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pulse: New pulseaudiosink element to handle format changes

Browse source 
This introduces a new bin which wraps around pulsesink and depending on
the formats supported by the sink, plugs in/out a decodebin2 as
required. This allows users to switch sinks on the stream and adapts
accordingly (for example, you could watch a movie in passthrough mode on
your receiver which supports AC3 decode, then plug out and switch to a
non-digital profile to continue uninterrupted on analog output).

The bin is required because doing the same with playbin2/playsink will
require API changes that cannot be made in 0.10. With 0.11/1.0, we
should be able to ask for upstream caps renegotiation to deal with all
this.

https://bugzilla.gnome.org/show_bug.cgi?id=657179
This commit is contained in:
Arun Raghavan 2011-03-29 12:09:18 +05:30
parent 11b0a0effc
commit 8ca420f547
6 changed files with 1053 additions and 58 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

4
ext/pulse/Makefile.am
View file

@ -7,12 +7,14 @@ libgstpulse_la_SOURCES = \
pulsemixertrack.c \
pulseprobe.c \
pulsesink.c \
pulseaudiosink.c \
pulsesrc.c \
pulseutil.c
libgstpulse_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(PULSE_CFLAGS)
libgstpulse_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) \
-lgstinterfaces-$(GST_MAJORMINOR) $(GST_BASE_LIBS) $(GST_LIBS) $(PULSE_LIBS)
-lgstinterfaces-$(GST_MAJORMINOR) -lgstpbutils-$(GST_MAJORMINOR) \
$(GST_BASE_LIBS) $(GST_LIBS) $(PULSE_LIBS)
libgstpulse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstpulse_la_LIBTOOLFLAGS = --tag=disable-static

6
ext/pulse/plugin.c
View file

@ -49,6 +49,12 @@ plugin_init (GstPlugin * plugin)
GST_TYPE_PULSESRC))
return FALSE;
#ifdef HAVE_PULSE_1_0
if (!gst_element_register (plugin, "pulseaudiosink", GST_RANK_PRIMARY + 11,
GST_TYPE_PULSE_AUDIO_SINK))
return FALSE;
#endif
if (!gst_element_register (plugin, "pulsemixer", GST_RANK_NONE,
GST_TYPE_PULSEMIXER))
return FALSE;

