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gstreamer/ext/pulse/pulsesink.c
Wim Taymans 20a6908dfd pulsesink: bps is signed int to avoid overflow 
Keep bps as gint instead of guint because we will be doing signed math with it
later on and we don't want weird results.
2009-04-10 12:01:27 +02:00

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/* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
* (c) 2009 Wim Taymans
*
* 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-pulsesink
* @see_also: pulsesrc, pulsemixer
*
* This element outputs audio to a
* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink
* ]| Play an Ogg/Vorbis file.
* |[
* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink
* ]| Play a 440Hz sine wave.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdio.h>
#include <gst/base/gstbasesink.h>
#include <gst/gsttaglist.h>
#include "pulsesink.h"
#include "pulseutil.h"
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
/* according to
* http://www.pulseaudio.org/ticket/314
* we need pulse-0.9.12 to use sink volume properties
*/
enum
{
PROP_SERVER = 1,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_VOLUME
};
#define GST_TYPE_PULSERING_BUFFER \
(gst_pulseringbuffer_get_type())
#define GST_PULSERING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PULSERING_BUFFER,GstPulseRingBuffer))
#define GST_PULSERING_BUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PULSERING_BUFFER,GstPulseRingBufferClass))
#define GST_PULSERING_BUFFER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_PULSERING_BUFFER, GstPulseRingBufferClass))
#define GST_PULSERING_BUFFER_CAST(obj) \
((GstPulseRingBuffer *)obj)
#define GST_IS_PULSERING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PULSERING_BUFFER))
#define GST_IS_PULSERING_BUFFER_CLASS(klass)\
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PULSERING_BUFFER))
typedef struct _GstPulseRingBuffer GstPulseRingBuffer;
typedef struct _GstPulseRingBufferClass GstPulseRingBufferClass;
/* We keep a custom ringbuffer that is backed up by data allocated by
* pulseaudio. We must also overide the commit function to write into
* pulseaudio memory instead. */
struct _GstPulseRingBuffer
{
GstRingBuffer object;
gchar *stream_name;
pa_context *context;
pa_stream *stream;
pa_sample_spec sample_spec;
gint64 offset;
gboolean corked;
gboolean in_commit;
gboolean paused;
guint required;
};
struct _GstPulseRingBufferClass
{
GstRingBufferClass parent_class;
};
static void gst_pulseringbuffer_class_init (GstPulseRingBufferClass * klass);
static void gst_pulseringbuffer_init (GstPulseRingBuffer * ringbuffer,
GstPulseRingBufferClass * klass);
static void gst_pulseringbuffer_finalize (GObject * object);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_pulseringbuffer_open_device (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_close_device (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec);
static gboolean gst_pulseringbuffer_release (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_start (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_pause (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_stop (GstRingBuffer * buf);
static guint gst_pulseringbuffer_commit (GstRingBuffer * buf,
guint64 * sample, guchar * data, gint in_samples, gint out_samples,
gint * accum);
/* ringbuffer abstract base class */
static GType
gst_pulseringbuffer_get_type (void)
{
static GType ringbuffer_type = 0;
if (!ringbuffer_type) {
static const GTypeInfo ringbuffer_info = {
sizeof (GstPulseRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_pulseringbuffer_class_init,
NULL,
NULL,
sizeof (GstPulseRingBuffer),
0,
(GInstanceInitFunc) gst_pulseringbuffer_init,
NULL
};
ringbuffer_type =
g_type_register_static (GST_TYPE_RING_BUFFER, "GstPulseSinkRingBuffer",
&ringbuffer_info, 0);
}
return ringbuffer_type;
}
static void
gst_pulseringbuffer_class_init (GstPulseRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstobject_class = (GstObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_finalize);
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_pulseringbuffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_pulseringbuffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_pulseringbuffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_pulseringbuffer_release);
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_start);
gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_pause);
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_stop);
gstringbuffer_class->commit = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_commit);
}
static void
gst_pulseringbuffer_init (GstPulseRingBuffer * pbuf,
GstPulseRingBufferClass * g_class)
{
pbuf->stream_name = NULL;
pbuf->context = NULL;
pbuf->stream = NULL;
#if HAVE_PULSE_0_9_13
pa_sample_spec_init (&pbuf->sample_spec);
#else
pbuf->sample_spec.format = PA_SAMPLE_INVALID;
pbuf->sample_spec.rate = 0;
pbuf->sample_spec.channels = 0;
#endif
pbuf->paused = FALSE;
pbuf->corked = TRUE;
}
static void
gst_pulsering_destroy_stream (GstPulseRingBuffer * pbuf)
{
if (pbuf->stream) {
pa_stream_disconnect (pbuf->stream);
/* Make sure we don't get any further callbacks */
pa_stream_set_state_callback (pbuf->stream, NULL, NULL);
pa_stream_set_write_callback (pbuf->stream, NULL, NULL);
pa_stream_unref (pbuf->stream);
pbuf->stream = NULL;
}
g_free (pbuf->stream_name);
pbuf->stream_name = NULL;
}
static void
gst_pulsering_destroy_context (GstPulseRingBuffer * pbuf)
{
gst_pulsering_destroy_stream (pbuf);
if (pbuf->context) {
pa_context_disconnect (pbuf->context);
/* Make sure we don't get any further callbacks */
pa_context_set_state_callback (pbuf->context, NULL, NULL);
pa_context_set_subscribe_callback (pbuf->context, NULL, NULL);
pa_context_unref (pbuf->context);
pbuf->context = NULL;
}
}
static void
gst_pulseringbuffer_finalize (GObject * object)
{
GstPulseRingBuffer *ringbuffer;
ringbuffer = GST_PULSERING_BUFFER_CAST (object);
gst_pulsering_destroy_context (ringbuffer);
G_OBJECT_CLASS (ring_parent_class)->finalize (object);
}
static gboolean
gst_pulsering_is_dead (GstPulseSink * psink, GstPulseRingBuffer * pbuf)
{
if (!pbuf->context
|| !PA_CONTEXT_IS_GOOD (pa_context_get_state (pbuf->context))
|| !pbuf->stream
|| !PA_STREAM_IS_GOOD (pa_stream_get_state (pbuf->stream))) {
const gchar *err_str = pbuf->context ?
