static void
gst_pulsesrc_stream_latency_update_cb (pa_stream * s, void *userdata)
{
const pa_timing_info *info;
pa_usec_t source_usec;
info = pa_stream_get_timing_info (s);
if (!info) {
GST_LOG_OBJECT (GST_PULSESRC_CAST (userdata),
"latency update (information unknown)");
return;
}
#ifdef HAVE_PULSE_0_9_11
source_usec = info->configured_source_usec;
#else
source_usec = 0;
#endif
GST_LOG_OBJECT (GST_PULSESRC_CAST (userdata),
"latency_update, %" G_GUINT64_FORMAT ", %d:%" G_GINT64_FORMAT ", %d:%"
G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT,
GST_TIMEVAL_TO_TIME (info->timestamp), info->write_index_corrupt,
info->write_index, info->read_index_corrupt, info->read_index,
info->source_usec, source_usec);
}
static void
gst_pulsesrc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
switch (prop_id) {
case PROP_SERVER:
g_free (pulsesrc->server);
pulsesrc->server = g_value_dup_string (value);
if (pulsesrc->probe)
gst_pulseprobe_set_server (pulsesrc->probe, pulsesrc->server);
break;
case PROP_DEVICE:
g_free (pulsesrc->device);
pulsesrc->device = g_value_dup_string (value);
break;
case PROP_STREAM_PROPERTIES:
if (pulsesrc->properties)
gst_structure_free (pulsesrc->properties);
pulsesrc->properties =
gst_structure_copy (gst_value_get_structure (value));
if (pulsesrc->proplist)
pa_proplist_free (pulsesrc->proplist);
pulsesrc->proplist = gst_pulse_make_proplist (pulsesrc->properties);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesrc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
switch (prop_id) {
case PROP_SERVER:
g_value_set_string (value, pulsesrc->server);
break;
case PROP_DEVICE:
g_value_set_string (value, pulsesrc->device);
break;
case PROP_DEVICE_NAME:
g_value_take_string (value, gst_pulsesrc_device_description (pulsesrc));
break;
case PROP_STREAM_PROPERTIES:
gst_value_set_structure (value, pulsesrc->properties);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesrc_finalize (GObject * object)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
g_free (pulsesrc->server);
g_free (pulsesrc->device);
if (pulsesrc->properties)
gst_structure_free (pulsesrc->properties);
if (pulsesrc->proplist)
pa_proplist_free (pulsesrc->proplist);
if (pulsesrc->mixer) {
gst_pulsemixer_ctrl_free (pulsesrc->mixer);
pulsesrc->mixer = NULL;
}
if (pulsesrc->probe) {
gst_pulseprobe_free (pulsesrc->probe);
pulsesrc->probe = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_pulsesrc_success_cb (pa_stream * s, int success, void *userdata)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
pulsesrc->operation_success = !!success;
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
}
static gboolean
gst_pulsesrc_close (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
gst_pulsesrc_destroy_context (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
}
static void
gst_pulsesrc_stream_request_cb (pa_stream * s, size_t length, void *userdata)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
GST_LOG_OBJECT (pulsesrc, "got request for length %" G_GSIZE_FORMAT, length);
if (pulsesrc->in_read) {
/* only signal when reading */
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
}
}
static void
gst_pulsesrc_reset (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_operation *o = NULL;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
GST_DEBUG_OBJECT (pulsesrc, "reset");
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto unlock_and_fail;
if (!(o =
pa_stream_flush (pulsesrc->stream, gst_pulsesrc_success_cb,
pulsesrc))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_flush() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pulsesrc->paused = TRUE;
/* Inform anyone waiting in _write() call that it shall wakeup */
if (pulsesrc->in_read) {
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
}
pulsesrc->operation_success = FALSE;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto unlock_and_fail;
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
if (!pulsesrc->operation_success) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Flush failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
unlock_and_fail:
if (o) {
pa_operation_cancel (o);
pa_operation_unref (o);
}
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
}
static gboolean
gst_pulsesrc_interface_supported (GstImplementsInterface *
iface, GType interface_type)
{
GstPulseSrc *this = GST_PULSESRC_CAST (iface);
if (interface_type == GST_TYPE_MIXER && this->mixer)
