#include <CoreAudio/CoreAudio.h>
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/DefaultAudioOutput.h>
+
+/** describes one input buffer for the osx writer */
struct osx_buffer {
+ /** pointer to the beginning of the buffer */
float *buffer;
+ /** the size of this buffer */
long size;
- float *ptr; /* Where in the buffer are we? */
+ /* current position in the buffer */
+ float *ptr;
+ /** number of floats not yet consuned */
long remaining;
+ /** pointer to the next audio buffer */
struct osx_buffer *next;
};
-typedef struct osx_buffer osx_buffer;
+/** data specific to the osx writer */
struct private_osx_write_data {
- long size;
- short *ptr;
- AudioUnit output;
+ /** the main control structure for audio data manipulation */
+ AudioUnit audio_unit;
+ /** non-zero if playback has started */
char play;
- osx_buffer *from; /* Current buffers */
- osx_buffer *to;
+ /** callback reads audio data from this buffer */
+ struct osx_buffer *from;
+ /* the post_select writes audio data here */
+ struct osx_buffer *to;
+ /** sample rate of the current audio stream */
unsigned samplerate;
+ /** number of channels of the current audio stream */
unsigned channels;
};
-
-/*
- * Tried with 3 buffers, but then any little window move is sufficient to
- * stop the sound (OK, on a G3 400 with a Public Beta. Perhaps now we can
- * go down to 2 buffers). With 16 buffers we have 1.5 seconds music
- * buffered (or, if you're pessimistic, 1.5 seconds latency). Note 0
- * buffers don't work much further than the Bus error.
- */
-#define NUMBER_BUFFERS 2
-
static void destroy_buffers(struct private_osx_write_data *powd)
{
- osx_buffer *ptr;
- osx_buffer *ptr2;
+ struct osx_buffer *ptr;
+ struct osx_buffer *ptr2;
ptr = powd->to->next;
powd->to->next = NULL;
while (ptr) {
}
}
-static void init_buffers(struct private_osx_write_data *powd)
+static void init_buffers(struct writer_node *wn)
{
+ struct private_osx_write_data *powd = wn->private_data;
+ struct osx_write_args_info *conf = wn->conf;
+ struct osx_buffer **ptrptr;
int i;
- osx_buffer ** ptrptr;
ptrptr = &powd->to;
- for (i = 0; i < NUMBER_BUFFERS; i++) {
- *ptrptr = malloc(sizeof(osx_buffer));
+ for (i = 0; i < conf->numbuffers_arg; i++) {
+ *ptrptr = malloc(sizeof(struct osx_buffer));
(*ptrptr)->size = 0;
(*ptrptr)->remaining = 0;
(*ptrptr)->buffer = NULL;
*ptrptr = powd->from = powd->to;
}
-static void fill_buffer(osx_buffer *b, short *source, long size)
+static void fill_buffer(struct osx_buffer *b, short *source, long size)
{
float *dest;
if (b->remaining) /* Non empty buffer, must still be playing */
return;
- PARA_INFO_LOG("%ld\n", size);
if (b->size != size) {
b->buffer = para_realloc(b->buffer, size * sizeof(float));
b->size = size;
/* wait for the results */
usleep(2000);
}
- PARA_INFO_LOG("buf %p: n = %ld, m= %ld\n", powd->from->buffer, n, m);
+// PARA_INFO_LOG("buf %p: n = %ld, m= %ld\n", powd->from->buffer, n, m);
/*
* we dump what we can. In fact, just the necessary
* should be sufficient
if (!comp)
goto e0;
ret = -E_OPEN_COMP;
- if (OpenAComponent(comp, &powd->output))
+ if (OpenAComponent(comp, &powd->audio_unit))
goto e0;
ret = -E_UNIT_INIT;
- if (AudioUnitInitialize(powd->output))
+ if (AudioUnitInitialize(powd->audio_unit))
goto e1;
- powd->size = 0;
- powd->ptr = NULL;
powd->play = 0;
/* Hmmm, let's choose PCM format */
/* We tell the Output Unit what format we're going to supply data to it.
/* one of the most constant constants of the whole computer history */
format.mBitsPerChannel = sizeof(float) * 8;
ret = -E_STREAM_FORMAT;
- if (AudioUnitSetProperty(powd->output, kAudioUnitProperty_StreamFormat,
+ if (AudioUnitSetProperty(powd->audio_unit, kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &format,
sizeof(AudioStreamBasicDescription)))
goto e2;
- init_buffers(powd);
+ init_buffers(wn);
ret = -E_ADD_CALLBACK;
- if (AudioUnitSetProperty(powd->output, kAudioUnitProperty_SetRenderCallback,
+ if (AudioUnitSetProperty(powd->audio_unit, kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Input, 0, &inputCallback,
sizeof(inputCallback)) < 0)
goto e3;
e3:
destroy_buffers(powd);
e2:
- AudioUnitUninitialize(powd->output);
+ AudioUnitUninitialize(powd->audio_unit);
e1:
- CloseComponent(powd->output);
+ CloseComponent(powd->audio_unit);
e0:
return ret;
}
struct private_osx_write_data *powd = wn->private_data;
PARA_INFO_LOG("closing writer node %p\n", wn);
- AudioOutputUnitStop(powd->output);
- AudioUnitUninitialize(powd->output);
- CloseComponent(powd->output);
+ AudioOutputUnitStop(powd->audio_unit);
+ AudioUnitUninitialize(powd->audio_unit);
+ CloseComponent(powd->audio_unit);
destroy_buffers(powd);
free(powd);
}
powd->to = powd->to->next;
wn->written = *wng->loaded;
if (!powd->play) {
- if (AudioOutputUnitStart(powd->output))
+ if (AudioOutputUnitStart(powd->audio_unit))
return -E_UNIT_START;
powd->play = 1;
}
static int osx_write_pre_select(struct sched *s, __a_unused struct writer_node *wn)
{
+ struct private_osx_write_data *powd = wn->private_data;
+ struct writer_node_group *wng = wn->wng;
+ size_t numbytes = powd->to->remaining * sizeof(short);
+ struct timeval tmp = {.tv_sec = 1, .tv_usec = 0}, delay = tmp;
+ unsigned long divisor;
+
+ if (!numbytes && *wng->loaded >= sizeof(short))
+ goto min_delay; /* there's a buffer to fill */
+ if (!numbytes)
+ return 1;
+ divisor = powd->samplerate * powd->channels * 2 / numbytes;
+ if (divisor)
+ tv_divide(divisor, &tmp, &delay);
+ if (tv_diff(&s->timeout, &delay, NULL) > 0)
+ s->timeout = delay;
+// PARA_DEBUG_LOG("delay: %lu:%lu\n", (long unsigned) s->timeout.tv_sec,
+// (long unsigned) s->timeout.tv_usec);
+ return 1;
+min_delay:
+ PARA_DEBUG_LOG("%s\n", "minimal delay");
s->timeout.tv_sec = 0;
- s->timeout.tv_usec = 20;
+ s->timeout.tv_usec = 1;
return 1;
}
+/** the init function of the osx writer */
void osx_write_init(struct writer *w)
{
w->open = osx_write_open;
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