/**
* gst_audio_clock_new:
* @name: the name of the clock
* @func: a function
* @user_data: user data
*
* Create a new #GstAudioClock instance. Whenever the clock time should be
* calculated it will call @func with @user_data. When @func returns
* #GST_CLOCK_TIME_NONE, the clock will return the last reported time.
*
* Returns: a new #GstAudioClock casted to a #GstClock.
*/
GstClock *
gst_audio_clock_new (gchar * name, GstAudioClockGetTimeFunc func,
gpointer user_data)
{
GstAudioClock *aclock =
GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, "name", name, NULL));
aclock->func = func;
aclock->user_data = user_data;
return (GstClock *) aclock;
}
static GstClockTime
gst_audio_clock_get_internal_time (GstClock * clock)
{
GstAudioClock *aclock;
GstClockTime result;
aclock = GST_AUDIO_CLOCK (clock);
result = aclock->func (clock, aclock->user_data);
if (result == GST_CLOCK_TIME_NONE)
result = aclock->last_time;
else {
result += aclock->abidata.ABI.time_offset;
aclock->last_time = result;
}
return result;
}
static GstStateChangeReturn
alsaspdifsink_change_state (GstElement * element, GstStateChange transition)
{
AlsaSPDIFSink *sink = ALSASPDIFSINK (element);
GstStateChangeReturn ret;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
sink->frames = 0;
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->clock), 0);
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
if (!alsaspdifsink_open (sink)) {
GST_WARNING_OBJECT (sink, "Failed to open alsa device");
return GST_STATE_CHANGE_FAILURE;
}
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
GST_INFO_OBJECT (sink, "Parent change_state returned %d", ret);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
alsaspdifsink_close (sink);
break;
default:
break;
}
return ret;
}
static GstStateChangeReturn
gst_base_audio_src_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
GST_DEBUG_OBJECT (src, "NULL->READY");
GST_OBJECT_LOCK (src);
if (src->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
}
GST_OBJECT_UNLOCK (src);
if (!gst_ring_buffer_open_device (src->ringbuffer))
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_DEBUG_OBJECT (src, "READY->PAUSED");
src->next_sample = -1;
gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
gst_ring_buffer_may_start (src->ringbuffer, TRUE);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_DEBUG_OBJECT (src, "PLAYING->PAUSED");
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
gst_ring_buffer_pause (src->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
gst_ring_buffer_release (src->ringbuffer);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
GST_DEBUG_OBJECT (src, "READY->NULL");
gst_ring_buffer_close_device (src->ringbuffer);
GST_OBJECT_LOCK (src);
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
GST_OBJECT_UNLOCK (src);
break;
default:
break;
}
return ret;
/* ERRORS */
open_failed:
{
/* subclass must post a meaningfull error message */
GST_DEBUG_OBJECT (src, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
}
static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_stream_flags_t flags;
pa_operation *o;
GstAudioClock *clock;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
{
GstAudioRingBufferSpec s = *spec;
const pa_channel_map *m;
m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &s);
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(pulsesrc)->ringbuffer, s.info.position);
}
/* enable event notifications */
GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
if (!(o = pa_context_subscribe (pulsesrc->context,
PA_SUBSCRIPTION_MASK_SOURCE_OUTPUT, NULL, NULL))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_subscribe() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pa_operation_unref (o);
wanted.maxlength = -1;
wanted.tlength = -1;
wanted.prebuf = 0;
wanted.minreq = -1;
wanted.fragsize = spec->segsize;
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d", wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", wanted.prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d", wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d", wanted.fragsize);
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
PA_STREAM_START_CORKED;
if (pulsesrc->mute_set && pulsesrc->mute)
flags |= PA_STREAM_START_MUTED;
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
flags) < 0) {
goto connect_failed;
}
/* our clock will now start from 0 again */
clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
gst_audio_clock_reset (clock, 0);
pulsesrc->corked = TRUE;
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state (pulsesrc->stream);
if (!PA_STREAM_IS_GOOD (state))
goto stream_is_bad;
if (state == PA_STREAM_READY)
break;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
pulsesrc->stream_connected = TRUE;
/* store the source output index so it can be accessed via a property */
pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream);
g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
if (pulsesrc->volume_set) {
gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
pulsesrc->volume_set = FALSE;
}
/* get the actual buffering properties now */
actual = pa_stream_get_buffer_attr (pulsesrc->stream);
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d (wanted: %d)",
actual->tlength, wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", actual->prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d (wanted %d)", actual->minreq,
wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d (wanted %d)",
actual->fragsize, wanted.fragsize);
if (actual->fragsize >= wanted.fragsize) {
spec->segsize = actual->fragsize;
} else {
spec->segsize = actual->fragsize * (wanted.fragsize / actual->fragsize);
}
spec->segtotal = actual->maxlength / spec->segsize;
if (!pulsesrc->paused) {
GST_DEBUG_OBJECT (pulsesrc, "uncorking because we are playing");
gst_pulsesrc_set_corked (pulsesrc, FALSE, FALSE);
}
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
/* ERRORS */
connect_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
stream_is_bad:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
unlock_and_fail:
{
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return FALSE;
}
}
static GstStateChangeReturn
gst_base_audio_src_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
GST_DEBUG_OBJECT (src, "NULL->READY");
GST_OBJECT_LOCK (src);
if (src->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
}
GST_OBJECT_UNLOCK (src);
if (!gst_ring_buffer_open_device (src->ringbuffer))
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_DEBUG_OBJECT (src, "READY->PAUSED");
src->next_sample = -1;
gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
/* Only post clock-provide messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) &&
GST_AUDIO_CLOCK_CAST (src->clock)->func ==
(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
src->clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
gst_ring_buffer_may_start (src->ringbuffer, TRUE);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_DEBUG_OBJECT (src, "PLAYING->PAUSED");
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
gst_ring_buffer_pause (src->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
/* Only post clock-lost messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) &&
GST_AUDIO_CLOCK_CAST (src->clock)->func ==
(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element), src->clock));
gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
gst_ring_buffer_release (src->ringbuffer);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
GST_DEBUG_OBJECT (src, "READY->NULL");
gst_ring_buffer_close_device (src->ringbuffer);
GST_OBJECT_LOCK (src);
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
GST_OBJECT_UNLOCK (src);
break;
default:
break;
}
return ret;
/* ERRORS */
open_failed:
{
/* subclass must post a meaningfull error message */
GST_DEBUG_OBJECT (src, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
}
