mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-27 10:40:34 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/ext/pulse/pulsesink.c
Tim-Philipp Müller 2270d3359a pulsesink: printf format fix
2009-11-05 23:46:58 +00:00

2225 lines
62 KiB
C
Raw Blame History

/*-*- Mode: C; c-basic-offset: 2 -*-*/
/* 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 <gst/interfaces/streamvolume.h>
#include <gst/gst-i18n-plugin.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
*/
#define DEFAULT_SERVER NULL
#define DEFAULT_DEVICE NULL
#define DEFAULT_DEVICE_NAME NULL
#define DEFAULT_VOLUME 1.0
#define DEFAULT_MUTE FALSE
#define MAX_VOLUME 10.0
enum
{
PROP_0,
PROP_SERVER,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_VOLUME,
PROP_MUTE,
PROP_LAST
};
#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;
gboolean corked:1;
gboolean in_commit:1;
gboolean paused:1;
};
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 void gst_pulseringbuffer_clear (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;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) 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->clear_all =
GST_DEBUG_FUNCPTR (gst_pulseringbuffer_clear);
gstringbuffer_class->commit = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_commit);
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
g_type_class_ref (GST_TYPE_PULSERING_BUFFER);
}
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->corked = TRUE;
pbuf->in_commit = FALSE;
pbuf->paused = FALSE;
}
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_set_underflow_callback (pbuf->stream, NULL, NULL);
pa_stream_set_overflow_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);
#if HAVE_PULSE_0_9_12
pa_context_set_subscribe_callback (pbuf->context, NULL, NULL);
#endif
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 && (length >= rbuf->spec.segsize)) {
/* only signal when we are waiting in the commit thread
* and got request for atleast a segment */
pa_threaded_mainloop_signal (psink->mainloop, 0);
}
}
static void
gst_pulsering_stream_underflow_cb (pa_stream * s, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_WARNING_OBJECT (psink, "Got underflow");
}
static void
gst_pulsering_stream_overflow_cb (pa_stream * s, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_WARNING_OBJECT (psink, "Got overflow");
}
static void
gst_pulsering_stream_latency_cb (pa_stream * s, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
const pa_timing_info *info;
pa_usec_t sink_usec;
info = pa_stream_get_timing_info (s);
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
if (!info) {
GST_LOG_OBJECT (psink, "latency update (information unknown)");
return;
}
#if HAVE_PULSE_0_9_11
sink_usec = info->configured_sink_usec;
#else
sink_usec = 0;
#endif
GST_LOG_OBJECT (psink,
"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->sink_usec, sink_usec);
}
#if HAVE_PULSE_0_9_15
static void
gst_pulsering_stream_event_cb (pa_stream * p, const char *name,
pa_proplist * pl, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
if (!strcmp (name, PA_STREAM_EVENT_REQUEST_CORK)) {
/* the stream wants to PAUSE, post a message for the application. */
GST_DEBUG_OBJECT (psink, "got request for CORK");
gst_element_post_message (GST_ELEMENT_CAST (psink),
gst_message_new_request_state (GST_OBJECT_CAST (psink),
GST_STATE_PAUSED));
} else if (!strcmp (name, PA_STREAM_EVENT_REQUEST_UNCORK)) {
GST_DEBUG_OBJECT (psink, "got request for UNCORK");
gst_element_post_message (GST_ELEMENT_CAST (psink),
gst_message_new_request_state (GST_OBJECT_CAST (psink),
GST_STATE_PLAYING));
} else {
GST_DEBUG_OBJECT (psink, "got unknown event %s", name);
}
}
#endif
/* 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 wanted;
const pa_buffer_attr *actual;
pa_channel_map channel_map;
pa_operation *o = NULL;
#if HAVE_PULSE_0_9_20
pa_cvolume v;
#endif
pa_cvolume *pv = NULL;
pa_stream_flags_t flags;
const gchar *name;
GstAudioClock *clock;
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);
pa_stream_set_underflow_callback (pbuf->stream,
gst_pulsering_stream_underflow_cb, pbuf);
pa_stream_set_overflow_callback (pbuf->stream,
gst_pulsering_stream_overflow_cb, pbuf);
pa_stream_set_latency_update_callback (pbuf->stream,
gst_pulsering_stream_latency_cb, pbuf);
#if HAVE_PULSE_0_9_15
pa_stream_set_event_callback (pbuf->stream,
gst_pulsering_stream_event_cb, pbuf);
#endif
/* buffering requirements. When setting prebuf to 0, the stream will not pause
* when we cause an underrun, which causes time to continue. */
memset (&wanted, 0, sizeof (wanted));
wanted.tlength = spec->segtotal * spec->segsize;
wanted.maxlength = -1;
wanted.prebuf = 0;
wanted.minreq = spec->segsize;
GST_INFO_OBJECT (psink, "tlength: %d", wanted.tlength);
GST_INFO_OBJECT (psink, "maxlength: %d", wanted.maxlength);
GST_INFO_OBJECT (psink, "prebuf: %d", wanted.prebuf);
GST_INFO_OBJECT (psink, "minreq: %d", wanted.minreq);
#if HAVE_PULSE_0_9_20
/* 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;
}
#endif
/* 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;
#if HAVE_PULSE_0_9_12
if (psink->mute_set && psink->mute)
flags |= PA_STREAM_START_MUTED;
#endif
/* 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,
&wanted, 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);
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);
}
/* After we passed the volume off of to PA we never want to set it
again, since it is PA's job to save/restore volumes. */
psink->volume_set = psink->mute_set = FALSE;
GST_LOG_OBJECT (psink, "stream is acquired now");
/* get the actual buffering properties now */