927
ext/pulse/pulseaudiosink.c Normal file
View file

@ -0,0 +1,927 @@
/*-*- Mode: C; c-basic-offset: 2 -*-*/
/* GStreamer pulseaudio plugin
*
* Copyright (c) 2011 Intel Corporation
* 2011 Collabora
* 2011 Arun Raghavan <arun.raghavan@collabora.co.uk>
* 2011 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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-pulseaudiosink
* @see_also: pulsesink, pulsesrc, pulsemixer
*
* This element outputs audio to a
* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink> via
* the @pulsesink element. It transparently takes care of passing compressed
* format as-is if the sink supports it, decoding if necessary, and changes
* to supported formats at runtime.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! pulseaudiosink
* ]| Decode and play an Ogg/Vorbis file.
* |[
* gst-launch -v filesrc location=test.mp3 ! mp3parse ! pulseaudiosink stream-properties="props,media.title=test"
* ]| Play an MP3 file on a sink that supports decoding directly, plug in a
* decoder if/when required.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#ifdef HAVE_PULSE_1_0
#include <gst/pbutils/pbutils.h>
#include <gst/gst-i18n-plugin.h>
#include <gst/audio/gstaudioiec61937.h>
#include "pulsesink.h"
GST_DEBUG_CATEGORY (pulseaudiosink_debug);
#define GST_CAT_DEFAULT (pulseaudiosink_debug)
#define GST_PULSE_AUDIO_SINK_LOCK(obj) G_STMT_START { \
GST_LOG_OBJECT (obj, \
"locking from thread %p", \
g_thread_self ()); \
g_mutex_lock (GST_PULSE_AUDIO_SINK_CAST(obj)->lock); \
GST_LOG_OBJECT (obj, \
"locked from thread %p", \
g_thread_self ()); \
} G_STMT_END
#define GST_PULSE_AUDIO_SINK_UNLOCK(obj) G_STMT_START { \
GST_LOG_OBJECT (obj, \
"unlocking from thread %p", \
g_thread_self ()); \
g_mutex_unlock (GST_PULSE_AUDIO_SINK_CAST(obj)->lock); \
} G_STMT_END
typedef struct
{
GstBin parent;
GMutex *lock;
GstPad *sinkpad;
GstPad *sink_proxypad;
GstPadEventFunction sinkpad_old_eventfunc;
GstPadEventFunction proxypad_old_eventfunc;
GstPulseSink *psink;
GstElement *dbin2;
GstSegment segment;
guint event_probe_id;
gulong pad_added_id;
gboolean format_lost;
} GstPulseAudioSink;
typedef struct
{
GstBinClass parent_class;
guint n_prop_own;
guint n_prop_total;
} GstPulseAudioSinkClass;
static void gst_pulse_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_pulse_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_pulse_audio_sink_dispose (GObject * object);
static gboolean gst_pulse_audio_sink_src_event (GstPad * pad, GstEvent * event);
static gboolean gst_pulse_audio_sink_sink_event (GstPad * pad,
GstEvent * event);
static gboolean gst_pulse_audio_sink_sink_acceptcaps (GstPad * pad,
GstCaps * caps);
static gboolean gst_pulse_audio_sink_sink_setcaps (GstPad * pad,
GstCaps * caps);
static GstStateChangeReturn
gst_pulse_audio_sink_change_state (GstElement * element,
GstStateChange transition);
static void
gst_pulse_audio_sink_do_init (GType type)
{
GST_DEBUG_CATEGORY_INIT (pulseaudiosink_debug, "pulseaudiosink", 0,
"Bin that wraps pulsesink for handling compressed formats");
}
GST_BOILERPLATE_FULL (GstPulseAudioSink, gst_pulse_audio_sink, GstBin,
GST_TYPE_BIN, gst_pulse_audio_sink_do_init);
static GstStaticPadTemplate sink_template =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS (PULSE_SINK_TEMPLATE_CAPS));
static void
gst_pulse_audio_sink_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details_simple (element_class,
"Bin wrapping pulsesink", "Sink/Audio/Bin",
"Correctly handles sink changes when streaming compressed formats to "
"pulsesink", "Arun Raghavan <arun.raghavan@collabora.co.uk>");
}
static GParamSpec *
param_spec_copy (GParamSpec * spec)
{
const char *name, *nick, *blurb;
GParamFlags flags;
name = g_param_spec_get_name (spec);
nick = g_param_spec_get_nick (spec);
blurb = g_param_spec_get_blurb (spec);
flags = spec->flags;
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_BOOLEAN) {
return g_param_spec_boolean (name, nick, blurb,
