* <title>Example pipelines</title>
* |[
* gst-launch -v pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
- * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis.
+ * ]| Record from a sound card using pulseaudio and encode to Ogg/Vorbis.
* </refsect2>
*/
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
+#define DEFAULT_SERVER NULL
+#define DEFAULT_DEVICE NULL
+#define DEFAULT_DEVICE_NAME NULL
+
enum
{
- PROP_SERVER = 1,
+ PROP_0,
+ PROP_SERVER,
PROP_DEVICE,
- PROP_DEVICE_NAME
+ PROP_DEVICE_NAME,
+ PROP_CLIENT,
+ PROP_STREAM_PROPERTIES,
+ PROP_SOURCE_OUTPUT_INDEX,
+ PROP_LAST
};
static void gst_pulsesrc_destroy_stream (GstPulseSrc * pulsesrc);
-
static void gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc);
static void gst_pulsesrc_set_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_pulsesrc_finalize (GObject * object);
+static gboolean gst_pulsesrc_set_corked (GstPulseSrc * psrc, gboolean corked,
+ gboolean wait);
static gboolean gst_pulsesrc_open (GstAudioSrc * asrc);
static gboolean gst_pulsesrc_close (GstAudioSrc * asrc);
static gboolean gst_pulsesrc_prepare (GstAudioSrc * asrc,
- GstRingBufferSpec * spec);
+ GstAudioRingBufferSpec * spec);
static gboolean gst_pulsesrc_unprepare (GstAudioSrc * asrc);
static GstStateChangeReturn gst_pulsesrc_change_state (GstElement *
element, GstStateChange transition);
-static void gst_pulsesrc_init_interfaces (GType type);
-
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
-# define ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
+# define FORMATS "{ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, U8 }"
#else
-# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
+# define FORMATS "{ S16BE, S16LE, F32BE, F32LE, S32BE, S32LE, U8 }"
#endif
-GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS (GstPulseSrc, gst_pulsesrc);
-GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseSrc, gst_pulsesrc);
-GST_BOILERPLATE_FULL (GstPulseSrc, gst_pulsesrc, GstAudioSrc,
- GST_TYPE_AUDIO_SRC, gst_pulsesrc_init_interfaces);
-
-static gboolean
-gst_pulsesrc_interface_supported (GstImplementsInterface *
- iface, GType interface_type)
-{
- GstPulseSrc *this = GST_PULSESRC (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 GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " FORMATS ", "
+ "rate = (int) [ 1, MAX ], "
+ "channels = (int) [ 1, 32 ];"
+ "audio/x-alaw, "
+ "rate = (int) [ 1, MAX], "
+ "channels = (int) [ 1, 32 ];"
+ "audio/x-mulaw, "
+ "rate = (int) [ 1, MAX], " "channels = (int) [ 1, 32 ]")
+ );
-static void
-gst_pulsesrc_implements_interface_init (GstImplementsInterfaceClass * klass)
-{
- klass->supported = gst_pulsesrc_interface_supported;
-}
-
-static void
-gst_pulsesrc_init_interfaces (GType type)
-{
- static const GInterfaceInfo implements_iface_info = {
- (GInterfaceInitFunc) gst_pulsesrc_implements_interface_init,
- NULL,
- NULL,
- };
- static const GInterfaceInfo mixer_iface_info = {
- (GInterfaceInitFunc) gst_pulsesrc_mixer_interface_init,
- NULL,
- NULL,
- };
- static const GInterfaceInfo probe_iface_info = {
- (GInterfaceInitFunc) gst_pulsesrc_property_probe_interface_init,
- NULL,
- NULL,
- };
-
- g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
- &implements_iface_info);
- g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
- g_type_add_interface_static (type, GST_TYPE_PROPERTY_PROBE,
- &probe_iface_info);
-}
-static void
-gst_pulsesrc_base_init (gpointer g_class)
-{
+GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS (GstPulseSrc, gst_pulsesrc);
+GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseSrc, gst_pulsesrc);
- static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) { " ENDIANNESS " }, "
- "signed = (boolean) TRUE, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-raw-float, "
- "endianness = (int) { " ENDIANNESS " }, "
