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990 lines
28 KiB
C
990 lines
28 KiB
C
/* GStreamer
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* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a
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* copy of this software and associated documentation files (the "Software"),
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* to deal in the Software without restriction, including without limitation
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* the rights to use, copy, modify, merge, publish, distribute, sublicense,
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* and/or sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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* DEALINGS IN THE SOFTWARE.
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*
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* Alternatively, the contents of this file may be used under the
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* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
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* which case the following provisions apply instead of the ones
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* mentioned above:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-jackaudiosrc
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* @title: jackaudiosrc
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* @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer
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*
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* A Src that inputs data from Jack ports.
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*
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* It will create N Jack ports named in_<name>_<num> where
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* <name> is the element name and <num> is starting from 1.
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* Each port corresponds to a gstreamer channel.
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*
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* The samplerate as exposed on the caps is always the same as the samplerate of
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* the jack server.
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*
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* When the #GstJackAudioSrc:connect property is set to auto, this element
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* will try to connect each input port to a random physical jack output pin.
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*
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* When the #GstJackAudioSrc:connect property is set to none, the element will
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* accept any number of output channels and will create (but not connect) an
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* input port for each channel.
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*
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* The element will generate an error when the Jack server is shut down when it
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* was PAUSED or PLAYING. This element does not support dynamic rate and buffer
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* size changes at runtime.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0
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* ]| Get audio input into gstreamer from jack.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst-i18n-plugin.h>
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#include <stdlib.h>
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#include <string.h>
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#include <gst/audio/audio.h>
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#include "gstjackaudiosrc.h"
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#include "gstjackringbuffer.h"
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#include "gstjackutil.h"
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GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug);
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#define GST_CAT_DEFAULT gst_jack_audio_src_debug
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static gboolean
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gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels)
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{
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jack_client_t *client;
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client = gst_jack_audio_client_get_client (src->client);
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/* remove ports we don't need */
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while (src->port_count > channels)
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jack_port_unregister (client, src->ports[--src->port_count]);
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/* alloc enough input ports */
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src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels);
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src->buffers = g_realloc (src->buffers, sizeof (sample_t *) * channels);
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/* create an input port for each channel */
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while (src->port_count < channels) {
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gchar *name;
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/* port names start from 1 and are local to the element */
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name =
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g_strdup_printf ("in_%s_%d", GST_ELEMENT_NAME (src),
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src->port_count + 1);
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src->ports[src->port_count] =
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jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
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JackPortIsInput, 0);
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if (src->ports[src->port_count] == NULL)
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return FALSE;
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src->port_count++;
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g_free (name);
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}
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return TRUE;
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}
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static void
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gst_jack_audio_src_free_channels (GstJackAudioSrc * src)
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{
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gint res, i = 0;
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jack_client_t *client;
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client = gst_jack_audio_client_get_client (src->client);
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/* get rid of all ports */
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while (src->port_count) {
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GST_LOG_OBJECT (src, "unregister port %d", i);
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if ((res = jack_port_unregister (client, src->ports[i++])))
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GST_DEBUG_OBJECT (src, "unregister of port failed (%d)", res);
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src->port_count--;
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}
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g_free (src->ports);
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src->ports = NULL;
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g_free (src->buffers);
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src->buffers = NULL;
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}
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/* ringbuffer abstract base class */
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static GType
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gst_jack_ring_buffer_get_type (void)
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{
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static volatile gsize ringbuffer_type = 0;
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if (g_once_init_enter (&ringbuffer_type)) {
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static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_jack_ring_buffer_class_init,
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NULL,
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NULL,
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sizeof (GstJackRingBuffer),
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0,
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(GInstanceInitFunc) gst_jack_ring_buffer_init,
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NULL
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};
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GType tmp = g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,
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"GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
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g_once_init_leave (&ringbuffer_type, tmp);
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}
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return (GType) ringbuffer_type;
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}
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static void
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gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
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{
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GstAudioRingBufferClass *gstringbuffer_class;
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gstringbuffer_class = (GstAudioRingBufferClass *) klass;
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ring_parent_class = g_type_class_peek_parent (klass);
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gstringbuffer_class->open_device =
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GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
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gstringbuffer_class->close_device =
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GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
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gstringbuffer_class->acquire =
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GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
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gstringbuffer_class->release =
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GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
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gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
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gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
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gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
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gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
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gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
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}
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/* this is the callback of jack. This should be RT-safe.
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* Writes samples from the jack input port's buffer to the gst ring buffer.