pa_strerror (pa_context_errno (pbuf->context)) : NULL;
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("Disconnected: %s",
err_str), (NULL));
return TRUE;
}
return FALSE;
}
static void
gst_pulsering_context_state_cb (pa_context * c, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_context_state_t state;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
state = pa_context_get_state (c);
GST_LOG_OBJECT (psink, "got new context state %d", state);
switch (state) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
GST_LOG_OBJECT (psink, "signaling");
pa_threaded_mainloop_signal (psink->mainloop, 0);
break;
case PA_CONTEXT_UNCONNECTED:
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
}
}
#if HAVE_PULSE_0_9_12
static void
gst_pulsering_context_subscribe_cb (pa_context * c,
pa_subscription_event_type_t t, uint32_t idx, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_LOG_OBJECT (psink, "type %d, idx %u", t, idx);
if (t != (PA_SUBSCRIPTION_EVENT_SINK_INPUT | PA_SUBSCRIPTION_EVENT_CHANGE) &&
t != (PA_SUBSCRIPTION_EVENT_SINK_INPUT | PA_SUBSCRIPTION_EVENT_NEW))
return;
if (!pbuf->stream)
return;
if (idx != pa_stream_get_index (pbuf->stream))
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 (&psink->notify, 0, 1);
}
#endif
/* will be called when the device should be opened. In this case we will connect
* to the server. We should not try to open any streams in this state. */
static gboolean
gst_pulseringbuffer_open_device (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
gchar *name;
pa_mainloop_api *api;
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
pbuf = GST_PULSERING_BUFFER_CAST (buf);
g_assert (!pbuf->context);
g_assert (!pbuf->stream);
name = gst_pulse_client_name ();
pa_threaded_mainloop_lock (psink->mainloop);
/* get the mainloop api and create a context */
GST_LOG_OBJECT (psink, "new context with name %s", GST_STR_NULL (name));
api = pa_threaded_mainloop_get_api (psink->mainloop);
if (!(pbuf->context = pa_context_new (api, name)))
goto create_failed;
/* register some essential callbacks */
pa_context_set_state_callback (pbuf->context,
gst_pulsering_context_state_cb, pbuf);
#if HAVE_PULSE_0_9_12
pa_context_set_subscribe_callback (pbuf->context,
gst_pulsering_context_subscribe_cb, pbuf);
#endif
/* try to connect to the server and wait for completioni, we don't want to
* autospawn a deamon */
GST_LOG_OBJECT (psink, "connect to server %s", GST_STR_NULL (psink->server));
if (pa_context_connect (pbuf->context, psink->server, PA_CONTEXT_NOAUTOSPAWN,
NULL) < 0)
goto connect_failed;
for (;;) {
pa_context_state_t state;
state = pa_context_get_state (pbuf->context);
GST_LOG_OBJECT (psink, "context state is now %d", state);
if (!PA_CONTEXT_IS_GOOD (state))
goto connect_failed;
if (state == PA_CONTEXT_READY)
break;
/* Wait until the context is ready */
GST_LOG_OBJECT (psink, "waiting..");
pa_threaded_mainloop_wait (psink->mainloop);
}
GST_LOG_OBJECT (psink, "opened the device");
pa_threaded_mainloop_unlock (psink->mainloop);
g_free (name);
return TRUE;
/* ERRORS */
unlock_and_fail:
{
gst_pulsering_destroy_context (pbuf);
pa_threaded_mainloop_unlock (psink->mainloop);
g_free (name);
return FALSE;
}
create_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("Failed to create context"), (NULL));
goto unlock_and_fail;
}
connect_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock_and_fail;
}
}
/* close the device */
static gboolean
gst_pulseringbuffer_close_device (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
GST_LOG_OBJECT (psink, "closing device");
pa_threaded_mainloop_lock (psink->mainloop);
gst_pulsering_destroy_context (pbuf);
pa_threaded_mainloop_unlock (psink->mainloop);
GST_LOG_OBJECT (psink, "closed device");
return TRUE;
}
static void
gst_pulsering_stream_state_cb (pa_stream * s, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_stream_state_t state;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
state = pa_stream_get_state (s);
GST_LOG_OBJECT (psink, "got new stream state %d", state);
switch (state) {
case PA_STREAM_READY:
case PA_STREAM_FAILED:
case PA_STREAM_TERMINATED:
GST_LOG_OBJECT (psink, "signaling");
pa_threaded_mainloop_signal (psink->mainloop, 0);
break;
case PA_STREAM_UNCONNECTED:
case PA_STREAM_CREATING:
break;
}
}
static void
gst_pulsering_stream_request_cb (pa_stream * s, size_t length, void *userdata)
{
GstPulseSink *psink;
GstRingBuffer *rbuf;
GstPulseRingBuffer *pbuf;
rbuf = GST_RING_BUFFER_CAST (userdata);
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_LOG_OBJECT (psink, "got request for length %" G_GSIZE_FORMAT, length);
if (pbuf->in_commit) {
/* only signal when we are waiting in the commit thread */
pa_threaded_mainloop_signal (psink->mainloop, 0);
}
}
