static int pulseaudio_allocate_voice(ALLEGRO_VOICE *voice)
{
PULSEAUDIO_VOICE *pv = al_malloc(sizeof(PULSEAUDIO_VOICE));
pa_sample_spec ss;
pa_buffer_attr ba;
ss.channels = al_get_channel_count(voice->chan_conf);
ss.rate = voice->frequency;
if (voice->depth == ALLEGRO_AUDIO_DEPTH_UINT8)
ss.format = PA_SAMPLE_U8;
else if (voice->depth == ALLEGRO_AUDIO_DEPTH_INT16)
ss.format = PA_SAMPLE_S16NE;
#if PA_API_VERSION > 11
else if (voice->depth == ALLEGRO_AUDIO_DEPTH_INT24)
ss.format = PA_SAMPLE_S24NE;
#endif
else if (voice->depth == ALLEGRO_AUDIO_DEPTH_FLOAT32)
ss.format = PA_SAMPLE_FLOAT32NE;
else {
ALLEGRO_ERROR("Unsupported PulseAudio sound format.\n");
al_free(pv);
return 1;
}
ba.maxlength = 0x10000; // maximum length of buffer
ba.tlength = 0x2000; // target length of buffer
ba.prebuf = 0; // minimum data size required before playback starts
ba.minreq = 0; // minimum size of request
ba.fragsize = -1; // fragment size (recording)
pv->s = pa_simple_new(NULL, // Use the default server.
al_get_app_name(),
PA_STREAM_PLAYBACK,
NULL, // Use the default device.
"Allegro Voice",
&ss,
NULL, // Use default channel map
&ba,
NULL // Ignore error code.
);
if (!pv->s) {
al_free(pv);
return 1;
}
voice->extra = pv;
pv->frame_size = ss.channels * al_get_audio_depth_size(voice->depth);
pv->status = PV_STOPPED;
pv->buffer_mutex = al_create_mutex();
pv->poll_thread = al_create_thread(pulseaudio_update, (void*)voice);
al_start_thread(pv->poll_thread);
return 0;
}
static int pulseaudio_allocate_recorder(ALLEGRO_AUDIO_RECORDER *r)
{
PULSEAUDIO_RECORDER *pa;
pa = al_calloc(1, sizeof(*pa));
if (!pa) {
ALLEGRO_ERROR("Unable to allocate memory for PULSEAUDIO_RECORDER.\n");
return 1;
}
pa->ss.channels = al_get_channel_count(r->chan_conf);
pa->ss.rate = r->frequency;
if (r->depth == ALLEGRO_AUDIO_DEPTH_UINT8)
pa->ss.format = PA_SAMPLE_U8;
else if (r->depth == ALLEGRO_AUDIO_DEPTH_INT16)
pa->ss.format = PA_SAMPLE_S16NE;
#if PA_API_VERSION > 11
else if (r->depth == ALLEGRO_AUDIO_DEPTH_INT24)
pa->ss.format = PA_SAMPLE_S24NE;
#endif
else if (r->depth == ALLEGRO_AUDIO_DEPTH_FLOAT32)
pa->ss.format = PA_SAMPLE_FLOAT32NE;
else {
ALLEGRO_ERROR("Unsupported PulseAudio sound format (depth).\n");
al_free(pa);
return 1;
}
/* maximum length of the PulseAudio buffer. -1 => let the server decide. */
pa->ba.maxlength = -1;
/* fragment size (bytes) controls how much data is returned back per read.
The documentation recommends -1 for default behavior, but that sets a
latency of around 2 seconds. Lower value decreases latency but increases
overhead.
The following attempts to set it (the base latency) to 1/8 of a second.