return TRUE;
if (interface_type == GST_TYPE_PROPERTY_PROBE && this->probe)
return TRUE;
return FALSE;
}
static void
gst_pulsesrc_source_info_cb (pa_context * c, const pa_source_info * i, int eol,
void *userdata)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
if (!i)
goto done;
g_free (pulsesrc->device_description);
pulsesrc->device_description = g_strdup (i->description);
done:
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
}
static gboolean
gst_pulsesrc_unprepare (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
return TRUE;
}
static void
gst_pulsesrc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
switch (prop_id) {
case PROP_SERVER:
g_free (pulsesrc->server);
pulsesrc->server = g_value_dup_string (value);
if (pulsesrc->probe)
gst_pulseprobe_set_server (pulsesrc->probe, pulsesrc->server);
break;
case PROP_DEVICE:
g_free (pulsesrc->device);
pulsesrc->device = g_value_dup_string (value);
break;
case PROP_CLIENT_NAME:
g_free (pulsesrc->client_name);
if (!g_value_get_string (value)) {
GST_WARNING_OBJECT (pulsesrc,
"Empty PulseAudio client name not allowed. Resetting to default value");
pulsesrc->client_name = gst_pulse_client_name ();
} else
pulsesrc->client_name = g_value_dup_string (value);
break;
case PROP_STREAM_PROPERTIES:
if (pulsesrc->properties)
gst_structure_free (pulsesrc->properties);
pulsesrc->properties =
gst_structure_copy (gst_value_get_structure (value));
if (pulsesrc->proplist)
pa_proplist_free (pulsesrc->proplist);
pulsesrc->proplist = gst_pulse_make_proplist (pulsesrc->properties);
break;
case PROP_VOLUME:
gst_pulsesrc_set_stream_volume (pulsesrc, g_value_get_double (value));
break;
case PROP_MUTE:
gst_pulsesrc_set_stream_mute (pulsesrc, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesrc_source_info_cb (pa_context * c, const pa_source_info * i, int eol,
void *userdata)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
if (!i)
return;
if (!pulsesrc->stream)
return;
g_assert (i->index == pa_stream_get_device_index (pulsesrc->stream));
g_free (pulsesrc->device_description);
pulsesrc->device_description = g_strdup (i->description);
}
static void
gst_pulsesrc_stream_state_cb (pa_stream * s, void *userdata)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
switch (pa_stream_get_state (s)) {
case PA_STREAM_READY:
case PA_STREAM_FAILED:
case PA_STREAM_TERMINATED:
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
break;
case PA_STREAM_UNCONNECTED:
case PA_STREAM_CREATING:
break;
}
}
static void
gst_pulsesrc_context_state_cb (pa_context * c, void *userdata)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
switch (pa_context_get_state (c)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
break;
case PA_CONTEXT_UNCONNECTED:
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
}
}
static void
gst_pulsesrc_source_output_info_cb (pa_context * c,
const pa_source_output_info * i, int eol, void *userdata)
{
GstPulseSrc *psrc;
psrc = GST_PULSESRC_CAST (userdata);
if (!i)
goto done;
/* If the index doesn't match our current stream,
* it implies we just recreated the stream (caps change)
*/
if (i->index == psrc->source_output_idx) {
psrc->volume = pa_sw_volume_to_linear (pa_cvolume_max (&i->volume));
psrc->mute = i->mute;
}
done:
pa_threaded_mainloop_signal (psrc->mainloop, 0);
}
static void
gst_pulsesrc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
switch (prop_id) {
case PROP_SERVER:
g_value_set_string (value, pulsesrc->server);
break;
case PROP_DEVICE:
g_value_set_string (value, pulsesrc->device);
break;
case PROP_DEVICE_NAME:
g_value_take_string (value, gst_pulsesrc_device_description (pulsesrc));
break;
case PROP_CLIENT_NAME:
g_value_set_string (value, pulsesrc->client_name);
break;
case PROP_STREAM_PROPERTIES:
gst_value_set_structure (value, pulsesrc->properties);
break;
case PROP_SOURCE_OUTPUT_INDEX:
g_value_set_uint (value, pulsesrc->source_output_idx);
break;
case PROP_VOLUME:
g_value_set_double (value, gst_pulsesrc_get_stream_volume (pulsesrc));
break;
case PROP_MUTE:
g_value_set_boolean (value, gst_pulsesrc_get_stream_mute (pulsesrc));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* return the delay in samples */
static guint
gst_pulsesrc_delay (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_usec_t t;
int negative, res;
guint result;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto server_dead;
/* get the latency, this can fail when we don't have a latency update yet.