- "width = (int) 32, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-raw-int, "
- "endianness = (int) { " ENDIANNESS " }, "
- "signed = (boolean) TRUE, "
- "width = (int) 32, "
- "depth = (int) 32, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-raw-int, "
- "signed = (boolean) FALSE, "
- "width = (int) 8, "
- "depth = (int) 8, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-alaw, "
- "rate = (int) [ 1, MAX], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-mulaw, "
- "rate = (int) [ 1, MAX], " "channels = (int) [ 1, 32 ]")
- );
-
- GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_set_details_simple (element_class,
- "PulseAudio Audio Source",
- "Source/Audio",
- "Captures audio from a PulseAudio server", "Lennart Poettering");
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&pad_template));
-}
+#define gst_pulsesrc_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstPulseSrc, gst_pulsesrc, GST_TYPE_AUDIO_SRC,
+ G_IMPLEMENT_INTERFACE (GST_TYPE_MIXER, gst_pulsesrc_mixer_interface_init);
+ G_IMPLEMENT_INTERFACE (GST_TYPE_PROPERTY_PROBE,
+ gst_pulsesrc_property_probe_interface_init));
static void
gst_pulsesrc_class_init (GstPulseSrcClass * klass)
GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass);
GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+ gchar *clientname;
- gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulsesrc_finalize);
- gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsesrc_set_property);
- gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_pulsesrc_get_property);
+ gobject_class->finalize = gst_pulsesrc_finalize;
+ gobject_class->set_property = gst_pulsesrc_set_property;
+ gobject_class->get_property = gst_pulsesrc_get_property;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_pulsesrc_change_state);
g_object_class_install_property (gobject_class,
PROP_SERVER,
g_param_spec_string ("server", "Server",
- "The PulseAudio server to connect to", NULL,
+ "The PulseAudio server to connect to", DEFAULT_SERVER,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
g_object_class_install_property (gobject_class, PROP_DEVICE,
- g_param_spec_string ("device", "Source",
- "The PulseAudio source device to connect to", NULL,
+ g_param_spec_string ("device", "Device",
+ "The PulseAudio source device to connect to", DEFAULT_DEVICE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
- "Human-readable name of the sound device", NULL,
+ "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+ clientname = gst_pulse_client_name ();
+ /**
+ * GstPulseSink:client
+ *
+ * The PulseAudio client name to use.
+ *
+ * Since: 0.10.27
+ */
+ g_object_class_install_property (gobject_class,
+ PROP_CLIENT,
+ g_param_spec_string ("client", "Client",
+ "The PulseAudio client_name_to_use", clientname,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+ GST_PARAM_MUTABLE_READY));
+ g_free (clientname);
+
+ /**
+ * GstPulseSrc:stream-properties
+ *
+ * List of pulseaudio stream properties. A list of defined properties can be
+ * found in the <ulink href="http://0pointer.de/lennart/projects/pulseaudio/doxygen/proplist_8h.html">pulseaudio api docs</ulink>.
+ *
+ * Below is an example for registering as a music application to pulseaudio.
+ * |[
+ * GstStructure *props;
+ *
+ * props = gst_structure_from_string ("props,media.role=music", NULL);
+ * g_object_set (pulse, "stream-properties", props, NULL);
+ * gst_structure_free (props);
+ * ]|
+ *
+ * Since: 0.10.26
+ */
+ g_object_class_install_property (gobject_class,
+ PROP_STREAM_PROPERTIES,
+ g_param_spec_boxed ("stream-properties", "stream properties",
+ "list of pulseaudio stream properties",
+ GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ /**
+ * GstPulseSrc:source-output-index
+ *
+ * The index of the PulseAudio source output corresponding to this element.
