static void
gst_dasf_src_init (GstDasfSrc* self, GstDasfSrcClass* klass)
{
GST_DEBUG ("");
GstBaseAudioSrc *baseaudiosrc;
baseaudiosrc = GST_BASE_AUDIO_SRC (self);
GstDasfSrcPrivate* priv = GST_DASF_SRC_GET_PRIVATE (self);
priv->num_output_buffers = NUM_OUTPUT_BUFFERS_DEFAULT;
priv->incount = 0;
priv->outcount = 0;
priv->volume = 100;
priv->mute = FALSE;
baseaudiosrc->clock = gst_audio_clock_new ("GstDasfSrcClock", (GstAudioClockGetTimeFunc) gst_dasf_src_get_time, self);
self->component = NULL;
self->tracks = NULL;
GstMixerTrack* track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
track->label = g_strdup ("DASF Volume");
track->num_channels = 1;
track->min_volume = 0;
track->max_volume = 100;
track->flags = GST_MIXER_TRACK_MASTER | GST_MIXER_TRACK_MUTE;
self->tracks = g_slist_append (self->tracks, track);
return;
}
static void
gst_base_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
src->buffer_time = g_value_get_int64 (value);
break;
case PROP_LATENCY_TIME:
src->latency_time = g_value_get_int64 (value);
break;
case PROP_PROVIDE_CLOCK:
gst_base_audio_src_set_provide_clock (src, g_value_get_boolean (value));
break;
case PROP_SLAVE_METHOD:
gst_base_audio_src_set_slave_method (src, g_value_get_enum (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
gboolean res;
res = TRUE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
GST_DEBUG_OBJECT (bsrc, "flush-start");
gst_ring_buffer_pause (src->ringbuffer);
gst_ring_buffer_clear_all (src->ringbuffer);
break;
case GST_EVENT_FLUSH_STOP:
GST_DEBUG_OBJECT (bsrc, "flush-stop");
/* always resync on sample after a flush */
src->next_sample = -1;
gst_ring_buffer_clear_all (src->ringbuffer);
break;
case GST_EVENT_SEEK:
GST_DEBUG_OBJECT (bsrc, "refuse to seek");
res = FALSE;
break;
default:
GST_DEBUG_OBJECT (bsrc, "dropping event %p", event);
break;
}
return res;
}
static void
gst_pulsesrc_init (GstPulseSrc * pulsesrc, GstPulseSrcClass * klass)
{
pulsesrc->server = NULL;
pulsesrc->device = NULL;
pulsesrc->device_description = NULL;
pulsesrc->context = NULL;
pulsesrc->stream = NULL;
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
#ifdef HAVE_PULSE_0_9_13
pa_sample_spec_init (&pulsesrc->sample_spec);
#else
pulsesrc->sample_spec.format = PA_SAMPLE_INVALID;
pulsesrc->sample_spec.rate = 0;
pulsesrc->sample_spec.channels = 0;
#endif
pulsesrc->operation_success = FALSE;
pulsesrc->paused = FALSE;
pulsesrc->in_read = FALSE;
pulsesrc->mixer = NULL;
pulsesrc->probe = gst_pulseprobe_new (G_OBJECT (pulsesrc), G_OBJECT_GET_CLASS (pulsesrc), PROP_DEVICE, pulsesrc->server, FALSE, TRUE); /* FALSE for sinks, TRUE for sources */
/* this should be the default but it isn't yet */
gst_base_audio_src_set_slave_method (GST_BASE_AUDIO_SRC (pulsesrc),
GST_BASE_AUDIO_SRC_SLAVE_SKEW);
}
static gboolean
gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
GstRingBufferSpec *spec;
spec = &src->ringbuffer->spec;
spec->buffer_time = src->buffer_time;
spec->latency_time = src->latency_time;
if (!gst_ring_buffer_parse_caps (spec, caps))
goto parse_error;
/* calculate suggested segsize and segtotal */
spec->segsize =
spec->rate * spec->bytes_per_sample * spec->latency_time / GST_MSECOND;
spec->segtotal = spec->buffer_time / spec->latency_time;
GST_DEBUG ("release old ringbuffer");
gst_ring_buffer_release (src->ringbuffer);
gst_ring_buffer_debug_spec_buff (spec);
GST_DEBUG ("acquire new ringbuffer");
if (!gst_ring_buffer_acquire (src->ringbuffer, spec))
