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gstreamer/ext/pulse/pulsesink.c
Tim-Philipp Müller f325935314 pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling 
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.

g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.

Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
2011-04-16 18:15:43 +01:00

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/*-*- Mode: C; c-basic-offset: 2 -*-*/
/* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
* (c) 2009 Wim Taymans
*
* gst-pulse is free software; you can redistribute it and/or modify
* it under the terms of the GNU Lesser General Public License as
* published by the Free Software Foundation; either version 2.1 of the
* License, or (at your option) any later version.
*
* gst-pulse is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
/**
* SECTION:element-pulsesink
* @see_also: pulsesrc, pulsemixer
*
* This element outputs audio to a
* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink
* ]| Play an Ogg/Vorbis file.
* |[
* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink
* ]| Play a 440Hz sine wave.
* |[
* gst-launch -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".
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdio.h>
#include <gst/base/gstbasesink.h>
#include <gst/gsttaglist.h>
#include <gst/interfaces/streamvolume.h>
#include <gst/gst-i18n-plugin.h>
#include <gst/pbutils/pbutils.h> /* only used for GST_PLUGINS_BASE_VERSION_* */
#include "pulsesink.h"
#include "pulseutil.h"
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
/* according to
* http://www.pulseaudio.org/ticket/314
* we need pulse-0.9.12 to use sink volume properties
*/
#define DEFAULT_SERVER NULL
#define DEFAULT_DEVICE NULL
#define DEFAULT_DEVICE_NAME NULL
#define DEFAULT_VOLUME 1.0
#define DEFAULT_MUTE FALSE
#define MAX_VOLUME 10.0
enum
{
PROP_0,
PROP_SERVER,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_VOLUME,
PROP_MUTE,
PROP_CLIENT,
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;
/* 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 = NULL;
/* We keep a custom ringbuffer that is backed up by data allocated by
* pulseaudio. We must also overide the commit function to write into
* pulseaudio memory instead. */
struct _GstPulseRingBuffer
{
GstRingBuffer object;
gchar *context_name;
gchar *stream_name;
pa_context *context;
pa_stream *stream;
pa_sample_spec sample_spec;
#ifdef HAVE_PULSE_0_9_16
void *m_data;
size_t m_towrite;
size_t m_writable;
gint64 m_offset;
gint64 m_lastoffset;
#endif
gboolean corked:1;
gboolean in_commit:1;
gboolean paused:1;
};
struct _GstPulseRingBufferClass
{
GstRingBufferClass parent_class;
};
static GType gst_pulseringbuffer_get_type (void);
static void gst_pulseringbuffer_finalize (GObject * object);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_pulseringbuffer_open_device (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_close_device (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec);
static gboolean gst_pulseringbuffer_release (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_start (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_pause (GstRingBuffer * buf);
static gboolean gst_pulseringbuffer_stop (GstRingBuffer * buf);
static void gst_pulseringbuffer_clear (GstRingBuffer * buf);
static guint gst_pulseringbuffer_commit (GstRingBuffer * buf,
guint64 * sample, guchar * data, gint in_samples, gint out_samples,
gint * accum);
G_DEFINE_TYPE (GstPulseRingBuffer, gst_pulseringbuffer, GST_TYPE_RING_BUFFER);
static void
gst_pulsesink_init_contexts (void)
{
g_assert (pa_shared_resource_mutex == NULL);
pa_shared_resource_mutex = g_mutex_new ();
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;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) 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;
#ifdef HAVE_PULSE_0_9_13
pa_sample_spec_init (&pbuf->sample_spec);
#else
pbuf->sample_spec.format = PA_SAMPLE_INVALID;
pbuf->sample_spec.rate = 0;
pbuf->sample_spec.channels = 0;
#endif
#ifdef HAVE_PULSE_0_9_16
pbuf->m_data = NULL;
pbuf->m_towrite = 0;
pbuf->m_writable = 0;
pbuf->m_offset = 0;
pbuf->m_lastoffset = 0;
#endif
pbuf->corked = TRUE;
pbuf->in_commit = FALSE;
pbuf->paused = FALSE;
}
static void
gst_pulsering_destroy_stream (GstPulseRingBuffer * pbuf)
{
if (pbuf->stream) {
#ifdef HAVE_PULSE_0_9_16
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;
}
#endif
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);
#ifdef HAVE_PULSE_0_9_12
pa_context_set_subscribe_callback (pctx->context, NULL, NULL);
#endif
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;
}
}
#ifdef HAVE_PULSE_0_9_12
static void
gst_pulsering_context_subscribe_cb (pa_context * c,
pa_subscription_event_type_t t, uint32_t idx, void *userdata)
{
GstPulseSink *psink;