G_PARAM_SPEC_BOOLEAN (spec)->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_BOXED) {
return g_param_spec_boxed (name, nick, blurb, spec->value_type, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_CHAR) {
GParamSpecChar *cspec = G_PARAM_SPEC_CHAR (spec);
return g_param_spec_char (name, nick, blurb, cspec->minimum,
cspec->maximum, cspec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_DOUBLE) {
GParamSpecDouble *dspec = G_PARAM_SPEC_DOUBLE (spec);
return g_param_spec_double (name, nick, blurb, dspec->minimum,
dspec->maximum, dspec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_ENUM) {
return g_param_spec_enum (name, nick, blurb, spec->value_type,
G_PARAM_SPEC_ENUM (spec)->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_FLAGS) {
return g_param_spec_flags (name, nick, blurb, spec->value_type,
G_PARAM_SPEC_ENUM (spec)->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_FLOAT) {
GParamSpecFloat *fspec = G_PARAM_SPEC_FLOAT (spec);
return g_param_spec_double (name, nick, blurb, fspec->minimum,
fspec->maximum, fspec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_GTYPE) {
return g_param_spec_gtype (name, nick, blurb,
G_PARAM_SPEC_GTYPE (spec)->is_a_type, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_INT) {
GParamSpecInt *ispec = G_PARAM_SPEC_INT (spec);
return g_param_spec_int (name, nick, blurb, ispec->minimum,
ispec->maximum, ispec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_INT64) {
GParamSpecInt64 *ispec = G_PARAM_SPEC_INT64 (spec);
return g_param_spec_int64 (name, nick, blurb, ispec->minimum,
ispec->maximum, ispec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_LONG) {
GParamSpecLong *lspec = G_PARAM_SPEC_LONG (spec);
return g_param_spec_long (name, nick, blurb, lspec->minimum,
lspec->maximum, lspec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_OBJECT) {
return g_param_spec_object (name, nick, blurb, spec->value_type, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_PARAM) {
return g_param_spec_param (name, nick, blurb, spec->value_type, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_POINTER) {
return g_param_spec_pointer (name, nick, blurb, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_STRING) {
return g_param_spec_string (name, nick, blurb,
G_PARAM_SPEC_STRING (spec)->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_UCHAR) {
GParamSpecUChar *cspec = G_PARAM_SPEC_UCHAR (spec);
return g_param_spec_uchar (name, nick, blurb, cspec->minimum,
cspec->maximum, cspec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_UINT) {
GParamSpecUInt *ispec = G_PARAM_SPEC_UINT (spec);
return g_param_spec_uint (name, nick, blurb, ispec->minimum,
ispec->maximum, ispec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_UINT64) {
GParamSpecUInt64 *ispec = G_PARAM_SPEC_UINT64 (spec);
return g_param_spec_uint64 (name, nick, blurb, ispec->minimum,
ispec->maximum, ispec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_ULONG) {
GParamSpecULong *lspec = G_PARAM_SPEC_ULONG (spec);
return g_param_spec_ulong (name, nick, blurb, lspec->minimum,
lspec->maximum, lspec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_UNICHAR) {
return g_param_spec_unichar (name, nick, blurb,
G_PARAM_SPEC_UNICHAR (spec)->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_VARIANT) {
GParamSpecVariant *vspec = G_PARAM_SPEC_VARIANT (spec);
return g_param_spec_variant (name, nick, blurb, vspec->type,
vspec->default_value, flags);
}
if (G_PARAM_SPEC_TYPE (spec) == GST_TYPE_PARAM_MINI_OBJECT) {
return gst_param_spec_mini_object (name, nick, blurb, spec->value_type,
flags);
}
g_warning ("Unknown param type %ld for '%s'",
(long) G_PARAM_SPEC_TYPE (spec), name);
g_assert_not_reached ();
}
static void
gst_pulse_audio_sink_class_init (GstPulseAudioSinkClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) klass;
GstPulseSinkClass *psink_class =
GST_PULSESINK_CLASS (g_type_class_ref (GST_TYPE_PULSESINK));
GParamSpec **specs;
guint n, i, j;
gobject_class->get_property = gst_pulse_audio_sink_get_property;
