mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-08 10:31:05 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/ext/pulse/pulsesrc.c
Arun Raghavan 379049809c pulsesrc: Add a source-output-index property 
This exposes the source output index of the record stream that we open
so that clients can use this with the introspection if they want (to
move the stream, for example).
2011-07-29 00:07:52 +05:30

1324 lines
37 KiB
C
Raw Blame History

/*
* 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>
#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
enum
{
PROP_0,
PROP_SERVER,
PROP_DEVICE,
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,
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;
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)
{
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)
{
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 ();
/**
* 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));
}
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;
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);
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;
}
}
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
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;
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");
}
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);
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;
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_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_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
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);
}
/* 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;
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;
}
Reference in a new issue View git blame Copy permalink