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 %d, idx %u", t, idx);
if (!pbuf->stream)
continue;
if (idx != pa_stream_get_index (pbuf->stream))
continue;
/* Actually this event is also triggered when other properties of
* the stream change that are unrelated to the volume. However it is
* probably cheaper to signal the change here and check for the
* volume when the GObject property is read instead of querying it always. */
/* inform streaming thread to notify */
g_atomic_int_compare_and_exchange (&psink->notify, 0, 1);
}
}
#endif
/* will be called when the device should be opened. In this case we will connect
* to the server. We should not try to open any streams in this state. */
static gboolean
gst_pulseringbuffer_open_device (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
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);
#ifdef HAVE_PULSE_0_9_12
pa_context_set_subscribe_callback (pctx->context,
gst_pulsering_context_subscribe_cb, pctx);
#endif
/* 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);
}
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;
}
}
/* close the device */
static gboolean
gst_pulseringbuffer_close_device (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
GST_LOG_OBJECT (psink, "closing device");
pa_threaded_mainloop_lock (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;
GstRingBuffer *rbuf;
GstPulseRingBuffer *pbuf;
rbuf = GST_RING_BUFFER_CAST (userdata);
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_LOG_OBJECT (psink, "got request for length %" G_GSIZE_FORMAT, length);
if (pbuf->in_commit && (length >= rbuf->spec.segsize)) {
/* only signal when we are waiting in the commit thread
* and got request for atleast a segment */
pa_threaded_mainloop_signal (mainloop, 0);
}
}
static void
gst_pulsering_stream_underflow_cb (pa_stream * s, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_WARNING_OBJECT (psink, "Got underflow");
}
static void
gst_pulsering_stream_overflow_cb (pa_stream * s, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
GST_WARNING_OBJECT (psink, "Got overflow");
}
static void
gst_pulsering_stream_latency_cb (pa_stream * s, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
const pa_timing_info *info;
pa_usec_t sink_usec;
info = pa_stream_get_timing_info (s);
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
if (!info) {
GST_LOG_OBJECT (psink, "latency update (information unknown)");
return;
}
#ifdef HAVE_PULSE_0_9_11
sink_usec = info->configured_sink_usec;
#else
sink_usec = 0;
#endif
GST_LOG_OBJECT (psink,
"latency_update, %" G_GUINT64_FORMAT ", %d:%" G_GINT64_FORMAT ", %d:%"
G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT,
GST_TIMEVAL_TO_TIME (info->timestamp), info->write_index_corrupt,
info->write_index, info->read_index_corrupt, info->read_index,
info->sink_usec, sink_usec);
}
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");
}
#ifdef HAVE_PULSE_0_9_11
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");
}
#endif
#ifdef HAVE_PULSE_0_9_15
static void
gst_pulsering_stream_event_cb (pa_stream * p, const char *name,
pa_proplist * pl, void *userdata)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
if (!strcmp (name, PA_STREAM_EVENT_REQUEST_CORK)) {
/* the stream wants to PAUSE, post a message for the application. */
GST_DEBUG_OBJECT (psink, "got request for CORK");
gst_element_post_message (GST_ELEMENT_CAST (psink),
gst_message_new_request_state (GST_OBJECT_CAST (psink),
GST_STATE_PAUSED));
} else if (!strcmp (name, PA_STREAM_EVENT_REQUEST_UNCORK)) {
GST_DEBUG_OBJECT (psink, "got request for UNCORK");
gst_element_post_message (GST_ELEMENT_CAST (psink),
gst_message_new_request_state (GST_OBJECT_CAST (psink),
GST_STATE_PLAYING));
} else {
GST_DEBUG_OBJECT (psink, "got unknown event %s", name);
}
}
#endif
/* This method should create a new stream of the given @spec. No playback should
* start yet so we start in the corked state. */
static gboolean
gst_pulseringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_buffer_attr wanted;
const pa_buffer_attr *actual;
pa_channel_map channel_map;
pa_operation *o = NULL;
#ifdef HAVE_PULSE_0_9_20
pa_cvolume v;
#endif
pa_cvolume *pv = NULL;
pa_stream_flags_t flags;
const gchar *name;
GstAudioClock *clock;
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
pbuf = GST_PULSERING_BUFFER_CAST (buf);
GST_LOG_OBJECT (psink, "creating sample spec");
/* convert the gstreamer sample spec to the pulseaudio format */
if (!gst_pulse_fill_sample_spec (spec, &pbuf->sample_spec))
goto invalid_spec;
pa_threaded_mainloop_lock (mainloop);
/* we need a context and a no stream */
g_assert (pbuf->context);
g_assert (!pbuf->stream);
/* enable event notifications */
GST_LOG_OBJECT (psink, "subscribing to context events");
if (!(o = pa_context_subscribe (pbuf->context,
PA_SUBSCRIPTION_MASK_SINK_INPUT, NULL, NULL)))
goto subscribe_failed;