/* This method should create a new stream of the given @spec. No playback should
* start yet so we start in the corked state. */
static gboolean
gst_pulseringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_buffer_attr buf_attr;
const pa_buffer_attr *buf_attr_ptr;
pa_channel_map channel_map;
pa_operation *o = NULL;
pa_cvolume v, *pv;
pa_stream_flags_t flags;
const gchar *name;
GstAudioClock *clock;
gint64 time_offset;
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
pbuf = GST_PULSERING_BUFFER_CAST (buf);
GST_LOG_OBJECT (psink, "creating sample spec");
/* convert the gstreamer sample spec to the pulseaudio format */
if (!gst_pulse_fill_sample_spec (spec, &pbuf->sample_spec))
goto invalid_spec;
pa_threaded_mainloop_lock (psink->mainloop);
/* we need a context and a no stream */
g_assert (pbuf->context);
g_assert (!pbuf->stream);
/* enable event notifications */
GST_LOG_OBJECT (psink, "subscribing to context events");
if (!(o = pa_context_subscribe (pbuf->context,
PA_SUBSCRIPTION_MASK_SINK_INPUT, NULL, NULL)))
goto subscribe_failed;
pa_operation_unref (o);
/* initialize the channel map */
gst_pulse_gst_to_channel_map (&channel_map, spec);
/* find a good name for the stream */
if (psink->stream_name)
name = psink->stream_name;
else
name = "Playback Stream";
/* create a stream */
GST_LOG_OBJECT (psink, "creating stream with name %s", name);
if (!(pbuf->stream = pa_stream_new (pbuf->context,
name, &pbuf->sample_spec, &channel_map)))
goto stream_failed;
/* install essential callbacks */
pa_stream_set_state_callback (pbuf->stream,
gst_pulsering_stream_state_cb, pbuf);
pa_stream_set_write_callback (pbuf->stream,
gst_pulsering_stream_request_cb, pbuf);
/* buffering requirements. When setting prebuf to 0, the stream will not pause
* when we cause an underrun, which causes time to continue. */
memset (&buf_attr, 0, sizeof (buf_attr));
buf_attr.tlength = spec->segtotal * spec->segsize;
buf_attr.maxlength = buf_attr.tlength * 2;
buf_attr.prebuf = 0;
buf_attr.minreq = spec->segsize;
GST_INFO_OBJECT (psink, "tlength: %d", buf_attr.tlength);
GST_INFO_OBJECT (psink, "maxlength: %d", buf_attr.maxlength);
GST_INFO_OBJECT (psink, "prebuf: %d", buf_attr.prebuf);
GST_INFO_OBJECT (psink, "minreq: %d", buf_attr.minreq);
/* configure volume when we changed it, else we leave the default */
if (psink->volume_set) {
GST_LOG_OBJECT (psink, "have volume of %f", psink->volume);
pv = &v;
gst_pulse_cvolume_from_linear (pv, pbuf->sample_spec.channels,
psink->volume);
} else {
pv = NULL;
}
/* construct the flags */
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
#if HAVE_PULSE_0_9_11
PA_STREAM_ADJUST_LATENCY |
#endif
PA_STREAM_START_CORKED;
/* we always start corked (see flags above) */
pbuf->corked = TRUE;
/* try to connect now */
GST_LOG_OBJECT (psink, "connect for playback to device %s",
GST_STR_NULL (psink->device));
if (pa_stream_connect_playback (pbuf->stream, psink->device,
&buf_attr, flags, pv, NULL) < 0)
goto connect_failed;
/* our clock will now start from 0 again */
clock = GST_AUDIO_CLOCK (GST_BASE_AUDIO_SINK (psink)->provided_clock);
gst_audio_clock_reset (clock, 0);
time_offset = clock->abidata.ABI.time_offset;
GST_LOG_OBJECT (psink, "got time offset %" GST_TIME_FORMAT,
GST_TIME_ARGS (time_offset));
/* calculate the sample offset for 0 */
if (time_offset > 0)
pbuf->offset = gst_util_uint64_scale_int (time_offset,
pbuf->sample_spec.rate, GST_SECOND);
else
pbuf->offset = -gst_util_uint64_scale_int (-time_offset,
pbuf->sample_spec.rate, GST_SECOND);
GST_LOG_OBJECT (psink, "sample offset %" G_GINT64_FORMAT, pbuf->offset);
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state (pbuf->stream);
GST_LOG_OBJECT (psink, "stream state is now %d", state);
if (!PA_STREAM_IS_GOOD (state))
goto connect_failed;
if (state == PA_STREAM_READY)
break;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (psink->mainloop);
}
GST_LOG_OBJECT (psink, "stream is acquired now");
/* get the actual buffering properties now */
buf_attr_ptr = pa_stream_get_buffer_attr (pbuf->stream);
GST_INFO_OBJECT (psink, "tlength: %d", buf_attr_ptr->tlength);
GST_INFO_OBJECT (psink, "maxlength: %d", buf_attr_ptr->maxlength);
GST_INFO_OBJECT (psink, "prebuf: %d", buf_attr_ptr->prebuf);
GST_INFO_OBJECT (psink, "minreq: %d", buf_attr_ptr->minreq);
spec->segsize = buf_attr.minreq;
spec->segtotal = buf_attr.tlength / spec->segsize;
pa_threaded_mainloop_unlock (psink->mainloop);
return TRUE;
/* ERRORS */
unlock_and_fail:
{
gst_pulsering_destroy_stream (pbuf);
pa_threaded_mainloop_unlock (psink->mainloop);
return FALSE;
}
invalid_spec:
{
GST_ELEMENT_ERROR (psink, RESOURCE, SETTINGS,
("Invalid sample specification."), (NULL));
return FALSE;
}
