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gstreamer/ext/pulse/pulsesrc.c

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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., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301
* USA.
*/
/**
* SECTION:element-pulsesrc
* @see_also: pulsesink
*
* This element captures audio from a
* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 -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 <gst/audio/audio.h>
#include "pulsesrc.h"
#include "pulseutil.h"
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
#define DEFAULT_SERVER NULL
#define DEFAULT_DEVICE NULL
#define DEFAULT_CURRENT_DEVICE NULL
#define DEFAULT_DEVICE_NAME NULL
#define DEFAULT_VOLUME 1.0
#define DEFAULT_MUTE FALSE
#define MAX_VOLUME 10.0
/* See the pulsesink code for notes on how we interact with the PA mainloop
* thread. */
enum
{
PROP_0,
PROP_SERVER,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_CURRENT_DEVICE,
PROP_CLIENT_NAME,
PROP_STREAM_PROPERTIES,
PROP_SOURCE_OUTPUT_INDEX,
PROP_VOLUME,
PROP_MUTE,
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_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,
GstAudioRingBufferSpec * spec);
static gboolean gst_pulsesrc_unprepare (GstAudioSrc * asrc);
static guint gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data,
guint length, GstClockTime * timestamp);
static guint gst_pulsesrc_delay (GstAudioSrc * asrc);
static void gst_pulsesrc_reset (GstAudioSrc * src);
static gboolean gst_pulsesrc_negotiate (GstBaseSrc * basesrc);
static gboolean gst_pulsesrc_event (GstBaseSrc * basesrc, GstEvent * event);
static GstStateChangeReturn gst_pulsesrc_change_state (GstElement *
element, GstStateChange transition);
static GstClockTime gst_pulsesrc_get_time (GstClock * clock, GstPulseSrc * src);
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (_PULSE_CAPS_PCM)
);
#define gst_pulsesrc_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstPulseSrc, gst_pulsesrc, GST_TYPE_AUDIO_SRC,
G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL));
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->event = GST_DEBUG_FUNCPTR (gst_pulsesrc_event);
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_CURRENT_DEVICE,
g_param_spec_string ("current-device", "Current Device",
"The current PulseAudio source device", DEFAULT_CURRENT_DEVICE,
G_PARAM_READABLE | 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-name
*
* The PulseAudio client name to use.
*/
g_object_class_install_property (gobject_class,
PROP_CLIENT_NAME,
g_param_spec_string ("client-name", "Client Name",
"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);
* ]|
*/
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.
*/
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_static_metadata (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));
/**
* GstPulseSrc:volume:
*
* The volume of the record stream.
*/
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.
*/
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));
}
static void
gst_pulsesrc_init (GstPulseSrc * pulsesrc)
{
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;
pa_sample_spec_init (&pulsesrc->sample_spec);
pulsesrc->operation_success = FALSE;
pulsesrc->paused = TRUE;
pulsesrc->in_read = FALSE;
pulsesrc->volume = DEFAULT_VOLUME;
pulsesrc->volume_set = FALSE;
pulsesrc->mute = DEFAULT_MUTE;
pulsesrc->mute_set = FALSE;
pulsesrc->notify = 0;
pulsesrc->properties = NULL;
pulsesrc->proplist = NULL;
/* this should be the default but it isn't yet */
gst_audio_base_src_set_slave_method (GST_AUDIO_BASE_SRC (pulsesrc),
GST_AUDIO_BASE_SRC_SLAVE_SKEW);
/* override with a custom clock */
if (GST_AUDIO_BASE_SRC (pulsesrc)->clock)
gst_object_unref (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
GST_AUDIO_BASE_SRC (pulsesrc)->clock =
gst_audio_clock_new ("GstPulseSrcClock",
(GstAudioClockGetTimeFunc) gst_pulsesrc_get_time, pulsesrc, NULL);
}
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->stream_connected = FALSE;
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);
pa_context_set_subscribe_callback (pulsesrc->context, NULL, NULL);