pa_operation_unref (o);
/* initialize the channel map */
gst_pulse_gst_to_channel_map (&channel_map, spec);
/* find a good name for the stream */
if (psink->stream_name)
name = psink->stream_name;
else
name = "Playback Stream";
/* create a stream */
GST_LOG_OBJECT (psink, "creating stream with name %s", name);
if (!(pbuf->stream = pa_stream_new_with_proplist (pbuf->context, name,
&pbuf->sample_spec, &channel_map, 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);
#ifdef HAVE_PULSE_0_9_11
pa_stream_set_started_callback (pbuf->stream,
gst_pulsering_stream_started_cb, pbuf);
#endif
#ifdef HAVE_PULSE_0_9_15
pa_stream_set_event_callback (pbuf->stream,
gst_pulsering_stream_event_cb, pbuf);
#endif
/* buffering requirements. When setting prebuf to 0, the stream will not pause
* when we cause an underrun, which causes time to continue. */
memset (&wanted, 0, sizeof (wanted));
wanted.tlength = spec->segtotal * spec->segsize;
wanted.maxlength = -1;
wanted.prebuf = 0;
wanted.minreq = spec->segsize;
GST_INFO_OBJECT (psink, "tlength: %d", wanted.tlength);
GST_INFO_OBJECT (psink, "maxlength: %d", wanted.maxlength);
GST_INFO_OBJECT (psink, "prebuf: %d", wanted.prebuf);
GST_INFO_OBJECT (psink, "minreq: %d", wanted.minreq);
#ifdef HAVE_PULSE_0_9_20
/* configure volume when we changed it, else we leave the default */
if (psink->volume_set) {
GST_LOG_OBJECT (psink, "have volume of %f", psink->volume);
pv = &v;
gst_pulse_cvolume_from_linear (pv, pbuf->sample_spec.channels,
psink->volume);
} else {
pv = NULL;
}
#endif
/* construct the flags */
flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
#ifdef HAVE_PULSE_0_9_11
PA_STREAM_ADJUST_LATENCY |
#endif
PA_STREAM_START_CORKED;
#ifdef HAVE_PULSE_0_9_12
if (psink->mute_set && psink->mute)
flags |= PA_STREAM_START_MUTED;
#endif
/* we always start corked (see flags above) */
pbuf->corked = TRUE;
/* try to connect now */
GST_LOG_OBJECT (psink, "connect for playback to device %s",
GST_STR_NULL (psink->device));
if (pa_stream_connect_playback (pbuf->stream, psink->device,
&wanted, flags, pv, NULL) < 0)
goto connect_failed;
/* our clock will now start from 0 again */
clock = GST_AUDIO_CLOCK (GST_BASE_AUDIO_SINK (psink)->provided_clock);
gst_audio_clock_reset (clock, 0);
for (;;) {
pa_stream_state_t state;
state = pa_stream_get_state (pbuf->stream);
GST_LOG_OBJECT (psink, "stream state is now %d", state);
if (!PA_STREAM_IS_GOOD (state))
goto connect_failed;
if (state == PA_STREAM_READY)
break;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (mainloop);
}
/* After we passed the volume off of to PA we never want to set it
again, since it is PA's job to save/restore volumes. */
psink->volume_set = psink->mute_set = FALSE;
GST_LOG_OBJECT (psink, "stream is acquired now");
/* get the actual buffering properties now */
actual = pa_stream_get_buffer_attr (pbuf->stream);
GST_INFO_OBJECT (psink, "tlength: %d (wanted: %d)", actual->tlength,
wanted.tlength);
GST_INFO_OBJECT (psink, "maxlength: %d", actual->maxlength);
GST_INFO_OBJECT (psink, "prebuf: %d", actual->prebuf);
GST_INFO_OBJECT (psink, "minreq: %d (wanted %d)", actual->minreq,
wanted.minreq);
spec->segsize = actual->minreq;
spec->segtotal = actual->tlength / spec->segsize;
pa_threaded_mainloop_unlock (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 (GstRingBuffer * buf)
{
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
pa_threaded_mainloop_lock (mainloop);
gst_pulsering_destroy_stream (pbuf);
pa_threaded_mainloop_unlock (mainloop);
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));
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 (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (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);
}
static void
mainloop_enter_defer_cb (pa_mainloop_api * api, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
GstMessage *message;
GValue val = { 0 };
g_value_init (&val, G_TYPE_POINTER);
g_value_set_pointer (&val, g_thread_self ());
GST_DEBUG_OBJECT (pulsesink, "posting ENTER stream status");
message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT (pulsesink));
gst_message_set_stream_status_object (message, &val);
gst_element_post_message (GST_ELEMENT (pulsesink), message);
/* signal the waiter */
pulsesink->pa_defer_ran = TRUE;
pa_threaded_mainloop_signal (mainloop, 0);
}
/* start/resume playback ASAP, we don't uncork here but in the commit method */
static gboolean
gst_pulseringbuffer_start (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (mainloop);
GST_DEBUG_OBJECT (psink, "scheduling stream status");
psink->pa_defer_ran = FALSE;
pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
mainloop_enter_defer_cb, psink);
GST_DEBUG_OBJECT (psink, "starting");
pbuf->paused = FALSE;
/* EOS needs running clock */
if (GST_BASE_SINK_CAST (psink)->eos ||
g_atomic_int_get (&GST_BASE_AUDIO_SINK (psink)->abidata.ABI.
