/*-*- 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., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301
* USA.
*/
/**
* SECTION:element-pulsesink
* @title: pulsesink
* @see_also: pulsesrc
*
* This element outputs audio to a
* PulseAudio sound server.
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink
* ]| Play an Ogg/Vorbis file.
* |[
* gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink
* ]| Play a 440Hz sine wave.
* |[
* gst-launch-1.0 -v audiotestsrc ! pulsesink stream-properties="props,media.title=test"
* ]| Play a sine wave and set a stream property. The property can be checked
* with "pactl list".
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include
#include
#include
#include
#include
#include
#include /* only used for GST_PLUGINS_BASE_VERSION_* */
#include
#include "pulsesink.h"
#include "pulseutil.h"
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
#define DEFAULT_SERVER NULL
#define DEFAULT_DEVICE NULL
#define DEFAULT_CURRENT_DEVICE NULL
#define DEFAULT_DEVICE_NAME NULL
#define DEFAULT_VOLUME 1.0
#define DEFAULT_MUTE FALSE
#define MAX_VOLUME 10.0
enum
{
PROP_0,
PROP_SERVER,
PROP_DEVICE,
PROP_CURRENT_DEVICE,
PROP_DEVICE_NAME,
PROP_VOLUME,
PROP_MUTE,
PROP_CLIENT_NAME,
PROP_STREAM_PROPERTIES,
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;
typedef struct _GstPulseContext GstPulseContext;
/* A note on threading.
*
* We use a pa_threaded_mainloop to interact with the PulseAudio server. This
* starts up a separate thread that runs a mainloop to carry back events,
* messages and timing updates from the PulseAudio server.
*
* In most cases, the PulseAudio API we use communicates with the server and
* processes replies asynchronously. Operations on PA objects that result in
* such communication are protected with a pa_threaded_mainloop_lock() and
* pa_threaded_mainloop_unlock(). These guarantee mutual exclusion with the
* mainloop thread -- when an iteration of the mainloop thread begins, it first
* tries to acquire this lock, and cannot do so if our code also holds that
* lock.
*
* When we need to complete an operation synchronously, we use
* pa_threaded_mainloop_wait() and pa_threaded_mainloop_signal(). These work
* much as pthread conditionals do. pa_threaded_mainloop_wait() is called with
* the mainloop lock held. It releases the lock (thereby allowing the mainloop
* to execute), and waits till one of our callbacks to be executed by the
* mainloop thread calls pa_threaded_mainloop_signal(). At the end of the
* mainloop iteration, the pa_threaded_mainloop_wait() will reacquire the
* mainloop lock and return control to the caller.
*/
/* Store the PA contexts in a hash table to allow easy sharing among
* multiple instances of the sink. Keys are $context_name@$server_name
* (strings) and values should be GstPulseContext pointers.
*/
struct _GstPulseContext
{
pa_context *context;
GSList *ring_buffers;
};
static GHashTable *gst_pulse_shared_contexts = NULL;
/* use one static main-loop for all instances
* this is needed to make the context sharing work as the contexts are
* released when releasing their parent main-loop
*/
static pa_threaded_mainloop *mainloop = NULL;
static guint mainloop_ref_ct = 0;
/* lock for access to shared resources */
static GMutex pa_shared_resource_mutex;
/* 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
{
GstAudioRingBuffer object;
gchar *context_name;
gchar *stream_name;
pa_context *context;
pa_stream *stream;
pa_stream *probe_stream;
pa_format_info *format;
guint channels;
gboolean is_pcm;
void *m_data;
size_t m_towrite;
size_t m_writable;
gint64 m_offset;
gint64 m_lastoffset;
gboolean corked:1;
gboolean in_commit:1;
gboolean paused:1;
};
struct _GstPulseRingBufferClass
{
GstAudioRingBufferClass parent_class;
};
static GType gst_pulseringbuffer_get_type (void);
static void gst_pulseringbuffer_finalize (GObject * object);
static GstAudioRingBufferClass *ring_parent_class = NULL;
static gboolean gst_pulseringbuffer_open_device (GstAudioRingBuffer * buf);
static gboolean gst_pulseringbuffer_close_device (GstAudioRingBuffer * buf);
static gboolean gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec);
static gboolean gst_pulseringbuffer_release (GstAudioRingBuffer * buf);
static gboolean gst_pulseringbuffer_start (GstAudioRingBuffer * buf);
static gboolean gst_pulseringbuffer_pause (GstAudioRingBuffer * buf);
static gboolean gst_pulseringbuffer_stop (GstAudioRingBuffer * buf);
static void gst_pulseringbuffer_clear (GstAudioRingBuffer * buf);
static guint gst_pulseringbuffer_commit (GstAudioRingBuffer * buf,
guint64 * sample, guchar * data, gint in_samples, gint out_samples,
gint * accum);
G_DEFINE_TYPE (GstPulseRingBuffer, gst_pulseringbuffer,
GST_TYPE_AUDIO_RING_BUFFER);
static void
gst_pulsesink_init_contexts (void)
{
g_mutex_init (&pa_shared_resource_mutex);
gst_pulse_shared_contexts = g_hash_table_new_full (g_str_hash, g_str_equal,
g_free, NULL);
}
static void
gst_pulseringbuffer_class_init (GstPulseRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstAudioRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstringbuffer_class = (GstAudioRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = 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);
}
static void
gst_pulseringbuffer_init (GstPulseRingBuffer * pbuf)
{
pbuf->stream_name = NULL;
pbuf->context = NULL;
pbuf->stream = NULL;
pbuf->probe_stream = NULL;
pbuf->format = NULL;
pbuf->channels = 0;
pbuf->is_pcm = FALSE;
pbuf->m_data = NULL;
pbuf->m_towrite = 0;
pbuf->m_writable = 0;
pbuf->m_offset = 0;
pbuf->m_lastoffset = 0;
pbuf->corked = TRUE;
pbuf->in_commit = FALSE;
pbuf->paused = FALSE;
}
/* Call with mainloop lock held if wait == TRUE) */
static void
gst_pulse_destroy_stream (pa_stream * stream, gboolean wait)
{
/* Make sure we don't get any further callbacks */
pa_stream_set_write_callback (stream, NULL, NULL);
pa_stream_set_underflow_callback (stream, NULL, NULL);
pa_stream_set_overflow_callback (stream, NULL, NULL);
pa_stream_disconnect (stream);
if (wait)
pa_threaded_mainloop_wait (mainloop);
pa_stream_set_state_callback (stream, NULL, NULL);
pa_stream_unref (stream);
}
static void
gst_pulsering_destroy_stream (GstPulseRingBuffer * pbuf)
{
if (pbuf->probe_stream) {
gst_pulse_destroy_stream (pbuf->probe_stream, FALSE);
pbuf->probe_stream = NULL;
}
if (pbuf->stream) {
if (pbuf->m_data) {
/* drop shm memory buffer */
pa_stream_cancel_write (pbuf->stream);
/* reset internal variables */
pbuf->m_data = NULL;
pbuf->m_towrite = 0;
pbuf->m_writable = 0;
pbuf->m_offset = 0;
pbuf->m_lastoffset = 0;
}
if (pbuf->format) {
pa_format_info_free (pbuf->format);
pbuf->format = NULL;
pbuf->channels = 0;
pbuf->is_pcm = FALSE;
}
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)
{
g_mutex_lock (&pa_shared_resource_mutex);
GST_DEBUG_OBJECT (pbuf, "destroying ringbuffer %p", pbuf);
gst_pulsering_destroy_stream (pbuf);
if (pbuf->context) {
pa_context_unref (pbuf->context);
pbuf->context = NULL;
}
if (pbuf->context_name) {
GstPulseContext *pctx;
pctx = g_hash_table_lookup (gst_pulse_shared_contexts, pbuf->context_name);
GST_DEBUG_OBJECT (pbuf, "releasing context with name %s, pbuf=%p, pctx=%p",