+ *
+ * Since: 0.10.31
+ */
+ g_object_class_install_property (gobject_class,
+ PROP_SOURCE_OUTPUT_INDEX,
+ g_param_spec_uint ("source-output-index", "source output index",
+ "The index of the PulseAudio source output corresponding to this "
+ "record stream", 0, G_MAXUINT, PA_INVALID_INDEX,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+ gst_element_class_set_details_simple (gstelement_class,
+ "PulseAudio Audio Source",
+ "Source/Audio",
+ "Captures audio from a PulseAudio server", "Lennart Poettering");
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&pad_template));
}
static void
-gst_pulsesrc_init (GstPulseSrc * pulsesrc, GstPulseSrcClass * klass)
+gst_pulsesrc_init (GstPulseSrc * pulsesrc)
{
- int e;
-
- pulsesrc->server = pulsesrc->device = pulsesrc->device_description = NULL;
+ pulsesrc->server = NULL;
+ pulsesrc->device = NULL;
+ pulsesrc->client_name = gst_pulse_client_name ();
+ pulsesrc->device_description = NULL;
pulsesrc->context = NULL;
pulsesrc->stream = NULL;
+ pulsesrc->stream_connected = FALSE;
+ pulsesrc->source_output_idx = PA_INVALID_INDEX;
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
-#if 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->did_reset = FALSE;
+ pulsesrc->paused = TRUE;
pulsesrc->in_read = FALSE;
- pulsesrc->mainloop = pa_threaded_mainloop_new ();
- g_assert (pulsesrc->mainloop);
+ pulsesrc->mixer = NULL;
- e = pa_threaded_mainloop_start (pulsesrc->mainloop);
- g_assert (e == 0);
+ pulsesrc->properties = NULL;
+ pulsesrc->proplist = NULL;
- 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 */
- pulsesrc->probe = gst_pulseprobe_new (G_OBJECT (pulsesrc), G_OBJECT_GET_CLASS (pulsesrc), PROP_DEVICE, pulsesrc->device, 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 void
pa_stream_disconnect (pulsesrc->stream);
pa_stream_unref (pulsesrc->stream);
pulsesrc->stream = NULL;
+ pulsesrc->stream_connected = FALSE;
+ pulsesrc->source_output_idx = PA_INVALID_INDEX;
+ g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
}
g_free (pulsesrc->device_description);
static void
gst_pulsesrc_finalize (GObject * object)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (object);
-
- pa_threaded_mainloop_stop (pulsesrc->mainloop);
-
- gst_pulsesrc_destroy_context (pulsesrc);
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
g_free (pulsesrc->server);
g_free (pulsesrc->device);
+ g_free (pulsesrc->client_name);
- pa_threaded_mainloop_free (pulsesrc->mainloop);
+ 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);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
+#define CONTEXT_OK(c) ((c) && PA_CONTEXT_IS_GOOD (pa_context_get_state ((c))))
+#define STREAM_OK(s) ((s) && PA_STREAM_IS_GOOD (pa_stream_get_state ((s))))
+
static gboolean
-gst_pulsesrc_is_dead (GstPulseSrc * pulsesrc)
+gst_pulsesrc_is_dead (GstPulseSrc * pulsesrc, gboolean check_stream)
{
+ if (!CONTEXT_OK (pulsesrc->context))
+ goto error;
+
+ if (check_stream && !STREAM_OK (pulsesrc->stream))
+ goto error;
+
+ return FALSE;
- if (!pulsesrc->context
- || !PA_CONTEXT_IS_GOOD (pa_context_get_state (pulsesrc->context))
- || !pulsesrc->stream
- || !PA_STREAM_IS_GOOD (pa_stream_get_state (pulsesrc->stream))) {
+error:
+ {
const gchar *err_str = pulsesrc->context ?
pa_strerror (pa_context_errno (pulsesrc->context)) : NULL;
-
GST_ELEMENT_ERROR ((pulsesrc), RESOURCE, FAILED, ("Disconnected: %s",
err_str), (NULL));
return TRUE;
}
-
- return FALSE;
-}
-
-
-static void
-gst_pulsesrc_set_property (GObject * object,
- guint prop_id, const GValue * value, GParamSpec * pspec)
-{
-
- GstPulseSrc *pulsesrc = GST_PULSESRC (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;
-
- 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 (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));
+ goto done;
g_free (pulsesrc->device_description);
pulsesrc->device_description = g_strdup (i->description);
+
+done:
+ pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
}
static gchar *
pa_operation *o = NULL;
gchar *t;
- pa_threaded_mainloop_lock (pulsesrc->mainloop);
+ if (!pulsesrc->mainloop)
+ goto no_mainloop;
- if (!pulsesrc->stream)
- goto unlock;
+ pa_threaded_mainloop_lock (pulsesrc->mainloop);
- if (!(o = pa_context_get_source_info_by_index (pulsesrc->context,
- pa_stream_get_device_index (pulsesrc->stream),
- gst_pulsesrc_source_info_cb, pulsesrc))) {