eos_rendering))
gst_pulsering_set_corked (pbuf, FALSE, FALSE);
pa_threaded_mainloop_unlock (mainloop);
return TRUE;
}
/* pause/stop playback ASAP */
static gboolean
gst_pulseringbuffer_pause (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
gboolean res;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (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;
}
static void
mainloop_leave_defer_cb (pa_mainloop_api * api, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
GstMessage *message;
GValue val = { 0 };
g_value_init (&val, G_TYPE_POINTER);
g_value_set_pointer (&val, g_thread_self ());
GST_DEBUG_OBJECT (pulsesink, "posting LEAVE stream status");
message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT (pulsesink));
gst_message_set_stream_status_object (message, &val);
gst_element_post_message (GST_ELEMENT (pulsesink), message);
pulsesink->pa_defer_ran = TRUE;
pa_threaded_mainloop_signal (mainloop, 0);
gst_object_unref (pulsesink);
}
/* stop playback, we flush everything. */
static gboolean
gst_pulseringbuffer_stop (GstRingBuffer * buf)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
gboolean res = FALSE;
pa_operation *o = NULL;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
pa_threaded_mainloop_lock (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 (strcmp (psink->pa_version, "0.9.12")) {
/* then try to flush, it's not fatal when this fails */
GST_DEBUG_OBJECT (psink, "flushing");
if ((o = pa_stream_flush (pbuf->stream, gst_pulsering_success_cb, pbuf))) {
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
GST_DEBUG_OBJECT (psink, "wait for completion");
pa_threaded_mainloop_wait (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);
}
GST_DEBUG_OBJECT (psink, "scheduling stream status");
psink->pa_defer_ran = FALSE;
gst_object_ref (psink);
pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
mainloop_leave_defer_cb, psink);
GST_DEBUG_OBJECT (psink, "waiting for stream status");
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, bps); \
s += bps; \
*accum += outr; \
if ((*accum << 1) >= inr) { \
*accum -= inr; \
d += bps; \
} \
} \
in_samples -= (s - sb)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
/* out_samples > in_samples, for rates smaller than 1.0 */
#define FWD_DOWN_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = s, *db = d; \
while (s <= se && d < de) { \
memcpy (d, s, bps); \
d += bps; \
*accum += inr; \
if ((*accum << 1) >= outr) { \
*accum -= outr; \
s += bps; \
} \
} \
in_samples -= (s - sb)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
#define REV_UP_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
memcpy (d, se, bps); \
se -= bps; \
*accum += outr; \
while (d < de && (*accum << 1) >= inr) { \
*accum -= inr; \
d += bps; \
} \
} \
in_samples -= (sb - se)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
#define REV_DOWN_SAMPLES(s,se,d,de) \
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
memcpy (d, se, bps); \
d += bps; \
*accum += inr; \
while (s <= se && (*accum << 1) >= outr) { \
*accum -= outr; \
se -= bps; \
} \
} \
in_samples -= (sb - se)/bps; \
out_samples -= (d - db)/bps; \
GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
/* our custom commit function because we write into the buffer of pulseaudio
* instead of keeping our own buffer */
static guint
gst_pulseringbuffer_commit (GstRingBuffer * buf, guint64 * sample,
guchar * data, gint in_samples, gint out_samples, gint * accum)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
guint result;
guint8 *data_end;
gboolean reverse;
gint *toprocess;
gint inr, outr, bps;
gint64 offset;
guint bufsize;
pbuf = GST_PULSERING_BUFFER_CAST (buf);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
/* FIXME post message rather than using a signal (as mixer interface) */
if (g_atomic_int_compare_and_exchange (&psink->notify, 1, 0)) {
g_object_notify (G_OBJECT (psink), "volume");
g_object_notify (G_OBJECT (psink), "mute");
}
/* make sure the ringbuffer is started */
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
GST_RING_BUFFER_STATE_STARTED)) {
/* see if we are allowed to start it */
if (G_UNLIKELY (g_atomic_int_get (&buf->abidata.ABI.may_start) == FALSE))
goto no_start;
GST_DEBUG_OBJECT (buf, "start!");
if (!gst_ring_buffer_start (buf))
goto start_failed;
}
pa_threaded_mainloop_lock (mainloop);
GST_DEBUG_OBJECT (psink, "entering commit");
pbuf->in_commit = TRUE;
bps = buf->spec.bytes_per_sample;
bufsize = buf->spec.segsize * buf->spec.segtotal;
/* our toy resampler for trick modes */
reverse = out_samples < 0;
out_samples = ABS (out_samples);
if (in_samples >= out_samples)
toprocess = &in_samples;
else
toprocess = &out_samples;
inr = in_samples - 1;
outr = out_samples - 1;
GST_DEBUG_OBJECT (psink, "in %d, out %d", inr, outr);
/* data_end points to the last sample we have to write, not past it. This is
* needed to properly handle reverse playback: it points to the last sample. */
data_end = data + (bps * inr);
