mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
3369 lines
94 KiB
C
3369 lines
94 KiB
C
/*-*- 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
|
|
* @see_also: pulsesrc
|
|
*
|
|
* This element outputs audio to a
|
|
* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
|
|
*
|
|
* <refsect2>
|
|
* <title>Example pipelines</title>
|
|
* |[
|
|
* 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".
|
|
* </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/audio/audio.h>
|
|
#include <gst/gst-i18n-plugin.h>
|
|
|
|
#include <gst/pbutils/pbutils.h> /* only used for GST_PLUGINS_BASE_VERSION_* */
|
|
|
|
#include <gst/glib-compat-private.h>
|
|
|
|
#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;
|
|
|
|
/* 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);
|
|
}
|
|
|
|
/* called from pulse 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);
|
|
}
|
|
|
|
/* 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, "scheduling stream status");
|
|
psink->defer_pending++;
|
|
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_AUDIO_BASE_SINK (psink)->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 (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;
|
|
}
|
|
|
|
/* called from pulse 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);
|
|
}
|
|
|
|
/* 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);
|
|
}
|
|
|
|
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);
|
|
|
|
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);
|
|
|
|
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS (PULSE_SINK_TEMPLATE_CAPS));
|
|
|
|
#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);
|
|
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 <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
|
|
* ]|
|
|
*/
|
|
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");
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&pad_template));
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
static gboolean
|
|
gst_pulse_format_info_int_prop_to_value (pa_format_info * format,
|
|
const char *key, GValue * value)
|
|
{
|
|
GValue v = { 0, };
|
|
int i;
|
|
int *a, n;
|
|
int min, max;
|
|
|
|
if (pa_format_info_get_prop_int (format, key, &i) == 0) {
|
|
g_value_init (value, G_TYPE_INT);
|
|
g_value_set_int (value, i);
|
|
|
|
} else if (pa_format_info_get_prop_int_array (format, key, &a, &n) == 0) {
|
|
g_value_init (value, GST_TYPE_LIST);
|
|
g_value_init (&v, G_TYPE_INT);
|
|
|
|
for (i = 0; i < n; i++) {
|
|
g_value_set_int (&v, a[i]);
|
|
gst_value_list_append_value (value, &v);
|
|
}
|
|
|
|
pa_xfree (a);
|
|
|
|
} else if (pa_format_info_get_prop_int_range (format, key, &min, &max) == 0) {
|
|
g_value_init (value, GST_TYPE_INT_RANGE);
|
|
gst_value_set_int_range (value, min, max);
|
|
|
|
} else {
|
|
/* Property not available or is not an int type */
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_pulse_format_info_to_caps (pa_format_info * format)
|
|
{
|
|
GstCaps *ret = NULL;
|
|
GValue v = { 0, };
|
|
pa_sample_spec ss;
|
|
|
|
switch (format->encoding) {
|
|
case PA_ENCODING_PCM:{
|
|
char *tmp = NULL;
|
|
|
|
pa_format_info_to_sample_spec (format, &ss, NULL);
|
|
|
|
if (pa_format_info_get_prop_string (format,
|
|
PA_PROP_FORMAT_SAMPLE_FORMAT, &tmp)) {
|
|
/* No specific sample format means any sample format */
|
|
ret = gst_caps_from_string (_PULSE_CAPS_PCM);
|
|
goto out;
|
|
|
|
} else if (ss.format == PA_SAMPLE_ALAW) {
|
|
ret = gst_caps_from_string (_PULSE_CAPS_ALAW);
|
|
|
|
} else if (ss.format == PA_SAMPLE_ULAW) {
|
|
ret = gst_caps_from_string (_PULSE_CAPS_MP3);
|
|
|
|
} else {
|
|
/* Linear PCM format */
|
|
const char *sformat =
|
|
gst_pulse_sample_format_to_caps_format (ss.format);
|
|
|
|
ret = gst_caps_from_string (_PULSE_CAPS_LINEAR);
|
|
|
|
if (sformat)
|
|
gst_caps_set_simple (ret, "format", G_TYPE_STRING, NULL);
|
|
}
|
|
|
|
pa_xfree (tmp);
|
|
break;
|
|
}
|
|
|
|
case PA_ENCODING_AC3_IEC61937:
|
|
ret = gst_caps_from_string (_PULSE_CAPS_AC3);
|
|
break;
|
|
|
|
case PA_ENCODING_EAC3_IEC61937:
|
|
ret = gst_caps_from_string (_PULSE_CAPS_EAC3);
|
|
break;
|
|
|
|
case PA_ENCODING_DTS_IEC61937:
|
|
ret = gst_caps_from_string (_PULSE_CAPS_DTS);
|
|
break;
|
|
|
|
case PA_ENCODING_MPEG_IEC61937:
|
|
ret = gst_caps_from_string (_PULSE_CAPS_MP3);
|
|
break;
|
|
|
|
default:
|
|
GST_WARNING ("Found a PA format that we don't support yet");
|
|
goto out;
|
|
}
|
|
|
|
if (gst_pulse_format_info_int_prop_to_value (format, PA_PROP_FORMAT_RATE, &v))
|
|
gst_caps_set_value (ret, "rate", &v);
|
|
|
|
g_value_unset (&v);
|
|
|
|
if (gst_pulse_format_info_int_prop_to_value (format, PA_PROP_FORMAT_CHANNELS,
|
|
&v))
|
|
gst_caps_set_value (ret, "channels", &v);
|
|
|
|
out:
|
|
return ret;
|
|
}
|
|
|
|
/* 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;
|
|
}
|
|
|
|
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);
|
|
if (!pbuf->probe_stream) {
|
|
GST_WARNING_OBJECT (psink, "Could not create probe stream");
|
|
goto unlock;
|
|
}
|
|
|
|
pa_format_info_free (format);
|
|
|
|
stream = pbuf->probe_stream;
|
|
}
|
|
|
|
ret = gst_caps_new_empty ();
|
|
|
|
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)) {
|
|
gst_caps_append (ret,
|
|
gst_pulse_format_info_to_caps ((pa_format_info *) i->data));
|
|
}
|
|
|
|
if (filter) {
|
|
GstCaps *tmp = gst_caps_intersect_full (filter, ret,
|
|
GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (ret);
|
|
ret = tmp;
|
|
}
|
|
|
|
unlock:
|
|
pa_threaded_mainloop_unlock (mainloop);
|
|
/* FIXME: this could be freed after device_name is got */
|
|
GST_OBJECT_UNLOCK (pbuf);
|
|
|
|
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);
|
|
|
|
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");
|
|
goto unlock;
|
|
}
|
|
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;
|
|
|
|
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);
|
|
return TRUE;
|
|
} else {
|
|
return FALSE;
|
|
}
|
|
}
|
|
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;
|
|
}
|
|
}
|