+ if (!(o = pa_context_get_source_info_by_name (pulsesrc->context,
+ pulsesrc->device, gst_pulsesrc_source_info_cb, pulsesrc))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_get_source_info() failed: %s",
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
- if (gst_pulsesrc_is_dead (pulsesrc))
+ if (gst_pulsesrc_is_dead (pulsesrc, FALSE))
goto unlock;
pa_threaded_mainloop_wait (pulsesrc->mainloop);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return t;
+
+no_mainloop:
+ {
+ GST_DEBUG_OBJECT (pulsesrc, "have no mainloop");
+ return NULL;
+ }
+}
+
+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:
+ 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;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
}
static void
guint prop_id, GValue * value, GParamSpec * pspec)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (object);
+ 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:{
- char *t = gst_pulsesrc_device_description (pulsesrc);
- g_value_set_string (value, t);
- g_free (t);
+ case PROP_DEVICE_NAME:
+ g_value_take_string (value, gst_pulsesrc_device_description (pulsesrc));
+ break;
+ case PROP_CLIENT:
+ 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;
- }
-
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
static void
gst_pulsesrc_context_state_cb (pa_context * c, void *userdata)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
switch (pa_context_get_state (c)) {
case PA_CONTEXT_READY:
static void
gst_pulsesrc_stream_state_cb (pa_stream * s, void *userdata)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
switch (pa_stream_get_state (s)) {
static void
gst_pulsesrc_stream_request_cb (pa_stream * s, size_t length, void *userdata)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
- pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
+ 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_stream_latency_update_cb (pa_stream * s, void *userdata)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);
+ const pa_timing_info *info;
+ pa_usec_t source_usec;
- pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
+ info = pa_stream_get_timing_info (s);
+
+ if (!info) {
+ GST_LOG_OBJECT (GST_PULSESRC_CAST (userdata),
+ "latency update (information unknown)");
+ return;
+ }
+ source_usec = info->configured_source_usec;
+
+ 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_stream_underflow_cb (pa_stream * s, void *userdata)
+{
+ GST_WARNING_OBJECT (GST_PULSESRC_CAST (userdata), "Got underflow");
+}
+
+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_open (GstAudioSrc * asrc)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
- gchar *name = gst_pulse_client_name ();
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
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))) {
+ pulsesrc->client_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));
/* 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);
- gst_pulsesrc_destroy_context (pulsesrc);
-
- pa_threaded_mainloop_unlock (pulsesrc->mainloop);
+ pa_threaded_mainloop_unlock (pulsesrc->mainloop);
- g_free (name);
- return FALSE;
+ return FALSE;
+ }
}
static gboolean
gst_pulsesrc_close (GstAudioSrc * asrc)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
gst_pulsesrc_destroy_context (pulsesrc);
static gboolean
gst_pulsesrc_unprepare (GstAudioSrc * asrc)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
gst_pulsesrc_destroy_stream (pulsesrc);
static guint
gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
+ 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;
- if (!pulsesrc->read_buffer) {
+ 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))
+ 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) {
- GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
- ("pa_stream_peek() failed: %s",
- pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
- goto unlock_and_fail;
- }
+ &pulsesrc->read_buffer_length) < 0)
+ goto peek_failed;
- if (pulsesrc->read_buffer)
- break;
+ GST_LOG_OBJECT (pulsesrc, "have data of %" G_GSIZE_FORMAT " bytes",
+ pulsesrc->read_buffer_length);
- if (pulsesrc->did_reset)
- goto unlock_and_fail;
+ /* 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;
}
}
- g_assert (pulsesrc->read_buffer && pulsesrc->read_buffer_length);
-
l = pulsesrc->read_buffer_length >
length ? length : pulsesrc->read_buffer_length;
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;