if (pbuf->paused)
goto was_paused;
/* offset is in bytes */
offset = *sample * bps;
while (*toprocess > 0) {
size_t avail;
guint towrite;
GST_LOG_OBJECT (psink,
"need to write %d samples at offset %" G_GINT64_FORMAT, *toprocess,
offset);
#ifdef HAVE_PULSE_0_9_16
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 * bps;
/* Only ever write segsize bytes at once. This will
* also limit the PA shm buffer to segsize
*/
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 / bps, 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 (pbuf->m_writable == (size_t) - 1)
goto writable_size_failed;
pbuf->m_writable /= bps;
pbuf->m_writable *= bps; /* 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 a single byte, wait a bit */
GST_LOG_OBJECT (psink, "waiting for free space");
pa_threaded_mainloop_wait (mainloop);
if (pbuf->paused)
goto was_paused;
}
/* make sure we only buffer up latency-time samples */
if (pbuf->m_writable > buf->spec.segsize) {
/* limit buffering to latency-time value */
pbuf->m_writable = buf->spec.segsize;
GST_LOG_OBJECT (psink, "Limiting buffering to %" G_GSIZE_FORMAT,
pbuf->m_writable);
}
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);
/* Just to make sure that we didn't get more than requested */
if (pbuf->m_writable > buf->spec.segsize) {
/* limit buffering to latency-time value */
pbuf->m_writable = buf->spec.segsize;
}
}
if (pbuf->m_writable < towrite)
towrite = pbuf->m_writable;
avail = towrite / bps;
GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
(guint) avail, offset);
if (G_LIKELY (inr == outr && !reverse)) {
/* no rate conversion, simply write out the samples */
/* 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 / bps;
}
/* 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 / bps, 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 */
}
#else
for (;;) {
/* FIXME, this is not quite right */
if ((avail = pa_stream_writable_size (pbuf->stream)) == (size_t) - 1)
goto writable_size_failed;
/* We always try to satisfy a request for data */
GST_LOG_OBJECT (psink, "writable bytes %" G_GSIZE_FORMAT, avail);
/* convert to samples, we can only deal with multiples of the
* sample size */
avail /= bps;
if (avail > 0)
break;
/* see if we need to uncork because we have no free space */
if (pbuf->corked) {
if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
goto uncork_failed;
}
/* we can't write a single byte, wait a bit */
GST_LOG_OBJECT (psink, "waiting for free space");
pa_threaded_mainloop_wait (mainloop);
if (pbuf->paused)
goto was_paused;
}
if (avail > out_samples)
avail = out_samples;
towrite = avail * bps;
GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
(guint) avail, offset);
if (G_LIKELY (inr == outr && !reverse)) {
/* no rate conversion, simply write out the samples */
if (pa_stream_write (pbuf->stream, data, towrite, NULL, offset,
PA_SEEK_ABSOLUTE) < 0)
goto write_failed;
data += towrite;
in_samples -= avail;
out_samples -= avail;
} else {
guint8 *dest, *d, *d_end;
/* we need to allocate a temporary buffer to resample the data into,
* FIXME, we should have a pulseaudio API to allocate this buffer for us
* from the shared memory. */
dest = d = g_malloc (towrite);
d_end = d + towrite;
if (!reverse) {
if (inr >= outr)
/* forward speed up */
FWD_UP_SAMPLES (data, data_end, d, d_end);
else
/* forward slow down */
FWD_DOWN_SAMPLES (data, data_end, d, d_end);
} else {
if (inr >= outr)
/* reverse speed up */
REV_UP_SAMPLES (data, data_end, d, d_end);
else
/* reverse slow down */
REV_DOWN_SAMPLES (data, data_end, d, d_end);
}
/* see what we have left to write */
towrite = (d - dest);
if (pa_stream_write (pbuf->stream, dest, towrite,
g_free, offset, PA_SEEK_ABSOLUTE) < 0)
goto write_failed;
avail = towrite / bps;
}
#endif /* HAVE_PULSE_0_9_16 */
*sample += avail;
offset += avail * bps;
#ifdef HAVE_PULSE_0_9_16
pbuf->m_lastoffset = offset;
#endif
/* check if we need to uncork after writing the samples */
if (pbuf->corked) {
const pa_timing_info *info;
if ((info = pa_stream_get_timing_info (pbuf->stream))) {
GST_LOG_OBJECT (psink,
"read_index at %" G_GUINT64_FORMAT ", offset %" G_GINT64_FORMAT,
info->read_index, offset);
/* we uncork when the read_index is too far behind the offset we need
* to write to. */
if (info->read_index + bufsize <= offset) {
if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
goto uncork_failed;
}
} else {
GST_LOG_OBJECT (psink, "no timing info available yet");
}
}
}
/* we consumed all samples here */
data = data_end + bps;
pbuf->in_commit = FALSE;
pa_threaded_mainloop_unlock (mainloop);
done:
result = inr - ((data_end - data) / bps);
GST_LOG_OBJECT (psink, "wrote %d samples", result);
return result;
/* ERRORS */
unlock_and_fail:
{
pbuf->in_commit = FALSE;
GST_LOG_OBJECT (psink, "we are reset");
pa_threaded_mainloop_unlock (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)
{
#ifdef HAVE_PULSE_0_9_16
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 bps;