*/
pa->ba.fragsize = (r->sample_size * r->frequency) / 8;
pa->s = pa_simple_new(NULL, al_get_app_name(), PA_STREAM_RECORD, NULL, "Allegro Audio Recorder", &pa->ss, NULL, &pa->ba, NULL);
if (!pa->s) {
ALLEGRO_ERROR("pa_simple_new() failed.\n");
al_free(pa);
return 1;
}
r->thread = al_create_thread(pulse_audio_update_recorder, r);
r->extra = pa;
return 0;
};
/* _al_kcm_refill_stream:
* Called by the mixer when the current buffer has been used up. It should
* point to the next pending buffer and reset the sample position.
* Returns true if the next buffer is available and set up.
* Otherwise returns false.
*/
bool _al_kcm_refill_stream(ALLEGRO_AUDIO_STREAM *stream)
{
ALLEGRO_SAMPLE_INSTANCE *spl = &stream->spl;
void *old_buf = spl->spl_data.buffer.ptr;
void *new_buf;
size_t i;
if (old_buf) {
/* Slide the buffers down one position and put the
* completed buffer into the used array to be refilled.
*/
for (i = 0;
i < stream->buf_count-1 && stream->pending_bufs[i];
i++) {
stream->pending_bufs[i] = stream->pending_bufs[i+1];
}
stream->pending_bufs[i] = NULL;
for (i = 0; stream->used_bufs[i]; i++)
;
stream->used_bufs[i] = old_buf;
}
new_buf = stream->pending_bufs[0];
stream->spl.spl_data.buffer.ptr = new_buf;
if (!new_buf) {
ALLEGRO_WARN("Out of buffers\n");
return false;
}
/* Copy the last MAX_LAG sample values to the front of the new buffer
* for interpolation.
*/
if (old_buf) {
const int bytes_per_sample =
al_get_channel_count(spl->spl_data.chan_conf) *
al_get_audio_depth_size(spl->spl_data.depth);
memcpy(
(char *) new_buf - bytes_per_sample * MAX_LAG,
(char *) old_buf + bytes_per_sample * (spl->pos-MAX_LAG),
bytes_per_sample * MAX_LAG);
stream->consumed_fragments++;
}
stream->spl.pos = 0;
return true;
}
/* stream_read:
* This passes the next waiting stream buffer to the voice via vbuf.
*/
static void stream_read(void *source, void **vbuf, unsigned int *samples,
ALLEGRO_AUDIO_DEPTH buffer_depth, size_t dest_maxc)
{
ALLEGRO_AUDIO_STREAM *stream = (ALLEGRO_AUDIO_STREAM*)source;
unsigned int len = stream->spl.spl_data.len;
unsigned int pos = stream->spl.pos;
if (!stream->spl.is_playing) {
*vbuf = NULL;
*samples = 0;
return;
}
if (*samples > len)
*samples = len;
if (pos >= len) {
_al_kcm_refill_stream(stream);
if (!stream->pending_bufs[0]) {
if (stream->is_draining) {
stream->spl.is_playing = false;
}
*vbuf = NULL;
*samples = 0;
return;
}
*vbuf = stream->pending_bufs[0];
pos = *samples;
_al_kcm_emit_stream_events(stream);
}
else {
int bytes = pos * al_get_channel_count(stream->spl.spl_data.chan_conf)
* al_get_audio_depth_size(stream->spl.spl_data.depth);
*vbuf = ((char *)stream->pending_bufs[0]) + bytes;
if (pos + *samples > len)
*samples = len - pos;
pos += *samples;
}
stream->spl.pos = pos;
(void)dest_maxc;
(void)buffer_depth;
}
static void reset_stopped_stream(ALLEGRO_AUDIO_STREAM *stream)
{
const int bytes_per_sample =
al_get_channel_count(stream->spl.spl_data.chan_conf) *
al_get_audio_depth_size(stream->spl.spl_data.depth);
const int fragment_buffer_size =
bytes_per_sample * (stream->spl.spl_data.len + MAX_LAG);
size_t i, n;
/* Write silence to the "invisible" part in between fragment buffers to
* avoid interpolation artifacts. It's tempting to zero the complete
* memory block in one go but some of the buffers might be getting
* refilled. So they are currently "owned" by the library user and
* should not be overwritten. But zeroing the parts not visible to the
* user should be OK.