goto acquire_error;
/* calculate actual latency and buffer times */
spec->latency_time =
spec->segsize * GST_MSECOND / (spec->rate * spec->bytes_per_sample);
spec->buffer_time =
spec->segtotal * spec->segsize * GST_MSECOND / (spec->rate *
spec->bytes_per_sample);
gst_ring_buffer_debug_spec_buff (spec);
g_object_notify (G_OBJECT (src), "actual-buffer-time");
g_object_notify (G_OBJECT (src), "actual-latency-time");
return TRUE;
/* ERRORS */
parse_error:
{
GST_DEBUG ("could not parse caps");
return FALSE;
}
acquire_error:
{
GST_DEBUG ("could not acquire ringbuffer");
return FALSE;
}
}
static gboolean
gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min_latency, max_latency;
GstRingBufferSpec *spec;
GST_OBJECT_LOCK (src);
if (G_UNLIKELY (src->ringbuffer == NULL
|| src->ringbuffer->spec.rate == 0)) {
GST_OBJECT_UNLOCK (src);
goto done;
}
spec = &src->ringbuffer->spec;
/* we have at least 1 segment of latency */
min_latency =
gst_util_uint64_scale_int (spec->segsize, GST_SECOND,
spec->rate * spec->bytes_per_sample);
/* we cannot delay more than the buffersize else we lose data */
max_latency =
gst_util_uint64_scale_int (spec->segtotal * spec->segsize, GST_SECOND,
spec->rate * spec->bytes_per_sample);
GST_OBJECT_UNLOCK (src);
GST_DEBUG_OBJECT (src,
"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
/* we are always live, the min latency is 1 segment and the max latency is
* the complete buffer of segments. */
gst_query_set_latency (query, TRUE, min_latency, max_latency);
res = TRUE;
break;
}
default:
res = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
break;
}
done:
return res;
}
static void
gst_base_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
g_value_set_int64 (value, src->buffer_time);
break;
case PROP_LATENCY_TIME:
g_value_set_int64 (value, src->latency_time);
break;
case PROP_ACTUAL_BUFFER_TIME:
GST_OBJECT_LOCK (src);
if (src->ringbuffer && src->ringbuffer->acquired)
g_value_set_int64 (value, src->ringbuffer->spec.buffer_time);
else
g_value_set_int64 (value, DEFAULT_ACTUAL_BUFFER_TIME);
GST_OBJECT_UNLOCK (src);
break;
case PROP_ACTUAL_LATENCY_TIME:
GST_OBJECT_LOCK (src);
if (src->ringbuffer && src->ringbuffer->acquired)
g_value_set_int64 (value, src->ringbuffer->spec.latency_time);
else
g_value_set_int64 (value, DEFAULT_ACTUAL_LATENCY_TIME);
GST_OBJECT_UNLOCK (src);
break;
case PROP_PROVIDE_CLOCK:
g_value_set_boolean (value, gst_base_audio_src_get_provide_clock (src));
break;
case PROP_SLAVE_METHOD:
g_value_set_enum (value, gst_base_audio_src_get_slave_method (src));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
gst_ring_buffer_pause (src->ringbuffer);
gst_ring_buffer_clear_all (src->ringbuffer);
break;
case GST_EVENT_FLUSH_STOP:
/* always resync on sample after a flush */
src->next_sample = -1;
gst_ring_buffer_clear_all (src->ringbuffer);
break;
default:
break;
}
return TRUE;
}
static void
gst_base_audio_src_dispose (GObject * object)
{
GstBaseAudioSrc *src;
src = GST_BASE_AUDIO_SRC (object);
GST_OBJECT_LOCK (src);
if (src->clock)
gst_object_unref (src->clock);
src->clock = NULL;
if (src->ringbuffer) {
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
}
GST_OBJECT_UNLOCK (src);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static GstClock *
gst_base_audio_src_provide_clock (GstElement * elem)
{
GstBaseAudioSrc *src;
GstClock *clock;
src = GST_BASE_AUDIO_SRC (elem);