pbuf->context_name, pbuf, pctx);
if (pctx) {
pctx->ring_buffers = g_slist_remove (pctx->ring_buffers, pbuf);
if (pctx->ring_buffers == NULL) {
GST_DEBUG_OBJECT (pbuf,
"destroying final context with name %s, pbuf=%p, pctx=%p",
pbuf->context_name, pbuf, pctx);
pa_context_disconnect (pctx->context);
/* Make sure we don't get any further callbacks */
pa_context_set_state_callback (pctx->context, NULL, NULL);
pa_context_set_subscribe_callback (pctx->context, NULL, NULL);
g_hash_table_remove (gst_pulse_shared_contexts, pbuf->context_name);
pa_context_unref (pctx->context);
g_slice_free (GstPulseContext, pctx);
}
}
g_free (pbuf->context_name);
pbuf->context_name = NULL;
}
g_mutex_unlock (&pa_shared_resource_mutex);
}
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);
}
#define CONTEXT_OK(c) ((c) && PA_CONTEXT_IS_GOOD (pa_context_get_state ((c))))
#define STREAM_OK(s) ((s) && PA_STREAM_IS_GOOD (pa_stream_get_state ((s))))
static gboolean
gst_pulsering_is_dead (GstPulseSink * psink, GstPulseRingBuffer * pbuf,
gboolean check_stream)
{
if (!CONTEXT_OK (pbuf->context))
goto error;
if (check_stream && !STREAM_OK (pbuf->stream))
goto error;
return FALSE;
error:
{
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;
}
}
static void
gst_pulsering_context_state_cb (pa_context * c, void *userdata)
{
pa_context_state_t state;
pa_threaded_mainloop *mainloop = (pa_threaded_mainloop *) userdata;
state = pa_context_get_state (c);
GST_LOG ("got new context state %d", state);
switch (state) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
GST_LOG ("signaling");
pa_threaded_mainloop_signal (mainloop, 0);
break;
case PA_CONTEXT_UNCONNECTED:
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
}
}
static void
gst_pulsering_context_subscribe_cb (pa_context * c,
pa_subscription_event_type_t t, uint32_t idx, void *userdata)
{
GstPulseSink *psink;
GstPulseContext *pctx = (GstPulseContext *) userdata;
GSList *walk;
if (t != (PA_SUBSCRIPTION_EVENT_SINK_INPUT | PA_SUBSCRIPTION_EVENT_CHANGE) &&
t != (PA_SUBSCRIPTION_EVENT_SINK_INPUT | PA_SUBSCRIPTION_EVENT_NEW))
return;
for (walk = pctx->ring_buffers; walk; walk = g_slist_next (walk)) {
GstPulseRingBuffer *pbuf = (GstPulseRingBuffer *) walk->data;
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_LOG_OBJECT (psink, "type %04x, idx %u", t, idx);
if (!pbuf->stream)
continue;
if (idx != pa_stream_get_index (pbuf->stream))
continue;
if (psink->device && pbuf->is_pcm &&
!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");
/* FIXME: send reconfigure event instead and let decodebin/playbin
* handle that. Also take care of ac3 alignment. See "pulse-format-lost" */
renego = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_new_empty ("pulse-sink-changed"));
if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego))
GST_DEBUG_OBJECT (psink, "Emitted sink-changed - nobody was listening");
}
/* 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);
}
}
/* 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 (GstAudioRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
GstPulseContext *pctx;
pa_mainloop_api *api;
gboolean need_unlock_shared;
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
pbuf = GST_PULSERING_BUFFER_CAST (buf);
g_assert (!pbuf->stream);
g_assert (psink->client_name);
if (psink->server)
pbuf->context_name = g_strdup_printf ("%s@%s", psink->client_name,
psink->server);
else
pbuf->context_name = g_strdup (psink->client_name);
pa_threaded_mainloop_lock (mainloop);
g_mutex_lock (&pa_shared_resource_mutex);
need_unlock_shared = TRUE;
pctx = g_hash_table_lookup (gst_pulse_shared_contexts, pbuf->context_name);
if (pctx == NULL) {
pctx = g_slice_new0 (GstPulseContext);
/* get the mainloop api and create a context */
GST_INFO_OBJECT (psink, "new context with name %s, pbuf=%p, pctx=%p",
pbuf->context_name, pbuf, pctx);
api = pa_threaded_mainloop_get_api (mainloop);
if (!(pctx->context = pa_context_new (api, pbuf->context_name)))
goto create_failed;
pctx->ring_buffers = g_slist_prepend (pctx->ring_buffers, pbuf);
g_hash_table_insert (gst_pulse_shared_contexts,
g_strdup (pbuf->context_name), (gpointer) pctx);
/* register some essential callbacks */
pa_context_set_state_callback (pctx->context,
gst_pulsering_context_state_cb, mainloop);
pa_context_set_subscribe_callback (pctx->context,
gst_pulsering_context_subscribe_cb, pctx);
/* try to connect to the server and wait for completion, 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 (pctx->context, psink->server,
PA_CONTEXT_NOAUTOSPAWN, NULL) < 0)
goto connect_failed;
} else {
GST_INFO_OBJECT (psink,
"reusing shared context with name %s, pbuf=%p, pctx=%p",
pbuf->context_name, pbuf, pctx);
pctx->ring_buffers = g_slist_prepend (pctx->ring_buffers, pbuf);
}
g_mutex_unlock (&pa_shared_resource_mutex);
need_unlock_shared = FALSE;
/* context created or shared okay */
pbuf->context = pa_context_ref (pctx->context);
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 (mainloop);
}
if (pa_context_get_server_protocol_version (pbuf->context) < 22) {
/* We need PulseAudio >= 1.0 on the server side for the extended API */
goto bad_server_version;
}
GST_LOG_OBJECT (psink, "opened the device");
pa_threaded_mainloop_unlock (mainloop);
return TRUE;
/* ERRORS */
unlock_and_fail:
{
if (need_unlock_shared)
g_mutex_unlock (&pa_shared_resource_mutex);
gst_pulsering_destroy_context (pbuf);
pa_threaded_mainloop_unlock (mainloop);
return FALSE;
}
create_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("Failed to create context"), (NULL));
g_slice_free (GstPulseContext, pctx);
goto unlock_and_fail;
}
connect_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pctx->context))), (NULL));
goto unlock_and_fail;
}
bad_server_version:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("PulseAudio server version "
"is too old."), (NULL));
goto unlock_and_fail;
}
}
/* close the device */
static gboolean
gst_pulseringbuffer_close_device (GstAudioRingBuffer * 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 (mainloop);
gst_pulsering_destroy_context (pbuf);
pa_threaded_mainloop_unlock (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 (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;
GstAudioRingBuffer *rbuf;
GstPulseRingBuffer *pbuf;
rbuf = GST_AUDIO_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 (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;
GstAudioRingBuffer *ringbuf;
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));
ringbuf = GST_AUDIO_RING_BUFFER (pbuf);
if (!info) {
GST_LOG_OBJECT (psink, "latency update (information unknown)");
return;
}
if (!info->read_index_corrupt) {
/* Update segdone based on the read index. segdone is of segment
* granularity, while the read index is at byte granularity. We take the
* ceiling while converting the latter to the former since it is more
* conservative to report that we've read more than we have than to report
* less. One concern here is that latency updates happen every 100ms, which
* means segdone is not updated very often, but increasing the update
* frequency would mean more communication overhead. */
g_atomic_int_set (&ringbuf->segdone,
(int) gst_util_uint64_scale_ceil (info->read_index, 1,