bps = (GST_RING_BUFFER_CAST (pbuf))->spec.bytes_per_sample;
GST_LOG_OBJECT (psink,
"flushing %u samples at offset %" G_GINT64_FORMAT,
(guint) pbuf->m_towrite / bps, 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;
}
#endif
}
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 GstStateChangeReturn gst_pulsesink_change_state (GstElement * element,
GstStateChange transition);
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) \
gst_pulsesink_init_contexts (); \
gst_pulsesink_init_interfaces (type);
GST_BOILERPLATE_FULL (GstPulseSink, gst_pulsesink, GstBaseAudioSink,
GST_TYPE_BASE_AUDIO_SINK, _do_init);
static gboolean
gst_pulsesink_interface_supported (GstImplementsInterface *
iface, GType interface_type)
{
GstPulseSink *this = GST_PULSESINK_CAST (iface);
if (interface_type == GST_TYPE_PROPERTY_PROBE && this->probe)
return TRUE;
if (interface_type == GST_TYPE_STREAM_VOLUME)
return TRUE;
return FALSE;
}
static void
gst_pulsesink_implements_interface_init (GstImplementsInterfaceClass * klass)
{
klass->supported = gst_pulsesink_interface_supported;
}
static void
gst_pulsesink_init_interfaces (GType type)
{
static const GInterfaceInfo implements_iface_info = {
(GInterfaceInitFunc) gst_pulsesink_implements_interface_init,
NULL,
NULL,
};
static const GInterfaceInfo probe_iface_info = {
(GInterfaceInitFunc) gst_pulsesink_property_probe_interface_init,
NULL,
NULL,
};
#ifdef HAVE_PULSE_0_9_12
static const GInterfaceInfo svol_iface_info = {
NULL, NULL, NULL
};
g_type_add_interface_static (type, GST_TYPE_STREAM_VOLUME, &svol_iface_info);
#endif
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
&implements_iface_info);
g_type_add_interface_static (type, GST_TYPE_PROPERTY_PROBE,
&probe_iface_info);
}
static void
gst_pulsesink_base_init (gpointer g_class)
{
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 32 ];"
"audio/x-raw-float, "
"endianness = (int) { " ENDIANNESS " }, "
"width = (int) 32, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 32 ];"
"audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 32, "
"depth = (int) 32, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
#ifdef HAVE_PULSE_0_9_15
"audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 24, "
"depth = (int) 24, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 32 ];"
"audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 32, "
"depth = (int) 24, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
#endif
"audio/x-raw-int, "
"signed = (boolean) FALSE, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 32 ];"
"audio/x-alaw, "
"rate = (int) [ 1, MAX], "
"channels = (int) [ 1, 32 ];"
"audio/x-mulaw, "
"rate = (int) [ 1, MAX], " "channels = (int) [ 1, 32 ]")
);
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details_simple (element_class,
"PulseAudio Audio Sink",
"Sink/Audio", "Plays audio to a PulseAudio server", "Lennart Poettering");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&pad_template));
}
static GstRingBuffer *
gst_pulsesink_create_ringbuffer (GstBaseAudioSink * sink)
{
GstRingBuffer *buffer;
GST_DEBUG_OBJECT (sink, "creating ringbuffer");
buffer = g_object_new (GST_TYPE_PULSERING_BUFFER, NULL);
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
return buffer;
}
static void
gst_pulsesink_class_init (GstPulseSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
GstBaseSinkClass *bc;
GstBaseAudioSinkClass *gstaudiosink_class = GST_BASE_AUDIO_SINK_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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);
/* 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);
/* Overwrite GObject fields */
g_object_class_install_property (gobject_class,
PROP_SERVER,
g_param_spec_string ("server", "Server",
"The PulseAudio server to connect to", DEFAULT_SERVER,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"The PulseAudio sink device to connect to", DEFAULT_DEVICE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
#ifdef HAVE_PULSE_0_9_12
g_object_class_install_property (gobject_class,
PROP_VOLUME,
g_param_spec_double ("volume", "Volume",
"Linear volume of this stream, 1.0=100%", 0.0, MAX_VOLUME,
DEFAULT_VOLUME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MUTE,
g_param_spec_boolean ("mute", "Mute",
"Mute state of this stream", DEFAULT_MUTE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
#endif
/**
* GstPulseSink:client
*
* The PulseAudio client name to use.
*
* Since: 0.10.25
*/
g_object_class_install_property (gobject_class,
PROP_CLIENT,
g_param_spec_string ("client", "Client",
"The PulseAudio client name to use", gst_pulse_client_name (),
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY));
/**
* GstPulseSink:stream-properties
*
* List of pulseaudio stream properties. A list of defined properties can be
* found in the <ulink url="http://0pointer.de/lennart/projects/pulseaudio/doxygen/proplist_8h.html">pulseaudio api docs</ulink>.
*
* Below is an example for registering as a music application to pulseaudio.