subscribe_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_context_subscribe() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock_and_fail;
}
stream_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("Failed to create stream: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock_and_fail;
}
connect_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock_and_fail;
}
}
/* free the stream that we acquired before */
static gboolean
gst_pulseringbuffer_release (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
pbuf = GST_PULSERING_BUFFER_CAST (buf);
pa_threaded_mainloop_lock (psink->mainloop);
gst_pulsering_destroy_stream (pbuf);
pa_threaded_mainloop_unlock (psink->mainloop);
return TRUE;
}
static void
gst_pulsering_success_cb (pa_stream * s, int success, void *userdata)
{
GstPulseRingBuffer *pbuf;
GstPulseSink *psink;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_signal (psink->mainloop, 0);
}
/* update the corked state of a stream, must be called with the mainloop
* lock */
static gboolean
gst_pulsering_set_corked (GstPulseRingBuffer * pbuf, gboolean corked,
gboolean wait)
{
pa_operation *o = NULL;
GstPulseSink *psink;
gboolean res = FALSE;
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_DEBUG_OBJECT (psink, "setting corked state to %d", corked);
if (pbuf->corked != corked) {
if (!(o = pa_stream_cork (pbuf->stream, corked,
gst_pulsering_success_cb, pbuf)))
goto cork_failed;
while (wait && pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (psink->mainloop);
if (gst_pulsering_is_dead (psink, pbuf))
goto server_dead;
}
pbuf->corked = corked;
}
res = TRUE;
cleanup:
if (o)
pa_operation_unref (o);
return res;
/* ERRORS */
server_dead:
{
GST_DEBUG_OBJECT (psink, "the server is dead");
goto cleanup;
}
cork_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_cork() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto cleanup;
}
}
/* start/resume playback ASAP, we don't uncork here but in the commit method */
static gboolean
gst_pulseringbuffer_start (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (psink->mainloop);
GST_DEBUG_OBJECT (psink, "starting");
pbuf->paused = FALSE;
pa_threaded_mainloop_unlock (psink->mainloop);
return TRUE;
}
/* pause/stop playback ASAP */
static gboolean
gst_pulseringbuffer_pause (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
gboolean res;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (psink->mainloop);
GST_DEBUG_OBJECT (psink, "pausing and corking");
/* make sure the commit method stops writing */
pbuf->paused = TRUE;
res = gst_pulsering_set_corked (pbuf, TRUE, FALSE);
if (pbuf->in_commit) {
/* we are waiting in a commit, signal */
GST_DEBUG_OBJECT (psink, "signal commit");
pa_threaded_mainloop_signal (psink->mainloop, 0);
}
pa_threaded_mainloop_unlock (psink->mainloop);
return res;
}
/* stop playback, we flush everything. */
static gboolean
gst_pulseringbuffer_stop (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
gboolean res = FALSE;
pa_operation *o = NULL;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (psink->mainloop);
pbuf->paused = TRUE;
res = gst_pulsering_set_corked (pbuf, TRUE, TRUE);
/* Inform anyone waiting in _commit() call that it shall wakeup */
if (pbuf->in_commit) {
GST_DEBUG_OBJECT (psink, "signal commit thread");
pa_threaded_mainloop_signal (psink->mainloop, 0);
}
/* then try to flush, it's not fatal when this fails */
GST_DEBUG_OBJECT (psink, "flushing");
if ((o = pa_stream_flush (pbuf->stream, gst_pulsering_success_cb, pbuf))) {
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
GST_DEBUG_OBJECT (psink, "wait for completion");
pa_threaded_mainloop_wait (psink->mainloop);
if (gst_pulsering_is_dead (psink, pbuf))
goto server_dead;
}
GST_DEBUG_OBJECT (psink, "flush completed");
}
res = TRUE;
cleanup:
if (o) {
pa_operation_cancel (o);
pa_operation_unref (o);
}
pa_threaded_mainloop_unlock (psink->mainloop);
return res;
/* ERRORS */
server_dead:
{
GST_DEBUG_OBJECT (psink, "the server is dead");
goto cleanup;
}
}
/* in_samples >= out_samples, rate > 1.0 */
#define FWD_UP_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = s, *db = d; \
while (s <= se && d < de) { \
memcpy (d, s, bps); \
s += bps; \
*accum += outr; \
if ((*accum << 1) >= inr) { \
*accum -= inr; \
d += bps; \
} \
} \
in_samples -= (s - sb)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
/* out_samples > in_samples, for rates smaller than 1.0 */
#define FWD_DOWN_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = s, *db = d; \
while (s <= se && d < de) { \
memcpy (d, s, bps); \
d += bps; \
*accum += inr; \
if ((*accum << 1) >= outr) { \
*accum -= outr; \
s += bps; \
} \
} \
in_samples -= (s - sb)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
#define REV_UP_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