actual = pa_stream_get_buffer_attr (pbuf->stream);
GST_INFO_OBJECT (psink, "tlength: %d (wanted: %d)", actual->tlength,
wanted.tlength);
GST_INFO_OBJECT (psink, "maxlength: %d", actual->maxlength);
GST_INFO_OBJECT (psink, "prebuf: %d", actual->prebuf);
GST_INFO_OBJECT (psink, "minreq: %d (wanted %d)", actual->minreq,
wanted.minreq);
spec->segsize = actual->minreq;
spec->segtotal = actual->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;
} else {
GST_DEBUG_OBJECT (psink, "skipping, already in requested state");
}
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;
}
}
static void
gst_pulseringbuffer_clear (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (psink->mainloop);
GST_DEBUG_OBJECT (psink, "clearing");
if (pbuf->stream) {
/* don't wait for the flush to complete */
if ((o = pa_stream_flush (pbuf->stream, NULL, pbuf)))
pa_operation_unref (o);
}
pa_threaded_mainloop_unlock (psink->mainloop);
}
static void
mainloop_enter_defer_cb (pa_mainloop_api * api, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
GstMessage *message;
GValue val = { 0 };
g_value_init (&val, G_TYPE_POINTER);
g_value_set_pointer (&val, g_thread_self ());
GST_DEBUG_OBJECT (pulsesink, "posting ENTER stream status");
message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT (pulsesink));
gst_message_set_stream_status_object (message, &val);
gst_element_post_message (GST_ELEMENT (pulsesink), message);
/* signal the waiter */
pulsesink->pa_defer_ran = TRUE;
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
}
/* 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, "scheduling stream status");
psink->pa_defer_ran = FALSE;
pa_mainloop_api_once (pa_threaded_mainloop_get_api (psink->mainloop),
mainloop_enter_defer_cb, psink);
GST_DEBUG_OBJECT (psink, "starting");
pbuf->paused = FALSE;
gst_pulsering_set_corked (pbuf, FALSE, 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;
}
static void
mainloop_leave_defer_cb (pa_mainloop_api * api, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
GstMessage *message;
GValue val = { 0 };
g_value_init (&val, G_TYPE_POINTER);
g_value_set_pointer (&val, g_thread_self ());
GST_DEBUG_OBJECT (pulsesink, "posting LEAVE stream status");
message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT (pulsesink));
gst_message_set_stream_status_object (message, &val);
gst_element_post_message (GST_ELEMENT (pulsesink), message);
pulsesink->pa_defer_ran = TRUE;
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
}
/* 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);
}
if (strcmp (psink->pa_version, "0.9.12")) {
/* 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);
}
GST_DEBUG_OBJECT (psink, "scheduling stream status");
psink->pa_defer_ran = FALSE;
pa_mainloop_api_once (pa_threaded_mainloop_get_api (psink->mainloop),
mainloop_leave_defer_cb, psink);
GST_DEBUG_OBJECT (psink, "waiting for stream status");
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 (d < de && (*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 (s <= se && (*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");
g_object_notify (G_OBJECT (psink), "mute");
}
/* 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;
GST_DEBUG_OBJECT (psink, "in %d, out %d", inr, outr);
/* 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;
/* offset is in bytes */
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 %u samples at offset %" G_GUINT64_FORMAT,
(guint) 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 += avail * bps;
/* check if we need to uncork after writing the samples */
if (pbuf->corked) {
const pa_timing_info *info;
if ((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;
}
} else {
GST_LOG_OBJECT (psink, "no timing info available yet");
}
}
}
/* 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;
if (interface_type == GST_TYPE_STREAM_VOLUME)
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,
};
#if HAVE_PULSE_0_9_12
static const GInterfaceInfo svol_iface_info = {
NULL, NULL, NULL
};
g_type_add_interface_static (type, GST_TYPE_STREAM_VOLUME, &svol_iface_info);
#endif
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
&implements_iface_info);
g_type_add_interface_static (type, GST_TYPE_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 ];"
#if HAVE_PULSE_0_9_15
"audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 24, "
"depth = (int) 24, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 32 ];"
"audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 32, "
"depth = (int) 24, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
#endif
"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", 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 sink device to connect to", DEFAULT_DEVICE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
#if HAVE_PULSE_0_9_12
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));
g_object_class_install_property (gobject_class,
PROP_MUTE,
g_param_spec_boolean ("mute", "Mute",
"Mute state of this stream", DEFAULT_MUTE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
#endif
}
/* returns the current time of the sink ringbuffer */
static GstClockTime
gst_pulsesink_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 (gst_pulsering_is_dead (psink, pbuf))
goto server_dead;
/* 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;
/* ERRORS */
server_dead:
{
GST_DEBUG_OBJECT (psink, "the server is dead");
pa_threaded_mainloop_unlock (psink->mainloop);
return GST_CLOCK_TIME_NONE;
}
}
static void
gst_pulsesink_init (GstPulseSink * pulsesink, GstPulseSinkClass * klass)
{
guint res;
pulsesink->server = NULL;
