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gstreamer/ext/pulse/pulsesrc.c
Arun Raghavan 1f4bb68794 pulsesrc: Implement GstStreamVolume interface 
PulseAudio 1.0 supports per-source-output volumes, and this exposes the
functionality via the GstStreamVolume interface.

When compiled against pre-1.0 PulseAudio, the interface is not
implemented, and the "volume" or "mute" properties are not available.
This bit of ugliness will go away when we can depend on PulseAudio 1.0
or greater.

https://bugzilla.gnome.org/show_bug.cgi?id=595055
2011-11-25 22:30:41 +05:30

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/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse is free software; you can redistribute it and/or modify
* it under the terms of the GNU Lesser General Public License as
* published by the Free Software Foundation; either version 2.1 of the
* License, or (at your option) any later version.
*
* gst-pulse is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
/**
* SECTION:element-pulsesrc
* @see_also: pulsesink, pulsemixer
*
* This element captures audio from a
* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
* ]| Record from a sound card using pulseaudio and encode to Ogg/Vorbis.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdio.h>
#include <gst/base/gstbasesrc.h>
#include <gst/gsttaglist.h>
#ifdef HAVE_PULSE_1_0
#include <gst/interfaces/streamvolume.h>
#endif
#include "pulsesrc.h"
#include "pulseutil.h"
#include "pulsemixerctrl.h"
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
#ifdef HAVE_PULSE_1_0
#define DEFAULT_VOLUME 1.0
#define DEFAULT_MUTE FALSE
#define MAX_VOLUME 10.0
#endif
enum
{
PROP_0,
PROP_SERVER,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_CLIENT,
PROP_STREAM_PROPERTIES,
PROP_SOURCE_OUTPUT_INDEX,
#ifdef HAVE_PULSE_1_0
PROP_VOLUME,
PROP_MUTE,
#endif
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,
const GValue * value, GParamSpec * pspec);
static void gst_pulsesrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_pulsesrc_finalize (GObject * object);
static gboolean gst_pulsesrc_open (GstAudioSrc * asrc);
static gboolean gst_pulsesrc_close (GstAudioSrc * asrc);
static gboolean gst_pulsesrc_prepare (GstAudioSrc * asrc,
GstRingBufferSpec * spec);
static gboolean gst_pulsesrc_unprepare (GstAudioSrc * asrc);
static guint gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data,
guint length);
static guint gst_pulsesrc_delay (GstAudioSrc * asrc);
static void gst_pulsesrc_reset (GstAudioSrc * src);
static gboolean gst_pulsesrc_negotiate (GstBaseSrc * basesrc);
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"
#else
# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
#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_CAST (iface);
if (interface_type == GST_TYPE_MIXER && this->mixer)
return TRUE;
if (interface_type == GST_TYPE_PROPERTY_PROBE && this->probe)
return TRUE;
#ifdef HAVE_PULSE_1_0
if (interface_type == GST_TYPE_STREAM_VOLUME)
return TRUE;
#endif
return FALSE;
}
static void
gst_pulsesrc_implements_interface_init (GstImplementsInterfaceClass * klass)
{
klass->supported = gst_pulsesrc_interface_supported;
}
static void
gst_pulsesrc_init_interfaces (GType type)
{
#ifdef HAVE_PULSE_1_0
static const GInterfaceInfo svol_iface_info = {
NULL, NULL, NULL,
};
#endif
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,
};
#ifdef HAVE_PULSE_1_0
g_type_add_interface_static (type, GST_TYPE_STREAM_VOLUME, &svol_iface_info);
#endif
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)
{
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));
}
static void
gst_pulsesrc_class_init (GstPulseSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (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_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);
gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_pulsesrc_negotiate);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_pulsesrc_open);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_pulsesrc_close);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_pulsesrc_prepare);
gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_pulsesrc_unprepare);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_pulsesrc_read);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_pulsesrc_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_pulsesrc_reset);
/* Overwrite GObject fields */
g_object_class_install_property (gobject_class,
PROP_SERVER,
g_param_spec_string ("server", "Server",
"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", "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", DEFAULT_DEVICE_NAME,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
clientname = gst_pulse_client_name ();
/**
* GstPulseSrc: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));
#ifdef HAVE_PULSE_1_0
/**
* GstPulseSrc:volume
*
* The volume of the record stream. Only works when using PulseAudio 1.0 or
* later.