/* we have no ringbuffer (must be NULL state) */
if (src->ringbuffer == NULL)
goto wrong_state;
if (!gst_ring_buffer_is_acquired (src->ringbuffer))
goto wrong_state;
GST_OBJECT_LOCK (src);
if (!src->priv->provide_clock)
goto clock_disabled;
clock = GST_CLOCK_CAST (gst_object_ref (src->clock));
GST_OBJECT_UNLOCK (src);
return clock;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (src, "ringbuffer not acquired");
return NULL;
}
clock_disabled:
{
GST_DEBUG_OBJECT (src, "clock provide disabled");
GST_OBJECT_UNLOCK (src);
return NULL;
}
}
static GstFlowReturn
gst_dasf_src_create (GstAudioSrc *audiosrc,
guint64 offset,
guint length,
GstBuffer **buffer)
{
GstDasfSrc* self = GST_DASF_SRC (audiosrc);
GstBaseAudioSrc *baseaudiosrc = GST_BASE_AUDIO_SRC (self);
GstBuffer* gst_buffer = NULL;
OMX_BUFFERHEADERTYPE* omx_buffer = NULL;
GstDasfSrcPrivate* priv = GST_DASF_SRC_GET_PRIVATE (self);
GstGooAudioFilter* me = self->peer_element;
GST_DEBUG ("");
if (me->component->cur_state != OMX_StateExecuting)
{
return GST_FLOW_UNEXPECTED;
}
GST_DEBUG ("goo stuff");
{
omx_buffer = goo_port_grab_buffer (me->outport);
if (gst_pad_alloc_buffer (GST_BASE_SRC_PAD (self),
priv->outcount,
omx_buffer->nFilledLen,
GST_PAD_CAPS (GST_BASE_SRC_PAD (self)),
&gst_buffer) == GST_FLOW_OK)
{
if (GST_IS_GOO_BUFFER (gst_buffer))
{
memcpy (GST_BUFFER_DATA (gst_buffer),
omx_buffer->pBuffer,
omx_buffer->nFilledLen);
goo_component_release_buffer (me->component,
omx_buffer);
}
else
{
gst_buffer_unref (gst_buffer);
gst_buffer = (GstBuffer*) gst_goo_buffer_new ();
gst_goo_buffer_set_data (gst_buffer,
me->component,
omx_buffer);
}
}
else
{
goto fail;
}
}
GST_DEBUG ("gst stuff");
{
GstClock* clock = NULL;
GstClockTime timestamp, duration;
clock = gst_element_get_clock (GST_ELEMENT (self));
timestamp = gst_clock_get_time (clock);
timestamp -= gst_element_get_base_time (GST_ELEMENT (self));
gst_object_unref (clock);
GST_BUFFER_TIMESTAMP (gst_buffer) = gst_util_uint64_scale_int (GST_SECOND, priv->outcount, 50);
/* Set 20 millisecond duration */
duration = gst_util_uint64_scale_int
(GST_SECOND,
1,
50);
GST_BUFFER_DURATION (gst_buffer) = duration;
GST_BUFFER_OFFSET (gst_buffer) = priv->outcount++;
GST_BUFFER_OFFSET_END (gst_buffer) = priv->outcount;
gst_buffer_set_caps (gst_buffer,
GST_PAD_CAPS (GST_BASE_SRC_PAD (self)));
}
beach:
*buffer = gst_buffer;
return GST_FLOW_OK;
fail:
if (G_LIKELY (*buffer))
{
gst_buffer_unref (*buffer);
}
return GST_FLOW_ERROR;
}
static GstFlowReturn
gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GstBuffer ** outbuf)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
GstBuffer *buf;
guchar *data;
guint samples, total_samples;
guint64 sample;
gint bps;
GstRingBuffer *ringbuffer;
GstRingBufferSpec *spec;
guint read;
GstClockTime timestamp, duration;
GstClock *clock;
ringbuffer = src->ringbuffer;
spec = &ringbuffer->spec;
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuffer)))
goto wrong_state;
bps = spec->bytes_per_sample;
if ((length == 0 && bsrc->blocksize == 0) || length == -1)
/* no length given, use the default segment size */
length = spec->segsize;
else
/* make sure we round down to an integral number of samples */
length -= length % bps;
/* figure out the offset in the ringbuffer */
if (G_UNLIKELY (offset != -1)) {
sample = offset / bps;
/* if a specific offset was given it must be the next sequential
* offset we expect or we fail for now. */
if (src->next_sample != -1 && sample != src->next_sample)