pulsesink->device = NULL;
pulsesink->device_description = NULL;
pulsesink->volume = DEFAULT_VOLUME;
pulsesink->volume_set = FALSE;
pulsesink->mute = DEFAULT_MUTE;
pulsesink->mute_set = FALSE;
pulsesink->notify = 0;
/* needed for conditional execution */
pulsesink->pa_version = pa_get_library_version ();
GST_DEBUG_OBJECT (pulsesink, "using pulseaudio version %s",
pulsesink->pa_version);
pulsesink->mainloop = pa_threaded_mainloop_new ();
g_assert (pulsesink->mainloop != NULL);
res = pa_threaded_mainloop_start (pulsesink->mainloop);
g_assert (res == 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_pulsesink_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);
g_free (pulsesink->device_description);
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;
uint32_t idx;
pa_threaded_mainloop_lock (psink->mainloop);
GST_DEBUG_OBJECT (psink, "setting volume to %f", volume);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
goto no_index;
gst_pulse_cvolume_from_linear (&v, pbuf->sample_spec.channels, volume);
if (!(o = pa_context_set_sink_input_volume (pbuf->context, 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 (psink->mainloop);
return;
/* ERRORS */
no_buffer:
{
psink->volume = volume;
psink->volume_set = TRUE;
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
no_index:
{
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
goto unlock;
}
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_set_mute (GstPulseSink * psink, gboolean mute)
{
pa_operation *o = NULL;
GstPulseRingBuffer *pbuf;
uint32_t idx;
pa_threaded_mainloop_lock (psink->mainloop);
GST_DEBUG_OBJECT (psink, "setting mute state to %d", mute);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
goto no_index;
if (!(o = pa_context_set_sink_input_mute (pbuf->context, 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 (psink->mainloop);
return;
/* ERRORS */
no_buffer:
{
psink->mute = mute;
psink->mute_set = TRUE;
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
no_index:
{
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
goto unlock;
}
mute_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_set_sink_input_mute() 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)
goto done;
if (!pbuf->stream)
goto done;
/* If the index doesn't match our current stream,
* it implies we just recreated the stream (caps change)
*/
if (i->index == pa_stream_get_index (pbuf->stream)) {
psink->volume = pa_sw_volume_to_linear (pa_cvolume_max (&i->volume));
psink->mute = i->mute;
}
done:
pa_threaded_mainloop_signal (psink->mainloop, 0);
}
static gdouble
gst_pulsesink_get_volume (GstPulseSink * psink)
{
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
gdouble v = DEFAULT_VOLUME;
uint32_t idx;
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 ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
goto no_index;
if (!(o = pa_context_get_sink_input_info (pbuf->context, idx,
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;
}
v = psink->volume;
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (psink->mainloop);
if (v > MAX_VOLUME) {
GST_WARNING_OBJECT (psink, "Clipped volume from %f to %f", v, MAX_VOLUME);
v = MAX_VOLUME;
}
return v;
/* ERRORS */
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
no_index:
{
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
goto unlock;
}
info_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_context_get_sink_input_info() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
static gboolean
gst_pulsesink_get_mute (GstPulseSink * psink)
{
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
uint32_t idx;
gboolean mute = FALSE;
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 ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
goto no_index;
if (!(o = pa_context_get_sink_input_info (pbuf->context, idx,
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;
}
mute = psink->mute;
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (psink->mainloop);
return mute;
/* ERRORS */
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
no_index:
{
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
goto unlock;
}
info_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_context_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)
goto done;
if (!pbuf->stream)
goto done;
g_assert (i->index == pa_stream_get_device_index (pbuf->stream));
g_free (psink->device_description);
psink->device_description = g_strdup (i->description);
done:
pa_threaded_mainloop_signal (psink->mainloop, 0);
}
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_context_get_sink_info_by_index() 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;
case PROP_MUTE:
gst_pulsesink_set_mute (pulsesink, g_value_get_boolean (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:
g_value_take_string (value, gst_pulsesink_device_description (pulsesink));
break;
#if HAVE_PULSE_0_9_12
case PROP_VOLUME:
g_value_set_double (value, gst_pulsesink_get_volume (pulsesink));
break;
case PROP_MUTE:
g_value_set_boolean (value, gst_pulsesink_get_mute (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);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
g_free (pbuf->stream_name);
pbuf->stream_name = g_strdup (t);
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,
/* might get overriden in the next iteration by GST_TAG_ARTIST */
GST_TAG_PERFORMER, PA_PROP_MEDIA_ARTIST,
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 (!artist)
gst_tag_list_get_string (l, GST_TAG_PERFORMER, &artist);
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);
}
Reference in a new issue View git blame Copy permalink