gobject_class->set_property = gst_pulse_audio_sink_set_property;
gobject_class->dispose = gst_pulse_audio_sink_dispose;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_change_state);
/* Find out how many properties we already have */
specs = g_object_class_list_properties (gobject_class, &klass->n_prop_own);
g_free (specs);
/* Proxy pulsesink's properties */
specs = g_object_class_list_properties (G_OBJECT_CLASS (psink_class), &n);
for (i = 0, j = klass->n_prop_own; i < n; i++) {
if (g_object_class_find_property (gobject_class,
g_param_spec_get_name (specs[i]))) {
/* We already inherited this property from a parent, skip */
j--;
} else {
g_object_class_install_property (gobject_class, i + j + 1,
param_spec_copy (specs[i]));
}
}
klass->n_prop_total = i + j;
g_free (specs);
g_type_class_unref (psink_class);
}
static GstPad *
get_proxypad (GstPad * sinkpad)
{
GstIterator *iter = NULL;
GstPad *proxypad = NULL;
iter = gst_pad_iterate_internal_links (sinkpad);
if (iter) {
if (gst_iterator_next (iter, (gpointer) & proxypad) != GST_ITERATOR_OK)
proxypad = NULL;
gst_iterator_free (iter);
}
return proxypad;
}
static void
post_missing_element_message (GstPulseAudioSink * pbin, const gchar * name)
{
GstMessage *msg;
msg = gst_missing_element_message_new (GST_ELEMENT_CAST (pbin), name);
gst_element_post_message (GST_ELEMENT_CAST (pbin), msg);
}
static void
notify_cb (GObject * selector, GParamSpec * pspec, GstPulseAudioSink * pbin)
{
g_object_notify (G_OBJECT (pbin), g_param_spec_get_name (pspec));
}
static void
gst_pulse_audio_sink_init (GstPulseAudioSink * pbin,
GstPulseAudioSinkClass * klass)
{
GstPad *pad = NULL;
GParamSpec **specs;
GString *prop;
guint i;
pbin->lock = g_mutex_new ();
gst_segment_init (&pbin->segment, GST_FORMAT_UNDEFINED);
pbin->psink = GST_PULSESINK (gst_element_factory_make ("pulsesink",
"pulseaudiosink-sink"));
g_assert (pbin->psink != NULL);
if (!gst_bin_add (GST_BIN (pbin), GST_ELEMENT (pbin->psink))) {
GST_ERROR_OBJECT (pbin, "Failed to add pulsesink to bin");
goto error;
}
pad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
pbin->sinkpad = gst_ghost_pad_new_from_template ("sink", pad,
gst_static_pad_template_get (&sink_template));
pbin->sinkpad_old_eventfunc = GST_PAD_EVENTFUNC (pbin->sinkpad);
gst_pad_set_event_function (pbin->sinkpad,
GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_sink_event));
gst_pad_set_setcaps_function (pbin->sinkpad,
GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_sink_setcaps));
gst_pad_set_acceptcaps_function (pbin->sinkpad,
GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_sink_acceptcaps));
gst_element_add_pad (GST_ELEMENT (pbin), pbin->sinkpad);
if (!(pbin->sink_proxypad = get_proxypad (pbin->sinkpad)))
GST_ERROR_OBJECT (pbin, "Failed to get proxypad of srcpad");
else {
pbin->proxypad_old_eventfunc = GST_PAD_EVENTFUNC (pbin->sink_proxypad);
gst_pad_set_event_function (pbin->sink_proxypad,
GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_src_event));
}
/* Now proxy all the notify::* signals */
specs = g_object_class_list_properties (G_OBJECT_CLASS (klass), &i);
prop = g_string_sized_new (30);
for (i--; i >= klass->n_prop_own; i--) {
g_string_printf (prop, "notify::%s", g_param_spec_get_name (specs[i]));
g_signal_connect (pbin->psink, prop->str, G_CALLBACK (notify_cb), pbin);
}
g_string_free (prop, TRUE);
g_free (specs);
pbin->format_lost = FALSE;
out:
if (pad)
gst_object_unref (pad);
return;
error:
if (pbin->psink)
gst_object_unref (pbin->psink);
goto out;
}
static void
gst_pulse_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (object);
GstPulseAudioSinkClass *klass =
GST_PULSE_AUDIO_SINK_CLASS (G_OBJECT_GET_CLASS (object));
g_return_if_fail (prop_id <= klass->n_prop_total);
g_object_set_property (G_OBJECT (pbin->psink), g_param_spec_get_name (pspec),
value);
}
static void
gst_pulse_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (object);
GstPulseAudioSinkClass *klass =