- if (pa_stream_drop (pulsesrc->stream) < 0) {
- GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
- ("pa_stream_drop() failed: %s",
- pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
- goto unlock_and_fail;
- }
-
+ /* reset pointer to data */
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
}
}
- pulsesrc->did_reset = FALSE;
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);
- pulsesrc->did_reset = FALSE;
- pulsesrc->in_read = FALSE;
-
- pa_threaded_mainloop_unlock (pulsesrc->mainloop);
- return (guint) - 1;
+ return (guint) - 1;
+ }
}
+/* return the delay in samples */
static guint
gst_pulsesrc_delay (GstAudioSrc * asrc)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
-
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_usec_t t;
-
- int negative;
+ int negative, res;
+ guint result;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
+ if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
+ goto server_dead;
- for (;;) {
- if (gst_pulsesrc_is_dead (pulsesrc))
- goto unlock_and_fail;
-
- if (pa_stream_get_latency (pulsesrc->stream, &t, &negative) >= 0)
- break;
-
- if (pa_context_errno (pulsesrc->context) != PA_ERR_NODATA) {
- GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
- ("pa_stream_get_latency() failed: %s",
- pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
- goto unlock_and_fail;
- }
-
- pa_threaded_mainloop_wait (pulsesrc->mainloop);
- }
-
- if (negative)
- t = 0;
+ /* 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);
- return (guint) ((t * pulsesrc->sample_spec.rate) / 1000000LL);
-
-unlock_and_fail:
+ 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;
- pa_threaded_mainloop_unlock (pulsesrc->mainloop);
- return 0;
+ /* ERRORS */
+server_dead:
+ {
+ GST_DEBUG_OBJECT (pulsesrc, "the server is dead");
+ pa_threaded_mainloop_unlock (pulsesrc->mainloop);
+ return 0;
+ }
}
static gboolean
pa_channel_map channel_map;
GstStructure *s;
gboolean need_channel_layout = FALSE;
- GstRingBufferSpec spec;
+ GstAudioRingBufferSpec spec;
+ const gchar *name;
- memset (&spec, 0, sizeof (GstRingBufferSpec));
+ memset (&spec, 0, sizeof (GstAudioRingBufferSpec));
spec.latency_time = GST_SECOND;
- if (!gst_ring_buffer_parse_caps (&spec, caps)) {
- GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
- ("Can't parse caps."), (NULL));
- goto fail;
- }
+ if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
+ goto invalid_caps;
+
/* Keep the refcount of the caps at 1 to make them writable */
gst_caps_unref (spec.caps);
- if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec)) {
- GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
- ("Invalid sample specification."), (NULL));
- goto fail;
- }
+ if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec))
+ goto invalid_spec;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
- if (!pulsesrc->context) {
- GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context"), (NULL));
- goto unlock_and_fail;
- }
+ if (!pulsesrc->context)
+ goto bad_context;
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_has_field (s, "channel-layout") ||
!gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
- if (spec.channels == 1)
+ if (spec.info.channels == 1)
pa_channel_map_init_mono (&channel_map);
- else if (spec.channels == 2)
+ else if (spec.info.channels == 2)
pa_channel_map_init_stereo (&channel_map);
else
need_channel_layout = TRUE;
}
- if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
- "Record Stream",
- &pulsesrc->sample_spec,
- (need_channel_layout) ? NULL : &channel_map))) {
- GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
- ("Failed to create stream: %s",
- pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
- goto unlock_and_fail;
- }
+ name = "Record Stream";
+ if (pulsesrc->proplist) {
+ if (!(pulsesrc->stream = pa_stream_new_with_proplist (pulsesrc->context,
+ name, &pulsesrc->sample_spec,
+ (need_channel_layout) ? NULL : &channel_map,
+ pulsesrc->proplist)))
+ goto create_failed;
+
+ } else if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
+ name, &pulsesrc->sample_spec,
+ (need_channel_layout) ? NULL : &channel_map)))
+ goto create_failed;
if (need_channel_layout) {
const pa_channel_map *m = pa_stream_get_channel_map (pulsesrc->stream);
pulsesrc);
pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
pulsesrc);