ringbuf->spec.segsize));
}
sink_usec = info->configured_sink_usec;
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);
}
static void
gst_pulsering_stream_suspended_cb (pa_stream * p, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
if (pa_stream_is_suspended (p))
GST_DEBUG_OBJECT (psink, "stream suspended");
else
GST_DEBUG_OBJECT (psink, "stream resumed");
}
static void
gst_pulsering_stream_started_cb (pa_stream * p, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_DEBUG_OBJECT (psink, "stream started");
}
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 if (!strcmp (name, PA_STREAM_EVENT_FORMAT_LOST)) {
GstEvent *renego;
if (g_atomic_int_get (&psink->format_lost)) {
/* Duplicate event before we're done reconfiguring, discard */
return;
}
GST_DEBUG_OBJECT (psink, "got FORMAT LOST");
g_atomic_int_set (&psink->format_lost, 1);
psink->format_lost_time = g_ascii_strtoull (pa_proplist_gets (pl,
"stream-time"), NULL, 0) * 1000;
g_free (psink->device);
psink->device = g_strdup (pa_proplist_gets (pl, "device"));
/* FIXME: send reconfigure event instead and let decodebin/playbin
* handle that. Also take care of ac3 alignment */
renego = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_new_empty ("pulse-format-lost"));
#if 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->dbin ? "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");
}
#endif
if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego)) {
/* Nobody handled the format change - emit an error */
GST_ELEMENT_ERROR (psink, STREAM, FORMAT, ("Sink format changed"),
("Sink format changed"));
}
} else {
GST_DEBUG_OBJECT (psink, "got unknown event %s", name);
}
}
/* Called with the mainloop locked */
static gboolean
gst_pulsering_wait_for_stream_ready (GstPulseSink * psink, pa_stream * stream)
{
pa_stream_state_t state;
for (;;) {
state = pa_stream_get_state (stream);
GST_LOG_OBJECT (psink, "stream state is now %d", state);
if (!PA_STREAM_IS_GOOD (state))
return FALSE;
if (state == PA_STREAM_READY)
return TRUE;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (mainloop);
}
}
/* 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 (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
pa_channel_map channel_map;
pa_operation *o = NULL;
pa_cvolume v;
pa_cvolume *pv = NULL;
pa_stream_flags_t flags;
const gchar *name;
GstAudioClock *clock;
pa_format_info *formats[1];
#ifndef GST_DISABLE_GST_DEBUG
gchar print_buf[PA_FORMAT_INFO_SNPRINT_MAX];
#endif
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_format_info (spec, &pbuf->format, &pbuf->channels))
goto invalid_spec;
pbuf->is_pcm = pa_format_info_is_pcm (pbuf->format);
pa_threaded_mainloop_lock (mainloop);
/* we need a context and a no stream */
g_assert (pbuf->context);
g_assert (!pbuf->stream);
/* if we have a probe, disconnect it first so that if we're creating a
* compressed stream, it doesn't get blocked by a PCM stream */
if (pbuf->probe_stream) {
gst_pulse_destroy_stream (pbuf->probe_stream, TRUE);
pbuf->probe_stream = NULL;
}
/* 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 */
if (pbuf->is_pcm && gst_pulse_gst_to_channel_map (&channel_map, spec))
pa_format_info_set_channel_map (pbuf->format, &channel_map);
/* find a good name for the stream */
if (psink->stream_name)
name = psink->stream_name;
else
name = "Playback Stream";
/* create a stream */
formats[0] = pbuf->format;
if (!(pbuf->stream = pa_stream_new_extended (pbuf->context, name, formats, 1,
psink->proplist)))
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);
pa_stream_set_suspended_callback (pbuf->stream,
gst_pulsering_stream_suspended_cb, pbuf);
pa_stream_set_started_callback (pbuf->stream,
gst_pulsering_stream_started_cb, pbuf);
pa_stream_set_event_callback (pbuf->stream,
gst_pulsering_stream_event_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 (&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);
/* 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;
if (pbuf->is_pcm)
gst_pulse_cvolume_from_linear (pv, pbuf->channels, psink->volume);
else {
GST_DEBUG_OBJECT (psink, "passthrough stream, not setting volume");
pv = NULL;
}
} else {
pv = NULL;
}
/* construct the flags */
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_ADJUST_LATENCY | PA_STREAM_START_CORKED;
if (psink->mute_set) {
if (psink->mute)
flags |= PA_STREAM_START_MUTED;
else
flags |= PA_STREAM_START_UNMUTED;
}
/* 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_AUDIO_BASE_SINK (psink)->provided_clock);
gst_audio_clock_reset (clock, 0);
if (!gst_pulsering_wait_for_stream_ready (psink, pbuf->stream))
goto connect_failed;
g_free (psink->device);
psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
#ifndef GST_DISABLE_GST_DEBUG
pa_format_info_snprint (print_buf, sizeof (print_buf),
pa_stream_get_format_info (pbuf->stream));
GST_INFO_OBJECT (psink, "negotiated to: %s", print_buf);
#endif
/* 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 (mainloop);
return TRUE;
/* ERRORS */
unlock_and_fail:
{
gst_pulsering_destroy_stream (pbuf);
pa_threaded_mainloop_unlock (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 (GstAudioRingBuffer * buf)
{
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
pa_threaded_mainloop_lock (mainloop);
gst_pulsering_destroy_stream (pbuf);
pa_threaded_mainloop_unlock (mainloop);
{
GstPulseSink *psink;
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
g_atomic_int_set (&psink->format_lost, FALSE);
psink->format_lost_time = GST_CLOCK_TIME_NONE;
}
return TRUE;
}
static void
gst_pulsering_success_cb (pa_stream * s, int success, void *userdata)
{
pa_threaded_mainloop_signal (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));
if (g_atomic_int_get (&psink->format_lost)) {
/* Sink format changed, stream's gone so fake being paused */
return TRUE;
}
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 (mainloop);
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
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 (GstAudioRingBuffer * 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 (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 (mainloop);
}
#if 0
/* called from pulse thread with the mainloop lock */
static void
mainloop_enter_defer_cb (pa_mainloop_api * api, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
GstMessage *message;
GValue val = { 0 };
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));
g_value_init (&val, GST_TYPE_G_THREAD);
g_value_set_boxed (&val, g_thread_self ());
gst_message_set_stream_status_object (message, &val);
g_value_unset (&val);
gst_element_post_message (GST_ELEMENT (pulsesink), message);
g_return_if_fail (pulsesink->defer_pending);
pulsesink->defer_pending--;
pa_threaded_mainloop_signal (mainloop, 0);
}
#endif
/* start/resume playback ASAP, we don't uncork here but in the commit method */
static gboolean
gst_pulseringbuffer_start (GstAudioRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (mainloop);
GST_DEBUG_OBJECT (psink, "starting");
pbuf->paused = FALSE;
/* EOS needs running clock */
if (GST_BASE_SINK_CAST (psink)->eos ||
g_atomic_int_get (&GST_AUDIO_BASE_SINK (psink)->eos_rendering))
gst_pulsering_set_corked (pbuf, FALSE, FALSE);
#if 0