* We don't want to wait for latency updates here but we just return 0. */
res = pa_stream_get_latency (pulsesrc->stream, &t, &negative);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
if (res > 0) {
GST_DEBUG_OBJECT (pulsesrc, "could not get latency");
result = 0;
} else {
if (negative)
result = 0;
else
result = (guint) ((t * pulsesrc->sample_spec.rate) / 1000000LL);
}
return result;
/* ERRORS */
server_dead:
{
GST_DEBUG_OBJECT (pulsesrc, "the server is dead");
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return 0;
}
}
static GstStateChangeReturn
gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstPulseSrc *this = GST_PULSESRC_CAST (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!(this->mainloop = pa_threaded_mainloop_new ()))
goto mainloop_failed;
if (pa_threaded_mainloop_start (this->mainloop) < 0) {
pa_threaded_mainloop_free (this->mainloop);
this->mainloop = NULL;
goto mainloop_start_failed;
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
GST_AUDIO_BASE_SRC (this)->clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* uncork and start recording */
gst_pulsesrc_play (this);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* stop recording ASAP by corking */
pa_threaded_mainloop_lock (this->mainloop);
GST_DEBUG_OBJECT (this, "corking");
gst_pulsesrc_set_corked (this, TRUE, FALSE);
pa_threaded_mainloop_unlock (this->mainloop);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* now make sure we get out of the _read method */
gst_pulsesrc_pause (this);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (this->mainloop)
pa_threaded_mainloop_stop (this->mainloop);
gst_pulsesrc_destroy_context (this);
if (this->mainloop) {
pa_threaded_mainloop_free (this->mainloop);
this->mainloop = NULL;
}
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* format_lost is reset in release() in baseaudiosink */
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
GST_AUDIO_BASE_SRC (this)->clock));
break;
default:
break;
}
return ret;
/* ERRORS */
mainloop_failed:
{
GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
("pa_threaded_mainloop_new() failed"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
mainloop_start_failed:
{
GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
("pa_threaded_mainloop_start() failed"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
}
static guint
gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
size_t sum = 0;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
pulsesrc->in_read = TRUE;
if (pulsesrc->paused)
goto was_paused;
while (length > 0) {
size_t l;
GST_LOG_OBJECT (pulsesrc, "reading %u bytes", length);
/*check if we have a leftover buffer */
if (!pulsesrc->read_buffer) {
for (;;) {
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto unlock_and_fail;
/* read all available data, we keep a pointer to the data and the length
* and take from it what we need. */
if (pa_stream_peek (pulsesrc->stream, &pulsesrc->read_buffer,
&pulsesrc->read_buffer_length) < 0)
goto peek_failed;
GST_LOG_OBJECT (pulsesrc, "have data of %" G_GSIZE_FORMAT " bytes",
pulsesrc->read_buffer_length);
/* if we have data, process if */
if (pulsesrc->read_buffer && pulsesrc->read_buffer_length)
break;
/* now wait for more data to become available */
GST_LOG_OBJECT (pulsesrc, "waiting for data");
pa_threaded_mainloop_wait (pulsesrc->mainloop);
if (pulsesrc->paused)
goto was_paused;
}
}
l = pulsesrc->read_buffer_length >
length ? length : pulsesrc->read_buffer_length;
memcpy (data, pulsesrc->read_buffer, l);
pulsesrc->read_buffer = (const guint8 *) pulsesrc->read_buffer + l;
pulsesrc->read_buffer_length -= l;
data = (guint8 *) data + l;
length -= l;
sum += l;
if (pulsesrc->read_buffer_length <= 0) {
/* we copied all of the data, drop it now */
if (pa_stream_drop (pulsesrc->stream) < 0)
goto drop_failed;
/* reset pointer to data */
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
}
}
pulsesrc->in_read = FALSE;
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return sum;
/* ERRORS */
was_paused:
{
GST_LOG_OBJECT (pulsesrc, "we are paused");
goto unlock_and_fail;
}
peek_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_peek() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
drop_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_drop() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
unlock_and_fail:
{
pulsesrc->in_read = FALSE;
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return (guint) - 1;
}
}
static GstStateChangeReturn
gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstPulseSrc *this = GST_PULSESRC_CAST (element);
int e;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
this->mainloop = pa_threaded_mainloop_new ();
g_assert (this->mainloop);
e = pa_threaded_mainloop_start (this->mainloop);
g_assert (e == 0);
if (!this->mixer)
this->mixer =
gst_pulsemixer_ctrl_new (G_OBJECT (this), this->server,
this->device, GST_PULSEMIXER_SOURCE);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* uncork and start recording */