*/
for (i = 0; i < stream->buf_count; ++i) {
al_fill_silence((char *)stream->main_buffer + i * fragment_buffer_size,
MAX_LAG, stream->spl.spl_data.depth, stream->spl.spl_data.chan_conf);
}
/* Get the current number of entries in the used_buf list. */
for (n = 0; n < stream->buf_count && stream->used_bufs[n]; n++)
;
/* Move everything from pending_bufs to used_bufs. */
i = 0;
while (i < stream->buf_count &&
n < stream->buf_count &&
stream->pending_bufs[i])
{
stream->used_bufs[n] = stream->pending_bufs[i];
stream->pending_bufs[i] = NULL;
n++;
i++;
}
/* No fragment buffer is currently playing. */
stream->spl.spl_data.buffer.ptr = NULL;
stream->spl.pos = stream->spl.spl_data.len;
stream->spl.pos_bresenham_error = 0;
stream->consumed_fragments = 0;
}
/* Function: al_attach_audio_stream_to_voice
*/
bool al_attach_audio_stream_to_voice(ALLEGRO_AUDIO_STREAM *stream,
ALLEGRO_VOICE *voice)
{
bool ret;
ASSERT(voice);
ASSERT(stream);
if (voice->attached_stream) {
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Attempted to attach to a voice that already has an attachment");
return false;
}
if (stream->spl.parent.u.ptr) {
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Attempted to attach a stream that is already attached");
return false;
}
if (voice->chan_conf != stream->spl.spl_data.chan_conf ||
voice->frequency != stream->spl.spl_data.frequency ||
voice->depth != stream->spl.spl_data.depth)
{
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Stream settings do not match voice settings");
return false;
}
al_lock_mutex(voice->mutex);
voice->attached_stream = &stream->spl;
_al_kcm_stream_set_mutex(&stream->spl, voice->mutex);
stream->spl.parent.u.voice = voice;
stream->spl.parent.is_voice = true;
voice->is_streaming = true;
voice->num_buffers = stream->buf_count;
voice->buffer_size = (stream->spl.spl_data.len) *
al_get_channel_count(stream->spl.spl_data.chan_conf) *
al_get_audio_depth_size(stream->spl.spl_data.depth);
ASSERT(stream->spl.spl_read == NULL);
stream->spl.spl_read = stream_read;
if (voice->driver->start_voice(voice) != 0) {
voice->attached_stream = NULL;
_al_kcm_stream_set_mutex(&stream->spl, NULL);
stream->spl.parent.u.voice = NULL;
stream->spl.spl_read = NULL;
_al_set_error(ALLEGRO_GENERIC_ERROR, "Unable to start stream");
ret = false;
}
else {
ret = true;
}
al_unlock_mutex(voice->mutex);
return ret;
}
/* Function: al_attach_sample_instance_to_voice
*/
bool al_attach_sample_instance_to_voice(ALLEGRO_SAMPLE_INSTANCE *spl,
ALLEGRO_VOICE *voice)
{
bool ret;
ASSERT(voice);
ASSERT(spl);
if (voice->attached_stream) {
ALLEGRO_WARN(
"Attempted to attach to a voice that already has an attachment\n");
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Attempted to attach to a voice that already has an attachment");
return false;
}
if (spl->parent.u.ptr) {
ALLEGRO_WARN("Attempted to attach a sample that is already attached\n");
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Attempted to attach a sample that is already attached");
return false;
}
if (voice->chan_conf != spl->spl_data.chan_conf ||