GST_PULSE_AUDIO_SINK_CLASS (G_OBJECT_GET_CLASS (object));
g_return_if_fail (prop_id <= klass->n_prop_total);
g_object_get_property (G_OBJECT (pbin->psink), g_param_spec_get_name (pspec),
value);
}
static void
gst_pulse_audio_sink_free_dbin2 (GstPulseAudioSink * pbin)
{
g_signal_handler_disconnect (pbin->dbin2, pbin->pad_added_id);
gst_element_set_state (pbin->dbin2, GST_STATE_NULL);
gst_bin_remove (GST_BIN (pbin), pbin->dbin2);
pbin->dbin2 = NULL;
}
static void
gst_pulse_audio_sink_dispose (GObject * object)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (object);
if (pbin->lock) {
g_mutex_free (pbin->lock);
pbin->lock = NULL;
}
if (pbin->sink_proxypad) {
gst_object_unref (pbin->sink_proxypad);
pbin->sink_proxypad = NULL;
}
if (pbin->dbin2) {
g_signal_handler_disconnect (pbin->dbin2, pbin->pad_added_id);
pbin->dbin2 = NULL;
}
pbin->sinkpad = NULL;
pbin->psink = NULL;
}
static gboolean
gst_pulse_audio_sink_update_sinkpad (GstPulseAudioSink * pbin, GstPad * sinkpad)
{
gboolean ret;
ret = gst_ghost_pad_set_target (GST_GHOST_PAD_CAST (pbin->sinkpad), sinkpad);
if (!ret)
GST_WARNING_OBJECT (pbin, "Could not update ghostpad target");
return ret;
}
static void
distribute_running_time (GstElement * element, const GstSegment * segment)
{
GstEvent *event;
GstPad *pad;
pad = gst_element_get_static_pad (element, "sink");
/* FIXME: Some decoders collect newsegments and send them out at once, making
* them lose accumulator events (and thus making dbin2_event_probe() hard to
* do right if we're sending these as well. We can get away with not sending
* these at the moment, but this should be fixed! */
#if 0
if (segment->accum) {
event = gst_event_new_new_segment_full (FALSE, segment->rate,
segment->applied_rate, segment->format, 0, segment->accum, 0);
gst_pad_send_event (pad, event);
}
#endif
event = gst_event_new_new_segment_full (FALSE, segment->rate,
segment->applied_rate, segment->format,
segment->start, segment->stop, segment->time);
gst_pad_send_event (pad, event);
gst_object_unref (pad);
}
static gboolean
dbin2_event_probe (GstPad * pad, GstMiniObject * obj, gpointer data)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (data);
GstEvent *event = GST_EVENT (obj);
if (GST_EVENT_TYPE (event) == GST_EVENT_NEWSEGMENT) {
GST_DEBUG_OBJECT (pbin, "Got newsegment - dropping");
gst_pad_remove_event_probe (pad, pbin->event_probe_id);
gst_object_unref (pbin);
return FALSE;
}
return TRUE;
}
static void
pad_added_cb (GstElement * dbin2, GstPad * pad, gpointer * data)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (data);
GstPad *sinkpad = NULL;
pbin = GST_PULSE_AUDIO_SINK (data);
sinkpad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
GST_PULSE_AUDIO_SINK_LOCK (pbin);
if (gst_pad_link (pad, sinkpad) != GST_PAD_LINK_OK)
GST_ERROR_OBJECT (pbin, "Failed to link decodebin2 to pulsesink");
else
GST_DEBUG_OBJECT (pbin, "Linked new pad to pulsesink");
GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
gst_object_unref (sinkpad);
}
/* Called with pbin lock held */
static void
gst_pulse_audio_sink_add_dbin2 (GstPulseAudioSink * pbin)
{
GstPad *sinkpad = NULL;
g_assert (pbin->dbin2 == NULL);
pbin->dbin2 = gst_element_factory_make ("decodebin2", "pulseaudiosink-dbin2");
if (!pbin->dbin2) {
post_missing_element_message (pbin, "decodebin2");
GST_ELEMENT_WARNING (pbin, CORE, MISSING_PLUGIN,
(_("Missing element '%s' - check your GStreamer installation."),
"decodebin2"), ("audio playback might fail"));
goto out;
}
if (!gst_bin_add (GST_BIN (pbin), pbin->dbin2)) {
GST_ERROR_OBJECT (pbin, "Failed to add decodebin2 to bin");
goto out;
}
pbin->pad_added_id = g_signal_connect (pbin->dbin2, "pad-added",
G_CALLBACK (pad_added_cb), pbin);
if (!gst_element_sync_state_with_parent (pbin->dbin2)) {
GST_ERROR_OBJECT (pbin, "Failed to set decodebin2 to parent state");
goto out;
}
/* Trap the newsegment events that we feed the decodebin and discard them */
sinkpad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