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);
g_free (pulsesrc->current_source_name);
if (pulsesrc->properties)
gst_structure_free (pulsesrc->properties);
if (pulsesrc->proplist)
pa_proplist_free (pulsesrc->proplist);
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 (!pulsesrc->stream_connected)
return TRUE;
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_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;
psrc->current_source_idx = i->source;
if (G_UNLIKELY (psrc->volume > MAX_VOLUME)) {
GST_WARNING_OBJECT (psrc, "Clipped volume from %f to %f",
psrc->volume, MAX_VOLUME);
psrc->volume = MAX_VOLUME;
}
}
done:
pa_threaded_mainloop_signal (psrc->mainloop, 0);
}
static void
gst_pulsesrc_get_source_output_info (GstPulseSrc * pulsesrc, gdouble * volume,
gboolean * mute)
{
pa_operation *o = NULL;
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:
if (volume)
*volume = pulsesrc->volume;
if (mute)
*mute = pulsesrc->mute;
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return;
/* ERRORS */
no_mainloop:
{
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
if (volume)
*volume = pulsesrc->volume;
if (mute)
*mute = pulsesrc->mute;
return;
}
no_index:
{
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
if (volume)
*volume = pulsesrc->volume;
if (mute)
*mute = pulsesrc->mute;
return;
}
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_current_source_info_cb (pa_context * c, const pa_source_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->current_source_idx) {
g_free (psrc->current_source_name);
psrc->current_source_name = g_strdup (i->name);
}
done:
pa_threaded_mainloop_signal (psrc->mainloop, 0);
}
static gchar *
gst_pulsesrc_get_current_device (GstPulseSrc * pulsesrc)
{
pa_operation *o = NULL;
gchar *current_src;
if (!pulsesrc->mainloop)
goto no_mainloop;
if (pulsesrc->source_output_idx == PA_INVALID_INDEX)
goto no_index;
gst_pulsesrc_get_source_output_info (pulsesrc, NULL, NULL);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!(o = pa_context_get_source_info_by_index (pulsesrc->context,
pulsesrc->current_source_idx, gst_pulsesrc_current_source_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:
current_src = g_strdup (pulsesrc->current_source_name);
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return current_src;
/* ERRORS */
no_mainloop:
{
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return NULL;
}
no_index:
{
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return NULL;
}
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;
}
}
static void
gst_pulsesrc_set_stream_device (GstPulseSrc * pulsesrc, const gchar * device)
{
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 stream device to %s", device);
if (!(o = pa_context_move_source_output_by_name (pulsesrc->context,
pulsesrc->source_output_idx, device, NULL, NULL)))
goto move_failed;
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return;
/* ERRORS */
no_mainloop:
{
GST_DEBUG_OBJECT (pulsesrc, "we have no mainloop");
return;
}
no_index:
{
GST_DEBUG_OBJECT (pulsesrc, "we don't have a stream index");
return;
}
move_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_move_source_output_by_name(%s) failed: %s",
device, pa_strerror (pa_context_errno (pulsesrc->context))),
(NULL));
goto unlock;
}
}
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);
break;
case PROP_DEVICE:
g_free (pulsesrc->device);
pulsesrc->device = g_value_dup_string (value);
gst_pulsesrc_set_stream_device (pulsesrc, pulsesrc->device);
break;
case PROP_CLIENT_NAME:
g_free (pulsesrc->client_name);
if (!g_value_get_string (value)) {
GST_WARNING_OBJECT (pulsesrc,
"Empty PulseAudio client name not allowed. Resetting to default value");
pulsesrc->client_name = gst_pulse_client_name ();
} else
pulsesrc->client_name = g_value_dup_string (value);
break;
case PROP_STREAM_PROPERTIES:
if (pulsesrc->properties)
gst_structure_free (pulsesrc->properties);
pulsesrc->properties =
gst_structure_copy (gst_value_get_structure (value));
if (pulsesrc->proplist)
pa_proplist_free (pulsesrc->proplist);
pulsesrc->proplist = gst_pulse_make_proplist (pulsesrc->properties);
