,24 +257,37 @@ int sandbox_sdl_key_pressed(int keycode)
void sandbox_sdl_fill_audio(void *udata, Uint8 *stream, int len)
{
struct buf_info *buf;
int avail;
int i;
avail = sdl.audio_size - sdl.audio_pos;
if (avail < len)
len = avail;
for (i = 0; i < 2; i++) {
buf = &sdl.buf[sdl.cur_buf];
avail = buf->size - buf->pos;
if (avail <= 0) {
sdl.cur_buf = 1 - sdl.cur_buf;
continue;
}
if (avail > len)
avail = len;
SDL_MixAudio(stream, sdl.audio_data + sdl.audio_pos, len,
SDL_MIX_MAXVOLUME);
sdl.audio_pos += len;
SDL_MixAudio(stream, buf->data + buf->pos, avail,
SDL_MIX_MAXVOLUME);
buf->pos += avail;
len -= avail;
/* Loop if we are at the end */
if (sdl.audio_pos == sdl.audio_size)
sdl.audio_pos = 0;
/* Move to next buffer if we are at the end */
if (buf->pos == buf->size)
buf->size = 0;
else
break;
}
}
int sandbox_sdl_sound_init(void)
{
SDL_AudioSpec wanted;
int i;
if (sandbox_sdl_ensure_init())
return -1;
@ -276,13 +303,20 @@ int sandbox_sdl_sound_init(void)
wanted.callback = sandbox_sdl_fill_audio;
wanted.userdata = NULL;
sdl.audio_size = sizeof(uint16_t) * wanted.freq;
sdl.audio_data = malloc(sdl.audio_size);
if (!sdl.audio_data) {
printf("%s: Out of memory\n", __func__);
return -1;
for (i = 0; i < 2; i++) {
struct buf_info *buf = &sdl.buf[i];
buf->alloced = sizeof(uint16_t) * wanted.freq * wanted.channels;
buf->data = malloc(buf->alloced);
if (!buf->data) {
printf("%s: Out of memory\n", __func__);
if (i == 1)
free(sdl.buf[0].data);
return -1;
}
buf->pos = 0;
buf->size = 0;
}
sdl.audio_pos = 0;
if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
printf("Unable to initialize SDL audio: %s\n", SDL_GetError());
@ -296,23 +330,27 @@ int sandbox_sdl_sound_init(void)
}
sdl.audio_active = true;
sdl.sample_rate = wanted.freq;
sdl.cur_buf = 0;
return 0;
err:
free(sdl.audio_data);
for (i = 0; i < 2; i++)
free(sdl.buf[i].data);
return -1;
}
int sandbox_sdl_sound_start(uint frequency)
{
struct buf_info *buf = &sdl.buf[0];
if (!sdl.audio_active)
return -1;
sdl.frequency = frequency;
sound_create_square_wave(sdl.sample_rate,
(unsigned short *)sdl.audio_data,
sdl.audio_size, frequency);
sdl.audio_pos = 0;
sound_create_square_wave(sdl.sample_rate, (unsigned short *)buf->data,
buf->alloced, frequency);
buf->pos = 0;
buf->size = buf->alloced;
SDL_PauseAudio(0);
return 0;