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