break;
case PROP_VOLUME:
gst_pulsesrc_set_stream_volume (pulsesrc, g_value_get_double (value));
break;
case PROP_MUTE:
gst_pulsesrc_set_stream_mute (pulsesrc, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesrc_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_CURRENT_DEVICE:
{
gchar *current_device = gst_pulsesrc_get_current_device (pulsesrc);
if (current_device)
g_value_take_string (value, current_device);
else
g_value_set_string (value, "");
break;
}
case PROP_DEVICE_NAME:
g_value_take_string (value, gst_pulsesrc_device_description (pulsesrc));
break;
case PROP_CLIENT_NAME:
g_value_set_string (value, pulsesrc->client_name);
break;
case PROP_STREAM_PROPERTIES:
gst_value_set_structure (value, pulsesrc->properties);
break;
case PROP_SOURCE_OUTPUT_INDEX:
g_value_set_uint (value, pulsesrc->source_output_idx);
break;
case PROP_VOLUME:
{
gdouble volume;
gst_pulsesrc_get_source_output_info (pulsesrc, &volume, NULL);
g_value_set_double (value, volume);
break;
}
case PROP_MUTE:
{
gboolean mute;
gst_pulsesrc_get_source_output_info (pulsesrc, NULL, &mute);
g_value_set_boolean (value, mute);
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 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);
}
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);
pa_context_set_subscribe_callback (pulsesrc->context,
gst_pulsesrc_context_subscribe_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,
GstClockTime * timestamp)
{
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
size_t sum = 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");
g_object_notify (G_OBJECT (pulsesrc), "current-device");
}
pa_threaded_mainloop_lock (pulsesrc->mainloop);
pulsesrc->in_read = TRUE;
if (!pulsesrc->stream_connected)
goto not_connected;
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 */
not_connected:
{
GST_LOG_OBJECT (pulsesrc, "we are not connected");
goto unlock_and_fail;
}
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,
GstAudioRingBufferSpec * rspec)
{
pa_channel_map channel_map;
const pa_channel_map *m;
GstStructure *s;
gboolean need_channel_layout = FALSE;
GstAudioRingBufferSpec new_spec, *spec = NULL;
const gchar *name;
int i;
/* If we already have a stream (renegotiation), free it first */
if (pulsesrc->stream)
gst_pulsesrc_destroy_stream (pulsesrc);
if (rspec) {
/* Post-negotiation, we already have a ringbuffer spec, so we just need to
* use it to create a stream. */
spec = rspec;
/* At this point, we expect the channel-mask to be set in caps, so we just
* use that */
if (!gst_pulse_gst_to_channel_map (&channel_map, spec))
goto invalid_spec;
} else if (caps) {
/* At negotiation time, we get a fixed caps and use it to set up a stream */
s = gst_caps_get_structure (*caps, 0);
gst_structure_get_int (s, "channels", &new_spec.info.channels);
if (!gst_structure_has_field (s, "channel-mask")) {
if (new_spec.info.channels == 1) {
pa_channel_map_init_mono (&channel_map);
} else if (new_spec.info.channels == 2) {
pa_channel_map_init_stereo (&channel_map);
} else {
need_channel_layout = TRUE;
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
G_GUINT64_CONSTANT (0), NULL);
}
}
memset (&new_spec, 0, sizeof (GstAudioRingBufferSpec));
new_spec.latency_time = GST_SECOND;
if (!gst_audio_ring_buffer_parse_caps (&new_spec, *caps))
goto invalid_caps;
/* Keep the refcount of the caps at 1 to make them writable */
gst_caps_unref (new_spec.caps);
if (!need_channel_layout
&& !gst_pulse_gst_to_channel_map (&channel_map, &new_spec)) {
need_channel_layout = TRUE;
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
G_GUINT64_CONSTANT (0), NULL);
for (i = 0; i < G_N_ELEMENTS (new_spec.info.position); i++)
new_spec.info.position[i] = GST_AUDIO_CHANNEL_POSITION_INVALID;
}
spec = &new_spec;
} else {
/* !rspec && !caps */
g_assert_not_reached ();
}
if (!gst_pulse_fill_sample_spec (spec, &pulsesrc->sample_spec))
goto invalid_spec;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!pulsesrc->context)
goto bad_context;
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 (caps) {
m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &new_spec);