gst_pulsesrc_play (this);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* stop recording ASAP by corking */
pa_threaded_mainloop_lock (this->mainloop);
GST_DEBUG_OBJECT (this, "corking");
gst_pulsesrc_set_corked (this, TRUE, FALSE);
pa_threaded_mainloop_unlock (this->mainloop);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* now make sure we get out of the _read method */
gst_pulsesrc_pause (this);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (this->mixer) {
gst_pulsemixer_ctrl_free (this->mixer);
this->mixer = NULL;
}
if (this->mainloop)
pa_threaded_mainloop_stop (this->mainloop);
gst_pulsesrc_destroy_context (this);
if (this->mainloop) {
pa_threaded_mainloop_free (this->mainloop);
this->mainloop = NULL;
}
break;
default:
break;
}
return ret;
}
/* This is essentially gst_base_src_negotiate_default() but the caps
* are guaranteed to have a channel layout for > 2 channels
*/
static gboolean
gst_pulsesrc_negotiate (GstBaseSrc * basesrc)
{
GstCaps *thiscaps;
GstCaps *caps = NULL;
GstCaps *peercaps = NULL;
gboolean result = FALSE;
/* first see what is possible on our source pad */
thiscaps = gst_pad_get_caps_reffed (GST_BASE_SRC_PAD (basesrc));
GST_DEBUG_OBJECT (basesrc, "caps of src: %" GST_PTR_FORMAT, thiscaps);
/* nothing or anything is allowed, we're done */
if (thiscaps == NULL || gst_caps_is_any (thiscaps))
goto no_nego_needed;
/* get the peer caps */
peercaps = gst_pad_peer_get_caps_reffed (GST_BASE_SRC_PAD (basesrc));
GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps);
if (peercaps) {
/* get intersection */
caps = gst_caps_intersect (thiscaps, peercaps);
GST_DEBUG_OBJECT (basesrc, "intersect: %" GST_PTR_FORMAT, caps);
gst_caps_unref (thiscaps);
gst_caps_unref (peercaps);
} else {
/* no peer, work with our own caps then */
caps = thiscaps;
}
if (caps) {
/* take first (and best, since they are sorted) possibility */
caps = gst_caps_make_writable (caps);
gst_caps_truncate (caps);
/* now fixate */
if (!gst_caps_is_empty (caps)) {
gst_pad_fixate_caps (GST_BASE_SRC_PAD (basesrc), caps);
GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps);
if (gst_caps_is_any (caps)) {
/* hmm, still anything, so element can do anything and
* nego is not needed */
result = TRUE;
} else if (gst_caps_is_fixed (caps)) {
/* yay, fixed caps, use those then */
result = gst_pulsesrc_create_stream (GST_PULSESRC_CAST (basesrc), caps);
if (result)
result = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
}
}
gst_caps_unref (caps);
}
return result;
no_nego_needed:
{
GST_DEBUG_OBJECT (basesrc, "no negotiation needed");
if (thiscaps)
gst_caps_unref (thiscaps);
return TRUE;
}
}
static gboolean
gst_pulsesrc_open (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
gchar *name = gst_pulse_client_name ();
pa_threaded_mainloop_lock (pulsesrc->mainloop);
g_assert (!pulsesrc->context);
g_assert (!pulsesrc->stream);
GST_DEBUG_OBJECT (pulsesrc, "opening device");
if (!(pulsesrc->context =
pa_context_new (pa_threaded_mainloop_get_api (pulsesrc->mainloop),
name))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to create context"),
(NULL));
goto unlock_and_fail;
}
pa_context_set_state_callback (pulsesrc->context,
gst_pulsesrc_context_state_cb, pulsesrc);
GST_DEBUG_OBJECT (pulsesrc, "connect to server %s",
GST_STR_NULL (pulsesrc->server));
if (pa_context_connect (pulsesrc->context, pulsesrc->server, 0, NULL) < 0) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
for (;;) {
pa_context_state_t state;
state = pa_context_get_state (pulsesrc->context);
if (!PA_CONTEXT_IS_GOOD (state)) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
if (state == PA_CONTEXT_READY)
break;
/* Wait until the context is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
GST_DEBUG_OBJECT (pulsesrc, "connected");
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
g_free (name);
return TRUE;
/* ERRORS */
unlock_and_fail:
{
gst_pulsesrc_destroy_context (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
g_free (name);
return FALSE;
}
}
static void
gst_pulsesrc_stream_overflow_cb (pa_stream * s, void *userdata)
{
GST_WARNING_OBJECT (GST_PULSESRC_CAST (userdata), "Got overflow");
}
static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
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);
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
PA_STREAM_INTERPOLATE_TIMING |
PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_NOT_MONOTONOUS |
#ifdef HAVE_PULSE_0_9_11
PA_STREAM_ADJUST_LATENCY |
#endif
PA_STREAM_START_CORKED) < 0) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pulsesrc->corked = TRUE;
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state (pulsesrc->stream);
if (!PA_STREAM_IS_GOOD (state)) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
if (state == PA_STREAM_READY)
break;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
/* 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;
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
unlock_and_fail:
{
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return FALSE;
}
}
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;
}
}
