static gboolean
gst_audioringbuffer_open_device (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
gboolean result = TRUE;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
if (csink->open)
result = csink->open (sink);
if (!result)
goto could_not_open;
return result;
could_not_open:
{
GST_DEBUG_OBJECT (sink, "could not open device");
return FALSE;
}
}
static gboolean
gst_audio_sink_ring_buffer_close_device (GstAudioRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
gboolean result = TRUE;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
if (csink->close)
result = csink->close (sink);
if (!result)
goto could_not_close;
return result;
could_not_close:
{
GST_DEBUG_OBJECT (sink, "could not close device");
return FALSE;
}
}
static gboolean
gst_audioringbuffer_activate (GstRingBuffer * buf, gboolean active)
{
GstAudioSink *sink;
GstAudioRingBuffer *abuf;
GError *error = NULL;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
abuf = GST_AUDIORING_BUFFER_CAST (buf);
if (active) {
abuf->running = TRUE;
GST_DEBUG_OBJECT (sink, "starting thread");
sink->thread =
g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE,
&error);
if (!sink->thread || error != NULL)
goto thread_failed;
GST_DEBUG_OBJECT (sink, "waiting for thread");
/* the object lock is taken */
GST_AUDIORING_BUFFER_WAIT (buf);
GST_DEBUG_OBJECT (sink, "thread is started");
} else {
#ifndef GSTREAMER_LITE
abuf->running = FALSE;
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (buf);
/* join the thread */
g_thread_join (sink->thread);
GST_OBJECT_LOCK (buf);
#else // GSTREAMER_LITE
// We may get called with active set to FALSE several times.
// See gst_base_audio_sink_change_state()
if (abuf->running)
{
abuf->running = FALSE;
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (buf);
/* join the thread */
if (sink->thread != NULL)
{
g_thread_join (sink->thread);
sink->thread = NULL;
}
GST_OBJECT_LOCK (buf);
}
#endif // GSTREAMER_LITE
}
return TRUE;
/* ERRORS */
thread_failed:
{
if (error)
GST_ERROR_OBJECT (sink, "could not create thread %s", error->message);
else
GST_ERROR_OBJECT (sink, "could not create thread for unknown reason");
return FALSE;
}
}
/* this internal thread does nothing else but write samples to the audio device.
* It will write each segment in the ringbuffer and will update the play
* pointer.
* The start/stop methods control the thread.
*/
static void
audioringbuffer_thread_func (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioRingBuffer *abuf = GST_AUDIORING_BUFFER_CAST (buf);
WriteFunc writefunc;
GstMessage *message;
GValue val = { 0 };
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
GST_DEBUG_OBJECT (sink, "enter thread");
GST_OBJECT_LOCK (abuf);
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (abuf);
writefunc = csink->write;
if (writefunc == NULL)
goto no_function;
g_value_init (&val, G_TYPE_POINTER);
g_value_set_pointer (&val, sink->thread);
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (sink));
gst_message_set_stream_status_object (message, &val);
GST_DEBUG_OBJECT (sink, "posting ENTER stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
/* buffer must be started */
if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint written;
left = len;
do {
written = writefunc (sink, readptr, left);
GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d",
written, left, readseg);
if (written < 0 || written > left) {
/* might not be critical, it e.g. happens when aborting playback */
GST_WARNING_OBJECT (sink,
"error writing data in %s (reason: %s), skipping segment (left: %d, written: %d)",
GST_DEBUG_FUNCPTR_NAME (writefunc),
(errno > 1 ? g_strerror (errno) : "unknown"), left, written);
break;
}
left -= written;
readptr += written;
} while (left > 0);
/* clear written samples */
gst_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_ring_buffer_advance (buf, 1);
} else {
GST_OBJECT_LOCK (abuf);
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_DEBUG_OBJECT (sink, "wait for action");
#ifndef GSTREAMER_LITE
GST_AUDIORING_BUFFER_WAIT (buf);
#else // GSTREAMER_LITE
// In same cases we may have condition when we waiting here for ring buffer to start,
// while ring buffer is started and data is available. So, lets use wait with timeout
// and recheck if we good to go. wait_segment() will start ring buffer when data is available.