goto wrong_offset;
} else {
/* calculate the sequentially next sample we need to read. This can jump and
* create a DISCONT. */
sample = gst_base_audio_src_get_offset (src);
}
GST_DEBUG_OBJECT (src, "reading from sample %" G_GUINT64_FORMAT, sample);
/* get the number of samples to read */
total_samples = samples = length / bps;
/* FIXME, using a bufferpool would be nice here */
buf = gst_buffer_new_and_alloc (length);
data = GST_BUFFER_DATA (buf);
do {
read = gst_ring_buffer_read (ringbuffer, sample, data, samples);
GST_DEBUG_OBJECT (src, "read %u of %u", read, samples);
/* if we read all, we're done */
if (read == samples)
break;
/* else something interrupted us and we wait for playing again. */
GST_DEBUG_OBJECT (src, "wait playing");
if (gst_base_src_wait_playing (bsrc) != GST_FLOW_OK)
goto stopped;
GST_DEBUG_OBJECT (src, "continue playing");
/* read next samples */
sample += read;
samples -= read;
data += read * bps;
} while (TRUE);
/* mark discontinuity if needed */
if (G_UNLIKELY (sample != src->next_sample) && src->next_sample != -1) {
GST_WARNING_OBJECT (src,
"create DISCONT of %" G_GUINT64_FORMAT " samples at sample %"
G_GUINT64_FORMAT, sample - src->next_sample, sample);
GST_ELEMENT_WARNING (src, CORE, CLOCK,
(_("Can't record audio fast enough")),
("dropped %" G_GUINT64_FORMAT " samples", sample - src->next_sample));
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
}
src->next_sample = sample + samples;
/* get the normal timestamp to get the duration. */
timestamp = gst_util_uint64_scale_int (sample, GST_SECOND, spec->rate);
duration = gst_util_uint64_scale_int (src->next_sample, GST_SECOND,
spec->rate) - timestamp;
GST_OBJECT_LOCK (src);
if (!(clock = GST_ELEMENT_CLOCK (src)))
goto no_sync;
if (clock != src->clock) {
/* we are slaved, check how to handle this */
switch (src->priv->slave_method) {
case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE:
/* not implemented, use skew algorithm. This algorithm should
* work on the readout pointer and produces more or less samples based
* on the clock drift */
case GST_BASE_AUDIO_SRC_SLAVE_SKEW:
{
GstClockTime running_time;
GstClockTime base_time;
GstClockTime current_time;
guint64 running_time_sample;
gint running_time_segment;
gint current_segment;
gint segment_skew;
gint sps;
/* samples per segment */
sps = ringbuffer->samples_per_seg;
/* get the current time */
current_time = gst_clock_get_time (clock);
/* get the basetime */
base_time = GST_ELEMENT_CAST (src)->base_time;
/* get the running_time */
running_time = current_time - base_time;
/* the running_time converted to a sample (relative to the ringbuffer) */
running_time_sample =
gst_util_uint64_scale_int (running_time, spec->rate, GST_SECOND);
/* the segmentnr corrensponding to running_time, round down */
running_time_segment = running_time_sample / sps;
/* the segment currently read from the ringbuffer */
current_segment = sample / sps;
/* the skew we have between running_time and the ringbuffertime */
segment_skew = running_time_segment - current_segment;
GST_DEBUG_OBJECT (bsrc, "\n running_time = %" GST_TIME_FORMAT
"\n timestamp = %" GST_TIME_FORMAT
"\n running_time_segment = %d"
"\n current_segment = %d"
"\n segment_skew = %d",
GST_TIME_ARGS (running_time),
GST_TIME_ARGS (timestamp),
running_time_segment, current_segment, segment_skew);
/* Resync the ringbuffer if:
* 1. We get one segment into the future.