* |[
* GstStructure *props;
*
* props = gst_structure_from_string ("props,media.role=music", NULL);
* g_object_set (pulse, "stream-properties", props, NULL);
* gst_structure_free
* ]|
*
* Since: 0.10.26
*/
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));
}
/* returns the current time of the sink ringbuffer */
static GstClockTime
gst_pulsesink_get_time (GstClock * clock, GstBaseAudioSink * sink)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
pa_usec_t time;
if (!sink->ringbuffer || !sink->ringbuffer->acquired)
return GST_CLOCK_TIME_NONE;
pbuf = GST_PULSERING_BUFFER_CAST (sink->ringbuffer);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
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_init (GstPulseSink * pulsesink, GstPulseSinkClass * klass)
{
pulsesink->server = NULL;
pulsesink->device = NULL;
pulsesink->device_description = NULL;
pulsesink->client_name = gst_pulse_client_name ();
pulsesink->volume = DEFAULT_VOLUME;
pulsesink->volume_set = FALSE;
pulsesink->mute = DEFAULT_MUTE;
pulsesink->mute_set = FALSE;
pulsesink->notify = 0;
/* needed for conditional execution */
pulsesink->pa_version = pa_get_library_version ();
pulsesink->properties = NULL;
pulsesink->proplist = NULL;
GST_DEBUG_OBJECT (pulsesink, "using pulseaudio version %s",
pulsesink->pa_version);
/* override with a custom clock */
if (GST_BASE_AUDIO_SINK (pulsesink)->provided_clock)
gst_object_unref (GST_BASE_AUDIO_SINK (pulsesink)->provided_clock);
GST_BASE_AUDIO_SINK (pulsesink)->provided_clock =
gst_audio_clock_new ("GstPulseSinkClock",
(GstAudioClockGetTimeFunc) gst_pulsesink_get_time, pulsesink);
/* TRUE for sinks, FALSE for sources */
pulsesink->probe = gst_pulseprobe_new (G_OBJECT (pulsesink),
G_OBJECT_GET_CLASS (pulsesink), PROP_DEVICE, pulsesink->device,
TRUE, FALSE);
}
static void
gst_pulsesink_finalize (GObject * object)
{
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
g_free (pulsesink->server);
g_free (pulsesink->device);
g_free (pulsesink->device_description);
g_free (pulsesink->client_name);
if (pulsesink->properties)
gst_structure_free (pulsesink->properties);
if (pulsesink->proplist)
pa_proplist_free (pulsesink->proplist);
if (pulsesink->probe) {
gst_pulseprobe_free (pulsesink->probe);
pulsesink->probe = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
#ifdef HAVE_PULSE_0_9_12
static void
gst_pulsesink_set_volume (GstPulseSink * psink, gdouble volume)
{
pa_cvolume v;
pa_operation *o = NULL;
GstPulseRingBuffer *pbuf;
uint32_t idx;
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_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
goto no_index;
gst_pulse_cvolume_from_linear (&v, pbuf->sample_spec.channels, volume);
if (!(o = pa_context_set_sink_input_volume (pbuf->context, idx,
&v, NULL, NULL)))
goto volume_failed;
/* We don't really care about the result of this call */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (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_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
goto no_index;
if (!(o = pa_context_set_sink_input_mute (pbuf->context, idx,
mute, NULL, NULL)))
goto mute_failed;
/* We don't really care about the result of this call */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (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;
}
done:
pa_threaded_mainloop_signal (mainloop, 0);
}
static gdouble
gst_pulsesink_get_volume (GstPulseSink * psink)
{
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
gdouble v = DEFAULT_VOLUME;
uint32_t idx;
if (!mainloop)
goto no_mainloop;
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
goto no_index;
if (!(o = pa_context_get_sink_input_info (pbuf->context, idx,
gst_pulsesink_sink_input_info_cb, pbuf)))
goto info_failed;
while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait (mainloop);
if (gst_pulsering_is_dead (psink, pbuf, TRUE))
goto unlock;
}
unlock:
v = psink->volume;
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (mainloop);
if (v > MAX_VOLUME) {
GST_WARNING_OBJECT (psink, "Clipped volume from %f to %f", v, MAX_VOLUME);
v = MAX_VOLUME;
}
return v;
/* ERRORS */
no_mainloop:
{
v = psink->volume;
GST_DEBUG_OBJECT (psink, "we have no mainloop");
return v;
}
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
no_index:
{
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
goto unlock;
}
info_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_context_get_sink_input_info() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
static gboolean
gst_pulsesink_get_mute (GstPulseSink * psink)
{
GstPulseRingBuffer *pbuf;
pa_operation *o = NULL;
uint32_t idx;
gboolean mute = FALSE;
if (!mainloop)
goto no_mainloop;
pa_threaded_mainloop_lock (mainloop);
mute = psink->mute;
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
goto no_index;
if (!(o = pa_context_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 (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (mainloop);
return mute;
/* ERRORS */
no_mainloop:
{
mute = psink->mute;
GST_DEBUG_OBJECT (psink, "we have no mainloop");
return mute;
}
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
no_index:
{
GST_DEBUG_OBJECT (psink, "we don't have a stream index");
goto unlock;
}
info_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_context_get_sink_input_info() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
#endif
static void
gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
void *userdata)
{
GstPulseRingBuffer *pbuf;
GstPulseSink *psink;
pbuf = GST_PULSERING_BUFFER_CAST (userdata);
psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
if (!i)
goto done;
g_free (psink->device_description);
psink->device_description = g_strdup (i->description);
done:
pa_threaded_mainloop_signal (mainloop, 0);
}
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_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL)
goto no_buffer;
if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
psink->device, gst_pulsesink_sink_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, FALSE))
goto unlock;
}
unlock:
if (o)
pa_operation_unref (o);
t = g_strdup (psink->device_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_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
switch (prop_id) {
case PROP_SERVER:
g_free (pulsesink->server);
pulsesink->server = g_value_dup_string (value);
if (pulsesink->probe)
gst_pulseprobe_set_server (pulsesink->probe, pulsesink->server);
break;
case PROP_DEVICE:
g_free (pulsesink->device);
pulsesink->device = g_value_dup_string (value);
break;
#ifdef HAVE_PULSE_0_9_12
case PROP_VOLUME:
gst_pulsesink_set_volume (pulsesink, g_value_get_double (value));
break;
case PROP_MUTE:
gst_pulsesink_set_mute (pulsesink, g_value_get_boolean (value));
break;
#endif
case PROP_CLIENT:
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_DEVICE_NAME:
g_value_take_string (value, gst_pulsesink_device_description (pulsesink));
break;
#ifdef HAVE_PULSE_0_9_12
case PROP_VOLUME:
g_value_set_double (value, gst_pulsesink_get_volume (pulsesink));
break;
case PROP_MUTE:
g_value_set_boolean (value, gst_pulsesink_get_mute (pulsesink));
break;
#endif
case PROP_CLIENT:
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_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
g_free (pbuf->stream_name);
pbuf->stream_name = g_strdup (t);
if (!(o = pa_stream_set_name (pbuf->stream, pbuf->stream_name, NULL, NULL)))
goto name_failed;
/* We're not interested if this operation failed or not */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (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;
}
}
#ifdef HAVE_PULSE_0_9_11
static void
gst_pulsesink_change_props (GstPulseSink * psink, GstTagList * l)
{
static const gchar *const map[] = {
GST_TAG_TITLE, PA_PROP_MEDIA_TITLE,
/* might get overriden in the next iteration by GST_TAG_ARTIST */
GST_TAG_PERFORMER, PA_PROP_MEDIA_ARTIST,
GST_TAG_ARTIST, PA_PROP_MEDIA_ARTIST,
GST_TAG_LANGUAGE_CODE, PA_PROP_MEDIA_LANGUAGE,
GST_TAG_LOCATION, PA_PROP_MEDIA_FILENAME,
/* We might add more here later on ... */
NULL
};
pa_proplist *pl = NULL;
const gchar *const *t;
gboolean empty = TRUE;
pa_operation *o = NULL;
GstPulseRingBuffer *pbuf;
pl = pa_proplist_new ();
for (t = map; *t; t += 2) {
gchar *n = NULL;
if (gst_tag_list_get_string (l, *t, &n)) {
if (n && *n) {
pa_proplist_sets (pl, *(t + 1), n);
empty = FALSE;
}
g_free (n);
}
}
if (empty)
goto finish;
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
if (!(o = pa_stream_proplist_update (pbuf->stream, PA_UPDATE_REPLACE,
pl, NULL, NULL)))
goto update_failed;
/* We're not interested if this operation failed or not */
unlock:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (mainloop);
finish:
if (pl)
pa_proplist_free (pl);
return;
/* ERRORS */
no_buffer:
{
GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
goto unlock;
}
update_failed:
{
GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
("pa_stream_proplist_update() failed: %s",
pa_strerror (pa_context_errno (pbuf->context))), (NULL));
goto unlock;
}
}
#endif
static void
gst_pulsesink_flush_ringbuffer (GstPulseSink * psink)
{
GstPulseRingBuffer *pbuf;
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_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);
#ifdef HAVE_PULSE_0_9_11
gst_pulsesink_change_props (pulsesink, l);
#endif
break;
}
case GST_EVENT_EOS:
gst_pulsesink_flush_ringbuffer (pulsesink);
break;
default:
;
}
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
}
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;
mainloop_ref_ct = 1;
pa_threaded_mainloop_start (mainloop);
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_BASE_AUDIO_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:
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
GST_BASE_AUDIO_SINK (pulsesink)->provided_clock));
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (mainloop) {
g_mutex_lock (pa_shared_resource_mutex);
mainloop_ref_ct--;
if (!mainloop_ref_ct) {
GST_INFO_OBJECT (element, "terminating pa main loop thread");
pa_threaded_mainloop_stop (mainloop);
pa_threaded_mainloop_free (mainloop);
mainloop = NULL;
}
g_mutex_unlock (pa_shared_resource_mutex);
}
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;
}
state_failure:
{
if (transition == GST_STATE_CHANGE_NULL_TO_READY) {
/* Clear the PA mainloop if baseaudiosink failed to open the ring_buffer */
g_assert (mainloop);
g_mutex_lock (pa_shared_resource_mutex);
mainloop_ref_ct--;
if (!mainloop_ref_ct) {
GST_INFO_OBJECT (element, "terminating pa main loop thread");
pa_threaded_mainloop_stop (mainloop);
pa_threaded_mainloop_free (mainloop);
mainloop = NULL;
}
g_mutex_unlock (pa_shared_resource_mutex);
}
return ret;
}
}
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