+ pa_stream_set_underflow_callback (pulsesrc->stream,
+ gst_pulsesrc_stream_underflow_cb, pulsesrc);
+ pa_stream_set_overflow_callback (pulsesrc->stream,
+ gst_pulsesrc_stream_overflow_cb, pulsesrc);
pa_stream_set_latency_update_callback (pulsesrc->stream,
gst_pulsesrc_stream_latency_update_cb, pulsesrc);
return TRUE;
+ /* ERRORS */
+invalid_caps:
+ {
+ GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
+ ("Can't parse caps."), (NULL));
+ goto fail;
+ }
+invalid_spec:
+ {
+ GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
+ ("Invalid sample specification."), (NULL));
+ goto fail;
+ }
+bad_context:
+ {
+ GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context"), (NULL));
+ goto unlock_and_fail;
+ }
+create_failed:
+ {
+ GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
+ ("Failed to create stream: %s",
+ pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
+ goto unlock_and_fail;
+ }
unlock_and_fail:
- gst_pulsesrc_destroy_stream (pulsesrc);
+ {
+ gst_pulsesrc_destroy_stream (pulsesrc);
- pa_threaded_mainloop_unlock (pulsesrc->mainloop);
+ pa_threaded_mainloop_unlock (pulsesrc->mainloop);
-fail:
- return FALSE;
+ fail:
+ return FALSE;
+ }
}
/* This is essentially gst_base_src_negotiate_default() but the caps
static gboolean
gst_pulsesrc_negotiate (GstBaseSrc * basesrc)
{
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (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 (GST_BASE_SRC_PAD (basesrc));
+ thiscaps = gst_pad_get_caps (GST_BASE_SRC_PAD (basesrc), NULL);
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 (GST_BASE_SRC_PAD (basesrc));
+ peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc), NULL);
GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps);
if (peercaps) {
- GstCaps *icaps;
-
/* get intersection */
- icaps = gst_caps_intersect (thiscaps, peercaps);
- GST_DEBUG_OBJECT (basesrc, "intersect: %" GST_PTR_FORMAT, icaps);
+ caps = gst_caps_intersect (thiscaps, peercaps);
+ GST_DEBUG_OBJECT (basesrc, "intersect: %" GST_PTR_FORMAT, caps);
gst_caps_unref (thiscaps);
gst_caps_unref (peercaps);
- if (icaps) {
- /* take first (and best, since they are sorted) possibility */
- caps = gst_caps_copy_nth (icaps, 0);
- gst_caps_unref (icaps);
- }
} 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_BASE_SRC_CLASS (parent_class)->fixate (basesrc, caps);
GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps);
if (gst_caps_is_any (caps)) {
result = TRUE;
} else if (gst_caps_is_fixed (caps)) {
/* yay, fixed caps, use those then */
- result = gst_pulsesrc_create_stream (GST_PULSESRC (basesrc), caps);
+ result = gst_pulsesrc_create_stream (pulsesrc, caps);
if (result)
- gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
- result = TRUE;
+ result = gst_base_src_set_caps (basesrc, caps);
}
}
gst_caps_unref (caps);
}
static gboolean
-gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
+gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
- pa_buffer_attr buf_attr;
- GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
+ pa_buffer_attr wanted;
+ const pa_buffer_attr *actual;
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
- memset (&buf_attr, 0, sizeof (buf_attr));
- buf_attr.maxlength = spec->segtotal * spec->segsize * 2;
- buf_attr.fragsize = spec->segsize;
+ wanted.maxlength = -1;
+ wanted.tlength = -1;
+ wanted.prebuf = 0;
+ wanted.minreq = -1;
+ wanted.fragsize = spec->segsize;
- if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &buf_attr,
- PA_STREAM_INTERPOLATE_TIMING |
- PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_NOT_MONOTONOUS |
-#if 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;
- }
+ 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_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
+ PA_STREAM_START_CORKED) < 0)
+ goto connect_failed;
+
+ 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 (!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");
+
+ /* 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);
- gst_pulsesrc_destroy_stream (pulsesrc);
-
- pa_threaded_mainloop_unlock (pulsesrc->mainloop);
- return FALSE;
+ pa_threaded_mainloop_unlock (pulsesrc->mainloop);
+ return FALSE;
+ }
}
static void
gst_pulsesrc_success_cb (pa_stream * s, int success, void *userdata)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);
+ GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