memcpy (d, se, bps); \
se -= bps; \
*accum += outr; \
while ((*accum << 1) >= inr) { \
*accum -= inr; \
d += bps; \
} \
} \
in_samples -= (sb - se)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
#define REV_DOWN_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
memcpy (d, se, bps); \
d += bps; \
*accum += inr; \
while ((*accum << 1) >= outr) { \
*accum -= outr; \
se -= bps; \
} \
} \
in_samples -= (sb - se)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
/* our custom commit function because we write into the buffer of pulseaudio
* instead of keeping our own buffer */
static guint
gst_pulseringbuffer_commit (GstRingBuffer * buf, guint64 * sample,
guchar * data, gint in_samples, gint out_samples, gint * accum)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
guint result;
guint8 *data_end;
gboolean reverse;
gint *toprocess;
gint inr, outr, bps;
gint64 offset;
guint bufsize;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
/* FIXME post message rather than using a signal (as mixer interface) */
if (g_atomic_int_compare_and_exchange (&psink->notify, 1, 0))
g_object_notify (G_OBJECT (psink), "volume");
/* make sure the ringbuffer is started */
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
GST_RING_BUFFER_STATE_STARTED)) {
/* see if we are allowed to start it */
if (G_UNLIKELY (g_atomic_int_get (&buf->abidata.ABI.may_start) == FALSE))
goto no_start;
GST_DEBUG_OBJECT (buf, "start!");
if (!gst_ring_buffer_start (buf))
goto start_failed;
}
pa_threaded_mainloop_lock (psink->mainloop);
GST_DEBUG_OBJECT (psink, "entering commit");
pbuf->in_commit = TRUE;
bps = buf->spec.bytes_per_sample;
bufsize = buf->spec.segsize * buf->spec.segtotal;
/* our toy resampler for trick modes */
reverse = out_samples < 0;
out_samples = ABS (out_samples);
if (in_samples >= out_samples)
toprocess = &in_samples;
else
toprocess = &out_samples;
inr = in_samples - 1;
outr = out_samples - 1;
/* data_end points to the last sample we have to write, not past it. This is
* needed to properly handle reverse playback: it points to the last sample. */
data_end = data + (bps * inr);
if (pbuf->paused)
goto was_paused;
/* correct for sample offset against the internal clock */
offset = *sample;
if (pbuf->offset >= 0) {
if (offset > pbuf->offset)
offset -= pbuf->offset;
else
offset = 0;
} else {
if (offset > -pbuf->offset)
offset += pbuf->offset;
else
offset = 0;
}
offset = *sample * bps;
while (*toprocess > 0) {
size_t avail;
guint towrite;
GST_LOG_OBJECT (psink,
"need to write %d samples at offset %" G_GINT64_FORMAT, *toprocess,
offset);
for (;;) {
/* FIXME, this is not quite right */
if ((avail = pa_stream_writable_size (pbuf->stream)) == (size_t) - 1)
goto writable_size_failed;
/* We always try to satisfy a request for data */
GST_LOG_OBJECT (psink, "writable bytes %" G_GSIZE_FORMAT, avail);
/* convert to samples, we can only deal with multiples of the
* sample size */
avail /= bps;
if (avail > 0)
break;
/* see if we need to uncork because we have no free space */
if (pbuf->corked) {
if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
goto uncork_failed;
}
/* we can't write a single byte, wait a bit */
GST_LOG_OBJECT (psink, "waiting for free space");
pa_threaded_mainloop_wait (psink->mainloop);
if (pbuf->paused)
goto was_paused;
}
if (avail > out_samples)
avail = out_samples;
towrite = avail * bps;
GST_LOG_OBJECT (psink, "writing %d samples at offset %" G_GUINT64_FORMAT,
avail, offset);
if (G_LIKELY (inr == outr && !reverse)) {
/* no rate conversion, simply write out the samples */
if (pa_stream_write (pbuf->stream, data, towrite, NULL, offset,
PA_SEEK_ABSOLUTE) < 0)
goto write_failed;
data += towrite;
in_samples -= avail;
out_samples -= avail;
} else {
guint8 *dest, *d, *d_end;
/* we need to allocate a temporary buffer to resample the data into,
* FIXME, we should have a pulseaudio API to allocate this buffer for us
* from the shared memory. */
dest = d = g_malloc (towrite);
d_end = d + towrite;
if (!reverse) {
if (inr >= outr)
/* forward speed up */
FWD_UP_SAMPLES (data, data_end, d, d_end);
else
/* forward slow down */
FWD_DOWN_SAMPLES (data, data_end, d, d_end);
} else {
if (inr >= outr)
/* reverse speed up */
REV_UP_SAMPLES (data, data_end, d, d_end);
else
/* reverse slow down */
REV_DOWN_SAMPLES (data, data_end, d, d_end);
}
/* see what we have left to write */
towrite = (d - dest);
if (pa_stream_write (pbuf->stream, dest, towrite,
g_free, offset, PA_SEEK_ABSOLUTE) < 0)
goto write_failed;
avail = towrite / bps;
}
*sample += avail;
offset = *sample * bps;
/* check if we need to uncork after writing the samples */
if (pbuf->corked) {
const pa_timing_info *info = pa_stream_get_timing_info (pbuf->stream);
GST_LOG_OBJECT (psink,