GST_DEBUG_OBJECT (psink, "scheduling stream status");
psink->defer_pending++;
pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
mainloop_enter_defer_cb, psink);
/* Wait for the stream status message to be posted. This needs to be done
* synchronously because the callback will take the mainloop lock
* (implicitly) and then take the GST_OBJECT_LOCK. Everywhere else, we take
* the locks in the reverse order, so not doing this synchronously could
* cause a deadlock. */
GST_DEBUG_OBJECT (psink, "waiting for stream status (ENTER) to be posted");
pa_threaded_mainloop_wait (mainloop);
#endif
pa_threaded_mainloop_unlock (mainloop);
return TRUE;
}
/* pause/stop playback ASAP */
static gboolean
gst_pulseringbuffer_pause (GstAudioRingBuffer * 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 (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, TRUE);
if (pbuf->in_commit) {
/* we are waiting in a commit, signal */
GST_DEBUG_OBJECT (psink, "signal commit");
pa_threaded_mainloop_signal (mainloop, 0);
}
pa_threaded_mainloop_unlock (mainloop);
return res;
}
#if 0
/* called from pulse thread with the mainloop lock */
static void
mainloop_leave_defer_cb (pa_mainloop_api * api, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
GstMessage *message;
GValue val = { 0 };
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));
g_value_init (&val, GST_TYPE_G_THREAD);
g_value_set_boxed (&val, g_thread_self ());
gst_message_set_stream_status_object (message, &val);
g_value_unset (&val);
gst_element_post_message (GST_ELEMENT (pulsesink), message);
g_return_if_fail (pulsesink->defer_pending);
pulsesink->defer_pending--;
pa_threaded_mainloop_signal (mainloop, 0);
}
#endif
/* stop playback, we flush everything. */
static gboolean
gst_pulseringbuffer_stop (GstAudioRingBuffer * 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 (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 (mainloop, 0);
}
if (g_atomic_int_get (&psink->format_lost)) {
/* Don't try to flush, the stream's probably gone by now */
res = TRUE;
goto cleanup;
}
/* 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 (mainloop);
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
goto server_dead;
}
GST_DEBUG_OBJECT (psink, "flush completed");
}
res = TRUE;
cleanup:
if (o) {
pa_operation_cancel (o);
pa_operation_unref (o);
}
#if 0
GST_DEBUG_OBJECT (psink, "scheduling stream status");
psink->defer_pending++;
pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
mainloop_leave_defer_cb, psink);
/* Wait for the stream status message to be posted. This needs to be done
* synchronously because the callback will take the mainloop lock
* (implicitly) and then take the GST_OBJECT_LOCK. Everywhere else, we take
* the locks in the reverse order, so not doing this synchronously could
* cause a deadlock. */
GST_DEBUG_OBJECT (psink, "waiting for stream status (LEAVE) to be posted");
pa_threaded_mainloop_wait (mainloop);
#endif
pa_threaded_mainloop_unlock (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, bpf); \
s += bpf; \
*accum += outr; \
if ((*accum << 1) >= inr) { \
*accum -= inr; \
d += bpf; \
} \
} \
in_samples -= (s - sb)/bpf; \
out_samples -= (d - db)/bpf; \
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, bpf); \
d += bpf; \
*accum += inr; \
if ((*accum << 1) >= outr) { \
*accum -= outr; \
s += bpf; \
} \
} \
in_samples -= (s - sb)/bpf; \
out_samples -= (d - db)/bpf; \
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, bpf); \
se -= bpf; \
*accum += outr; \
while (d < de && (*accum << 1) >= inr) { \
*accum -= inr; \
d += bpf; \
} \
} \
in_samples -= (sb - se)/bpf; \
out_samples -= (d - db)/bpf; \
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, bpf); \
d += bpf; \
*accum += inr; \
while (s <= se && (*accum << 1) >= outr) { \
*accum -= outr; \
se -= bpf; \
} \
} \
in_samples -= (sb - se)/bpf; \
out_samples -= (d - db)/bpf; \
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 (GstAudioRingBuffer * 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, bpf;
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");
g_object_notify (G_OBJECT (psink), "current-device");
}
/* make sure the ringbuffer is started */
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
/* see if we are allowed to start it */
if (G_UNLIKELY (g_atomic_int_get (&buf->may_start) == FALSE))
goto no_start;
GST_DEBUG_OBJECT (buf, "start!");
if (!gst_audio_ring_buffer_start (buf))
goto start_failed;
}
pa_threaded_mainloop_lock (mainloop);
GST_DEBUG_OBJECT (psink, "entering commit");
pbuf->in_commit = TRUE;
bpf = GST_AUDIO_INFO_BPF (&buf->spec.info);
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 + (bpf * inr);
if (g_atomic_int_get (&psink->format_lost)) {
/* Sink format changed, drop the data and hope upstream renegotiates */
goto fake_done;
}
if (pbuf->paused)
goto was_paused;
/* offset is in bytes */
offset = *sample * bpf;
while (*toprocess > 0) {
size_t avail;
guint towrite;
GST_LOG_OBJECT (psink,
"need to write %d samples at offset %" G_GINT64_FORMAT, *toprocess,
offset);
if (offset != pbuf->m_lastoffset)
GST_LOG_OBJECT (psink, "discontinuity, offset is %" G_GINT64_FORMAT ", "
"last offset was %" G_GINT64_FORMAT, offset, pbuf->m_lastoffset);
towrite = out_samples * bpf;
/* Wait for at least segsize bytes to become available */
if (towrite > buf->spec.segsize)
towrite = buf->spec.segsize;
if ((pbuf->m_writable < towrite) || (offset != pbuf->m_lastoffset)) {
/* if no room left or discontinuity in offset,
we need to flush data and get a new buffer */
/* flush the buffer if possible */
if ((pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) {
GST_LOG_OBJECT (psink,
"flushing %u samples at offset %" G_GINT64_FORMAT,
(guint) pbuf->m_towrite / bpf, pbuf->m_offset);
if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
goto write_failed;
}
}
pbuf->m_towrite = 0;
pbuf->m_offset = offset; /* keep track of current offset */
/* get a buffer to write in for now on */
for (;;) {
pbuf->m_writable = pa_stream_writable_size (pbuf->stream);
if (g_atomic_int_get (&psink->format_lost)) {
/* Sink format changed, give up and hope upstream renegotiates */
goto fake_done;
}
if (pbuf->m_writable == (size_t) - 1)
goto writable_size_failed;
pbuf->m_writable /= bpf;
pbuf->m_writable *= bpf; /* handle only complete samples */
if (pbuf->m_writable >= towrite)
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 segsize bytes, wait a bit */
GST_LOG_OBJECT (psink, "waiting for free space");
pa_threaded_mainloop_wait (mainloop);
if (pbuf->paused)
goto was_paused;
}
/* Recalculate what we can write in the next chunk */
towrite = out_samples * bpf;
if (pbuf->m_writable > towrite)
pbuf->m_writable = towrite;
GST_LOG_OBJECT (psink, "requesting %" G_GSIZE_FORMAT " bytes of "
"shared memory", pbuf->m_writable);
if (pa_stream_begin_write (pbuf->stream, &pbuf->m_data,
&pbuf->m_writable) < 0) {
GST_LOG_OBJECT (psink, "pa_stream_begin_write() failed");
goto writable_size_failed;
}
GST_LOG_OBJECT (psink, "got %" G_GSIZE_FORMAT " bytes of shared memory",
pbuf->m_writable);
}
if (towrite > pbuf->m_writable)
towrite = pbuf->m_writable;
avail = towrite / bpf;
GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
(guint) avail, offset);