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*/
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static int
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jack_process_cb (jack_nframes_t nframes, void *arg)
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{
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GstJackAudioSrc *src;
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GstAudioRingBuffer *buf;
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gint len;
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guint8 *writeptr;
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gint writeseg;
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gint channels, i, j, flen;
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sample_t *data;
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buf = GST_AUDIO_RING_BUFFER_CAST (arg);
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src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
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/* get input buffers */
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for (i = 0; i < channels; i++)
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src->buffers[i] =
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(sample_t *) jack_port_get_buffer (src->ports[i], nframes);
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if (gst_audio_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) {
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flen = len / channels;
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/* the number of samples must be exactly the segment size */
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if (nframes * sizeof (sample_t) != flen)
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goto wrong_size;
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/* the samples in the jack input buffers have to be interleaved into the
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* ringbuffer */
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data = (sample_t *) writeptr;
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for (i = 0; i < nframes; ++i)
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for (j = 0; j < channels; ++j)
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*data++ = src->buffers[j][i];
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GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
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len / channels, channels);
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/* we wrote one segment */
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gst_audio_ring_buffer_advance (buf, 1);
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}
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return 0;
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/* ERRORS */
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wrong_size:
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{
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GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)",
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(gint) (nframes * sizeof (sample_t)), flen);
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return 1;
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}
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}
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/* we error out */
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static int
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jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
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{
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GstJackAudioSrc *src;
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GstJackRingBuffer *abuf;
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abuf = GST_JACK_RING_BUFFER_CAST (arg);
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src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
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if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
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goto not_supported;
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return 0;
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/* ERRORS */
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not_supported:
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{
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GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
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(NULL), ("Jack changed the sample rate, which is not supported"));
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return 1;
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}
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}
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/* we error out */
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static int
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jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
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{
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GstJackAudioSrc *src;
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GstJackRingBuffer *abuf;
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abuf = GST_JACK_RING_BUFFER_CAST (arg);
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src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
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if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
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goto not_supported;
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return 0;
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/* ERRORS */
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not_supported:
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{
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GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
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(NULL), ("Jack changed the buffer size, which is not supported"));
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return 1;
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}
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}
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static void
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jack_shutdown_cb (void *arg)
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{
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GstJackAudioSrc *src;
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src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
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GST_DEBUG_OBJECT (src, "shutdown");
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GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
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(NULL), ("Jack server shutdown"));
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}
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static void
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gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
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GstJackRingBufferClass * g_class)
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{
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buf->channels = -1;
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buf->buffer_size = -1;
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buf->sample_rate = -1;
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}
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/* the _open_device method should make a connection with the server
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*/
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static gboolean
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gst_jack_ring_buffer_open_device (GstAudioRingBuffer * buf)
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{
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GstJackAudioSrc *src;
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jack_status_t status = 0;
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const gchar *name;
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src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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GST_DEBUG_OBJECT (src, "open");
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if (src->client_name) {
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name = src->client_name;
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} else {
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name = g_get_application_name ();
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}
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if (!name)
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name = "GStreamer";
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src->client = gst_jack_audio_client_new (name, src->server,
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src->jclient,
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GST_JACK_CLIENT_SOURCE,
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jack_shutdown_cb,
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jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
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if (src->client == NULL)
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goto could_not_open;
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GST_DEBUG_OBJECT (src, "opened");
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return TRUE;
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/* ERRORS */
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could_not_open:
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{
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if (status & (JackServerFailed | JackFailure)) {
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GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
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(_("Jack server not found")),
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("Cannot connect to the Jack server (status %d)", status));
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} else {
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GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
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(NULL), ("Jack client open error (status %d)", status));
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}
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return FALSE;
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}
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}
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/* close the connection with the server
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*/
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static gboolean
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gst_jack_ring_buffer_close_device (GstAudioRingBuffer * buf)
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{
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GstJackAudioSrc *src;
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src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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GST_DEBUG_OBJECT (src, "close");
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gst_jack_audio_src_free_channels (src);
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gst_jack_audio_client_free (src->client);
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src->client = NULL;
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return TRUE;
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}
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/* allocate a buffer and setup resources to process the audio samples of
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* the format as specified in @spec.
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*
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* We allocate N jack ports, one for each channel. If we are asked to
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* automatically make a connection with physical ports, we connect as many
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* ports as there are physical ports, leaving leftover ports unconnected.
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*
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* It is assumed that samplerate and number of channels are acceptable since our
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* getcaps method will always provide correct values. If unacceptable caps are
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* received for some reason, we fail here.