voice->frequency != spl->spl_data.frequency ||
voice->depth != spl->spl_data.depth)
{
ALLEGRO_WARN("Sample settings do not match voice settings\n");
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Sample settings do not match voice settings");
return false;
}
al_lock_mutex(voice->mutex);
voice->attached_stream = spl;
voice->is_streaming = false;
voice->num_buffers = 1;
voice->buffer_size = (spl->spl_data.len) *
al_get_channel_count(voice->chan_conf) *
al_get_audio_depth_size(voice->depth);
spl->spl_read = NULL;
_al_kcm_stream_set_mutex(spl, voice->mutex);
spl->parent.u.voice = voice;
spl->parent.is_voice = true;
if (voice->driver->load_voice(voice, spl->spl_data.buffer.ptr) != 0 ||
(spl->is_playing && voice->driver->start_voice(voice) != 0))
{
voice->attached_stream = NULL;
spl->spl_read = NULL;
_al_kcm_stream_set_mutex(spl, NULL);
spl->parent.u.voice = NULL;
ALLEGRO_ERROR("Unable to load sample into voice\n");
ret = false;
}
else {
ret = true;
}
al_unlock_mutex(voice->mutex);
return ret;
}
static int oss_allocate_voice(ALLEGRO_VOICE *voice)
{
int format;
int chan_count;
OSS_VOICE *ex_data = calloc(1, sizeof(OSS_VOICE));
if (!ex_data)
return 1;
ex_data->fd = open(oss_audio_device, O_WRONLY/*, O_NONBLOCK*/);
if (ex_data->fd == -1) {
ALLEGRO_ERROR("Failed to open audio device '%s'.\n",
oss_audio_device);
ALLEGRO_ERROR("errno: %i -- %s\n", errno, strerror(errno));
free(ex_data);
return 1;
}
chan_count = al_get_channel_count(voice->chan_conf);
ex_data->frame_size = chan_count * al_get_depth_size(voice->depth);
if (!ex_data->frame_size)
goto Error;
ex_data->stop = true;
ex_data->stopped = true;
if (voice->depth == ALLEGRO_AUDIO_DEPTH_INT8)
format = AFMT_S8;
else if (voice->depth == ALLEGRO_AUDIO_DEPTH_UINT8)
format = AFMT_U8;
else if (voice->depth == ALLEGRO_AUDIO_DEPTH_INT16)
format = AFMT_S16_NE;
else if (voice->depth == ALLEGRO_AUDIO_DEPTH_UINT16)
format = AFMT_U16_NE;
#ifdef OSS_VER_4
else if (voice->depth == ALLEGRO_AUDIO_DEPTH_INT24)
format = AFMT_S24_NE;
else if (voice->depth == ALLEGRO_AUDIO_DEPTH_FLOAT32)
format = AFMT_FLOAT;
#endif
else {
ALLEGRO_ERROR("Unsupported OSS sound format.\n");
goto Error;
}
int tmp_format = format;
int tmp_chan_count = chan_count;
unsigned int tmp_freq = voice->frequency;
int tmp_oss_fragsize = oss_fragsize;
if (using_ver_4) {
#ifdef OSS_VER_4
int tmp_oss_timing_policy = oss_timing_policy;
if (ioctl(ex_data->fd, SNDCTL_DSP_POLICY, &tmp_oss_timing_policy) == -1) {
ALLEGRO_ERROR("Failed to set_timig policity to '%i'.\n",
tmp_oss_timing_policy);
ALLEGRO_ERROR("errno: %i -- %s\n", errno, strerror(errno));
goto Error;
}
ALLEGRO_INFO("Accepted timing policy value: %i\n", tmp_oss_timing_policy);
#endif
}
else {
if (ioctl(ex_data->fd, SNDCTL_DSP_SETFRAGMENT, &tmp_oss_fragsize) == -1) {
ALLEGRO_ERROR("Failed to set fragment size.\n");
ALLEGRO_ERROR("errno: %i -- %s\n", errno, strerror(errno));
goto Error;
}
}
if (ioctl(ex_data->fd, SNDCTL_DSP_SETFMT, &tmp_format) == -1) {
ALLEGRO_ERROR("Failed to set sample format.\n");
ALLEGRO_ERROR("errno: %i -- %s\n", errno, strerror(errno));
goto Error;