{
GTimeVal timeout;
g_get_current_time(&timeout);
g_time_val_add(&timeout, 100000); // 100 millisecond
GST_AUDIORING_BUFFER_TIMED_WAIT (buf, &timeout);
}
#endif // GSTREAMER_LITE
GST_DEBUG_OBJECT (sink, "got signal");
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (sink, "continue running");
GST_OBJECT_UNLOCK (abuf);
}
}
/* Will never be reached */
g_assert_not_reached ();
return;
/* ERROR */
no_function:
{
GST_DEBUG_OBJECT (sink, "no write function, exit thread");
return;
}
stop_running:
{
GST_OBJECT_UNLOCK (abuf);
GST_DEBUG_OBJECT (sink, "stop running, exit thread");
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (sink));
gst_message_set_stream_status_object (message, &val);
GST_DEBUG_OBJECT (sink, "posting LEAVE stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
return;
}
}
/* this internal thread does nothing else but write samples to the audio device.
* It will write each segment in the ringbuffer and will update the play
* pointer.
* The start/stop methods control the thread.
*/
static void
audioringbuffer_thread_func (GstAudioRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioSinkRingBuffer *abuf = GST_AUDIO_SINK_RING_BUFFER_CAST (buf);
WriteFunc writefunc;
GstMessage *message;
GValue val = { 0 };
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
GST_DEBUG_OBJECT (sink, "enter thread");
GST_OBJECT_LOCK (abuf);
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (abuf);
writefunc = csink->write;
if (writefunc == NULL)
goto no_function;
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (sink));
g_value_init (&val, GST_TYPE_G_THREAD);
g_value_set_boxed (&val, g_thread_self ());
gst_message_set_stream_status_object (message, &val);
g_value_unset (&val);
GST_DEBUG_OBJECT (sink, "posting ENTER stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
/* buffer must be started */
if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint written;
left = len;
do {
written = writefunc (sink, readptr, left);
GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d",
written, left, readseg);
if (written < 0 || written > left) {
/* might not be critical, it e.g. happens when aborting playback */
GST_WARNING_OBJECT (sink,
"error writing data in %s (reason: %s), skipping segment (left: %d, written: %d)",
GST_DEBUG_FUNCPTR_NAME (writefunc),
(errno > 1 ? g_strerror (errno) : "unknown"), left, written);
break;
}
left -= written;
readptr += written;
} while (left > 0);
/* clear written samples */
gst_audio_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_audio_ring_buffer_advance (buf, 1);
} else {
GST_OBJECT_LOCK (abuf);
if (!abuf->running)
goto stop_running;
if (G_UNLIKELY (g_atomic_int_get (&buf->state) ==
GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
GST_OBJECT_UNLOCK (abuf);
continue;
}
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
GST_DEBUG_OBJECT (sink, "wait for action");
GST_AUDIO_SINK_RING_BUFFER_WAIT (buf);
GST_DEBUG_OBJECT (sink, "got signal");
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (sink, "continue running");
GST_OBJECT_UNLOCK (abuf);
}
}
/* Will never be reached */
g_assert_not_reached ();
return;
/* ERROR */
no_function:
{
GST_DEBUG_OBJECT (sink, "no write function, exit thread");
return;
}
stop_running:
{
GST_OBJECT_UNLOCK (abuf);
GST_DEBUG_OBJECT (sink, "stop running, exit thread");
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (sink));
g_value_init (&val, GST_TYPE_G_THREAD);
g_value_set_boxed (&val, g_thread_self ());
gst_message_set_stream_status_object (message, &val);