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*/
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static gboolean
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gst_jack_ring_buffer_acquire (GstAudioRingBuffer * buf,
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GstAudioRingBufferSpec * spec)
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{
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GstJackAudioSrc *src;
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GstJackRingBuffer *abuf;
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const char **ports;
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gint sample_rate, buffer_size;
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gint i, bpf, rate, channels, res;
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jack_client_t *client;
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src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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abuf = GST_JACK_RING_BUFFER_CAST (buf);
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GST_DEBUG_OBJECT (src, "acquire");
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client = gst_jack_audio_client_get_client (src->client);
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rate = GST_AUDIO_INFO_RATE (&spec->info);
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/* sample rate must be that of the server */
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sample_rate = jack_get_sample_rate (client);
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if (sample_rate != rate)
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goto wrong_samplerate;
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bpf = GST_AUDIO_INFO_BPF (&spec->info);
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channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
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if (!gst_jack_audio_src_allocate_channels (src, channels))
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goto out_of_ports;
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gst_jack_set_layout (buf, spec);
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buffer_size = jack_get_buffer_size (client);
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/* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
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* for all channels */
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spec->segsize = buffer_size * sizeof (gfloat) * channels;
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spec->latency_time = gst_util_uint64_scale (spec->segsize,
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(GST_SECOND / GST_USECOND), rate * bpf);
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/* segtotal based on buffer-time latency */
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spec->segtotal = spec->buffer_time / spec->latency_time;
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if (spec->segtotal < 2) {
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spec->segtotal = 2;
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spec->buffer_time = spec->latency_time * spec->segtotal;
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}
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GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec",
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spec->buffer_time);
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GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec",
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spec->latency_time);
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GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d",
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buffer_size, spec->segsize, spec->segtotal);
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/* allocate the ringbuffer memory now */
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buf->size = spec->segtotal * spec->segsize;
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buf->memory = g_malloc0 (buf->size);
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if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
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goto could_not_activate;
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/* if we need to automatically connect the ports, do so now. We must do this
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* after activating the client. */
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if (src->connect == GST_JACK_CONNECT_AUTO
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|| src->connect == GST_JACK_CONNECT_AUTO_FORCED) {
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/* find all the physical output ports. A physical output port is a port
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* associated with a hardware device. Someone needs connect to a physical
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* port in order to capture something. */
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if (src->port_pattern == NULL) {
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ports = jack_get_ports (client, NULL, NULL,
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JackPortIsPhysical | JackPortIsOutput);
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} else {
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ports = jack_get_ports (client, src->port_pattern, NULL,
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JackPortIsOutput);
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}