pbin->event_probe_id = gst_pad_add_event_probe (sinkpad,
G_CALLBACK (dbin2_event_probe), gst_object_ref (pbin));
gst_object_unref (sinkpad);
sinkpad = NULL;
GST_DEBUG_OBJECT (pbin, "Distributing running time to decodebin");
distribute_running_time (pbin->dbin2, &pbin->segment);
sinkpad = gst_element_get_static_pad (pbin->dbin2, "sink");
gst_pulse_audio_sink_update_sinkpad (pbin, sinkpad);
out:
if (sinkpad)
gst_object_unref (sinkpad);
}
static void
update_eac3_alignment (GstPulseAudioSink * pbin)
{
GstCaps *caps = gst_pad_peer_get_caps_reffed (pbin->sinkpad);
GstStructure *st;
if (!caps)
return;
st = gst_caps_get_structure (caps, 0);
if (g_str_equal (gst_structure_get_name (st), "audio/x-eac3")) {
GstStructure *event_st = gst_structure_new ("ac3parse-set-alignment",
"alignment", G_TYPE_STRING, pbin->dbin2 ? "frame" : "iec61937", NULL);
if (!gst_pad_push_event (pbin->sinkpad,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, event_st)))
GST_WARNING_OBJECT (pbin->sinkpad, "Could not update alignment");
}
gst_caps_unref (caps);
}
static void
proxypad_blocked_cb (GstPad * pad, gboolean blocked, gpointer data)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (data);
GstCaps *caps;
GstPad *sinkpad = NULL;
if (!blocked) {
/* Unblocked, don't need to do anything */
GST_DEBUG_OBJECT (pbin, "unblocked");
return;
}
GST_DEBUG_OBJECT (pbin, "blocked");
GST_PULSE_AUDIO_SINK_LOCK (pbin);
if (!pbin->format_lost) {
sinkpad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
caps = gst_pad_get_caps_reffed (pad);
if (gst_pad_accept_caps (sinkpad, caps)) {
if (pbin->dbin2) {
GST_DEBUG_OBJECT (pbin, "Removing decodebin");
gst_pulse_audio_sink_free_dbin2 (pbin);
gst_pulse_audio_sink_update_sinkpad (pbin, sinkpad);
} else
GST_DEBUG_OBJECT (pbin, "Doing nothing");
gst_caps_unref (caps);
gst_object_unref (sinkpad);
goto done;
}
/* pulsesink doesn't accept the incoming caps, so add a decodebin
* (potentially after removing the existing once, since decodebin2 can't
* renegotiate). */
} else {
/* Format lost, proceed to try plugging a decodebin */
pbin->format_lost = FALSE;
}
if (pbin->dbin2 != NULL) {
/* decodebin2 doesn't support reconfiguration, so throw this one away and
* create a new one. */
gst_pulse_audio_sink_free_dbin2 (pbin);
}
GST_DEBUG_OBJECT (pbin, "Adding decodebin");
gst_pulse_audio_sink_add_dbin2 (pbin);
done:
update_eac3_alignment (pbin);
gst_pad_set_blocked_async_full (pad, FALSE, proxypad_blocked_cb,
gst_object_ref (pbin), (GDestroyNotify) gst_object_unref);
GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
}
static gboolean
gst_pulse_audio_sink_src_event (GstPad * pad, GstEvent * event)
{
GstPulseAudioSink *pbin = NULL;
GstPad *ghostpad = NULL;
gboolean ret = FALSE;
ghostpad = GST_PAD_CAST (gst_pad_get_parent (pad));
if (G_UNLIKELY (!ghostpad)) {
GST_WARNING_OBJECT (pad, "Could not get ghostpad");
goto out;
}
pbin = GST_PULSE_AUDIO_SINK (gst_pad_get_parent (ghostpad));
if (G_UNLIKELY (!pbin)) {
GST_WARNING_OBJECT (pad, "Could not get pulseaudiosink");
goto out;
}
if (G_UNLIKELY (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) &&
(gst_event_has_name (event, "pulse-format-lost") ||
gst_event_has_name (event, "pulse-sink-changed"))) {
g_return_val_if_fail (pad->mode != GST_ACTIVATE_PULL, FALSE);
GST_PULSE_AUDIO_SINK_LOCK (pbin);
if (gst_event_has_name (event, "pulse-format-lost"))
pbin->format_lost = TRUE;
if (!gst_pad_is_blocked (pad))
gst_pad_set_blocked_async_full (pad, TRUE, proxypad_blocked_cb,
gst_object_ref (pbin), (GDestroyNotify) gst_object_unref);
GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
ret = TRUE;
} else if (pbin->proxypad_old_eventfunc) {
ret = pbin->proxypad_old_eventfunc (pad, event);
event = NULL;
}
out:
if (ghostpad)
gst_object_unref (ghostpad);
if (pbin)
gst_object_unref (pbin);
if (event)
gst_event_unref (event);
return ret;
}
static gboolean
gst_pulse_audio_sink_sink_event (GstPad * pad, GstEvent * event)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (gst_pad_get_parent (pad));