- pulsesrc->operation_success = !!success;
+ pulsesrc->operation_success = ! !success;
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
}
static void
gst_pulsesrc_reset (GstAudioSrc * asrc)
{
- GstPulseSrc *pulsesrc = GST_PULSESRC (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))
+ if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto unlock_and_fail;
if (!(o =
goto unlock_and_fail;
}
+ pulsesrc->paused = TRUE;
/* Inform anyone waiting in _write() call that it shall wakeup */
if (pulsesrc->in_read) {
- pulsesrc->did_reset = TRUE;
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))
+ if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto unlock_and_fail;
pa_threaded_mainloop_wait (pulsesrc->mainloop);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
}
-static void
-gst_pulsesrc_pause (GstPulseSrc * pulsesrc, gboolean b)
+/* update the corked state of a stream, must be called with the mainloop
+ * lock */
+static gboolean
+gst_pulsesrc_set_corked (GstPulseSrc * psrc, gboolean corked, gboolean wait)
{
pa_operation *o = NULL;
+ gboolean res = FALSE;
- pa_threaded_mainloop_lock (pulsesrc->mainloop);
-
- if (gst_pulsesrc_is_dead (pulsesrc))
- goto unlock;
+ GST_DEBUG_OBJECT (psrc, "setting corked state to %d", corked);
+ if (!psrc->stream_connected)
+ return TRUE;
- if (!(o = pa_stream_cork (pulsesrc->stream, b, NULL, NULL))) {
+ if (psrc->corked != corked) {
+ if (!(o = pa_stream_cork (psrc->stream, corked,
+ gst_pulsesrc_success_cb, psrc)))
+ goto cork_failed;
- GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
- ("pa_stream_cork() failed: %s",
- pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
- goto unlock;
+ while (wait && pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
+ pa_threaded_mainloop_wait (psrc->mainloop);
+ if (gst_pulsesrc_is_dead (psrc, TRUE))
+ goto server_dead;
+ }
+ psrc->corked = corked;
+ } else {
+ GST_DEBUG_OBJECT (psrc, "skipping, already in requested state");
}
+ res = TRUE;
- while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
+cleanup:
+ if (o)
+ pa_operation_unref (o);
- if (gst_pulsesrc_is_dead (pulsesrc))
- goto unlock;
+ return res;
- pa_threaded_mainloop_wait (pulsesrc->mainloop);
+ /* ERRORS */
+server_dead:
+ {
+ GST_DEBUG_OBJECT (psrc, "the server is dead");
+ goto cleanup;
+ }
+cork_failed:
+ {
+ GST_ELEMENT_ERROR (psrc, RESOURCE, FAILED,
+ ("pa_stream_cork() failed: %s",
+ pa_strerror (pa_context_errno (psrc->context))), (NULL));
+ goto cleanup;
}
+}
-unlock:
+/* start/resume playback ASAP */
+static gboolean
+gst_pulsesrc_play (GstPulseSrc * psrc)
+{
+ pa_threaded_mainloop_lock (psrc->mainloop);
+ GST_DEBUG_OBJECT (psrc, "playing");
+ psrc->paused = FALSE;
+ gst_pulsesrc_set_corked (psrc, FALSE, FALSE);
+ pa_threaded_mainloop_unlock (psrc->mainloop);
- if (o)
- pa_operation_unref (o);
+ return TRUE;
+}
- pa_threaded_mainloop_unlock (pulsesrc->mainloop);
+/* pause/stop playback ASAP */
+static gboolean
+gst_pulsesrc_pause (GstPulseSrc * psrc)
+{
+ pa_threaded_mainloop_lock (psrc->mainloop);
+ GST_DEBUG_OBJECT (psrc, "pausing");
+ /* make sure the commit method stops writing */
+ psrc->paused = TRUE;
+ if (psrc->in_read) {
+ /* we are waiting in a read, signal */
+ GST_DEBUG_OBJECT (psrc, "signal read");
+ pa_threaded_mainloop_signal (psrc->mainloop, 0);
+ }
+ pa_threaded_mainloop_unlock (psrc->mainloop);
+
+ return TRUE;
}
static GstStateChangeReturn
gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
{
- GstPulseSrc *this = GST_PULSESRC (element);
+ GstStateChangeReturn ret;
+ GstPulseSrc *this = GST_PULSESRC_CAST (element);
switch (transition) {
-
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- gst_pulsesrc_pause (this,
- GST_STATE_TRANSITION_NEXT (transition) == GST_STATE_PAUSED);
- break;
-
case GST_STATE_CHANGE_NULL_TO_READY:
+ this->mainloop = pa_threaded_mainloop_new ();
+ g_assert (this->mainloop);
+
+ pa_threaded_mainloop_start (this->mainloop);
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;
+ }
- case GST_STATE_CHANGE_READY_TO_NULL:
+ 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;
}
- break;
+ 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;
}
- if (GST_ELEMENT_CLASS (parent_class)->change_state)
- return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- return GST_STATE_CHANGE_SUCCESS;
+ return ret;
}
projects / platform / upstream / gstreamer.git / blobdiff