/* No trick modes for passthrough streams */
if (G_UNLIKELY (!pbuf->is_pcm && (inr != outr || reverse))) {
GST_WARNING_OBJECT (psink, "Passthrough stream can't run in trick mode");
goto unlock_and_fail;
}
if (G_LIKELY (inr == outr && !reverse)) {
/* no rate conversion, simply write out the samples */
/* copy the data into internal buffer */
memcpy ((guint8 *) pbuf->m_data + pbuf->m_towrite, data, towrite);
pbuf->m_towrite += towrite;
pbuf->m_writable -= towrite;
data += towrite;
in_samples -= avail;
out_samples -= avail;
} else {
guint8 *dest, *d, *d_end;
/* write into the PulseAudio shm buffer */
dest = d = (guint8 *) pbuf->m_data + pbuf->m_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);
pbuf->m_towrite += towrite;
pbuf->m_writable -= towrite;
avail = towrite / bpf;
}
/* flush the buffer if it's full */
if ((pbuf->m_data != NULL) && (pbuf->m_towrite > 0)
&& (pbuf->m_writable == 0)) {
GST_LOG_OBJECT (psink, "flushing %u samples at offset %" G_GINT64_FORMAT,
(guint) pbuf->m_towrite / bpf, pbuf->m_offset);
if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
goto write_failed;
}
pbuf->m_towrite = 0;
pbuf->m_offset = offset + towrite; /* keep track of current offset */
}
*sample += avail;
offset += avail * bpf;
pbuf->m_lastoffset = offset;
/* 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");
}
}
}
fake_done:
/* we consumed all samples here */
data = data_end + bpf;
pbuf->in_commit = FALSE;
pa_threaded_mainloop_unlock (mainloop);
done:
result = inr - ((data_end - data) / bpf);
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 (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 (mainloop);
goto done;
}
was_paused:
{
pbuf->in_commit = FALSE;
GST_LOG_OBJECT (psink, "we are paused");
pa_threaded_mainloop_unlock (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;
}
}
/* write pending local samples, must be called with the mainloop lock */
static void
gst_pulsering_flush (GstPulseRingBuffer * pbuf)
{
GstPulseSink *psink;
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_DEBUG_OBJECT (psink, "entering flush");
/* flush the buffer if possible */
if (pbuf->stream && (pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) {
#ifndef GST_DISABLE_GST_DEBUG
gint bpf;
bpf = (GST_AUDIO_RING_BUFFER_CAST (pbuf))->spec.info.bpf;
GST_LOG_OBJECT (psink,
"flushing %u samples at offset %" G_GINT64_FORMAT,
(guint) pbuf->m_towrite / bpf, pbuf->m_offset);
#endif
if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
goto write_failed;
}
pbuf->m_towrite = 0;
pbuf->m_offset += pbuf->m_towrite; /* keep track of current offset */
}
done:
return;
/* ERRORS */
write_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_write() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto done;
}
}
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 gboolean gst_pulsesink_query (GstBaseSink * sink, GstQuery * query);
static GstStateChangeReturn gst_pulsesink_change_state (GstElement * element,
GstStateChange transition);
#define gst_pulsesink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstPulseSink, gst_pulsesink, GST_TYPE_AUDIO_BASE_SINK,
gst_pulsesink_init_contexts ();
G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL)
);
static GstAudioRingBuffer *
gst_pulsesink_create_ringbuffer (GstAudioBaseSink * sink)
{
GstAudioRingBuffer *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 GstBuffer *
gst_pulsesink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
{
switch (sink->ringbuffer->spec.type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC:
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC:
{
/* FIXME: alloc memory from PA if possible */
gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
GstBuffer *out;
GstMapInfo inmap, outmap;
gboolean res;
if (framesize <= 0)
return NULL;
out = gst_buffer_new_and_alloc (framesize);
gst_buffer_map (buf, &inmap, GST_MAP_READ);
gst_buffer_map (out, &outmap, GST_MAP_WRITE);
res = gst_audio_iec61937_payload (inmap.data, inmap.size,
outmap.data, outmap.size, &sink->ringbuffer->spec, G_BIG_ENDIAN);
gst_buffer_unmap (buf, &inmap);
gst_buffer_unmap (out, &outmap);
if (!res) {
gst_buffer_unref (out);
return NULL;
}
gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_METADATA, 0, -1);
return out;
}
default:
return gst_buffer_ref (buf);
}
}
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;
GstAudioBaseSinkClass *gstaudiosink_class = GST_AUDIO_BASE_SINK_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gchar *clientname;
gobject_class->finalize = gst_pulsesink_finalize;
gobject_class->set_property = gst_pulsesink_set_property;
gobject_class->get_property = gst_pulsesink_get_property;
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_pulsesink_event);
gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_pulsesink_query);
/* 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);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_pulsesink_change_state);
gstaudiosink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_pulsesink_create_ringbuffer);
gstaudiosink_class->payload = GST_DEBUG_FUNCPTR (gst_pulsesink_payload);
/* 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_CURRENT_DEVICE,
g_param_spec_string ("current-device", "Current Device",
"The current PulseAudio sink device", DEFAULT_CURRENT_DEVICE,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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));
/**
* GstPulseSink:client-name:
*
* The PulseAudio client name to use.
*/
clientname = gst_pulse_client_name ();
g_object_class_install_property (gobject_class,
PROP_CLIENT_NAME,
g_param_spec_string ("client-name", "Client Name",
"The PulseAudio client name to use", clientname,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
g_free (clientname);
/**
* GstPulseSink:stream-properties:
*
* List of pulseaudio stream properties. A list of defined properties can be
* found in the pulseaudio api docs.
*
* Below is an example for registering as a music application to pulseaudio.
* |[
* GstStructure *props;
*
* props = gst_structure_from_string ("props,media.role=music", NULL);
* g_object_set (pulse, "stream-properties", props, NULL);
* gst_structure_free
* ]|
*/
g_object_class_install_property (gobject_class,
PROP_STREAM_PROPERTIES,
g_param_spec_boxed ("stream-properties", "stream properties",
"list of pulseaudio stream properties",
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_set_static_metadata (gstelement_class,
"PulseAudio Audio Sink",
"Sink/Audio", "Plays audio to a PulseAudio server", "Lennart Poettering");
caps =
gst_pulse_fix_pcm_caps (gst_caps_from_string (PULSE_SINK_TEMPLATE_CAPS));
gst_element_class_add_pad_template (gstelement_class,
gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, caps));
gst_caps_unref (caps);
}
static void
free_device_info (GstPulseDeviceInfo * device_info)
{
GList *l;
g_free (device_info->description);
for (l = g_list_first (device_info->formats); l; l = g_list_next (l))
pa_format_info_free ((pa_format_info *) l->data);
g_list_free (device_info->formats);
}
/* Returns the current time of the sink ringbuffer. The timing_info is updated
* on every data write/flush and every 100ms (PA_STREAM_AUTO_TIMING_UPDATE).