*
* Since: 0.10.36
*/
g_object_class_install_property (gobject_class,
PROP_VOLUME, g_param_spec_double ("volume", "Volume",
"Linear volume of this stream, 1.0=100%",
0.0, MAX_VOLUME, DEFAULT_VOLUME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstPulseSrc:mute
*
* Whether the stream is muted or not. Only works when using PulseAudio 1.0
* or later.
*
* Since: 0.10.36
*/
g_object_class_install_property (gobject_class,
PROP_MUTE, g_param_spec_boolean ("mute", "Mute",
"Mute state of this stream",
DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
#endif
}
static void
gst_pulsesrc_init (GstPulseSrc * pulsesrc, GstPulseSrcClass * klass)
{
pulsesrc->server = NULL;
pulsesrc->device = NULL;
pulsesrc->client_name = gst_pulse_client_name ();
pulsesrc->device_description = NULL;
pulsesrc->context = NULL;
pulsesrc->stream = NULL;
pulsesrc->source_output_idx = PA_INVALID_INDEX;
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
pa_sample_spec_init (&pulsesrc->sample_spec);
pulsesrc->operation_success = FALSE;
pulsesrc->paused = FALSE;
pulsesrc->in_read = FALSE;
#ifdef HAVE_PULSE_1_0
pulsesrc->volume = DEFAULT_VOLUME;
pulsesrc->volume_set = FALSE;
pulsesrc->mute = DEFAULT_MUTE;
pulsesrc->mute_set = FALSE;
pulsesrc->notify = 0;
#endif
pulsesrc->mixer = NULL;
pulsesrc->properties = NULL;
pulsesrc->proplist = 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 void
gst_pulsesrc_destroy_stream (GstPulseSrc * pulsesrc)
{
if (pulsesrc->stream) {
pa_stream_disconnect (pulsesrc->stream);
pa_stream_unref (pulsesrc->stream);
pulsesrc->stream = NULL;
pulsesrc->source_output_idx = PA_INVALID_INDEX;
g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
}
g_free (pulsesrc->device_description);
pulsesrc->device_description = NULL;
}
static void
gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc)
{
gst_pulsesrc_destroy_stream (pulsesrc);
if (pulsesrc->context) {
pa_context_disconnect (pulsesrc->context);
/* Make sure we don't get any further callbacks */
pa_context_set_state_callback (pulsesrc->context, NULL, NULL);
#ifdef HAVE_PULSE_1_0
pa_context_set_subscribe_callback (pulsesrc->context, NULL, NULL);
#endif
pa_context_unref (pulsesrc->context);
pulsesrc->context = NULL;
}
}
static void
gst_pulsesrc_finalize (GObject * object)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (object);
g_free (pulsesrc->server);
g_free (pulsesrc->device);
g_free (pulsesrc->client_name);
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);
}
#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, gboolean check_stream)
{
if (!CONTEXT_OK (pulsesrc->context))
goto error;
if (check_stream && !STREAM_OK (pulsesrc->stream))
goto error;
return FALSE;
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;
}
}
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 gchar *
gst_pulsesrc_device_description (GstPulseSrc * pulsesrc)
{
pa_operation *o = NULL;
gchar *t;
if (!pulsesrc->mainloop)
goto no_mainloop;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
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",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock;
}
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
if (gst_pulsesrc_is_dead (pulsesrc, FALSE))
goto unlock;
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
unlock:
if (o)
pa_operation_unref (o);
t = g_strdup (pulsesrc->device_description);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return t;
no_mainloop:
{
GST_DEBUG_OBJECT (pulsesrc, "have no mainloop");
return NULL;
}
}
#ifdef HAVE_PULSE_1_0
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 gdouble
gst_pulsesrc_get_stream_volume (GstPulseSrc * pulsesrc)
{
pa_operation *o = NULL;
gdouble v;
if (!pulsesrc->mainloop)
goto no_mainloop;
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
goto no_index;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!(o = pa_context_get_source_output_info (pulsesrc->context,
pulsesrc->source_output_idx, gst_pulsesrc_source_output_info_cb,
pulsesrc)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (pulsesrc->mainloop);
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto unlock;
}
unlock:
v = pulsesrc->volume;
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
if (v > MAX_VOLUME) {
GST_WARNING_OBJECT (pulsesrc, "Clipped volume from %f to %f", v,
MAX_VOLUME);
v = MAX_VOLUME;
}
return v;
/* ERRORS */
no_mainloop:
{
v = pulsesrc->volume;
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return v;
}
no_index:
{
v = pulsesrc->volume;
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return v;
}