"read_index at %" G_GUINT64_FORMAT ", offset %" G_GINT64_FORMAT,
info->read_index, offset);
/* we uncork when the read_index is too far behind the offset we need
* to write to. */
if (info->read_index + bufsize <= offset) {
if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
goto uncork_failed;
}
}
}
/* we consumed all samples here */
data = data_end + bps;
pbuf->in_commit = FALSE;
pa_threaded_mainloop_unlock (psink->mainloop);
done:
result = inr - ((data_end - data) / bps);
GST_LOG_OBJECT (psink, "wrote %d samples", result);
return result;
/* ERRORS */
unlock_and_fail:
{
pbuf->in_commit = FALSE;
GST_LOG_OBJECT (psink, "we are reset");
pa_threaded_mainloop_unlock (psink->mainloop);
goto done;
}
no_start:
{
GST_LOG_OBJECT (psink, "we can not start");
return 0;
}
start_failed:
{
GST_LOG_OBJECT (psink, "failed to start the ringbuffer");
return 0;
}
uncork_failed:
{
pbuf->in_commit = FALSE;
GST_ERROR_OBJECT (psink, "uncork failed");
pa_threaded_mainloop_unlock (psink->mainloop);
goto done;
}
was_paused:
{
pbuf->in_commit = FALSE;
GST_LOG_OBJECT (psink, "we are paused");
pa_threaded_mainloop_unlock (psink->mainloop);
goto done;
}
writable_size_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_writable_size() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock_and_fail;
}
write_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_write() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock_and_fail;
}
}
static void gst_pulsesink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_pulsesink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_pulsesink_finalize (GObject * object);
static gboolean gst_pulsesink_event (GstBaseSink * sink, GstEvent * event);
static void gst_pulsesink_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_PULSEPROBE_METHODS (GstPulseSink, gst_pulsesink);
GST_BOILERPLATE_FULL (GstPulseSink, gst_pulsesink, GstBaseAudioSink,
GST_TYPE_BASE_AUDIO_SINK, gst_pulsesink_init_interfaces);
static gboolean
gst_pulsesink_interface_supported (GstImplementsInterface *
iface, GType interface_type)
{
GstPulseSink *this = GST_PULSESINK_CAST (iface);
if (interface_type == GST_TYPE_PROPERTY_PROBE && this->probe)
return TRUE;
return FALSE;
}
static void
gst_pulsesink_implements_interface_init (GstImplementsInterfaceClass * klass)
{
klass->supported = gst_pulsesink_interface_supported;
}
static void
gst_pulsesink_init_interfaces (GType type)
{
static const GInterfaceInfo implements_iface_info = {
(GInterfaceInitFunc) gst_pulsesink_implements_interface_init,
NULL,
NULL,
};
static const GInterfaceInfo probe_iface_info = {
(GInterfaceInitFunc) gst_pulsesink_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_PROPERTY_PROBE,
&probe_iface_info);
}
static void
gst_pulsesink_base_init (gpointer g_class)
{
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
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 Sink",
"Sink/Audio", "Plays audio to a PulseAudio server", "Lennart Poettering");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&pad_template));
}
static GstRingBuffer *
gst_pulsesink_create_ringbuffer (GstBaseAudioSink * sink)
{
GstRingBuffer *buffer;
GST_DEBUG_OBJECT (sink, "creating ringbuffer");
buffer = g_object_new (GST_TYPE_PULSERING_BUFFER, NULL);
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
return buffer;
}
static void
gst_pulsesink_class_init (GstPulseSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
GstBaseSinkClass *bc;
GstBaseAudioSinkClass *gstaudiosink_class = GST_BASE_AUDIO_SINK_CLASS (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulsesink_finalize);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsesink_set_property);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_pulsesink_get_property);
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_pulsesink_event);
/* restore the original basesink pull methods */
bc = g_type_class_peek (GST_TYPE_BASE_SINK);
gstbasesink_class->activate_pull = GST_DEBUG_FUNCPTR (bc->activate_pull);
gstaudiosink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_pulsesink_create_ringbuffer);
/* Overwrite GObject fields */
g_object_class_install_property (gobject_class,
PROP_SERVER,
g_param_spec_string ("server", "Server",
"The PulseAudio server to connect to", NULL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Sink",
"The PulseAudio sink device to connect to", NULL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device", NULL,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
#if HAVE_PULSE_0_9_12
g_object_class_install_property (gobject_class,
PROP_VOLUME,
g_param_spec_double ("volume", "Volume",
"Volume of this stream", 0.0, 1000.0, 1.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
#endif
}
/* returns the current time of the sink ringbuffer */