*/
static GstClockTime
gst_pulsesink_get_time (GstClock * clock, GstAudioBaseSink * sink)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_usec_t time;
if (!sink->ringbuffer || !sink->ringbuffer->acquired)
return GST_CLOCK_TIME_NONE;
pbuf = GST_PULSERING_BUFFER_CAST (sink->ringbuffer);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
if (g_atomic_int_get (&psink->format_lost)) {
/* Stream was lost in a format change, it'll get set up again once
* upstream renegotiates */
return psink->format_lost_time;
}
pa_threaded_mainloop_lock (mainloop);
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
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 (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 (mainloop);
return GST_CLOCK_TIME_NONE;
}
}
static void
gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
void *userdata)
{
GstPulseDeviceInfo *device_info = (GstPulseDeviceInfo *) userdata;
guint8 j;
if (!i)
goto done;
device_info->description = g_strdup (i->description);
device_info->formats = NULL;
for (j = 0; j < i->n_formats; j++)
device_info->formats = g_list_prepend (device_info->formats,
pa_format_info_copy (i->formats[j]));
done:
pa_threaded_mainloop_signal (mainloop, 0);
}
/* Call with mainloop lock held */
static pa_stream *
gst_pulsesink_create_probe_stream (GstPulseSink * psink,
GstPulseRingBuffer * pbuf, pa_format_info * format)
{
pa_format_info *formats[1] = { format };
pa_stream *stream;
pa_stream_flags_t flags;
GST_LOG_OBJECT (psink, "Creating probe stream");
if (!(stream = pa_stream_new_extended (pbuf->context, "pulsesink probe",
formats, 1, psink->proplist)))
goto error;
/* construct the flags */
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_ADJUST_LATENCY | PA_STREAM_START_CORKED;
pa_stream_set_state_callback (stream, gst_pulsering_stream_state_cb, pbuf);
if (pa_stream_connect_playback (stream, psink->device, NULL, flags, NULL,
NULL) < 0)
goto error;
if (!gst_pulsering_wait_for_stream_ready (psink, stream))
goto error;
return stream;
error:
if (stream)
pa_stream_unref (stream);
return NULL;
}
static GstCaps *
gst_pulsesink_query_getcaps (GstPulseSink * psink, GstCaps * filter)
{
GstPulseRingBuffer *pbuf = NULL;
GstPulseDeviceInfo device_info = { NULL, NULL };
GstCaps *ret = NULL;
GList *i;
pa_operation *o = NULL;
pa_stream *stream;
GST_OBJECT_LOCK (psink);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf != NULL)
gst_object_ref (pbuf);
GST_OBJECT_UNLOCK (psink);
if (!pbuf) {
ret = gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (psink));
goto out;
}
GST_OBJECT_LOCK (pbuf);
pa_threaded_mainloop_lock (mainloop);
if (!pbuf->context) {
ret = gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (psink));
goto unlock;
}
ret = gst_caps_new_empty ();
if (pbuf->stream) {
/* We're in PAUSED or higher */
stream = pbuf->stream;
} else if (pbuf->probe_stream) {
/* We're not paused, but have a cached probe stream */
stream = pbuf->probe_stream;
} else {
/* We're not yet in PAUSED and still need to create a probe stream.
*
* FIXME: PA doesn't accept "any" format. We fix something reasonable since
* this is merely a probe. This should eventually be fixed in PA and
* hard-coding the format should be dropped. */
pa_format_info *format = pa_format_info_new ();
format->encoding = PA_ENCODING_PCM;
pa_format_info_set_sample_format (format, PA_SAMPLE_S16LE);
pa_format_info_set_rate (format, GST_AUDIO_DEF_RATE);
pa_format_info_set_channels (format, GST_AUDIO_DEF_CHANNELS);
pbuf->probe_stream = gst_pulsesink_create_probe_stream (psink, pbuf,
format);
pa_format_info_free (format);
if (!pbuf->probe_stream) {
GST_WARNING_OBJECT (psink, "Could not create probe stream");
goto unlock;
}
stream = pbuf->probe_stream;
}
if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
pa_stream_get_device_name (stream), gst_pulsesink_sink_info_cb,
&device_info)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (mainloop);
if (gst_pulsering_is_dead (psink, pbuf, FALSE))
goto unlock;
}
for (i = g_list_first (device_info.formats); i; i = g_list_next (i)) {
GstCaps *caps = gst_pulse_format_info_to_caps ((pa_format_info *) i->data);
if (caps)
gst_caps_append (ret, caps);
}
unlock:
pa_threaded_mainloop_unlock (mainloop);
/* FIXME: this could be freed after device_name is got */
GST_OBJECT_UNLOCK (pbuf);
if (filter) {
GstCaps *tmp = gst_caps_intersect_full (filter, ret,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (ret);
ret = tmp;
}
out:
free_device_info (&device_info);
if (o)
pa_operation_unref (o);
if (pbuf)
gst_object_unref (pbuf);
GST_DEBUG_OBJECT (psink, "caps %" GST_PTR_FORMAT, ret);
return ret;
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_query_acceptcaps (GstPulseSink * psink, GstCaps * caps)
{
GstPulseRingBuffer *pbuf = NULL;
GstPulseDeviceInfo device_info = { NULL, NULL };
GstCaps *pad_caps;
GstStructure *st;
gboolean ret = FALSE;
GstAudioRingBufferSpec spec = { 0 };
pa_operation *o = NULL;
pa_channel_map channel_map;
pa_format_info *format = NULL;
guint channels;
pad_caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (psink));
ret = gst_caps_is_subset (caps, pad_caps);
gst_caps_unref (pad_caps);
GST_DEBUG_OBJECT (psink, "caps %" GST_PTR_FORMAT, caps);
/* Template caps didn't match */
if (!ret)
goto done;
/* 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 done;
GST_OBJECT_LOCK (psink);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf != NULL)
gst_object_ref (pbuf);
GST_OBJECT_UNLOCK (psink);
/* We're still in NULL state */
if (pbuf == NULL)
goto done;
GST_OBJECT_LOCK (pbuf);
pa_threaded_mainloop_lock (mainloop);
if (pbuf->context == NULL)
goto out;
ret = FALSE;
spec.latency_time = GST_AUDIO_BASE_SINK (psink)->latency_time;
if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
goto out;
if (!gst_pulse_fill_format_info (&spec, &format, &channels))
goto out;
/* Make sure input is framed (one frame per buffer) and can be payloaded */
if (!pa_format_info_is_pcm (format)) {
gboolean framed = FALSE, parsed = FALSE;
st = gst_caps_get_structure (caps, 0);
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;
}
/* initialize the channel map */
if (pa_format_info_is_pcm (format) &&
gst_pulse_gst_to_channel_map (&channel_map, &spec))
pa_format_info_set_channel_map (format, &channel_map);
if (pbuf->stream || pbuf->probe_stream) {
/* We're already in PAUSED or above, so just reuse this stream to query
* sink formats and use those. */
GList *i;
const char *device_name = pa_stream_get_device_name (pbuf->stream ?