}
if (tmp_format != format) {
ALLEGRO_ERROR("Sample format not supported by the driver.\n");
goto Error;
}
if (ioctl(ex_data->fd, SNDCTL_DSP_CHANNELS, &tmp_chan_count)) {
ALLEGRO_ERROR("Failed to set channel count.\n");
ALLEGRO_ERROR("errno: %i -- %s\n", errno, strerror(errno));
goto Error;
}
if (tmp_chan_count != chan_count) {
ALLEGRO_ERROR("Requested sample channe count %i, got %i.\n",
tmp_chan_count, chan_count);
}
if (ioctl(ex_data->fd, SNDCTL_DSP_SPEED, &tmp_freq) == -1) {
ALLEGRO_ERROR("Failed to set sample rate.\n");
ALLEGRO_ERROR("errno: %i -- %s\n", errno, strerror(errno));
goto Error;
}
if (voice->frequency != tmp_freq) {
ALLEGRO_ERROR("Requested sample rate %u, got %iu.\n", voice->frequency,
tmp_freq);
}
voice->extra = ex_data;
ex_data->quit_poll_thread = false;
ex_data->poll_thread = al_create_thread(oss_update, (void*)voice);
al_start_thread(ex_data->poll_thread);
return 0;
Error:
close(ex_data->fd);
free(ex_data);
return 1;
}
int main(int argc, char **argv)
{
ALLEGRO_VOICE *voice;
ALLEGRO_MIXER *mixer;
ALLEGRO_SAMPLE_INSTANCE *sample;
int i;
char const **filenames;
int n;
if (argc < 2) {
n = 1;
filenames = malloc(sizeof *filenames);
filenames[0] = "data/welcome.wav";
}
else {
n = argc - 1;
filenames = malloc(sizeof *filenames * n);
for (i = 1; i < argc; ++i) {
filenames[i - 1] = argv[i];
}
}
if (!al_init()) {
abort_example("Could not init Allegro.\n");
}
open_log();
al_init_acodec_addon();
if (!al_install_audio()) {
abort_example("Could not init sound!\n");
}
voice = al_create_voice(44100, ALLEGRO_AUDIO_DEPTH_INT16,
ALLEGRO_CHANNEL_CONF_2);
if (!voice) {
abort_example("Could not create ALLEGRO_VOICE.\n");
}
mixer = al_create_mixer(44100, ALLEGRO_AUDIO_DEPTH_FLOAT32,
ALLEGRO_CHANNEL_CONF_2);
if (!mixer) {
abort_example("al_create_mixer failed.\n");
}
if (!al_attach_mixer_to_voice(mixer, voice)) {
abort_example("al_attach_mixer_to_voice failed.\n");
}
sample = al_create_sample_instance(NULL);
if (!sample) {
abort_example("al_create_sample failed.\n");
}
for (i = 0; i < n; ++i) {
ALLEGRO_SAMPLE *sample_data = NULL;
const char *filename = filenames[i];
float sample_time = 0;
/* A matrix that puts everything in the left channel. */
float mono_to_stereo[] = {1.0, 0.0};
/* Load the entire sound file from disk. */
sample_data = al_load_sample(filename);
if (!sample_data) {
abort_example("Could not load sample from '%s'!\n",
filename);
continue;
}
if (!al_set_sample(sample, sample_data)) {
abort_example("al_set_sample_instance_ptr failed.\n");
continue;
}
if (!al_attach_sample_instance_to_mixer(sample, mixer)) {
abort_example("al_attach_sample_instance_to_mixer failed.\n");
goto done;
}
/* Play sample in looping mode. */
al_set_sample_instance_playmode(sample, ALLEGRO_PLAYMODE_LOOP);
al_play_sample_instance(sample);
sample_time = al_get_sample_instance_time(sample);
log_printf("Playing '%s' (%.3f seconds) normally.\n", filename,
sample_time);
al_rest(sample_time);
if (al_get_channel_count(al_get_sample_instance_channels(sample)) == 1) {
if (!al_set_sample_instance_channel_matrix(sample, mono_to_stereo)) {