info_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_get_source_output_info() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock;
}
}
static gboolean
gst_pulsesrc_get_stream_mute (GstPulseSrc * pulsesrc)
{
pa_operation *o = NULL;
gboolean mute;
if (!pulsesrc->mainloop)
goto no_mainloop;
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
goto no_index;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!(o = pa_context_get_source_output_info (pulsesrc->context,
pulsesrc->source_output_idx, gst_pulsesrc_source_output_info_cb,
pulsesrc)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (pulsesrc->mainloop);
if (gst_pulsesrc_is_dead (pulsesrc, TRUE))
goto unlock;
}
unlock:
mute = pulsesrc->mute;
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return mute;
/* ERRORS */
no_mainloop:
{
mute = pulsesrc->mute;
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return mute;
}
no_index:
{
mute = pulsesrc->mute;
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return mute;
}
info_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_get_source_output_info() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock;
}
}
static void
gst_pulsesrc_set_stream_volume (GstPulseSrc * pulsesrc, gdouble volume)
{
pa_cvolume v;
pa_operation *o = NULL;
if (!pulsesrc->mainloop)
goto no_mainloop;
if (!pulsesrc->source_output_idx)
goto no_index;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
GST_DEBUG_OBJECT (pulsesrc, "setting volume to %f", volume);
gst_pulse_cvolume_from_linear (&v, pulsesrc->sample_spec.channels, volume);
if (!(o = pa_context_set_source_output_volume (pulsesrc->context,
pulsesrc->source_output_idx, &v, NULL, NULL)))
goto volume_failed;
/* We don't really care about the result of this call */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return;
/* ERRORS */
no_mainloop:
{
pulsesrc->volume = volume;
pulsesrc->volume_set = TRUE;
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return;
}
no_index:
{
pulsesrc->volume = volume;
pulsesrc->volume_set = TRUE;
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return;
}
volume_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_set_source_output_volume() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock;
}
}
static void
gst_pulsesrc_set_stream_mute (GstPulseSrc * pulsesrc, gboolean mute)
{
pa_operation *o = NULL;
if (!pulsesrc->mainloop)
goto no_mainloop;
if (!pulsesrc->source_output_idx)
goto no_index;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
GST_DEBUG_OBJECT (pulsesrc, "setting mute state to %d", mute);
if (!(o = pa_context_set_source_output_mute (pulsesrc->context,
pulsesrc->source_output_idx, mute, NULL, NULL)))
goto mute_failed;
/* We don't really care about the result of this call */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return;
/* ERRORS */
no_mainloop:
{
pulsesrc->mute = mute;
pulsesrc->mute_set = TRUE;
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return;
}
no_index:
{
pulsesrc->mute = mute;
pulsesrc->mute_set = TRUE;
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return;
}
mute_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_set_source_output_mute() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock;
}
}
#endif
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;
#ifdef HAVE_PULSE_1_0
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;
#endif
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_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;
#ifdef HAVE_PULSE_1_0
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;
#endif
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_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_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_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_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;
}
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");
}
#ifdef HAVE_PULSE_1_0
static void
gst_pulsesrc_context_subscribe_cb (pa_context * c,
pa_subscription_event_type_t t, uint32_t idx, void *userdata)
{
GstPulseSrc *psrc = GST_PULSESRC (userdata);
if (t != (PA_SUBSCRIPTION_EVENT_SOURCE_OUTPUT | PA_SUBSCRIPTION_EVENT_CHANGE)
&& t != (PA_SUBSCRIPTION_EVENT_SOURCE_OUTPUT | PA_SUBSCRIPTION_EVENT_NEW))
return;
if (idx != psrc->source_output_idx)
return;
/* Actually this event is also triggered when other properties of the stream
* change that are unrelated to the volume. However it is probably cheaper to
* signal the change here and check for the volume when the GObject property
* is read instead of querying it always. */