pbuf->stream : pbuf->probe_stream);
if (!(o = pa_context_get_sink_info_by_name (pbuf->context, device_name,
gst_pulsesink_sink_info_cb, &device_info)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (mainloop);
if (gst_pulsering_is_dead (psink, pbuf, FALSE))
goto out;
}
for (i = g_list_first (device_info.formats); i; i = g_list_next (i)) {
if (pa_format_info_is_compatible ((pa_format_info *) i->data, format)) {
ret = TRUE;
break;
}
}
} else {
/* We're in READY, let's connect a stream to see if the format is
* accepted by whatever sink we're routed to */
pbuf->probe_stream = gst_pulsesink_create_probe_stream (psink, pbuf,
format);
if (pbuf->probe_stream)
ret = TRUE;
}
out:
if (format)
pa_format_info_free (format);
free_device_info (&device_info);
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (mainloop);
GST_OBJECT_UNLOCK (pbuf);
gst_caps_replace (&spec.caps, NULL);
gst_object_unref (pbuf);
done:
return ret;
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 out;
}
}
static void
gst_pulsesink_init (GstPulseSink * pulsesink)
{
pulsesink->server = NULL;
pulsesink->device = NULL;
pulsesink->device_info.description = NULL;
pulsesink->client_name = gst_pulse_client_name ();
pulsesink->device_info.formats = NULL;
pulsesink->volume = DEFAULT_VOLUME;
pulsesink->volume_set = FALSE;
pulsesink->mute = DEFAULT_MUTE;
pulsesink->mute_set = FALSE;
pulsesink->notify = 0;
g_atomic_int_set (&pulsesink->format_lost, FALSE);
pulsesink->format_lost_time = GST_CLOCK_TIME_NONE;
pulsesink->properties = NULL;
pulsesink->proplist = NULL;
/* override with a custom clock */
if (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock)
gst_object_unref (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock);
GST_AUDIO_BASE_SINK (pulsesink)->provided_clock =
gst_audio_clock_new ("GstPulseSinkClock",
(GstAudioClockGetTimeFunc) gst_pulsesink_get_time, pulsesink, NULL);
}
static void
gst_pulsesink_finalize (GObject * object)
{
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
g_free (pulsesink->server);
g_free (pulsesink->device);
g_free (pulsesink->client_name);
g_free (pulsesink->current_sink_name);
free_device_info (&pulsesink->device_info);
if (pulsesink->properties)
gst_structure_free (pulsesink->properties);
if (pulsesink->proplist)
pa_proplist_free (pulsesink->proplist);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_pulsesink_set_volume (GstPulseSink * psink, gdouble volume)
{
pa_cvolume v;
pa_operation *o = NULL;
GstPulseRingBuffer *pbuf;
uint32_t idx;
if (!mainloop)
goto no_mainloop;
pa_threaded_mainloop_lock (mainloop);
GST_DEBUG_OBJECT (psink, "setting volume to %f", volume);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_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 (pbuf->is_pcm)
gst_pulse_cvolume_from_linear (&v, pbuf->channels, volume);
else
/* FIXME: this will eventually be superceded by checks to see if the volume
* is readable/writable */
goto unlock;
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 (mainloop);
return;
/* ERRORS */
no_mainloop:
{
psink->volume = volume;
psink->volume_set = TRUE;
GST_DEBUG_OBJECT (psink, "we have no mainloop");
return;
}
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;
if (!mainloop)
goto no_mainloop;
pa_threaded_mainloop_lock (mainloop);
GST_DEBUG_OBJECT (psink, "setting mute state to %d", mute);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_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 (mainloop);
return;
/* ERRORS */
no_mainloop:
{
psink->mute = mute;
psink->mute_set = TRUE;
GST_DEBUG_OBJECT (psink, "we have no mainloop");
return;
}
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;
psink->current_sink_idx = i->sink;
if (psink->volume > MAX_VOLUME) {
GST_WARNING_OBJECT (psink, "Clipped volume from %f to %f", psink->volume,
MAX_VOLUME);
psink->volume = MAX_VOLUME;
}
}
done:
pa_threaded_mainloop_signal (mainloop, 0);
}
static void
gst_pulsesink_get_sink_input_info (GstPulseSink * psink, gdouble * volume,
gboolean * mute)
{
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
uint32_t idx;
if (!mainloop)
goto no_mainloop;
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_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 (mainloop);
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
goto unlock;
}
unlock:
if (volume)
*volume = psink->volume;
if (mute)
*mute = psink->mute;
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (mainloop);
return;
/* ERRORS */
no_mainloop:
{
if (volume)
*volume = psink->volume;
if (mute)
*mute = psink->mute;
GST_DEBUG_OBJECT (psink, "we have no mainloop");
return;
}
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 void
gst_pulsesink_current_sink_info_cb (pa_context * c, const pa_sink_info * i,
int eol, void *userdata)
{
GstPulseSink *psink;
psink = GST_PULSESINK_CAST (userdata);
if (!i)
goto done;
/* If the index doesn't match our current stream,
* it implies we just recreated the stream (caps change)
*/
if (i->index == psink->current_sink_idx) {
g_free (psink->current_sink_name);
psink->current_sink_name = g_strdup (i->name);
}
done:
pa_threaded_mainloop_signal (mainloop, 0);
}
static gchar *
gst_pulsesink_get_current_device (GstPulseSink * pulsesink)
{
pa_operation *o = NULL;
GstPulseRingBuffer *pbuf;
gchar *current_sink;
if (!mainloop)
goto no_mainloop;
pbuf =
GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (pulsesink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
gst_pulsesink_get_sink_input_info (pulsesink, NULL, NULL);
pa_threaded_mainloop_lock (mainloop);
if (!(o = pa_context_get_sink_info_by_index (pbuf->context,
pulsesink->current_sink_idx, gst_pulsesink_current_sink_info_cb,
pulsesink)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (mainloop);
if (gst_pulsering_is_dead (pulsesink, pbuf, TRUE))
goto unlock;
}
unlock:
current_sink = g_strdup (pulsesink->current_sink_name);
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (mainloop);
return current_sink;
/* ERRORS */
no_mainloop:
{
GST_DEBUG_OBJECT (pulsesink, "we have no mainloop");
return NULL;
}
no_buffer:
{
GST_DEBUG_OBJECT (pulsesink, "we have no ringbuffer");
return NULL;
}
info_failed:
{
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("pa_context_get_sink_input_info() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
static gchar *
gst_pulsesink_device_description (GstPulseSink * psink)
{
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
gchar *t;
if (!mainloop)
goto no_mainloop;
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL)
goto no_buffer;
free_device_info (&psink->device_info);
if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
psink->device, gst_pulsesink_sink_info_cb, &psink->device_info)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (mainloop);
if (gst_pulsering_is_dead (psink, pbuf, FALSE))
goto unlock;
}
unlock:
if (o)
pa_operation_unref (o);
t = g_strdup (psink->device_info.description);
pa_threaded_mainloop_unlock (mainloop);
return t;
/* ERRORS */
no_mainloop:
{
GST_DEBUG_OBJECT (psink, "we have no mainloop");
return NULL;
}
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_stream_device (GstPulseSink * psink, const gchar * device)
{
pa_operation *o = NULL;
GstPulseRingBuffer *pbuf;
uint32_t idx;
if (!mainloop)
goto no_mainloop;