abort_example("Failed to set channel matrix.\n");
}
log_printf("Playing left channel only.\n");
al_rest(sample_time);
}
if (!al_set_sample_instance_gain(sample, 0.5)) {
abort_example("Failed to set gain.\n");
}
log_printf("Playing with gain 0.5.\n");
al_rest(sample_time);
if (!al_set_sample_instance_gain(sample, 0.25)) {
abort_example("Failed to set gain.\n");
}
log_printf("Playing with gain 0.25.\n");
al_rest(sample_time);
al_stop_sample_instance(sample);
log_printf("Done playing '%s'\n", filename);
/* Free the memory allocated. */
al_set_sample(sample, NULL);
al_destroy_sample(sample_data);
}
al_destroy_sample_instance(sample);
al_destroy_mixer(mixer);
al_destroy_voice(voice);
al_uninstall_audio();
done:
close_log(true);
return 0;
}
static int _dsound_open_recorder(ALLEGRO_AUDIO_RECORDER *r)
{
HRESULT hr;
if (capture_device != NULL) {
/* FIXME: It's wrong to assume only a single recording device, but since
there is no enumeration of devices, it doesn't matter for now. */
ALLEGRO_ERROR("Already recording.\n");
return 1;
}
ALLEGRO_INFO("Creating default capture device.\n");
/* FIXME: Use default device until we have device enumeration */
hr = DirectSoundCaptureCreate8(NULL, &capture_device, NULL);
if (FAILED(hr)) {
ALLEGRO_ERROR("DirectSoundCaptureCreate8 failed: %s\n", ds_get_error(hr));
return 1;
}
hr = device->SetCooperativeLevel(get_window(), DSSCL_PRIORITY);
if (FAILED(hr)) {
ALLEGRO_ERROR("SetCooperativeLevel failed: %s\n", ds_get_error(hr));
return 1;
}
DSOUND_RECORD_DATA *extra = (DSOUND_RECORD_DATA *) al_calloc(1, sizeof(*extra));
DSCCAPS dsccaps;
dsccaps.dwSize = sizeof(DSCCAPS);
hr = capture_device->GetCaps(&dsccaps);
if (FAILED(hr)) {
ALLEGRO_ERROR("DirectSoundCaptureCreate8::GetCaps failed: %s\n", ds_get_error(hr));
}
else {
ALLEGRO_INFO("caps: %lu %lu\n", dsccaps.dwFormats, dsccaps.dwFormats & WAVE_FORMAT_2M16);
}
memset(&extra->wave_fmt, 0, sizeof(extra->wave_fmt));
extra->wave_fmt.wFormatTag = WAVE_FORMAT_PCM;
extra->wave_fmt.nChannels = (WORD)al_get_channel_count(r->chan_conf);
extra->wave_fmt.nSamplesPerSec = r->frequency;
extra->wave_fmt.wBitsPerSample = (WORD)al_get_audio_depth_size(r->depth) * 8;
extra->wave_fmt.nBlockAlign = extra->wave_fmt.nChannels * extra->wave_fmt.wBitsPerSample / 8;
extra->wave_fmt.nAvgBytesPerSec = extra->wave_fmt.nSamplesPerSec * extra->wave_fmt.nBlockAlign;
memset(&extra->desc, 0, sizeof(extra->desc));
extra->desc.dwSize = sizeof(extra->desc);
extra->desc.lpwfxFormat = &extra->wave_fmt;
extra->desc.dwBufferBytes = extra->wave_fmt.nAvgBytesPerSec * 5;
hr = capture_device->CreateCaptureBuffer(&extra->desc, &extra->buffer, NULL);
if (FAILED(hr)) {
al_free(extra);
ALLEGRO_ERROR("Unable to create Capture Buffer\n");
return 1;
}
extra->buffer->QueryInterface(_al_IID_IDirectSoundCaptureBuffer8, (void **) &extra->buffer8);
r->extra = extra;
r->thread = al_create_thread(_dsound_update_recorder, r);
return 0;
}
/* _al_kcm_feed_stream:
* A routine running in another thread that feeds the stream buffers as
* neccesary, usually getting data from some file reader backend.