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if (ports == NULL) {
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/* no ports? fine then we don't do anything except for posting a warning
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* message. */
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GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
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("No physical output ports found, leaving ports unconnected"));
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goto done;
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}
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for (i = 0; i < channels; i++) {
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/* stop when all output ports are exhausted */
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if (ports[i] == NULL) {
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/* post a warning that we could not connect all ports */
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GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
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("No more physical ports, leaving some ports unconnected"));
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break;
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}
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GST_DEBUG_OBJECT (src, "try connecting to %s",
|
|
jack_port_name (src->ports[i]));
|
|
|
|
/* connect the physical port to a port */
|
|
res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
|
|
if (res != 0 && res != EEXIST)
|
|
goto cannot_connect;
|
|
}
|
|
jack_free (ports);
|
|
}
|
|
done:
|
|
|
|
abuf->sample_rate = sample_rate;
|
|
abuf->buffer_size = buffer_size;
|
|
abuf->channels = channels;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
wrong_samplerate:
|
|
{
|
|
GST_ELEMENT_ERROR (src, 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 (src, RESOURCE, SETTINGS, (NULL),
|
|
("Cannot allocate more Jack ports"));
|
|
return FALSE;
|
|
}
|
|
could_not_activate:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
("Could not activate client (%d:%s)", res, g_strerror (res)));
|
|
return FALSE;
|
|
}
|
|
cannot_connect:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
|
|
("Could not connect input ports to physical ports (%d:%s)",
|
|
res, g_strerror (res)));
|
|
jack_free (ports);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* function is called with LOCK */
|
|
static gboolean
|
|
gst_jack_ring_buffer_release (GstAudioRingBuffer * buf)
|
|
{
|
|
GstJackAudioSrc *src;
|
|
GstJackRingBuffer *abuf;
|
|
gint res;
|
|
|
|
abuf = GST_JACK_RING_BUFFER_CAST (buf);
|
|
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
|
|
GST_DEBUG_OBJECT (src, "release");
|
|
|
|
if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
|
|
/* we only warn, this means the server is probably shut down and the client
|
|
* is gone anyway. */
|
|
GST_ELEMENT_WARNING (src, 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)
|
|
{
|
|
GstJackAudioSrc *src;
|
|
|
|
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
|
|
GST_DEBUG_OBJECT (src, "start");
|
|
|
|
if (src->transport & GST_JACK_TRANSPORT_MASTER) {
|
|
jack_client_t *client;
|
|
|
|
client = gst_jack_audio_client_get_client (src->client);
|
|
jack_transport_start (client);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
|
|
{
|
|
GstJackAudioSrc *src;
|
|
|
|
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
|
|
GST_DEBUG_OBJECT (src, "pause");
|
|
|
|
if (src->transport & GST_JACK_TRANSPORT_MASTER) {
|
|
jack_client_t *client;
|
|
|
|
client = gst_jack_audio_client_get_client (src->client);
|
|
jack_transport_stop (client);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
|
|
{
|
|
GstJackAudioSrc *src;
|
|
|
|
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
|
|
GST_DEBUG_OBJECT (src, "stop");
|
|
|
|
if (src->transport & GST_JACK_TRANSPORT_MASTER) {
|
|
jack_client_t *client;
|
|
|
|
client = gst_jack_audio_client_get_client (src->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)
|
|
{
|
|
GstJackAudioSrc *src;
|
|
guint i, res = 0;
|
|
jack_latency_range_t range;
|
|
|
|
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
|
|
for (i = 0; i < src->port_count; i++) {
|
|
jack_port_get_latency_range (src->ports[i], JackCaptureLatency, &range);
|
|
if (range.max > res)
|
|
res = range.max;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src, "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)
|
|
{
|
|
GstJackAudioSrc *src;
|
|
guint i, res = 0;
|
|
guint latency;
|
|
jack_client_t *client;
|
|
|
|
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
|
|
|
|
client = gst_jack_audio_client_get_client (src->client);
|
|
|
|
for (i = 0; i < src->port_count; i++) {
|
|
latency = jack_port_get_total_latency (client, src->ports[i]);
|
|
if (latency > res)
|
|
res = latency;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src, "delay %u", res);
|
|
|
|
return res;
|
|
}
|
|
#endif
|
|
|
|
/* Audiosrc signals and args */
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
#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
|
|
#define DEFAULT_PROP_PORT_PATTERN NULL
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_CONNECT,
|
|
PROP_SERVER,
|
|
PROP_CLIENT,
|
|
PROP_CLIENT_NAME,
|
|
PROP_PORT_PATTERN,
|
|
PROP_TRANSPORT,
|
|
PROP_LAST
|
|
};
|
|
|
|
/* the capabilities of the inputs and outputs.
|
|
*
|
|
* describe the real formats here.
|
|
*/
|
|
|
|
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
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 ]")
|
|
);
|
|
|
|
#define gst_jack_audio_src_parent_class parent_class
|
|