gst_audio_channel_positions_to_valid_order (new_spec.info.position,
new_spec.info.channels);
gst_caps_unref (*caps);
*caps = gst_audio_info_to_caps (&new_spec.info);
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;
/* 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);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
fail:
return FALSE;
}
}
static gboolean
gst_pulsesrc_event (GstBaseSrc * basesrc, GstEvent * event)
{
GST_DEBUG_OBJECT (basesrc, "handle event %" GST_PTR_FORMAT, event);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_RECONFIGURE:
gst_pad_check_reconfigure (GST_BASE_SRC_PAD (basesrc));
break;
default:
break;
}
return GST_BASE_SRC_CLASS (parent_class)->event (basesrc, event);
}
/* 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)
{
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_query_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_query_caps (GST_BASE_SRC_PAD (basesrc), NULL);
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_truncate (caps);
/* now fixate */
if (!gst_caps_is_empty (caps)) {
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)) {
/* 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 (pulsesrc, &caps, NULL);
if (result)
result = gst_base_src_set_caps (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, GstAudioRingBufferSpec * spec)
{
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
pa_stream_flags_t flags;
pa_operation *o;
GstAudioClock *clock;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!pulsesrc->stream)
gst_pulsesrc_create_stream (pulsesrc, NULL, spec);
{
GstAudioRingBufferSpec s = *spec;
const pa_channel_map *m;
m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &s);
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(pulsesrc)->ringbuffer, s.info.position);
}
/* enable event notifications */
GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
if (!(o = pa_context_subscribe (pulsesrc->context,
PA_SUBSCRIPTION_MASK_SOURCE_OUTPUT, NULL, NULL))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_context_subscribe() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pa_operation_unref (o);
/* There's a bit of a disconnect here between the audio ringbuffer and what
* PulseAudio provides. The audio ringbuffer provide a total of buffer_time
* worth of buffering, divided into segments of latency_time size. We're
* asking PulseAudio to provide a total latency of latency_time, which, with
* PA_STREAM_ADJUST_LATENCY, effectively sets itself up as a ringbuffer with
* one segment being the hardware buffer, and the other the software buffer.
* This segment size is returned as the fragsize.
*
* Since the two concepts don't map very well, what we do is keep segsize as
* it is (unless fragsize is even larger, in which case we use that). We'll
* get data from PulseAudio in smaller chunks than we want to pass on as an
* element, and we coalesce those chunks in the ringbuffer memory and pass it
* on in the expected chunk size. */
wanted.maxlength = spec->segsize * spec->segtotal;
wanted.tlength = -1;
wanted.prebuf = 0;
wanted.minreq = -1;
wanted.fragsize = spec->segsize;
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d", wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", wanted.prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d", wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d", wanted.fragsize);
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
PA_STREAM_START_CORKED;
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
flags) < 0) {
goto connect_failed;
}
/* our clock will now start from 0 again */
clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
gst_audio_clock_reset (clock, 0);
pulsesrc->corked = TRUE;
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state (pulsesrc->stream);
if (!PA_STREAM_IS_GOOD (state))
goto stream_is_bad;
if (state == PA_STREAM_READY)
break;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
}
pulsesrc->stream_connected = TRUE;
/* store the source output index so it can be accessed via a property */
pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream);