*/
void *_al_kcm_feed_stream(ALLEGRO_THREAD *self, void *vstream)
{
ALLEGRO_AUDIO_STREAM *stream = vstream;
ALLEGRO_EVENT_QUEUE *queue;
ALLEGRO_EVENT event;
(void)self;
ALLEGRO_DEBUG("Stream feeder thread started.\n");
queue = al_create_event_queue();
al_register_event_source(queue, &stream->spl.es);
al_lock_mutex(stream->feed_thread_started_mutex);
stream->feed_thread_started = true;
al_broadcast_cond(stream->feed_thread_started_cond);
al_unlock_mutex(stream->feed_thread_started_mutex);
stream->quit_feed_thread = false;
while (!stream->quit_feed_thread) {
char *fragment;
ALLEGRO_EVENT event;
al_wait_for_event(queue, &event);
if (event.type == ALLEGRO_EVENT_AUDIO_STREAM_FRAGMENT
&& !stream->is_draining) {
unsigned long bytes;
unsigned long bytes_written;
ALLEGRO_MUTEX *stream_mutex;
fragment = al_get_audio_stream_fragment(stream);
if (!fragment) {
/* This is not an error. */
continue;
}
bytes = (stream->spl.spl_data.len) *
al_get_channel_count(stream->spl.spl_data.chan_conf) *
al_get_audio_depth_size(stream->spl.spl_data.depth);
stream_mutex = maybe_lock_mutex(stream->spl.mutex);
bytes_written = stream->feeder(stream, fragment, bytes);
maybe_unlock_mutex(stream_mutex);
if (stream->spl.loop == _ALLEGRO_PLAYMODE_STREAM_ONEDIR) {
/* Keep rewinding until the fragment is filled. */
while (bytes_written < bytes &&
stream->spl.loop == _ALLEGRO_PLAYMODE_STREAM_ONEDIR) {
size_t bw;
al_rewind_audio_stream(stream);
stream_mutex = maybe_lock_mutex(stream->spl.mutex);
bw = stream->feeder(stream, fragment + bytes_written,
bytes - bytes_written);
bytes_written += bw;
maybe_unlock_mutex(stream_mutex);
}
}
else if (bytes_written < bytes) {
/* Fill the rest of the fragment with silence. */
int silence_samples = (bytes - bytes_written) /
(al_get_channel_count(stream->spl.spl_data.chan_conf) *
al_get_audio_depth_size(stream->spl.spl_data.depth));
al_fill_silence(fragment + bytes_written, silence_samples,
stream->spl.spl_data.depth, stream->spl.spl_data.chan_conf);
}
if (!al_set_audio_stream_fragment(stream, fragment)) {
ALLEGRO_ERROR("Error setting stream buffer.\n");
continue;
}
/* The streaming source doesn't feed any more, drain buffers and quit. */
if (bytes_written != bytes &&
stream->spl.loop == _ALLEGRO_PLAYMODE_STREAM_ONCE) {
al_drain_audio_stream(stream);
stream->quit_feed_thread = true;
}
}
else if (event.type == _KCM_STREAM_FEEDER_QUIT_EVENT_TYPE) {
stream->quit_feed_thread = true;
}
}
event.user.type = ALLEGRO_EVENT_AUDIO_STREAM_FINISHED;
event.user.timestamp = al_get_time();
al_emit_user_event(&stream->spl.es, &event, NULL);
al_destroy_event_queue(queue);
ALLEGRO_DEBUG("Stream feeder thread finished.\n");
return NULL;
}
/* Function: al_create_audio_stream