gboolean ret;
ret = pbin->sinkpad_old_eventfunc (pad, gst_event_ref (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
GST_PULSE_AUDIO_SINK_LOCK (pbin);
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
GST_DEBUG_OBJECT (pbin,
"newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
GST_TIME_ARGS (time));
if (format == GST_FORMAT_TIME) {
/* Store the values for feeding to sub-elements */
gst_segment_set_newsegment_full (&pbin->segment, update,
rate, arate, format, start, stop, time);
} else {
GST_WARNING_OBJECT (pbin, "Got a non-TIME format segment");
gst_segment_init (&pbin->segment, GST_FORMAT_TIME);
}
GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
break;
}
case GST_EVENT_FLUSH_STOP:
GST_PULSE_AUDIO_SINK_LOCK (pbin);
gst_segment_init (&pbin->segment, GST_FORMAT_UNDEFINED);
GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
break;
default:
break;
}
gst_object_unref (pbin);
gst_event_unref (event);
return ret;
}
/* The bin's acceptcaps should be exactly equivalent to a pulsesink that is
* connected to a sink that supports all the formats in template caps. This
* means that upstream will have to have everything possibly upto a parser
* plugged and we plugin a decoder whenever required. */
static gboolean
gst_pulse_audio_sink_sink_acceptcaps (GstPad * pad, GstCaps * caps)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (gst_pad_get_parent (pad));
GstRingBufferSpec spec = { 0 };
const GstStructure *st;
GstCaps *pad_caps = NULL;
gboolean ret = FALSE;
pad_caps = gst_pad_get_caps_reffed (pad);
if (!pad_caps || !gst_caps_can_intersect (pad_caps, caps))
goto out;
/* If we've not got fixed caps, creating a stream might fail, so let's just
* return from here with default acceptcaps behaviour */
if (!gst_caps_is_fixed (caps))
goto out;
spec.latency_time = GST_BASE_AUDIO_SINK (pbin->psink)->latency_time;
if (!gst_ring_buffer_parse_caps (&spec, caps))
goto out;
/* Make sure non-raw input is framed (one frame per buffer) and can be
* payloaded */
st = gst_caps_get_structure (caps, 0);
if (!g_str_has_prefix (gst_structure_get_name (st), "audio/x-raw")) {
gboolean framed = FALSE, parsed = FALSE;
gst_structure_get_boolean (st, "framed", &framed);
gst_structure_get_boolean (st, "parsed", &parsed);
if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
goto out;
}
ret = TRUE;
out:
if (pad_caps)
gst_caps_unref (pad_caps);
gst_object_unref (pbin);
return ret;
}
static gboolean
gst_pulse_audio_sink_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (gst_pad_get_parent (pad));
gboolean ret = TRUE;
GST_PULSE_AUDIO_SINK_LOCK (pbin);
if (!gst_pad_is_blocked (pbin->sinkpad))
gst_pad_set_blocked_async_full (pbin->sink_proxypad, TRUE,
proxypad_blocked_cb, gst_object_ref (pbin),
(GDestroyNotify) gst_object_unref);
GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
gst_object_unref (pbin);
return ret;
}
static GstStateChangeReturn
gst_pulse_audio_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
/* Nothing to do for upward transitions */
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_PULSE_AUDIO_SINK_LOCK (pbin);
if (gst_pad_is_blocked (pbin->sinkpad)) {
gst_pad_set_blocked_async_full (pbin->sink_proxypad, FALSE,
proxypad_blocked_cb, gst_object_ref (pbin),
(GDestroyNotify) gst_object_unref);
}
GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret != GST_STATE_CHANGE_SUCCESS) {
GST_DEBUG_OBJECT (pbin, "Base class returned %d on state change", ret);
goto out;
}
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_PULSE_AUDIO_SINK_LOCK (pbin);
gst_segment_init (&pbin->segment, GST_FORMAT_UNDEFINED);
if (pbin->dbin2) {
GstPad *pad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink),
"sink");
gst_pulse_audio_sink_free_dbin2 (pbin);
gst_pulse_audio_sink_update_sinkpad (pbin, pad);
gst_object_unref (pad);
}
GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
break;
default:
break;
}
out:
return ret;
}
#endif /* HAVE_PULSE_1_0 */

