From a197901b82c22db9d8441eadba8063151c994b22 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 2 Jan 2011 15:11:52 +0000 Subject: [PATCH] jack: move plugin to gst-plugins-good https://bugzilla.gnome.org/show_bug.cgi?id=621929 --- LICENSE_readme | 2 - Makefile.am | 4 + configure.ac | 11 - docs/plugins/Makefile.am | 2 - .../plugins/gst-plugins-bad-plugins-docs.sgml | 3 - .../gst-plugins-bad-plugins-sections.txt | 30 - docs/plugins/gst-plugins-bad-plugins.args | 86 +- .../plugins/gst-plugins-bad-plugins.hierarchy | 5 - docs/plugins/inspect/plugin-jack.xml | 43 - ext/Makefile.am | 8 - ext/jack/.gitignore | 1 - ext/jack/Makefile.am | 12 - ext/jack/README | 4 - ext/jack/gstjack.c | 95 -- ext/jack/gstjack.h | 55 -- ext/jack/gstjackaudioclient.c | 525 ----------- ext/jack/gstjackaudioclient.h | 59 -- ext/jack/gstjackaudiosink.c | 852 ----------------- ext/jack/gstjackaudiosink.h | 78 -- ext/jack/gstjackaudiosrc.c | 874 ------------------ ext/jack/gstjackaudiosrc.h | 97 -- ext/jack/gstjackringbuffer.h | 88 -- ext/jack/gstjackutil.c | 114 --- ext/jack/gstjackutil.h | 30 - gst-plugins-bad.spec.in | 1 - tests/examples/Makefile.am | 9 +- tests/examples/jack/Makefile.am | 6 - tests/examples/jack/jack_client.c | 79 -- 28 files changed, 19 insertions(+), 3154 deletions(-) delete mode 100644 docs/plugins/inspect/plugin-jack.xml delete mode 100644 ext/jack/.gitignore delete mode 100644 ext/jack/Makefile.am delete mode 100644 ext/jack/README delete mode 100644 ext/jack/gstjack.c delete mode 100644 ext/jack/gstjack.h delete mode 100644 ext/jack/gstjackaudioclient.c delete mode 100644 ext/jack/gstjackaudioclient.h delete mode 100644 ext/jack/gstjackaudiosink.c delete mode 100644 ext/jack/gstjackaudiosink.h delete mode 100644 ext/jack/gstjackaudiosrc.c delete mode 100644 ext/jack/gstjackaudiosrc.h delete mode 100644 ext/jack/gstjackringbuffer.h delete mode 100644 ext/jack/gstjackutil.c delete mode 100644 ext/jack/gstjackutil.h delete mode 100644 tests/examples/jack/Makefile.am delete mode 100644 tests/examples/jack/jack_client.c diff --git a/LICENSE_readme b/LICENSE_readme index 718d6d23bd..3afe96ebc0 100644 --- a/LICENSE_readme +++ b/LICENSE_readme @@ -37,8 +37,6 @@ xvid libxvidcore (http://www.xvid.org/) Plugins derived from GPL code are as follows: dvdreadsrc libdvdread (http://www.dtek.chalmers.se/groups/dvd/) -jack libjack (http://jackit.sourceforge.net/) - Note libjack is LGPL, but plugin is GPL. monoscope None (Algorithm by Ralph Loader, Joerg Walter, Richard Boulton, and Andy Lo A Foe) rtjpeg None (Erik Walthinsen's algorithm) diff --git a/Makefile.am b/Makefile.am index 402b8b9575..a8ac17d49b 100644 --- a/Makefile.am +++ b/Makefile.am @@ -46,6 +46,7 @@ CRUFT_FILES = \ $(top_builddir)/common/shave-libtool \ $(top_builddir)/ext/alsaspdif/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/ext/ivorbis/.libs/*.{so,dll,DLL,dylib} \ + $(top_builddir)/ext/jack/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/aacparse/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/amrparse/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/flacparse/.libs/*.{so,dll,DLL,dylib} \ @@ -60,6 +61,7 @@ CRUFT_FILES = \ $(top_builddir)/tests/check/elements/selector \ $(top_builddir)/tests/check/elements/valve \ $(top_builddir)/tests/check/pipelines/metadata \ + $(top_builddir)/tests/examples/jack/jack_client \ $(top_builddir)/tests/examples/switch/switcher \ $(top_builddir)/tests/icles/output-selector-test \ $(top_builddir)/tests/icles/test-oss4 @@ -74,8 +76,10 @@ CRUFT_DIRS = \ $(top_srcdir)/gst/valve \ $(top_srcdir)/tests/examples/shapewipe \ $(top_srcdir)/tests/examples/switch \ + $(top_srcdir)/tests/examples/jack \ $(top_srcdir)/ext/alsaspdif \ $(top_srcdir)/ext/ivorbis \ + $(top_srcdir)/ext/jack \ $(top_srcdir)/ext/metadata include $(top_srcdir)/common/cruft.mak diff --git a/configure.ac b/configure.ac index 50f03565e7..31b09ed66a 100644 --- a/configure.ac +++ b/configure.ac @@ -888,14 +888,6 @@ AG_GST_CHECK_FEATURE(GSM, [GSM library], gsmenc gsmdec, [ AC_SUBST(GSM_LIBS) ]) -dnl *** Jack *** -translit(dnm, m, l) AM_CONDITIONAL(USE_JACK, true) -AG_GST_CHECK_FEATURE(JACK, Jack, jack, [ - PKG_CHECK_MODULES(JACK, jack >= 0.99.10, HAVE_JACK="yes", HAVE_JACK="no") - AC_SUBST(JACK_CFLAGS) - AC_SUBST(JACK_LIBS) -]) - dnl *** jp2k *** translit(dnm, m, l) AM_CONDITIONAL(USE_JP2K, true) AG_GST_CHECK_FEATURE(JP2K, [jp2k], jp2kdec jp2kenc, [ @@ -1607,7 +1599,6 @@ AM_CONDITIONAL(USE_FAAD, false) AM_CONDITIONAL(USE_FBDEV, false) AM_CONDITIONAL(USE_FLITE, false) AM_CONDITIONAL(USE_GSM, false) -AM_CONDITIONAL(USE_JACK, false) AM_CONDITIONAL(USE_JP2K, false) AM_CONDITIONAL(USE_KATE, false) AM_CONDITIONAL(USE_TIGER, false) @@ -1812,7 +1803,6 @@ tests/examples/camerabin2/Makefile tests/examples/directfb/Makefile tests/examples/mxf/Makefile tests/examples/scaletempo/Makefile -tests/examples/jack/Makefile tests/icles/Makefile ext/amrwbenc/Makefile ext/assrender/Makefile @@ -1830,7 +1820,6 @@ ext/faac/Makefile ext/faad/Makefile ext/flite/Makefile ext/gsm/Makefile -ext/jack/Makefile ext/jp2k/Makefile ext/kate/Makefile ext/ladspa/Makefile diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index 6a151ffea1..63719918fa 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -99,8 +99,6 @@ EXTRA_HFILES = \ $(top_srcdir)/ext/dts/gstdtsdec.h \ $(top_srcdir)/ext/faac/gstfaac.h \ $(top_srcdir)/ext/faad/gstfaad.h \ - $(top_srcdir)/ext/jack/gstjackaudiosrc.h \ - $(top_srcdir)/ext/jack/gstjackaudiosink.h \ $(top_srcdir)/ext/kate/gstkateenc.h \ $(top_srcdir)/ext/kate/gstkatedec.h \ $(top_srcdir)/ext/kate/gstkateparse.h \ diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml index f4b5455045..d102a0691d 100644 --- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml @@ -69,8 +69,6 @@ - - @@ -169,7 +167,6 @@ - diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt index 764f641af2..fe5104ad51 100644 --- a/docs/plugins/gst-plugins-bad-plugins-sections.txt +++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt @@ -806,36 +806,6 @@ GST_TYPE_IVF_PARSE gst_ivf_parse_get_type -
-element-jackaudiosrc -jackaudiosrc -GstJackAudioSrc - -GstJackAudioSrcClass -GST_JACK_AUDIO_SRC -GST_JACK_AUDIO_SRC_CLASS -GST_JACK_AUDIO_SRC_GET_CLASS -GST_IS_JACK_AUDIO_SRC -GST_IS_JACK_AUDIO_SRC_CLASS -GST_TYPE_JACK_AUDIO_SRC -gst_jack_audio_src_get_type -
- -
-element-jackaudiosink -jackaudiosink -GstJackAudioSink - -GstJackAudioSinkClass -GST_JACK_AUDIO_SINK -GST_JACK_AUDIO_SINK_CLASS -GST_JACK_AUDIO_SINK_GET_CLASS -GST_IS_JACK_AUDIO_SINK -GST_IS_JACK_AUDIO_SINK_CLASS -GST_TYPE_JACK_AUDIO_SINK -gst_jack_audio_sink_get_type -
-
element-jpegparse jpegparse diff --git a/docs/plugins/gst-plugins-bad-plugins.args b/docs/plugins/gst-plugins-bad-plugins.args index 9d4bdf5ca3..8ca527f247 100644 --- a/docs/plugins/gst-plugins-bad-plugins.args +++ b/docs/plugins/gst-plugins-bad-plugins.args @@ -1538,36 +1538,6 @@ 1 - -GstJackAudioSink::connect -GstJackConnect - -rw -Connect -Specify how the output ports will be connected. -Automatically connect ports to physical ports - - - -GstJackAudioSink::server -gchar* - -rw -Server -The Jack server to connect to (NULL = default). -NULL - - - -GstJackAudioSink::client -JackClient* - -rw -JackClient -Handle for jack client. - - - GstDvbSrc::bandwidth GstDvbSrcBandwidth @@ -21953,36 +21923,6 @@ 0 - -GstJackAudioSrc::connect -GstJackConnect - -rw -Connect -Specify how the input ports will be connected. -Automatically connect ports to physical ports - - - -GstJackAudioSrc::server -gchar* - -rw -Server -The Jack server to connect to (NULL = default). -NULL - - - -GstJackAudioSrc::client -JackClient* - -rw -JackClient -Handle for jack client. - - - GstDCCPClientSrc::caps GstCaps* @@ -26530,7 +26470,7 @@ rw physics water density: from 1 to 4. -7.75038e-304 +8.09774e-321 @@ -26570,7 +26510,7 @@ rw splash make a big splash in the center. -0 +2.34994e-310 @@ -26580,7 +26520,7 @@ rw splash make a big splash in the center. -4.77773e-299 +1.82574e-315 @@ -26610,7 +26550,7 @@ rw ratiox x-ratio. -0 +1.47273e-316 @@ -26620,7 +26560,7 @@ rw ratioy y-ratio. -0 +1.85891e-316 @@ -26630,7 +26570,7 @@ rw DelayTime the delay time. -0 +2.18476e-316 @@ -26660,7 +26600,7 @@ rw Color the color of the image. -0 +1.39669e-37 @@ -26680,7 +26620,7 @@ rw Color-R the color of the image. -0 +7.30424e-38 @@ -27010,7 +26950,7 @@ rw lredscale multiplier for downscaling non-edge brightness. -0 +3.40216e-111 @@ -27020,7 +26960,7 @@ rw lthresh threshold for edge lightening. -0 +6.9235e+228 @@ -27030,7 +26970,7 @@ rw lupscale multiplier for upscaling edge brightness. -0 +7.54985e-96 @@ -27200,7 +27140,7 @@ rw blend blend factor. -4.74303e-322 +4.77773e-299 @@ -27390,7 +27330,7 @@ rw HSync the hsync offset. -0 +1.86264e-09 diff --git a/docs/plugins/gst-plugins-bad-plugins.hierarchy b/docs/plugins/gst-plugins-bad-plugins.hierarchy index 511a79e541..84322e7d5a 100644 --- a/docs/plugins/gst-plugins-bad-plugins.hierarchy +++ b/docs/plugins/gst-plugins-bad-plugins.hierarchy @@ -31,7 +31,6 @@ GObject GstApExSink GstNasSink GstSDLAudioSink - GstJackAudioSink GstChecksumSink GstDCCPClientSink GstDCCPServerSink @@ -46,8 +45,6 @@ GObject GstDTMFSrc GstDataURISrc GstPushSrc - GstBaseAudioSrc - GstJackAudioSrc GstDCCPClientSrc GstDCCPServerSrc GstDc1394 @@ -490,8 +487,6 @@ GObject GstRegistry GstRingBuffer GstAudioSinkRingBuffer - GstJackAudioSinkRingBuffer - GstJackAudioSrcRingBuffer GstTask GstTaskPool GstSignalObject diff --git a/docs/plugins/inspect/plugin-jack.xml b/docs/plugins/inspect/plugin-jack.xml deleted file mode 100644 index baa370192a..0000000000 --- a/docs/plugins/inspect/plugin-jack.xml +++ /dev/null @@ -1,43 +0,0 @@ - - jack - Jack elements - ../../ext/jack/.libs/libgstjack.so - libgstjack.so - 0.10.20.1 - LGPL - gst-plugins-bad - GStreamer Bad Plug-ins git - Unknown package origin - - - jackaudiosink - Audio Sink (Jack) - Sink/Audio - Output to Jack - Wim Taymans <wim@fluendo.com> - - - sink - sink - always -
audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
- - - - - jackaudiosrc - Audio Source (Jack) - Source/Audio - Input from Jack - Tristan Matthews <tristan@sat.qc.ca> - - - src - source - always -
audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]
- - - - - \ No newline at end of file diff --git a/ext/Makefile.am b/ext/Makefile.am index a9a7c8b5f4..9b670d4dbb 100644 --- a/ext/Makefile.am +++ b/ext/Makefile.am @@ -136,12 +136,6 @@ endif HERMES_DIR= # endif -if USE_JACK -JACK_DIR=jack -else -JACK_DIR= -endif - if USE_JP2K JP2K_DIR = jp2k else @@ -398,7 +392,6 @@ SUBDIRS=\ $(GSM_DIR) \ $(G729_DIR) \ $(HERMES_DIR) \ - $(JACK_DIR) \ $(JP2K_DIR) \ $(KATE_DIR) \ $(LADSPA_DIR) \ @@ -453,7 +446,6 @@ DIST_SUBDIRS = \ gsettings \ gsm \ ladspa \ - jack \ jp2k \ kate \ libmms \ diff --git a/ext/jack/.gitignore b/ext/jack/.gitignore deleted file mode 100644 index 916f265c7a..0000000000 --- a/ext/jack/.gitignore +++ /dev/null @@ -1 +0,0 @@ -*.loT diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am deleted file mode 100644 index ede70594bd..0000000000 --- a/ext/jack/Makefile.am +++ /dev/null @@ -1,12 +0,0 @@ - -plugin_LTLIBRARIES = libgstjack.la - -libgstjack_la_SOURCES = gstjackutil.c gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c -libgstjack_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS) -libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS) -libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -libgstjack_la_LIBTOOLFLAGS = --tag=disable-static - -noinst_HEADERS = gstjackutil.h gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h - -EXTRA_DIST = README diff --git a/ext/jack/README b/ext/jack/README deleted file mode 100644 index 2bb97bebad..0000000000 --- a/ext/jack/README +++ /dev/null @@ -1,4 +0,0 @@ -to be written, la dee da - -jackit.sf.net - diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c deleted file mode 100644 index 371a9e934f..0000000000 --- a/ext/jack/gstjack.c +++ /dev/null @@ -1,95 +0,0 @@ -/* GStreamer Jack plugins - * Copyright (C) 2006 Wim Taymans - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "gstjackaudiosrc.h" -#include "gstjackaudiosink.h" - -GType -gst_jack_connect_get_type (void) -{ - static GType jack_connect_type = 0; - static const GEnumValue jack_connect[] = { - {GST_JACK_CONNECT_NONE, - "Don't automatically connect ports to physical ports", "none"}, - {GST_JACK_CONNECT_AUTO, - "Automatically connect ports to physical ports", "auto"}, - {GST_JACK_CONNECT_AUTO_FORCED, - "Automatically connect ports to as many physical ports as possible", - "auto-forced"}, - {0, NULL, NULL}, - }; - - if (!jack_connect_type) { - jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect); - } - return jack_connect_type; -} - - -static gpointer -gst_jack_client_copy (gpointer jclient) -{ - return jclient; -} - - -static void -gst_jack_client_free (gpointer jclient) -{ - return; -} - - -GType -gst_jack_client_get_type (void) -{ - static GType type; /* 0 */ - - if (type == 0) { - /* hackish, but makes it show up nicely in gst-inspect */ - type = g_boxed_type_register_static ("JackClient", - (GBoxedCopyFunc) gst_jack_client_copy, - (GBoxedFreeFunc) gst_jack_client_free); - } - - return type; -} - -static gboolean -plugin_init (GstPlugin * plugin) -{ - if (!gst_element_register (plugin, "jackaudiosrc", GST_RANK_PRIMARY, - GST_TYPE_JACK_AUDIO_SRC)) - return FALSE; - if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY, - GST_TYPE_JACK_AUDIO_SINK)) - return FALSE; - - return TRUE; -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "jack", - "Jack elements", - plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h deleted file mode 100644 index d923866dfa..0000000000 --- a/ext/jack/gstjack.h +++ /dev/null @@ -1,55 +0,0 @@ -/* GStreamer - * Copyright (C) 2006 Wim Taymans - * - * gstjack.h: - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef _GST_JACK_H_ -#define _GST_JACK_H_ - - -/** - * GstJackConnect: - * @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports. - * In this mode, the element will accept any number of input channels and will - * create (but not connect) an output port for each channel. - * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each - * output port to a random physical jack input pin. The sink will - * expose the number of physical channels on its pad caps. - * @GST_JACK_CONNECT_AUTO_FORCED: In this mode, the element will try to connect each - * output port to a random physical jack input pin. The element will accept any number - * of input channels. - * - * Specify how the output ports will be connected. - */ - -typedef enum { - GST_JACK_CONNECT_NONE, - GST_JACK_CONNECT_AUTO, - GST_JACK_CONNECT_AUTO_FORCED -} GstJackConnect; - -typedef jack_default_audio_sample_t sample_t; - -#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type()) -#define GST_TYPE_JACK_CLIENT (gst_jack_client_get_type ()) - -GType gst_jack_client_get_type(void); -GType gst_jack_connect_get_type(void); - -#endif // _GST_JACK_H_ diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c deleted file mode 100644 index 1789edb60d..0000000000 --- a/ext/jack/gstjackaudioclient.c +++ /dev/null @@ -1,525 +0,0 @@ -/* GStreamer - * Copyright (C) 2006 Wim Taymans - * - * gstjackaudioclient.c: jack audio client 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#include - -#include "gstjackaudioclient.h" - -GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_client_debug); -#define GST_CAT_DEFAULT gst_jack_audio_client_debug - -void -gst_jack_audio_client_init (void) -{ - GST_DEBUG_CATEGORY_INIT (gst_jack_audio_client_debug, "jackclient", 0, - "jackclient helpers"); -} - -/* a list of global connections indexed by id and server. */ -G_LOCK_DEFINE_STATIC (connections_lock); -static GList *connections; - -/* the connection to a server */ -typedef struct -{ - gint refcount; - GMutex *lock; - GCond *flush_cond; - - /* id/server pair and the connection */ - gchar *id; - gchar *server; - jack_client_t *client; - - /* lists of GstJackAudioClients */ - gint n_clients; - GList *src_clients; - GList *sink_clients; -} GstJackAudioConnection; - -/* an object sharing a jack_client_t connection. */ -struct _GstJackAudioClient -{ - GstJackAudioConnection *conn; - - GstJackClientType type; - gboolean active; - gboolean deactivate; - - void (*shutdown) (void *arg); - JackProcessCallback process; - JackBufferSizeCallback buffer_size; - JackSampleRateCallback sample_rate; - gpointer user_data; -}; - -typedef jack_default_audio_sample_t sample_t; - -typedef struct -{ - jack_nframes_t nframes; - gpointer user_data; -} JackCB; - -static int -jack_process_cb (jack_nframes_t nframes, void *arg) -{ - GstJackAudioConnection *conn = (GstJackAudioConnection *) arg; - GList *walk; - int res = 0; - - g_mutex_lock (conn->lock); - /* call sources first, then sinks. Sources will either push data into the - * ringbuffer of the sinks, which will then pull the data out of it, or - * sinks will pull the data from the sources. */ - for (walk = conn->src_clients; walk; walk = g_list_next (walk)) { - GstJackAudioClient *client = (GstJackAudioClient *) walk->data; - - /* only call active clients */ - if ((client->active || client->deactivate) && client->process) { - res = client->process (nframes, client->user_data); - if (client->deactivate) { - client->deactivate = FALSE; - g_cond_signal (conn->flush_cond); - } - } - } - for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) { - GstJackAudioClient *client = (GstJackAudioClient *) walk->data; - - /* only call active clients */ - if ((client->active || client->deactivate) && client->process) { - res = client->process (nframes, client->user_data); - if (client->deactivate) { - client->deactivate = FALSE; - g_cond_signal (conn->flush_cond); - } - } - } - g_mutex_unlock (conn->lock); - - return res; -} - -/* we error out */ -static int -jack_sample_rate_cb (jack_nframes_t nframes, void *arg) -{ - return 0; -} - -/* we error out */ -static int -jack_buffer_size_cb (jack_nframes_t nframes, void *arg) -{ - return 0; -} - -static void -jack_shutdown_cb (void *arg) -{ - GstJackAudioConnection *conn = (GstJackAudioConnection *) arg; - GList *walk; - - GST_DEBUG ("disconnect client %s from server %s", conn->id, - GST_STR_NULL (conn->server)); - - g_mutex_lock (conn->lock); - for (walk = conn->src_clients; walk; walk = g_list_next (walk)) { - GstJackAudioClient *client = (GstJackAudioClient *) walk->data; - - if (client->shutdown) - client->shutdown (client->user_data); - } - for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) { - GstJackAudioClient *client = (GstJackAudioClient *) walk->data; - - if (client->shutdown) - client->shutdown (client->user_data); - } - g_mutex_unlock (conn->lock); -} - -typedef struct -{ - const gchar *id; - const gchar *server; -} FindData; - -static gint -connection_find (GstJackAudioConnection * conn, FindData * data) -{ - /* id's must match */ - if (strcmp (conn->id, data->id)) - return 1; - - /* both the same or NULL */ - if (conn->server == data->server) - return 0; - - /* we cannot compare NULL */ - if (conn->server == NULL || data->server == NULL) - return 1; - - if (strcmp (conn->server, data->server)) - return 1; - - return 0; -} - -/* make a connection with @id and @server. Returns NULL on failure with the - * status set. */ -static GstJackAudioConnection * -gst_jack_audio_make_connection (const gchar * id, const gchar * server, - jack_client_t * jclient, jack_status_t * status) -{ - GstJackAudioConnection *conn; - jack_options_t options; - gint res; - - *status = 0; - - GST_DEBUG ("new client %s, connecting to server %s", id, - GST_STR_NULL (server)); - - /* never start a server */ - options = JackNoStartServer; - /* if we have a servername, use it */ - if (server != NULL) - options |= JackServerName; - /* open the client */ - if (jclient == NULL) - jclient = jack_client_open (id, options, status, server); - if (jclient == NULL) - goto could_not_open; - - /* now create object */ - conn = g_new (GstJackAudioConnection, 1); - conn->refcount = 1; - conn->lock = g_mutex_new (); - conn->flush_cond = g_cond_new (); - conn->id = g_strdup (id); - conn->server = g_strdup (server); - conn->client = jclient; - conn->n_clients = 0; - conn->src_clients = NULL; - conn->sink_clients = NULL; - - /* set our callbacks */ - jack_set_process_callback (jclient, jack_process_cb, conn); - /* these callbacks cause us to error */ - jack_set_buffer_size_callback (jclient, jack_buffer_size_cb, conn); - jack_set_sample_rate_callback (jclient, jack_sample_rate_cb, conn); - jack_on_shutdown (jclient, jack_shutdown_cb, conn); - - /* all callbacks are set, activate the client */ - if ((res = jack_activate (jclient))) - goto could_not_activate; - - GST_DEBUG ("opened connection %p", conn); - - return conn; - - /* ERRORS */ -could_not_open: - { - GST_DEBUG ("failed to open jack client, %d", *status); - return NULL; - } -could_not_activate: - { - GST_ERROR ("Could not activate client (%d)", res); - *status = JackFailure; - g_mutex_free (conn->lock); - g_free (conn->id); - g_free (conn->server); - g_free (conn); - return NULL; - } -} - -static GstJackAudioConnection * -gst_jack_audio_get_connection (const gchar * id, const gchar * server, - jack_client_t * jclient, jack_status_t * status) -{ - GstJackAudioConnection *conn; - GList *found; - FindData data; - - GST_DEBUG ("getting connection for id %s, server %s", id, - GST_STR_NULL (server)); - - data.id = id; - data.server = server; - - G_LOCK (connections_lock); - found = - g_list_find_custom (connections, &data, (GCompareFunc) connection_find); - if (found != NULL && jclient != NULL) { - /* we found it, increase refcount and return it */ - conn = (GstJackAudioConnection *) found->data; - conn->refcount++; - - GST_DEBUG ("found connection %p", conn); - } else { - /* make new connection */ - conn = gst_jack_audio_make_connection (id, server, jclient, status); - if (conn != NULL) { - GST_DEBUG ("created connection %p", conn); - /* add to list on success */ - connections = g_list_prepend (connections, conn); - } else { - GST_WARNING ("could not create connection"); - } - } - G_UNLOCK (connections_lock); - - return conn; -} - -static void -gst_jack_audio_unref_connection (GstJackAudioConnection * conn) -{ - gint res; - gboolean zero; - - GST_DEBUG ("unref connection %p refcnt %d", conn, conn->refcount); - - G_LOCK (connections_lock); - conn->refcount--; - if ((zero = (conn->refcount == 0))) { - GST_DEBUG ("closing connection %p", conn); - /* remove from list, we can release the mutex after removing the connection - * from the list because after that, nobody can access the connection anymore. */ - connections = g_list_remove (connections, conn); - } - G_UNLOCK (connections_lock); - - /* if we are zero, close and cleanup the connection */ - if (zero) { - /* don't use conn->lock here. two reasons: - * - * 1) its not necessary: jack_deactivate() will not return until the JACK thread - * associated with this connection is cleaned up by a thread join, hence - * no more callbacks can occur or be in progress. - * - * 2) it would deadlock anyway, because jack_deactivate() will sleep - * waiting for the JACK thread, and can thus cause deadlock in - * jack_process_cb() - */ - if ((res = jack_deactivate (conn->client))) { - /* we only warn, this means the server is probably shut down and the client - * is gone anyway. */ - GST_WARNING ("Could not deactivate Jack client (%d)", res); - } - /* close connection */ - if ((res = jack_client_close (conn->client))) { - /* we assume the client is gone. */ - GST_WARNING ("close failed (%d)", res); - } - - /* free resources */ - g_mutex_free (conn->lock); - g_cond_free (conn->flush_cond); - g_free (conn->id); - g_free (conn->server); - g_free (conn); - } -} - -static void -gst_jack_audio_connection_add_client (GstJackAudioConnection * conn, - GstJackAudioClient * client) -{ - g_mutex_lock (conn->lock); - switch (client->type) { - case GST_JACK_CLIENT_SOURCE: - conn->src_clients = g_list_append (conn->src_clients, client); - conn->n_clients++; - break; - case GST_JACK_CLIENT_SINK: - conn->sink_clients = g_list_append (conn->sink_clients, client); - conn->n_clients++; - break; - default: - g_warning ("trying to add unknown client type"); - break; - } - g_mutex_unlock (conn->lock); -} - -static void -gst_jack_audio_connection_remove_client (GstJackAudioConnection * conn, - GstJackAudioClient * client) -{ - g_mutex_lock (conn->lock); - switch (client->type) { - case GST_JACK_CLIENT_SOURCE: - conn->src_clients = g_list_remove (conn->src_clients, client); - conn->n_clients--; - break; - case GST_JACK_CLIENT_SINK: - conn->sink_clients = g_list_remove (conn->sink_clients, client); - conn->n_clients--; - break; - default: - g_warning ("trying to remove unknown client type"); - break; - } - g_mutex_unlock (conn->lock); -} - -/** - * gst_jack_audio_client_get: - * @id: the client id - * @server: the server to connect to or NULL for the default server - * @type: the client type - * @shutdown: a callback when the jack server shuts down - * @process: a callback when samples are available - * @buffer_size: a callback when the buffer_size changes - * @sample_rate: a callback when the sample_rate changes - * @user_data: user data passed to the callbacks - * @status: pointer to hold the jack status code in case of errors - * - * Get the jack client connection for @id and @server. Connections to the same - * @id and @server will receive the same physical Jack client connection and - * will therefore be scheduled in the same process callback. - * - * Returns: a #GstJackAudioClient. - */ -GstJackAudioClient * -gst_jack_audio_client_new (const gchar * id, const gchar * server, - jack_client_t * jclient, GstJackClientType type, - void (*shutdown) (void *arg), JackProcessCallback process, - JackBufferSizeCallback buffer_size, JackSampleRateCallback sample_rate, - gpointer user_data, jack_status_t * status) -{ - GstJackAudioClient *client; - GstJackAudioConnection *conn; - - g_return_val_if_fail (id != NULL, NULL); - g_return_val_if_fail (status != NULL, NULL); - - /* first get a connection for the id/server pair */ - conn = gst_jack_audio_get_connection (id, server, jclient, status); - if (conn == NULL) - goto no_connection; - - GST_INFO ("new client %s", id); - - /* make new client using the connection */ - client = g_new (GstJackAudioClient, 1); - client->active = client->deactivate = FALSE; - client->conn = conn; - client->type = type; - client->shutdown = shutdown; - client->process = process; - client->buffer_size = buffer_size; - client->sample_rate = sample_rate; - client->user_data = user_data; - - /* add the client to the connection */ - gst_jack_audio_connection_add_client (conn, client); - - return client; - - /* ERRORS */ -no_connection: - { - GST_DEBUG ("Could not get server connection (%d)", *status); - return NULL; - } -} - -/** - * gst_jack_audio_client_free: - * @client: a #GstJackAudioClient - * - * Free the resources used by @client. - */ -void -gst_jack_audio_client_free (GstJackAudioClient * client) -{ - GstJackAudioConnection *conn; - - g_return_if_fail (client != NULL); - - GST_INFO ("free client"); - - conn = client->conn; - - /* remove from connection first so that it's not scheduled anymore after this - * call */ - gst_jack_audio_connection_remove_client (conn, client); - gst_jack_audio_unref_connection (conn); - - g_free (client); -} - -/** - * gst_jack_audio_client_get_client: - * @client: a #GstJackAudioClient - * - * Get the jack audio client for @client. This function is used to perform - * operations on the jack server from this client. - * - * Returns: The jack audio client. - */ -jack_client_t * -gst_jack_audio_client_get_client (GstJackAudioClient * client) -{ - g_return_val_if_fail (client != NULL, NULL); - - /* no lock needed, the connection and the client does not change - * once the client is created. */ - return client->conn->client; -} - -/** - * gst_jack_audio_client_set_active: - * @client: a #GstJackAudioClient - * @active: new mode for the client - * - * Activate or deactive @client. When a client is activated it will receive - * callbacks when data should be processed. - * - * Returns: 0 if all ok. - */ -gint -gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active) -{ - g_return_val_if_fail (client != NULL, -1); - - /* make sure that we are not dispatching the client */ - g_mutex_lock (client->conn->lock); - if (client->active && !active) { - /* we need to process once more to flush the port */ - client->deactivate = TRUE; - - /* need to wait for process_cb run once more */ - while (client->deactivate) - g_cond_wait (client->conn->flush_cond, client->conn->lock); - } - client->active = active; - g_mutex_unlock (client->conn->lock); - - return 0; -} diff --git a/ext/jack/gstjackaudioclient.h b/ext/jack/gstjackaudioclient.h deleted file mode 100644 index 5fb7e35444..0000000000 --- a/ext/jack/gstjackaudioclient.h +++ /dev/null @@ -1,59 +0,0 @@ -/* GStreamer - * Copyright (C) 2006 Wim Taymans - * - * gstjackaudioclient.h: - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_JACK_AUDIO_CLIENT_H__ -#define __GST_JACK_AUDIO_CLIENT_H__ - -#include - -#include - -G_BEGIN_DECLS - -typedef enum -{ - GST_JACK_CLIENT_SOURCE, - GST_JACK_CLIENT_SINK -} GstJackClientType; - -typedef struct _GstJackAudioClient GstJackAudioClient; - -void gst_jack_audio_client_init (void); - - -GstJackAudioClient * gst_jack_audio_client_new (const gchar *id, const gchar *server, - jack_client_t *jclient, - GstJackClientType type, - void (*shutdown) (void *arg), - JackProcessCallback process, - JackBufferSizeCallback buffer_size, - JackSampleRateCallback sample_rate, - gpointer user_data, - jack_status_t *status); -void gst_jack_audio_client_free (GstJackAudioClient *client); - -jack_client_t * gst_jack_audio_client_get_client (GstJackAudioClient *client); - -gboolean gst_jack_audio_client_set_active (GstJackAudioClient *client, gboolean active); - -G_END_DECLS - -#endif /* __GST_JACK_AUDIO_CLIENT_H__ */ diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c deleted file mode 100644 index 32bf1af3e4..0000000000 --- a/ext/jack/gstjackaudiosink.c +++ /dev/null @@ -1,852 +0,0 @@ -/* GStreamer - * Copyright (C) 2006 Wim Taymans - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:element-jackaudiosink - * @see_also: #GstBaseAudioSink, #GstRingBuffer - * - * 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. - * - * - * Example launch line - * |[ - * gst-launch audiotestsrc ! jackaudiosink - * ]| Play a sine wave to using jack. - * - * - * Last reviewed on 2006-11-30 (0.10.4) - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include -#include -#include - -#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); - - /* 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; -} - -/* ringbuffer abstract base class */ -static GType -gst_jack_ring_buffer_get_type (void) -{ - static GType ringbuffer_type = 0; - - if (!