static GstClockTime
gst_pulse_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_usec_t time;
if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0)
return GST_CLOCK_TIME_NONE;
pbuf = GST_PULSERING_BUFFER_CAST (sink->ringbuffer);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (psink->mainloop);
/* if we don't have enough data to get a timestamp, just return NONE, which
* will return the last reported time */
if (pa_stream_get_time (pbuf->stream, &time) < 0) {
GST_DEBUG_OBJECT (psink, "could not get time");
time = GST_CLOCK_TIME_NONE;
} else
time *= 1000;
pa_threaded_mainloop_unlock (psink->mainloop);
GST_LOG_OBJECT (psink, "current time is %" GST_TIME_FORMAT,
GST_TIME_ARGS (time));
return time;
}
static void
gst_pulsesink_init (GstPulseSink * pulsesink, GstPulseSinkClass * klass)
{
pulsesink->server = NULL;
pulsesink->device = NULL;
pulsesink->device_description = NULL;
pulsesink->volume = 1.0;
pulsesink->volume_set = FALSE;
pulsesink->notify = 0;
g_assert ((pulsesink->mainloop = pa_threaded_mainloop_new ()));
g_assert (pa_threaded_mainloop_start (pulsesink->mainloop) == 0);
/* TRUE for sinks, FALSE for sources */
pulsesink->probe = gst_pulseprobe_new (G_OBJECT (pulsesink),
G_OBJECT_GET_CLASS (pulsesink), PROP_DEVICE, pulsesink->device,
TRUE, FALSE);
/* override with a custom clock */
if (GST_BASE_AUDIO_SINK (pulsesink)->provided_clock)
gst_object_unref (GST_BASE_AUDIO_SINK (pulsesink)->provided_clock);
GST_BASE_AUDIO_SINK (pulsesink)->provided_clock =
gst_audio_clock_new ("GstPulseSinkClock",
(GstAudioClockGetTimeFunc) gst_pulse_sink_get_time, pulsesink);
}
static void
gst_pulsesink_finalize (GObject * object)
{
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
pa_threaded_mainloop_stop (pulsesink->mainloop);
g_free (pulsesink->server);
g_free (pulsesink->device);
pa_threaded_mainloop_free (pulsesink->mainloop);
if (pulsesink->probe) {
gst_pulseprobe_free (pulsesink->probe);
pulsesink->probe = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
#if HAVE_PULSE_0_9_12
static void
gst_pulsesink_set_volume (GstPulseSink * psink, gdouble volume)
{
pa_cvolume v;
pa_operation *o = NULL;
GstPulseRingBuffer *pbuf;
pa_threaded_mainloop_lock (psink->mainloop);
GST_DEBUG_OBJECT (psink, "setting volume to %f", volume);
psink->volume = volume;
psink->volume_set = TRUE;
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto unlock;
gst_pulse_cvolume_from_linear (&v, pbuf->sample_spec.channels, volume);
if (!(o = pa_context_set_sink_input_volume (pbuf->context,
pa_stream_get_index (pbuf->stream), &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 (psink->mainloop);
return;
/* ERRORS */
volume_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_set_sink_input_volume() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
static void
gst_pulsesink_sink_input_info_cb (pa_context * c, const pa_sink_input_info * i,
int eol, void *userdata)
{
GstPulseRingBuffer *pbuf;
GstPulseSink *psink;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
if (!i)
return;
if (!pbuf->stream)
return;
g_assert (i->index == pa_stream_get_index (pbuf->stream));
psink->volume = pa_sw_volume_to_linear (pa_cvolume_max (&i->volume));
}
static gdouble
gst_pulsesink_get_volume (GstPulseSink * psink)
{
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
gdouble v;
pa_threaded_mainloop_lock (psink->mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if (!(o = pa_context_get_sink_input_info (pbuf->context,
pa_stream_get_index (pbuf->stream),
gst_pulsesink_sink_input_info_cb, pbuf)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (psink->mainloop);
if (gst_pulsering_is_dead (psink, pbuf))
goto unlock;
}
unlock:
if (o)
pa_operation_unref (o);
v = psink->volume;
pa_threaded_mainloop_unlock (psink->mainloop);
return v;
/* ERRORS */
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
info_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_get_sink_input_info() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
#endif
static void
gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
void *userdata)
{
GstPulseRingBuffer *pbuf;
GstPulseSink *psink;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
if (!i)
return;
if (!pbuf->stream)
return;
g_assert (i->index == pa_stream_get_device_index (pbuf->stream));
g_free (psink->device_description);
psink->device_description = g_strdup (i->description);
}
static gchar *
gst_pulsesink_device_description (GstPulseSink * psink)
{
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
gchar *t;
pa_threaded_mainloop_lock (psink->mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if (!(o = pa_context_get_sink_info_by_index (pbuf->context,
pa_stream_get_device_index (pbuf->stream),