/* inform streaming thread to notify */
g_atomic_int_compare_and_exchange (&psrc->notify, 0, 1);
}
#endif
static gboolean
gst_pulsesrc_open (GstAudioSrc * asrc)
{
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),
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);
#ifdef HAVE_PULSE_1_0
pa_context_set_subscribe_callback (pulsesrc->context,
gst_pulsesrc_context_subscribe_cb, pulsesrc);
#endif
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);
return TRUE;
/* ERRORS */
unlock_and_fail:
{
gst_pulsesrc_destroy_context (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return FALSE;
}
}
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 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 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;
#ifdef HAVE_PULSE_1_0
if (g_atomic_int_compare_and_exchange (&pulsesrc->notify, 1, 0)) {
g_object_notify (G_OBJECT (pulsesrc), "volume");
g_object_notify (G_OBJECT (pulsesrc), "mute");
}
#endif
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;
}
}
/* 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 gboolean
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps * caps)
{
pa_channel_map channel_map;
GstStructure *s;
gboolean need_channel_layout = FALSE;
GstRingBufferSpec spec;
const gchar *name;
memset (&spec, 0, sizeof (GstRingBufferSpec));
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;
}
/* 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;
}
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!pulsesrc->context) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context"), (NULL));
goto unlock_and_fail;
}
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)
pa_channel_map_init_mono (&channel_map);
else if (spec.channels == 2)
pa_channel_map_init_stereo (&channel_map);
else
need_channel_layout = TRUE;
}
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))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to create stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
} else if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
name, &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;
}
if (need_channel_layout) {
const pa_channel_map *m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &spec);
caps = spec.caps;
}
GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, caps);
pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
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);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
unlock_and_fail:
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
fail:
return FALSE;
}
/* 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_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_stream_flags_t flags;
#ifdef HAVE_PULSE_1_0
pa_operation *o;
#endif
pa_threaded_mainloop_lock (pulsesrc->mainloop);
#ifdef HAVE_PULSE_1_0
/* enable event notifications */
GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
if (!(o = pa_context_subscribe (pulsesrc->context,
PA_SUBSCRIPTION_MASK_SINK_INPUT, 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);
#endif
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;
#ifdef HAVE_PULSE_1_0
if (pulsesrc->mute_set && pulsesrc->mute)
flags |= PA_STREAM_START_MUTED;
#endif
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
flags) < 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);
}
/* 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");
#ifdef HAVE_PULSE_1_0
if (pulsesrc->volume_set) {
gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
pulsesrc->volume_set = FALSE;
}
#endif
/* 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 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 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);
}
/* 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;
GST_DEBUG_OBJECT (psrc, "setting corked state to %d", corked);
if (psrc->corked != corked) {
if (!(o = pa_stream_cork (psrc->stream, corked,
gst_pulsesrc_success_cb, psrc)))
goto cork_failed;
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;
cleanup:
if (o)
pa_operation_unref (o);
return res;
/* 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;
}
}
/* 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);
return TRUE;
}
/* 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)
{
GstStateChangeReturn ret;
GstPulseSrc *this = GST_PULSESRC_CAST (element);
switch (transition) {
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;
}
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;
}
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