G_DEFINE_TYPE (GstJackAudioSrc, gst_jack_audio_src, GST_TYPE_AUDIO_BASE_SRC);
|
|
|
|
static void gst_jack_audio_src_dispose (GObject * object);
|
|
static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec);
|
|
static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec);
|
|
|
|
static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc,
|
|
GstCaps * filter);
|
|
static GstAudioRingBuffer *gst_jack_audio_src_create_ringbuffer (GstAudioBaseSrc
|
|
* src);
|
|
|
|
/* GObject vmethod implementations */
|
|
|
|
/* initialize the jack_audio_src's class */
|
|
static void
|
|
gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
GstBaseSrcClass *gstbasesrc_class;
|
|
GstAudioBaseSrcClass *gstaudiobasesrc_class;
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_src_debug, "jacksrc", 0,
|
|
"jacksrc element");
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
gstbasesrc_class = (GstBaseSrcClass *) klass;
|
|
gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
|
|
|
|
gobject_class->dispose = gst_jack_audio_src_dispose;
|
|
gobject_class->set_property = gst_jack_audio_src_set_property;
|
|
gobject_class->get_property = gst_jack_audio_src_get_property;
|
|
|
|
g_object_class_install_property (gobject_class, PROP_CONNECT,
|
|
g_param_spec_enum ("connect", "Connect",
|
|
"Specify how the input 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));
|
|
|
|
/**
|
|
* GstJackAudioSrc: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));
|
|
/**
|
|
* GstJackAudioSrc:port-pattern
|
|
*
|
|
* autoconnect to ports matching pattern, when NULL connect to physical ports
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_PORT_PATTERN,
|
|
g_param_spec_string ("port-pattern", "port pattern",
|
|
"A pattern to select which ports to connect to (NULL = first physical ports)",
|
|
DEFAULT_PROP_PORT_PATTERN,
|
|
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_add_static_pad_template (gstelement_class, &src_factory);
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class,
|
|
"Audio Source (Jack)", "Source/Audio",
|
|
"Captures audio from a JACK server",
|
|
"Tristan Matthews <tristan@sat.qc.ca>");
|
|
|
|
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
|
|
gstaudiobasesrc_class->create_ringbuffer =
|
|
GST_DEBUG_FUNCPTR (gst_jack_audio_src_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_src_init (GstJackAudioSrc * src)
|
|
{
|
|
//gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
|
|
src->connect = DEFAULT_PROP_CONNECT;
|
|
src->server = g_strdup (DEFAULT_PROP_SERVER);
|
|
src->jclient = NULL;
|
|
src->ports = NULL;
|
|
src->port_count = 0;
|
|
src->buffers = NULL;
|
|
src->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
|
|
src->transport = DEFAULT_PROP_TRANSPORT;
|
|
}
|
|
|
|
static void
|
|
gst_jack_audio_src_dispose (GObject * object)
|
|
{
|
|
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
|
|
|
|
gst_caps_replace (&src->caps, NULL);
|
|
|
|
if (src->client_name != NULL) {
|
|
g_free (src->client_name);
|
|
src->client_name = NULL;
|
|
}
|
|
|
|
if (src->port_pattern != NULL) {
|
|
g_free (src->port_pattern);
|
|
src->port_pattern = NULL;
|
|
}
|
|
|
|
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
}
|
|
|
|
static void
|
|
gst_jack_audio_src_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_CLIENT_NAME:
|
|
g_free (src->client_name);
|
|
src->client_name = g_value_dup_string (value);
|
|
break;
|
|
case PROP_PORT_PATTERN:
|
|
g_free (src->port_pattern);
|
|
src->port_pattern = g_value_dup_string (value);
|
|
break;
|
|
case PROP_CONNECT:
|
|
src->connect = g_value_get_enum (value);
|
|
break;
|
|
case PROP_SERVER:
|
|
g_free (src->server);
|
|
src->server = g_value_dup_string (value);
|
|
break;
|
|
case PROP_CLIENT:
|
|
if (GST_STATE (src) == GST_STATE_NULL ||
|
|
GST_STATE (src) == GST_STATE_READY) {
|
|
src->jclient = g_value_get_boxed (value);
|
|
}
|
|
break;
|
|
case PROP_TRANSPORT:
|
|
src->transport = g_value_get_flags (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_jack_audio_src_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_CLIENT_NAME:
|
|
g_value_set_string (value, src->client_name);
|
|
break;
|
|
case PROP_PORT_PATTERN:
|
|
g_value_set_string (value, src->port_pattern);
|
|
break;
|
|
case PROP_CONNECT:
|
|
g_value_set_enum (value, src->connect);
|
|
break;
|
|
case PROP_SERVER:
|
|
g_value_set_string (value, src->server);
|
|
break;
|
|
case PROP_CLIENT:
|
|
g_value_set_boxed (value, src->jclient);
|
|
break;
|
|
case PROP_TRANSPORT:
|
|
g_value_set_flags (value, src->transport);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_jack_audio_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
|
|
{
|
|
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
|
|
const char **ports;
|
|
gint min, max;
|
|
gint rate;
|
|
jack_client_t *client;
|
|
|
|
if (src->client == NULL)
|
|
goto no_client;
|
|
|
|
client = gst_jack_audio_client_get_client (src->client);
|
|
|
|
if (src->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 | JackPortIsOutput);
|
|
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 (src, "got %d-%d ports, samplerate: %d", min, max, rate);
|
|
|
|
if (!src->caps) {
|
|
src->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 (src, "returning caps %" GST_PTR_FORMAT, src->caps);
|
|
|
|
return gst_caps_ref (src->caps);
|
|
|
|
/* ERRORS */
|
|
no_client:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "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_src_create_ringbuffer (GstAudioBaseSrc * src)
|
|
{
|
|
GstAudioRingBuffer *buffer;
|
|
|
|
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
|
|
GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);
|
|
|
|
return buffer;
|
|
}
|