g_value_unset (&val);
GST_DEBUG_OBJECT (sink, "posting LEAVE stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
return;
}
}
static gboolean
NAS_createFlow (GstNasSink * sink, GstRingBufferSpec * spec)
{
AuElement elements[2];
AuUint32 buf_samples;
unsigned char format;
format = gst_nas_sink_sink_get_format (spec);
if (format == 0) {
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Unable to get format %d", spec->format));
return FALSE;
}
GST_DEBUG_OBJECT (sink, "Format: %d %d\n", spec->format, format);
sink->flow = AuGetScratchFlow (sink->audio, NULL);
if (sink->flow == 0) {
GST_DEBUG_OBJECT (sink, "couldn't get flow");
return FALSE;
}
/* free old Elements and reconnet to server, needed to change samplerate */
{
AuBool clocked;
int num_elements;
AuStatus status;
AuElement *oldelems;
oldelems =
AuGetElements (sink->audio, sink->flow, &clocked, &num_elements,
&status);
if (num_elements > 0) {
GST_DEBUG_OBJECT (sink, "GetElements status: %i", status);
if (oldelems)
AuFreeElements (sink->audio, num_elements, oldelems);
gst_nas_sink_close (GST_AUDIO_SINK (sink));
gst_nas_sink_open (GST_AUDIO_SINK (sink));
sink->flow = AuGetScratchFlow (sink->audio, NULL);
if (sink->flow == 0) {
GST_DEBUG_OBJECT (sink, "couldn't get flow");
return FALSE;
}
}
}
/* free old Elements and reconnet to server, needed to change samplerate */
{
AuBool clocked;
int num_elements;
AuStatus status;
AuElement *oldelems;
oldelems =
AuGetElements (sink->audio, sink->flow, &clocked, &num_elements,
&status);
if (num_elements > 0) {
GST_DEBUG_OBJECT (sink, "GetElements status: %i", status);
if (oldelems)
AuFreeElements (sink->audio, num_elements, oldelems);
gst_nas_sink_close (GST_AUDIO_SINK (sink));
gst_nas_sink_open (GST_AUDIO_SINK (sink));
sink->flow = AuGetScratchFlow (sink->audio, NULL);
if (sink->flow == 0) {
GST_DEBUG_OBJECT (sink, "couldn't get flow");
return FALSE;
}
}
}
buf_samples = spec->rate * NAS_SOUND_PORT_DURATION;
/*
spec->segsize = gst_util_uint64_scale (buf_samples * spec->bytes_per_sample,
spec->latency_time, GST_SECOND / GST_USECOND);
spec->segsize -= spec->segsize % spec->bytes_per_sample;
spec->segtotal = spec->buffer_time / spec->latency_time;
*/
spec->segsize = buf_samples * spec->bytes_per_sample;
spec->segtotal = 1;
GST_DEBUG_OBJECT (sink, "Rate %d Format %d tracks %d bufs %d %d/%d w %d",
spec->rate, format, spec->channels, (gint) buf_samples, spec->segsize,
spec->segtotal, spec->width);
AuMakeElementImportClient (&elements[0], /* element */
spec->rate, /* rate */
format, /* format */
spec->channels, /* number of tracks */
AuTrue, /* discart */
buf_samples, /* max samples */
(AuUint32) (buf_samples / 100 * AuSoundPortLowWaterMark),
/* low water mark */
0, /* num actions */
NULL);
sink->device = NAS_getDevice (sink->audio, spec->channels);
if (sink->device == AuNone) {
GST_DEBUG_OBJECT (sink, "no device with %i tracks found", spec->channels);
return FALSE;
}
AuMakeElementExportDevice (&elements[1], /* element */
0, /* input */
sink->device, /* device */
spec->rate, /* rate */
AuUnlimitedSamples, /* num samples */
0, /* num actions */
NULL); /* actions */
AuSetElements (sink->audio, /* server */
sink->flow, /* flow ID */
AuTrue, /* clocked */
2, /* num elements */
elements, /* elements */
NULL);
AuRegisterEventHandler (sink->audio, /* server */
AuEventHandlerIDMask, /* value mask */
0, /* type */
sink->flow, /* flow ID */
NAS_EventHandler, /* callback */
(AuPointer) sink); /* data */
AuStartFlow (sink->audio, sink->flow, NULL);
return TRUE;
}