static size_t ogg_stream_update(ALLEGRO_AUDIO_STREAM *stream, void *data,
size_t buf_size)
{
AL_OV_DATA *extra = (AL_OV_DATA *) stream->extra;
#ifdef ALLEGRO_LITTLE_ENDIAN
const int endian = 0; /* 0 for Little-Endian, 1 for Big-Endian */
#else
const int endian = 1; /* 0 for Little-Endian, 1 for Big-Endian */
#endif
const int word_size = 2; /* 1 = 8bit, 2 = 16-bit. nothing else */
const int signedness = 1; /* 0 for unsigned, 1 for signed */
unsigned long pos = 0;
int read_length = buf_size;
#if !defined(ALLEGRO_GP2XWIZ) && !defined(ALLEGRO_IPHONE)
double ctime = ov_time_tell(extra->vf);
#else
double ctime = ov_time_tell(extra->vf)/1000.0;
#endif
double rate = extra->vi->rate;
double btime = ((double)buf_size / (double)word_size) / rate;
if (stream->spl.loop == _ALLEGRO_PLAYMODE_STREAM_ONEDIR) {
if (ctime + btime > extra->loop_end) {
read_length = (extra->loop_end - ctime) * rate * (double)word_size;
if (read_length < 0)
return 0;
read_length += read_length % word_size;
}
}
while (pos < (unsigned long)read_length) {
#if !defined(ALLEGRO_GP2XWIZ) && !defined(ALLEGRO_IPHONE)
unsigned long read = ov_read(extra->vf, (char *)data + pos,
read_length - pos, endian, word_size,
signedness, &extra->bitstream);
#else
(void)endian;
(void)signedness;
unsigned long read = ov_read(extra->vf, (char *)data + pos,
read_length - pos, &extra->bitstream);
#endif
pos += read;
/* If nothing read then now to silence from here to the end. */
if (read == 0) {
int silence = _al_kcm_get_silence(stream->spl.spl_data.depth);
memset((char *)data + pos, silence, buf_size - pos);
/* return the number of usefull byes written */
return pos;
}
}
return pos;
}
/* Custom routine which runs in another thread to periodically check if DirectSound
wants more data for a stream */
static void* _dsound_update(ALLEGRO_THREAD *self, void *arg)
{
ALLEGRO_VOICE *voice = (ALLEGRO_VOICE *)arg;
ALLEGRO_DS_DATA *ex_data = (ALLEGRO_DS_DATA*)voice->extra;
const int bytes_per_sample = ex_data->bits_per_sample / 8;
DWORD play_cursor = 0;
DWORD write_cursor;
DWORD saved_play_cursor = 0;
unsigned int samples;
LPVOID ptr1, ptr2;
DWORD block1_bytes, block2_bytes;
unsigned char *data;
HRESULT hr;
(void)self;
unsigned char *silence = (unsigned char *)al_malloc(buffer_size);
int silence_value = _al_kcm_get_silence(voice->depth);
memset(silence, silence_value, buffer_size);
/* Fill buffer */
hr = ex_data->ds8_buffer->Lock(0, buffer_size,
&ptr1, &block1_bytes, &ptr2, &block2_bytes,
DSBLOCK_ENTIREBUFFER);
if (!FAILED(hr)) {
samples = buffer_size / bytes_per_sample / ex_data->channels;
data = (unsigned char *) _al_voice_update(voice, &samples);
memcpy(ptr1, data, block1_bytes);
memcpy(ptr2, data + block1_bytes, block2_bytes);
ex_data->ds8_buffer->Unlock(ptr1, block1_bytes, ptr2, block2_bytes);
}
ex_data->ds8_buffer->Play(0, 0, DSBPLAY_LOOPING);
do {
if (!_dsound_voice_is_playing(voice)) {
ex_data->ds8_buffer->Play(0, 0, DSBPLAY_LOOPING);
}
ex_data->ds8_buffer->GetCurrentPosition(&play_cursor, &write_cursor);
/* We try to fill the gap between the saved_play_cursor and the
* play_cursor.