* This is clearly a lie, because we can't
* possibly have a buffer with timestamp 1 at
* time 0. (unless it has time-travelled...)
*
* 2. We are more than the length of the ringbuffer behind.
* The length of the ringbuffer then gets to dictate
* the threshold for what is concidered "too late"
*
* 3. If this is our first buffer.
* We know that we should catch up to running_time
* the first time we are ran.
*/
if ((segment_skew < 0) ||
(segment_skew >= ringbuffer->spec.segtotal) ||
(current_segment == 0)) {
gint segments_written;
gint first_segment;
gint last_segment;
gint new_last_segment;
gint segment_diff;
gint new_first_segment;
guint64 new_sample;
/* we are going to say that the last segment was captured at the current time
(running_time), minus one segment of creation-latency in the ringbuffer.
This can be thought of as: The segment arrived in the ringbuffer at time X, and
that means it was created at time X - (one segment). */
new_last_segment = running_time_segment - 1;
/* for better readablity */
first_segment = current_segment;
/* get the amount of segments written from the device by now */
segments_written = g_atomic_int_get (&ringbuffer->segdone);
/* subtract the base to segments_written to get the number of the
last written segment in the ringbuffer (one segment written = segment 0) */
last_segment = segments_written - ringbuffer->segbase - 1;
/* we see how many segments the ringbuffer was timeshifted */
segment_diff = new_last_segment - last_segment;
/* we move the first segment an equal amount */
new_first_segment = first_segment + segment_diff;
/* and we also move the segmentbase the same amount */
ringbuffer->segbase -= segment_diff;
/* we calculate the new sample value */
new_sample = ((guint64) new_first_segment) * sps;
/* and get the relative time to this -> our new timestamp */
timestamp =
gst_util_uint64_scale_int (new_sample, GST_SECOND, spec->rate);
/* we update the next sample accordingly */
src->next_sample = new_sample + samples;
GST_DEBUG_OBJECT (bsrc,
"Timeshifted the ringbuffer with %d segments: "
"Updating the timestamp to %" GST_TIME_FORMAT ", "
"and src->next_sample to %" G_GUINT64_FORMAT, segment_diff,
GST_TIME_ARGS (timestamp), src->next_sample);
}
break;
}
case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP:
{
GstClockTime base_time, latency;
/* We are slaved to another clock, take running time of the pipeline clock and
* timestamp against it. Somebody else in the pipeline should figure out the
* clock drift. We keep the duration we calculated above. */
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (src)->base_time;
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
/* subtract latency */
latency =
gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
if (timestamp > latency)
timestamp -= latency;
else
timestamp = 0;
}
case GST_BASE_AUDIO_SRC_SLAVE_NONE:
break;
}
} else {
GstClockTime base_time;
/* we are not slaved, subtract base_time */
base_time = GST_ELEMENT_CAST (src)->base_time;
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
}
no_sync:
GST_OBJECT_UNLOCK (src);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
GST_BUFFER_OFFSET (buf) = sample;
GST_BUFFER_OFFSET_END (buf) = sample + samples;
*outbuf = buf;
return GST_FLOW_OK;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (src, "ringbuffer in wrong state");
return GST_FLOW_WRONG_STATE;
}
wrong_offset:
{
GST_ELEMENT_ERROR (src, RESOURCE, SEEK,
(NULL), ("resource can only be operated on sequentially but offset %"
G_GUINT64_FORMAT " was given", offset));
return GST_FLOW_ERROR;
}
stopped:
{
gst_buffer_unref (buf);
GST_DEBUG_OBJECT (src, "ringbuffer stopped");
return GST_FLOW_WRONG_STATE;
}
}
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 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;
}
}