81
ext/pulse/pulsesink.c
View file

@ -420,6 +420,27 @@ gst_pulsering_context_subscribe_cb (pa_context * c,
if (idx != pa_stream_get_index (pbuf->stream))
continue;
#ifdef HAVE_PULSE_1_0
if (psink->device && pa_format_info_is_pcm (pbuf->format) &&
!g_str_equal (psink->device,
pa_stream_get_device_name (pbuf->stream))) {
/* Underlying sink changed. And this is not a passthrough stream. Let's
* see if someone upstream wants to try to renegotiate. */
GstEvent *renego;
g_free (psink->device);
psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
GST_INFO_OBJECT (psink, "emitting sink-changed");
renego = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_new ("pulse-sink-changed", NULL));
if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego))
GST_DEBUG_OBJECT (psink, "Emitted sink-changed - nobody was listening");
}
#endif
/* 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
@ -1719,12 +1740,6 @@ static GstStateChangeReturn gst_pulsesink_change_state (GstElement * element,
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);
#define _do_init(type) \
@ -1784,57 +1799,7 @@ 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, "
"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 ];"
"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 ];"
#ifdef HAVE_PULSE_1_0
"audio/x-ac3, framed = (boolean) true;"
"audio/x-eac3, framed = (boolean) true; "
"audio/x-dts, framed = (boolean) true, "
" block_size = (int) { 512, 1024, 2048 }; "
"audio/mpeg, mpegversion = (int)1, "
" mpegaudioversion = (int) [ 1, 2 ], parsed = (boolean) true; "
#endif
));
GST_STATIC_CAPS (PULSE_SINK_TEMPLATE_CAPS));
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
@ -2075,6 +2040,8 @@ done:
}
#ifdef HAVE_PULSE_1_0
/* NOTE: If you're making changes here, see if pulseaudiosink acceptcaps also
* needs to be changed accordingly. */
static gboolean
gst_pulsesink_pad_acceptcaps (GstPad * pad, GstCaps * caps)
{

89
ext/pulse/pulsesink.h
View file

@ -24,6 +24,10 @@
#ifndef __GST_PULSESINK_H__
#define __GST_PULSESINK_H__
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/gstaudiosink.h>
@ -88,6 +92,91 @@ struct _GstPulseSinkClass
GType gst_pulsesink_get_type (void);
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
#endif
#define _PULSE_SINK_CAPS_COMMON \
"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 ];" \
"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 ];"
#ifdef HAVE_PULSE_1_0
#define _PULSE_SINK_CAPS_1_0 \
"audio/x-ac3, framed = (boolean) true;" \
"audio/x-eac3, framed = (boolean) true; " \
"audio/x-dts, framed = (boolean) true, " \
"block-size = (int) { 512, 1024, 2048 }; " \
"audio/mpeg, mpegversion = (int) 1, " \
"mpegaudioversion = (int) [ 1, 2 ], parsed = (boolean) true;"
#else
#define _PULSE_SINK_CAPS_1_0 ""
#endif
#define PULSE_SINK_TEMPLATE_CAPS \
_PULSE_SINK_CAPS_COMMON \
_PULSE_SINK_CAPS_1_0
#ifdef HAVE_PULSE_1_0
#define GST_TYPE_PULSE_AUDIO_SINK \
(gst_pulse_audio_sink_get_type())
#define GST_PULSE_AUDIO_SINK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PULSE_AUDIO_SINK,GstPulseAudioSink))
#define GST_PULSE_AUDIO_SINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PULSE_AUDIO_SINK,GstPulseAudioSinkClass))
#define GST_IS_PULSE_AUDIO_SINK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PULSE_AUDIO_SINK))
#define GST_IS_PULSE_AUDIO_SINK_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PULSE_AUDIO_SINK))
#define GST_PULSE_AUDIO_SINK_CAST(obj) \
((GstPulseAudioSink *)(obj))
GType gst_pulse_audio_sink_get_type (void);
#endif /* HAVE_PULSE_1_0 */
G_END_DECLS
#endif /* __GST_PULSESINK_H__ */

4
ext/pulse/pulseutil.h
View file

@ -22,6 +22,10 @@
#ifndef __GST_PULSEUTIL_H__
#define __GST_PULSEUTIL_H__
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <pulse/pulseaudio.h>
#include <gst/audio/gstaudiosink.h>