gst_pulsesink_sink_info_cb, pbuf)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (psink->mainloop);
if (gst_pulsering_is_dead (psink, pbuf))
goto unlock;
}
unlock:
if (o)
pa_operation_unref (o);
t = g_strdup (psink->device_description);
pa_threaded_mainloop_unlock (psink->mainloop);
return t;
/* ERRORS */
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
info_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_get_sink_info() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
static void
gst_pulsesink_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
switch (prop_id) {
case PROP_SERVER:
g_free (pulsesink->server);
pulsesink->server = g_value_dup_string (value);
if (pulsesink->probe)
gst_pulseprobe_set_server (pulsesink->probe, pulsesink->server);
break;
case PROP_DEVICE:
g_free (pulsesink->device);
pulsesink->device = g_value_dup_string (value);
break;
#if HAVE_PULSE_0_9_12
case PROP_VOLUME:
gst_pulsesink_set_volume (pulsesink, g_value_get_double (value));
break;
#endif
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesink_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
switch (prop_id) {
case PROP_SERVER:
g_value_set_string (value, pulsesink->server);
break;
case PROP_DEVICE:
g_value_set_string (value, pulsesink->device);
break;
case PROP_DEVICE_NAME:{
char *t = gst_pulsesink_device_description (pulsesink);
g_value_set_string (value, t);
g_free (t);
break;
}
#if HAVE_PULSE_0_9_12
case PROP_VOLUME:
g_value_set_double (value, gst_pulsesink_get_volume (pulsesink));
break;
#endif
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesink_change_title (GstPulseSink * psink, const gchar * t)
{
pa_operation *o = NULL;
GstPulseRingBuffer *pbuf;
pa_threaded_mainloop_lock (psink->mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
g_free (pbuf->stream_name);
pbuf->stream_name = g_strdup (t);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if (!(o = pa_stream_set_name (pbuf->stream, pbuf->stream_name, NULL, NULL)))
goto name_failed;
/* We're not interested if this operation failed or not */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (psink->mainloop);
return;
/* ERRORS */
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
name_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_set_name() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
#if HAVE_PULSE_0_9_11
static void
gst_pulsesink_change_props (GstPulseSink * psink, GstTagList * l)
{
static const gchar *const map[] = {
GST_TAG_TITLE, PA_PROP_MEDIA_TITLE,
GST_TAG_ARTIST, PA_PROP_MEDIA_ARTIST,
GST_TAG_LANGUAGE_CODE, PA_PROP_MEDIA_LANGUAGE,
GST_TAG_LOCATION, PA_PROP_MEDIA_FILENAME,
/* We might add more here later on ... */
NULL
};
pa_proplist *pl = NULL;
const gchar *const *t;
gboolean empty = TRUE;
pa_operation *o = NULL;
GstPulseRingBuffer *pbuf;
pl = pa_proplist_new ();
for (t = map; *t; t += 2) {
gchar *n = NULL;
if (gst_tag_list_get_string (l, *t, &n)) {
if (n && *n) {
pa_proplist_sets (pl, *(t + 1), n);
empty = FALSE;
}
g_free (n);
}
}
if (empty)
goto finish;
pa_threaded_mainloop_lock (psink->mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if (!(o = pa_stream_proplist_update (pbuf->stream, PA_UPDATE_REPLACE,
pl, NULL, NULL)))
goto update_failed;
/* We're not interested if this operation failed or not */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (psink->mainloop);
finish:
if (pl)
pa_proplist_free (pl);
return;
/* ERRORS */
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
update_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_proplist_update() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
#endif
static gboolean
gst_pulsesink_event (GstBaseSink * sink, GstEvent * event)
{
GstPulseSink *pulsesink = GST_PULSESINK_CAST (sink);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_TAG:{
gchar *title = NULL, *artist = NULL, *location = NULL, *description =
NULL, *t = NULL, *buf = NULL;
GstTagList *l;
gst_event_parse_tag (event, &l);
gst_tag_list_get_string (l, GST_TAG_TITLE, &title);
gst_tag_list_get_string (l, GST_TAG_ARTIST, &artist);
gst_tag_list_get_string (l, GST_TAG_LOCATION, &location);
gst_tag_list_get_string (l, GST_TAG_DESCRIPTION, &description);
if (title && artist)
t = buf =
g_strdup_printf ("'%s' by '%s'", g_strstrip (title),
g_strstrip (artist));
else if (title)
t = g_strstrip (title);
else if (description)
t = g_strstrip (description);
else if (location)
t = g_strstrip (location);
if (t)
gst_pulsesink_change_title (pulsesink, t);
g_free (title);
g_free (artist);
g_free (location);
g_free (description);
g_free (buf);
#if HAVE_PULSE_0_9_11
gst_pulsesink_change_props (pulsesink, l);
#endif
break;
}
default:
;
}
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
}
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