g_object_notify (G_OBJECT (pulsesrc), "source-output-index");
/* Although source output stream muting is supported, there is a bug in
* PulseAudio that doesn't allow us to do this at startup, so we mute
* manually post-connect. This should be moved back pre-connect once things
* are fixed on the PulseAudio side. */
if (pulsesrc->mute_set && pulsesrc->mute) {
gst_pulsesrc_set_stream_mute (pulsesrc, pulsesrc->mute);
pulsesrc->mute_set = FALSE;
}
if (pulsesrc->volume_set) {
gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
pulsesrc->volume_set = FALSE;
}
/* get the actual buffering properties now */
actual = pa_stream_get_buffer_attr (pulsesrc->stream);
GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength);
GST_INFO_OBJECT (pulsesrc, "tlength: %d (wanted: %d)",
actual->tlength, wanted.tlength);
GST_INFO_OBJECT (pulsesrc, "prebuf: %d", actual->prebuf);
GST_INFO_OBJECT (pulsesrc, "minreq: %d (wanted %d)", actual->minreq,
wanted.minreq);
GST_INFO_OBJECT (pulsesrc, "fragsize: %d (wanted %d)",
actual->fragsize, wanted.fragsize);
if (actual->fragsize >= spec->segsize) {
spec->segsize = actual->fragsize;
} else {
/* fragsize is smaller than what we wanted, so let the read function
* coalesce the smaller chunks as they come in */
}
/* Fix up the total ringbuffer size based on what we actually got */
spec->segtotal = actual->maxlength / spec->segsize;
if (!pulsesrc->paused) {
GST_DEBUG_OBJECT (pulsesrc, "uncorking because we are playing");
gst_pulsesrc_set_corked (pulsesrc, FALSE, FALSE);
}
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
/* ERRORS */
connect_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
stream_is_bad:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
unlock_and_fail:
{
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return FALSE;
}
}
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->stream_connected)
return TRUE;
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:
if (!(this->mainloop = pa_threaded_mainloop_new ()))
goto mainloop_failed;
if (pa_threaded_mainloop_start (this->mainloop) < 0) {
pa_threaded_mainloop_free (this->mainloop);
this->mainloop = NULL;
goto mainloop_start_failed;
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
GST_AUDIO_BASE_SRC (this)->clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* uncork and start recording */
gst_pulsesrc_play (this);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* stop recording ASAP by corking */
pa_threaded_mainloop_lock (this->mainloop);
GST_DEBUG_OBJECT (this, "corking");
gst_pulsesrc_set_corked (this, TRUE, FALSE);
pa_threaded_mainloop_unlock (this->mainloop);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* now make sure we get out of the _read method */
gst_pulsesrc_pause (this);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (this->mainloop)
pa_threaded_mainloop_stop (this->mainloop);
gst_pulsesrc_destroy_context (this);
if (this->mainloop) {
pa_threaded_mainloop_free (this->mainloop);
this->mainloop = NULL;
}
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* format_lost is reset in release() in baseaudiosink */
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
GST_AUDIO_BASE_SRC (this)->clock));
break;
default:
break;
}
return ret;
/* ERRORS */
mainloop_failed:
{
GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
("pa_threaded_mainloop_new() failed"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
mainloop_start_failed:
{
GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
("pa_threaded_mainloop_start() failed"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
}
static GstClockTime
gst_pulsesrc_get_time (GstClock * clock, GstPulseSrc * src)
{
pa_usec_t time = 0;
if (src->mainloop == NULL)
goto out;
pa_threaded_mainloop_lock (src->mainloop);
if (!src->stream)
goto unlock_and_out;
if (gst_pulsesrc_is_dead (src, TRUE))
goto unlock_and_out;
if (pa_stream_get_time (src->stream, &time) < 0) {
GST_DEBUG_OBJECT (src, "could not get time");
time = GST_CLOCK_TIME_NONE;
} else {
time *= 1000;
}
unlock_and_out:
pa_threaded_mainloop_unlock (src->mainloop);
out:
return time;
}
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