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Diffstat (limited to 'ext/jack/gstjackaudiosink.c')
| -rwxr-xr-x | ext/jack/gstjackaudiosink.c | 941 |
1 files changed, 941 insertions, 0 deletions
diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c new file mode 100755 index 0000000..3a83567 --- /dev/null +++ b/ext/jack/gstjackaudiosink.c @@ -0,0 +1,941 @@ +/* GStreamer + * Copyright (C) 2006 Wim Taymans <wim@fluendo.com> + * + * gstjackaudiosink.c: jack audio sink implementation + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, + * Boston, MA 02110-1301, USA. + */ + +/** + * SECTION:element-jackaudiosink + * @see_also: #GstAudioBaseSink, #GstAudioRingBuffer + * + * A Sink that outputs data to Jack ports. + * + * It will create N Jack ports named out_<name>_<num> where + * <name> is the element name and <num> is starting from 1. + * Each port corresponds to a gstreamer channel. + * + * The samplerate as exposed on the caps is always the same as the samplerate of + * the jack server. + * + * When the #GstJackAudioSink:connect property is set to auto, this element + * will try to connect each output port to a random physical jack input pin. In + * this mode, the sink will expose the number of physical channels on its pad + * caps. + * + * When the #GstJackAudioSink:connect property is set to none, the element will + * accept any number of input channels and will create (but not connect) an + * output port for each channel. + * + * The element will generate an error when the Jack server is shut down when it + * was PAUSED or PLAYING. This element does not support dynamic rate and buffer + * size changes at runtime. + * + * <refsect2> + * <title>Example launch line</title> + * |[ + * gst-launch-1.0 audiotestsrc ! jackaudiosink + * ]| Play a sine wave to using jack. + * </refsect2> + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <gst/gst-i18n-plugin.h> +#include <stdlib.h> +#include <string.h> +#include <gst/audio/audio.h> + +#include "gstjackaudiosink.h" +#include "gstjackringbuffer.h" + +GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug); +#define GST_CAT_DEFAULT gst_jack_audio_sink_debug + +static gboolean +gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels) +{ + jack_client_t *client; + + client = gst_jack_audio_client_get_client (sink->client); + + /* remove ports we don't need */ + while (sink->port_count > channels) { + jack_port_unregister (client, sink->ports[--sink->port_count]); + } + + /* alloc enough output ports */ + sink->ports = g_realloc (sink->ports, sizeof (jack_port_t *) * channels); + sink->buffers = g_realloc (sink->buffers, sizeof (sample_t *) * channels); + + /* create an output port for each channel */ + while (sink->port_count < channels) { + gchar *name; + + /* port names start from 1 and are local to the element */ + name = + g_strdup_printf ("out_%s_%d", GST_ELEMENT_NAME (sink), + sink->port_count + 1); + sink->ports[sink->port_count] = + jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE, + JackPortIsOutput, 0); + if (sink->ports[sink->port_count] == NULL) + return FALSE; + + sink->port_count++; + + g_free (name); + } + return TRUE; +} + +static void +gst_jack_audio_sink_free_channels (GstJackAudioSink * sink) +{ + gint res, i = 0; + jack_client_t *client; + + client = gst_jack_audio_client_get_client (sink->client); + + /* get rid of all ports */ + while (sink->port_count) { + GST_LOG_OBJECT (sink, "unregister port %d", i); + if ((res = jack_port_unregister (client, sink->ports[i++]))) { + GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res); + } + sink->port_count--; + } + g_free (sink->ports); + sink->ports = NULL; + g_free (sink->buffers); + sink->buffers = NULL; +} + +/* ringbuffer abstract base class */ +static GType +gst_jack_ring_buffer_get_type (void) +{ + static volatile gsize ringbuffer_type = 0; + + if (g_once_init_enter (&ringbuffer_type)) { + static const GTypeInfo ringbuffer_info = { + sizeof (GstJackRingBufferClass), + NULL, + NULL, + (GClassInitFunc) gst_jack_ring_buffer_class_init, + NULL, + NULL, + sizeof (GstJackRingBuffer), + 0, + (GInstanceInitFunc) gst_jack_ring_buffer_init, + NULL + }; + GType tmp = g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER, + "GstJackAudioSinkRingBuffer", &ringbuffer_info, 0); + g_once_init_leave (&ringbuffer_type, tmp); + } + + return (GType) ringbuffer_type; +} + +static void +gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass) +{ + GstAudioRingBufferClass *gstringbuffer_class; + + gstringbuffer_class = (GstAudioRingBufferClass *) klass; + + ring_parent_class = g_type_class_peek_parent (klass); + + gstringbuffer_class->open_device = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device); + gstringbuffer_class->close_device = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device); + gstringbuffer_class->acquire = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire); + gstringbuffer_class->release = + GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release); + gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start); + gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause); + gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start); + gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop); + + gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay); +} + +/* this is the callback of jack. This should RT-safe. + */ +static int +jack_process_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioSink *sink; + GstAudioRingBuffer *buf; + gint readseg, len; + guint8 *readptr; + gint i, j, flen, channels; + sample_t *data; + + buf = GST_AUDIO_RING_BUFFER_CAST (arg); + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info); + + /* get target buffers */ + for (i = 0; i < channels; i++) { + sink->buffers[i] = + (sample_t *) jack_port_get_buffer (sink->ports[i], nframes); + } + + if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) { + flen = len / channels; + + /* the number of samples must be exactly the segment size */ + if (nframes * sizeof (sample_t) != flen) + goto wrong_size; + + GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels", + nframes, readptr, flen, channels); + data = (sample_t *) readptr; + + /* the samples in the ringbuffer have the channels interleaved, we need to + * deinterleave into the jack target buffers */ + for (i = 0; i < nframes; i++) { + for (j = 0; j < channels; j++) { + sink->buffers[j][i] = *data++; + } + } + + /* clear written samples in the ringbuffer */ + gst_audio_ring_buffer_clear (buf, readseg); + + /* we wrote one segment */ + gst_audio_ring_buffer_advance (buf, 1); + } else { + GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes); + /* We are not allowed to read from the ringbuffer, write silence to all + * jack output buffers */ + for (i = 0; i < channels; i++) { + memset (sink->buffers[i], 0, nframes * sizeof (sample_t)); + } + } + return 0; + + /* ERRORS */ +wrong_size: + { + GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)", + (gint) (nframes * sizeof (sample_t)), flen); + return 1; + } +} + +/* we error out */ +static int +jack_sample_rate_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioSink *sink; + GstJackRingBuffer *abuf; + + abuf = GST_JACK_RING_BUFFER_CAST (arg); + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg)); + + if (abuf->sample_rate != -1 && abuf->sample_rate != nframes) + goto not_supported; + + return 0; + + /* ERRORS */ +not_supported: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, + (NULL), ("Jack changed the sample rate, which is not supported")); + return 1; + } +} + +/* we error out */ +static int +jack_buffer_size_cb (jack_nframes_t nframes, void *arg) +{ + GstJackAudioSink *sink; + GstJackRingBuffer *abuf; + + abuf = GST_JACK_RING_BUFFER_CAST (arg); + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg)); + + if (abuf->buffer_size != -1 && abuf->buffer_size != nframes) + goto not_supported; + + return 0; + + /* ERRORS */ +not_supported: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, + (NULL), ("Jack changed the buffer size, which is not supported")); + return 1; + } +} + +static void +jack_shutdown_cb (void *arg) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg)); + + GST_DEBUG_OBJECT (sink, "shutdown"); + + GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, + (NULL), ("Jack server shutdown")); +} + +static void +gst_jack_ring_buffer_init (GstJackRingBuffer * buf, + GstJackRingBufferClass * g_class) +{ + buf->channels = -1; + buf->buffer_size = -1; + buf->sample_rate = -1; +} + +/* the _open_device method should make a connection with the server + */ +static gboolean +gst_jack_ring_buffer_open_device (GstAudioRingBuffer * buf) +{ + GstJackAudioSink *sink; + jack_status_t status = 0; + const gchar *name; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "open"); + + if (sink->client_name) { + name = sink->client_name; + } else { + name = g_get_application_name (); + } + if (!name) + name = "GStreamer"; + + sink->client = gst_jack_audio_client_new (name, sink->server, + sink->jclient, + GST_JACK_CLIENT_SINK, + jack_shutdown_cb, + jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status); + if (sink->client == NULL) + goto could_not_open; + + GST_DEBUG_OBJECT (sink, "opened"); + + return TRUE; + + /* ERRORS */ +could_not_open: + { + if (status & JackServerFailed) { + GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND, + (_("Jack server not found")), + ("Cannot connect to the Jack server (status %d)", status)); + } else { + GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, + (NULL), ("Jack client open error (status %d)", status)); + } + return FALSE; + } +} + +/* close the connection with the server + */ +static gboolean +gst_jack_ring_buffer_close_device (GstAudioRingBuffer * buf) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "close"); + + gst_jack_audio_sink_free_channels (sink); + gst_jack_audio_client_free (sink->client); + sink->client = NULL; + + return TRUE; +} + +/* allocate a buffer and setup resources to process the audio samples of + * the format as specified in @spec. + * + * We allocate N jack ports, one for each channel. If we are asked to + * automatically make a connection with physical ports, we connect as many + * ports as there are physical ports, leaving leftover ports unconnected. + * + * It is assumed that samplerate and number of channels are acceptable since our + * getcaps method will always provide correct values. If unacceptable caps are + * received for some reason, we fail here. + */ +static gboolean +gst_jack_ring_buffer_acquire (GstAudioRingBuffer * buf, + GstAudioRingBufferSpec * spec) +{ + GstJackAudioSink *sink; + GstJackRingBuffer *abuf; + const char **ports; + gint sample_rate, buffer_size; + gint i, rate, bpf, channels, res; + jack_client_t *client; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + abuf = GST_JACK_RING_BUFFER_CAST (buf); + + GST_DEBUG_OBJECT (sink, "acquire"); + + client = gst_jack_audio_client_get_client (sink->client); + + rate = GST_AUDIO_INFO_RATE (&spec->info); + + /* sample rate must be that of the server */ + sample_rate = jack_get_sample_rate (client); + if (sample_rate != rate) + goto wrong_samplerate; + + channels = GST_AUDIO_INFO_CHANNELS (&spec->info); + bpf = GST_AUDIO_INFO_BPF (&spec->info); + + if (!gst_jack_audio_sink_allocate_channels (sink, channels)) + goto out_of_ports; + + buffer_size = jack_get_buffer_size (client); + + /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats + * for all channels */ + spec->segsize = buffer_size * sizeof (gfloat) * channels; + spec->latency_time = gst_util_uint64_scale (spec->segsize, + (GST_SECOND / GST_USECOND), rate * bpf); + /* segtotal based on buffer-time latency */ + spec->segtotal = spec->buffer_time / spec->latency_time; + if (spec->segtotal < 2) { + spec->segtotal = 2; + spec->buffer_time = spec->latency_time * spec->segtotal; + } + + GST_DEBUG_OBJECT (sink, "buffer time: %" G_GINT64_FORMAT " usec", + spec->buffer_time); + GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec", + spec->latency_time); + GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d", + buffer_size, spec->segsize, spec->segtotal); + + /* allocate the ringbuffer memory now */ + buf->size = spec->segtotal * spec->segsize; + buf->memory = g_malloc0 (buf->size); + + if ((res = gst_jack_audio_client_set_active (sink->client, TRUE))) + goto could_not_activate; + + /* if we need to automatically connect the ports, do so now. We must do this + * after activating the client. */ + if (sink->connect == GST_JACK_CONNECT_AUTO + || sink->connect == GST_JACK_CONNECT_AUTO_FORCED) { + /* find all the physical input ports. A physical input port is a port + * associated with a hardware device. Someone needs connect to a physical + * port in order to hear something. */ + ports = jack_get_ports (client, NULL, NULL, + JackPortIsPhysical | JackPortIsInput); + if (ports == NULL) { + /* no ports? fine then we don't do anything except for posting a warning + * message. */ + GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL), + ("No physical input ports found, leaving ports unconnected")); + goto done; + } + + for (i = 0; i < channels; i++) { + /* stop when all input ports are exhausted */ + if (ports[i] == NULL) { + /* post a warning that we could not connect all ports */ + GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL), + ("No more physical ports, leaving some ports unconnected")); + break; + } + GST_DEBUG_OBJECT (sink, "try connecting to %s", + jack_port_name (sink->ports[i])); + /* connect the port to a physical port */ + res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]); + if (res != 0 && res != EEXIST) + goto cannot_connect; + } + free (ports); + } +done: + + abuf->sample_rate = sample_rate; + abuf->buffer_size = buffer_size; + abuf->channels = channels; + + return TRUE; + + /* ERRORS */ +wrong_samplerate: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), + ("Wrong samplerate, server is running at %d and we received %d", + sample_rate, rate)); + return FALSE; + } +out_of_ports: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), + ("Cannot allocate more Jack ports")); + return FALSE; + } +could_not_activate: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), + ("Could not activate client (%d:%s)", res, g_strerror (res))); + return FALSE; + } +cannot_connect: + { + GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL), + ("Could not connect output ports to physical ports (%d:%s)", + res, g_strerror (res))); + free (ports); + return FALSE; + } +} + +/* function is called with LOCK */ +static gboolean +gst_jack_ring_buffer_release (GstAudioRingBuffer * buf) +{ + GstJackAudioSink *sink; + GstJackRingBuffer *abuf; + gint res; + + abuf = GST_JACK_RING_BUFFER_CAST (buf); + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "release"); + + if ((res = gst_jack_audio_client_set_active (sink->client, FALSE))) { + /* we only warn, this means the server is probably shut down and the client + * is gone anyway. */ + GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL), + ("Could not deactivate Jack client (%d)", res)); + } + + abuf->channels = -1; + abuf->buffer_size = -1; + abuf->sample_rate = -1; + + /* free the buffer */ + g_free (buf->memory); + buf->memory = NULL; + + return TRUE; +} + +static gboolean +gst_jack_ring_buffer_start (GstAudioRingBuffer * buf) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "start"); + + if (sink->transport & GST_JACK_TRANSPORT_MASTER) { + jack_client_t *client; + + client = gst_jack_audio_client_get_client (sink->client); + jack_transport_start (client); + } + + return TRUE; +} + +static gboolean +gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "pause"); + + if (sink->transport & GST_JACK_TRANSPORT_MASTER) { + jack_client_t *client; + + client = gst_jack_audio_client_get_client (sink->client); + jack_transport_stop (client); + } + + return TRUE; +} + +static gboolean +gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + GST_DEBUG_OBJECT (sink, "stop"); + + if (sink->transport & GST_JACK_TRANSPORT_MASTER) { + jack_client_t *client; + + client = gst_jack_audio_client_get_client (sink->client); + jack_transport_stop (client); + } + + return TRUE; +} + +#if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7) +static guint +gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf) +{ + GstJackAudioSink *sink; + guint i, res = 0; + jack_latency_range_t range; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + + for (i = 0; i < sink->port_count; i++) { + jack_port_get_latency_range (sink->ports[i], JackPlaybackLatency, &range); + if (range.max > res) + res = range.max; + } + + GST_LOG_OBJECT (sink, "delay %u", res); + + return res; +} +#else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */ +static guint +gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf) +{ + GstJackAudioSink *sink; + guint i, res = 0; + guint latency; + jack_client_t *client; + + sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); + client = gst_jack_audio_client_get_client (sink->client); + + for (i = 0; i < sink->port_count; i++) { + latency = jack_port_get_total_latency (client, sink->ports[i]); + if (latency > res) + res = latency; + } + + GST_LOG_OBJECT (sink, "delay %u", res); + + return res; +} +#endif + +static GstStaticPadTemplate jackaudiosink_sink_factory = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) " GST_JACK_FORMAT_STR ", " + "layout = (string) interleaved, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") + ); + +/* AudioSink signals and args */ +enum +{ + /* FILL ME */ + SIGNAL_LAST +}; + +#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO +#define DEFAULT_PROP_SERVER NULL +#define DEFAULT_PROP_CLIENT_NAME NULL +#define DEFAULT_PROP_TRANSPORT GST_JACK_TRANSPORT_AUTONOMOUS + +enum +{ + PROP_0, + PROP_CONNECT, + PROP_SERVER, + PROP_CLIENT, + PROP_CLIENT_NAME, + PROP_TRANSPORT, + PROP_LAST +}; + +#define gst_jack_audio_sink_parent_class parent_class +G_DEFINE_TYPE (GstJackAudioSink, gst_jack_audio_sink, GST_TYPE_AUDIO_BASE_SINK); + +static void gst_jack_audio_sink_dispose (GObject * object); +static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink, + GstCaps * filter); +static GstAudioRingBuffer + * gst_jack_audio_sink_create_ringbuffer (GstAudioBaseSink * sink); + +static void +gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseSinkClass *gstbasesink_class; + GstAudioBaseSinkClass *gstaudiobasesink_class; + + GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, + "jacksink element"); + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasesink_class = (GstBaseSinkClass *) klass; + gstaudiobasesink_class = (GstAudioBaseSinkClass *) klass; + + gobject_class->dispose = gst_jack_audio_sink_dispose; + gobject_class->get_property = gst_jack_audio_sink_get_property; + gobject_class->set_property = gst_jack_audio_sink_set_property; + + g_object_class_install_property (gobject_class, PROP_CONNECT, + g_param_spec_enum ("connect", "Connect", + "Specify how the output ports will be connected", + GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_SERVER, + g_param_spec_string ("server", "Server", + "The Jack server to connect to (NULL = default)", + DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + /** + * GstJackAudioSink:client-name: + * + * The client name to use. + */ + g_object_class_install_property (gobject_class, PROP_CLIENT_NAME, + g_param_spec_string ("client-name", "Client name", + "The client name of the Jack instance (NULL = default)", + DEFAULT_PROP_CLIENT_NAME, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_CLIENT, + g_param_spec_boxed ("client", "JackClient", "Handle for jack client", + GST_TYPE_JACK_CLIENT, + GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | + G_PARAM_STATIC_STRINGS)); + + /** + * GstJackAudioSink:transport: + * + * The jack transport behaviour for the client. + */ + g_object_class_install_property (gobject_class, PROP_TRANSPORT, + g_param_spec_flags ("transport", "Transport mode", + "Jack transport behaviour of the client", + GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + gst_element_class_set_static_metadata (gstelement_class, "Audio Sink (Jack)", + "Sink/Audio", "Output audio to a JACK server", + "Wim Taymans <wim.taymans@gmail.com>"); + + gst_element_class_add_pad_template (gstelement_class, + gst_static_pad_template_get (&jackaudiosink_sink_factory)); + + gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps); + + gstaudiobasesink_class->create_ringbuffer = + GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer); + + /* ref class from a thread-safe context to work around missing bit of + * thread-safety in GObject */ + g_type_class_ref (GST_TYPE_JACK_RING_BUFFER); + + gst_jack_audio_client_init (); +} + +static void +gst_jack_audio_sink_init (GstJackAudioSink * sink) +{ + sink->connect = DEFAULT_PROP_CONNECT; + sink->server = g_strdup (DEFAULT_PROP_SERVER); + sink->jclient = NULL; + sink->ports = NULL; + sink->port_count = 0; + sink->buffers = NULL; + sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME); + sink->transport = DEFAULT_PROP_TRANSPORT; +} + +static void +gst_jack_audio_sink_dispose (GObject * object) +{ + GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object); + + gst_caps_replace (&sink->caps, NULL); + + if (sink->client_name != NULL) { + g_free (sink->client_name); + sink->client_name = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_jack_audio_sink_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (object); + + switch (prop_id) { + case PROP_CLIENT_NAME: + g_free (sink->client_name); + sink->client_name = g_value_dup_string (value); + break; + case PROP_CONNECT: + sink->connect = g_value_get_enum (value); + break; + case PROP_SERVER: + g_free (sink->server); + sink->server = g_value_dup_string (value); + break; + case PROP_CLIENT: + if (GST_STATE (sink) == GST_STATE_NULL || + GST_STATE (sink) == GST_STATE_READY) { + sink->jclient = g_value_get_boxed (value); + } + break; + case PROP_TRANSPORT: + sink->transport = g_value_get_flags (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_jack_audio_sink_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstJackAudioSink *sink; + + sink = GST_JACK_AUDIO_SINK (object); + + switch (prop_id) { + case PROP_CLIENT_NAME: + g_value_set_string (value, sink->client_name); + break; + case PROP_CONNECT: + g_value_set_enum (value, sink->connect); + break; + case PROP_SERVER: + g_value_set_string (value, sink->server); + break; + case PROP_CLIENT: + g_value_set_boxed (value, sink->jclient); + break; + case PROP_TRANSPORT: + g_value_set_flags (value, sink->transport); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstCaps * +gst_jack_audio_sink_getcaps (GstBaseSink * bsink, GstCaps * filter) +{ + GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink); + const char **ports; + gint min, max; + gint rate; + jack_client_t *client; + + if (sink->client == NULL) + goto no_client; + + client = gst_jack_audio_client_get_client (sink->client); + + if (sink->connect == GST_JACK_CONNECT_AUTO) { + /* get a port count, this is the number of channels we can automatically + * connect. */ + ports = jack_get_ports (client, NULL, NULL, + JackPortIsPhysical | JackPortIsInput); + max = 0; + if (ports != NULL) { + for (; ports[max]; max++); + free (ports); + } else + max = 0; + } else { + /* we allow any number of pads, something else is going to connect the + * pads. */ + max = G_MAXINT; + } + min = MIN (1, max); + + rate = jack_get_sample_rate (client); + + GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate); + + if (!sink->caps) { + sink->caps = gst_caps_new_simple ("audio/x-raw", + "format", G_TYPE_STRING, GST_JACK_FORMAT_STR, + "layout", G_TYPE_STRING, "interleaved", + "rate", G_TYPE_INT, rate, + "channels", GST_TYPE_INT_RANGE, min, max, NULL); + } + GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, sink->caps); + + return gst_caps_ref (sink->caps); + + /* ERRORS */ +no_client: + { + GST_DEBUG_OBJECT (sink, "device not open, using template caps"); + /* base class will get template caps for us when we return NULL */ + return NULL; + } +} + +static GstAudioRingBuffer * +gst_jack_audio_sink_create_ringbuffer (GstAudioBaseSink * sink) +{ + GstAudioRingBuffer *buffer; + + buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL); + GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer); + + return buffer; +} |