*/
ALLEGRO_AUDIO_STREAM *al_create_audio_stream(size_t fragment_count,
unsigned int frag_samples, unsigned int freq, ALLEGRO_AUDIO_DEPTH depth,
ALLEGRO_CHANNEL_CONF chan_conf)
{
ALLEGRO_AUDIO_STREAM *stream;
unsigned long bytes_per_sample;
unsigned long bytes_per_frag_buf;
size_t i;
if (!fragment_count) {
_al_set_error(ALLEGRO_INVALID_PARAM,
"Attempted to create stream with no buffers");
return NULL;
}
if (!frag_samples) {
_al_set_error(ALLEGRO_INVALID_PARAM,
"Attempted to create stream with no buffer size");
return NULL;
}
if (!freq) {
_al_set_error(ALLEGRO_INVALID_PARAM,
"Attempted to create stream with no frequency");
return NULL;
}
bytes_per_sample = al_get_channel_count(chan_conf) *
al_get_audio_depth_size(depth);
bytes_per_frag_buf = frag_samples * bytes_per_sample;
stream = al_calloc(1, sizeof(*stream));
if (!stream) {
_al_set_error(ALLEGRO_GENERIC_ERROR,
"Out of memory allocating stream object");
return NULL;
}
stream->spl.is_playing = true;
stream->is_draining = false;
stream->spl.loop = _ALLEGRO_PLAYMODE_STREAM_ONCE;
stream->spl.spl_data.depth = depth;
stream->spl.spl_data.chan_conf = chan_conf;
stream->spl.spl_data.frequency = freq;
stream->spl.speed = 1.0f;
stream->spl.gain = 1.0f;
stream->spl.pan = 0.0f;
stream->spl.step = 0;
stream->spl.pos = frag_samples;
stream->spl.spl_data.len = stream->spl.pos;
stream->buf_count = fragment_count;
stream->used_bufs = al_calloc(1, fragment_count * sizeof(void *) * 2);
if (!stream->used_bufs) {
al_free(stream->used_bufs);
al_free(stream);
_al_set_error(ALLEGRO_GENERIC_ERROR,
"Out of memory allocating stream buffer pointers");
return NULL;
}
stream->pending_bufs = stream->used_bufs + fragment_count;
/* The main_buffer holds all the buffer fragments in contiguous memory.
* To support interpolation across buffer fragments, we allocate extra
* MAX_LAG samples at the start of each buffer fragment, to hold the
* last few sample values which came before that fragment.
*/
stream->main_buffer = al_calloc(1,
(MAX_LAG * bytes_per_sample + bytes_per_frag_buf) * fragment_count);
if (!stream->main_buffer) {
al_free(stream->used_bufs);
al_free(stream);
_al_set_error(ALLEGRO_GENERIC_ERROR,
"Out of memory allocating stream buffer");
return NULL;
}
for (i = 0; i < fragment_count; i++) {
char *buffer = (char *)stream->main_buffer
+ i * (MAX_LAG * bytes_per_sample + bytes_per_frag_buf);
al_fill_silence(buffer, MAX_LAG, depth, chan_conf);
stream->used_bufs[i] = buffer + MAX_LAG * bytes_per_sample;
}
al_init_user_event_source(&stream->spl.es);
/* This can lead to deadlocks on shutdown, hence we don't do it. */
/* _al_kcm_register_destructor(stream, (void (*)(void *)) al_destroy_audio_stream); */
return stream;
}