*/
int d = play_cursor - saved_play_cursor;
if (d < 0)
d += buffer_size;
/* Don't fill small gaps. Let it accumulate to amortise the cost of
* mixing the samples and locking/unlocking the buffer.
*/
if (d < MIN_FILL) {
al_rest(0.005);
continue;
}
/* Don't generate too many samples at once. The buffer may underrun
* while we wait for _al_voice_update to complete.
*/
samples = d / bytes_per_sample / ex_data->channels;
if (samples > MAX_FILL) {
samples = MAX_FILL;
}
/* Generate the samples. */
data = (unsigned char *) _al_voice_update(voice, &samples);
if (data == NULL) {
data = silence;
}
hr = ex_data->ds8_buffer->Lock(saved_play_cursor,
samples * bytes_per_sample * ex_data->channels,
&ptr1, &block1_bytes, &ptr2, &block2_bytes, 0);
if (!FAILED(hr)) {
memcpy(ptr1, data, block1_bytes);
memcpy(ptr2, data + block1_bytes, block2_bytes);
hr = ex_data->ds8_buffer->Unlock(ptr1, block1_bytes,
ptr2, block2_bytes);
if (FAILED(hr)) {
ALLEGRO_ERROR("Unlock failed: %s\n", ds_get_error(hr));
}
}
saved_play_cursor += block1_bytes + block2_bytes;
saved_play_cursor %= buffer_size;
} while (!ex_data->stop_voice);
ex_data->ds8_buffer->Stop();
al_free(silence);
ex_data->stop_voice = 0;
al_broadcast_cond(voice->cond);
return NULL;
}
static void* oss_update(ALLEGRO_THREAD *self, void *arg)
{
ALLEGRO_VOICE *voice = arg;
OSS_VOICE *oss_voice = voice->extra;
(void)self;
while (!oss_voice->quit_poll_thread) {
/*
For possible eventual non-blocking mode:
audio_buf_info bi;
if (ioctl(oss_voice->fd, SNDCTL_DSP_GETOSPACE, &bi) == -1) {
TRACE(PREFIX_E "Error SNDCTL_DSP_GETOSPACE.\n");
TRACE(PREFIX_E "errno: %i -- %s\n", errno, strerror(errno));
return NULL;
}
len = bi.bytes;
*/
/* How many bytes are we supposed to try to write at once? */
unsigned int frames = 1024;
if (oss_voice->stop && !oss_voice->stopped) {
oss_voice->stopped = true;
}
if (!oss_voice->stop && oss_voice->stopped) {
oss_voice->stopped = false;
}
if (!voice->is_streaming && !oss_voice->stopped) {
void *buf;
int bytes = frames * oss_voice->frame_size;
oss_update_nonstream_voice(voice, &buf, &bytes);
frames = bytes / oss_voice->frame_size;
if (write(oss_voice->fd, buf, bytes) == -1) {
ALLEGRO_ERROR("errno: %i -- %s\n", errno, strerror(errno));
if (errno != EINTR)
return NULL;
}
}
else if (voice->is_streaming && !oss_voice->stopped) {
const void *data = _al_voice_update(voice, &frames);
if (data == NULL)
goto silence;
if (write(oss_voice->fd, data, frames * oss_voice->frame_size) == -1) {
ALLEGRO_ERROR("errno: %i -- %s\n", errno, strerror(errno));
if (errno != EINTR)
return NULL;
}
}
else {
silence:
/* If stopped just fill with silence. */
memset(sil_buf, _al_kcm_get_silence(voice->depth), SIL_BUF_SIZE);
if(write(oss_voice->fd, sil_buf, SIL_BUF_SIZE) == -1) {
ALLEGRO_ERROR("errno: %i -- %s\n", errno, strerror(errno));
}
}
}
return NULL;
}