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_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_DEBUG_OBJECT (psink, "setting stream device to %s", device);
if (!(o = pa_context_move_sink_input_by_name (pbuf->context, idx, device,
NULL, NULL)))
goto move_failed;
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (mainloop);
return;
/* ERRORS */
no_mainloop:
{
GST_DEBUG_OBJECT (psink, "we have no mainloop");
return;
}
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");
return;
}
move_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_context_move_sink_input_by_name(%s) failed: %s", device,
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);
break;
case PROP_DEVICE:
g_free (pulsesink->device);
pulsesink->device = g_value_dup_string (value);
gst_pulsesink_set_stream_device (pulsesink, pulsesink->device);
break;
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;
case PROP_CLIENT_NAME:
g_free (pulsesink->client_name);
if (!g_value_get_string (value)) {
GST_WARNING_OBJECT (pulsesink,
"Empty PulseAudio client name not allowed. Resetting to default value");
pulsesink->client_name = gst_pulse_client_name ();
} else
pulsesink->client_name = g_value_dup_string (value);
break;
case PROP_STREAM_PROPERTIES:
if (pulsesink->properties)
gst_structure_free (pulsesink->properties);
pulsesink->properties =
gst_structure_copy (gst_value_get_structure (value));
if (pulsesink->proplist)
pa_proplist_free (pulsesink->proplist);
pulsesink->proplist = gst_pulse_make_proplist (pulsesink->properties);
break;
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_CURRENT_DEVICE:
{
gchar *current_device = gst_pulsesink_get_current_device (pulsesink);
if (current_device)
g_value_take_string (value, current_device);
else
g_value_set_string (value, "");
break;
}
case PROP_DEVICE_NAME:
g_value_take_string (value, gst_pulsesink_device_description (pulsesink));
break;
case PROP_VOLUME:
{
gdouble volume;
gst_pulsesink_get_sink_input_info (pulsesink, &volume, NULL);
g_value_set_double (value, volume);
break;
}
case PROP_MUTE:
{
gboolean mute;
gst_pulsesink_get_sink_input_info (pulsesink, NULL, &mute);
g_value_set_boolean (value, mute);
break;
}
case PROP_CLIENT_NAME:
g_value_set_string (value, pulsesink->client_name);
break;
case PROP_STREAM_PROPERTIES:
gst_value_set_structure (value, pulsesink->properties);
break;
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 (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_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 (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;
}
}
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 (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
/* We're not interested if this operation failed or not */
if (!(o = pa_stream_proplist_update (pbuf->stream, PA_UPDATE_REPLACE,
pl, NULL, NULL))) {
GST_DEBUG_OBJECT (psink, "pa_stream_proplist_update() failed");
}
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (mainloop);
finish:
if (pl)
pa_proplist_free (pl);
return;
/* ERRORS */
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
}
static void
gst_pulsesink_flush_ringbuffer (GstPulseSink * psink)
{
GstPulseRingBuffer *pbuf;
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
gst_pulsering_flush (pbuf);
/* Uncork if we haven't already (happens when waiting to get enough data
* to send out the first time) */
if (pbuf->corked)
gst_pulsering_set_corked (pbuf, FALSE, FALSE);
/* We're not interested if this operation failed or not */
unlock:
pa_threaded_mainloop_unlock (mainloop);
return;
/* ERRORS */
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
}
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)
/* TRANSLATORS: 'song title' by 'artist name' */
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);
gst_pulsesink_change_props (pulsesink, l);
break;
}
case GST_EVENT_GAP:{
GstClockTime timestamp, duration;
gst_event_parse_gap (event, ×tamp, &duration);
if (duration == GST_CLOCK_TIME_NONE)
gst_pulsesink_flush_ringbuffer (pulsesink);
break;
}
case GST_EVENT_EOS:
gst_pulsesink_flush_ringbuffer (pulsesink);
break;
default:
;
}
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
}
static gboolean
gst_pulsesink_query (GstBaseSink * sink, GstQuery * query)
{
GstPulseSink *pulsesink = GST_PULSESINK_CAST (sink);
gboolean ret = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *caps, *filter;
gst_query_parse_caps (query, &filter);
caps = gst_pulsesink_query_getcaps (pulsesink, filter);
if (caps) {
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
ret = TRUE;
}
break;
}
case GST_QUERY_ACCEPT_CAPS:
{
GstCaps *caps;
gst_query_parse_accept_caps (query, &caps);
ret = gst_pulsesink_query_acceptcaps (pulsesink, caps);
gst_query_set_accept_caps_result (query, ret);
ret = TRUE;
break;
}
default:
ret = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
break;
}
return ret;
}
static void
gst_pulsesink_release_mainloop (GstPulseSink * psink)
{
if (!mainloop)
return;
pa_threaded_mainloop_lock (mainloop);
while (psink->defer_pending) {
GST_DEBUG_OBJECT (psink, "waiting for stream status message emission");
pa_threaded_mainloop_wait (mainloop);
}
pa_threaded_mainloop_unlock (mainloop);
g_mutex_lock (&pa_shared_resource_mutex);
mainloop_ref_ct--;
if (!mainloop_ref_ct) {
GST_INFO_OBJECT (psink, "terminating pa main loop thread");
pa_threaded_mainloop_stop (mainloop);
pa_threaded_mainloop_free (mainloop);
mainloop = NULL;
}
g_mutex_unlock (&pa_shared_resource_mutex);
}
static GstStateChangeReturn
gst_pulsesink_change_state (GstElement * element, GstStateChange transition)
{
GstPulseSink *pulsesink = GST_PULSESINK (element);
GstStateChangeReturn ret;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
g_mutex_lock (&pa_shared_resource_mutex);
if (!mainloop_ref_ct) {
GST_INFO_OBJECT (element, "new pa main loop thread");
if (!(mainloop = pa_threaded_mainloop_new ()))
goto mainloop_failed;
if (pa_threaded_mainloop_start (mainloop) < 0) {
pa_threaded_mainloop_free (mainloop);
goto mainloop_start_failed;
}
mainloop_ref_ct = 1;
g_mutex_unlock (&pa_shared_resource_mutex);
} else {
GST_INFO_OBJECT (element, "reusing pa main loop thread");
mainloop_ref_ct++;
g_mutex_unlock (&pa_shared_resource_mutex);
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
GST_AUDIO_BASE_SINK (pulsesink)->provided_clock, TRUE));
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
goto state_failure;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* format_lost is reset in release() in audiobasesink */
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
GST_AUDIO_BASE_SINK (pulsesink)->provided_clock));
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_pulsesink_release_mainloop (pulsesink);
break;
default:
break;
}
return ret;
/* ERRORS */
mainloop_failed:
{
g_mutex_unlock (&pa_shared_resource_mutex);
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("pa_threaded_mainloop_new() failed"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
mainloop_start_failed:
{
g_mutex_unlock (&pa_shared_resource_mutex);
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("pa_threaded_mainloop_start() failed"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
state_failure:
{
if (transition == GST_STATE_CHANGE_NULL_TO_READY) {
/* Clear the PA mainloop if audiobasesink failed to open the ring_buffer */
g_assert (mainloop);
gst_pulsesink_release_mainloop (pulsesink);
}
return ret;
}
}