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 - }; - - ringbuffer_type = - g_type_register_static (GST_TYPE_RING_BUFFER, - "GstJackAudioSinkRingBuffer", &ringbuffer_info, 0); - } - return ringbuffer_type; -} - -static void -gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass) -{ - GObjectClass *gobject_class; - GstObjectClass *gstobject_class; - GstRingBufferClass *gstringbuffer_class; - - gobject_class = (GObjectClass *) klass; - gstobject_class = (GstObjectClass *) klass; - gstringbuffer_class = (GstRingBufferClass *) 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; - GstRingBuffer *buf; - GstJackRingBuffer *abuf; - gint readseg, len; - guint8 *readptr; - gint i, j, flen, channels; - sample_t **buffers, *data; - - buf = GST_RING_BUFFER_CAST (arg); - abuf = GST_JACK_RING_BUFFER_CAST (arg); - sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); - - channels = buf->spec.channels; - - /* alloc pointers to samples */ - buffers = g_alloca (sizeof (sample_t *) * channels); - - /* get target buffers */ - for (i = 0; i < channels; i++) { - buffers[i] = (sample_t *) jack_port_get_buffer (sink->ports[i], nframes); - } - - if (gst_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++) { - buffers[j][i] = *data++; - } - } - - /* clear written samples in the ringbuffer */ - gst_ring_buffer_clear (buf, readseg); - - /* we wrote one segment */ - gst_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 (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 (GstRingBuffer * 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"); - - 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 (GstRingBuffer * 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 (GstRingBuffer * buf, GstRingBufferSpec * spec) -{ - GstJackAudioSink *sink; - GstJackRingBuffer *abuf; - const char **ports; - gint sample_rate, buffer_size; - gint i, 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); - - /* sample rate must be that of the server */ - sample_rate = jack_get_sample_rate (client); - if (sample_rate != spec->rate) - goto wrong_samplerate; - - channels = spec->channels; - - 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), spec->rate * spec->bytes_per_sample); - /* 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->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); - memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); - - 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 = spec->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, spec->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 (GstRingBuffer * 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 */ - gst_buffer_unref (buf->data); - buf->data = NULL; - - return TRUE; -} - -static gboolean -gst_jack_ring_buffer_start (GstRingBuffer * buf) -{ - GstJackAudioSink *sink; - - sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); - - GST_DEBUG_OBJECT (sink, "start"); - - return TRUE; -} - -static gboolean -gst_jack_ring_buffer_pause (GstRingBuffer * buf) -{ - GstJackAudioSink *sink; - - sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); - - GST_DEBUG_OBJECT (sink, "pause"); - - return TRUE; -} - -static gboolean -gst_jack_ring_buffer_stop (GstRingBuffer * buf) -{ - GstJackAudioSink *sink; - - sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf)); - - GST_DEBUG_OBJECT (sink, "stop"); - - return TRUE; -} - -static guint -gst_jack_ring_buffer_delay (GstRingBuffer * buf) -{ - GstJackAudioSink *sink; - guint i, res = 0, 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; -} - -static GstStaticPadTemplate jackaudiosink_sink_factory = -GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-float, " - "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, " - "width = (int) 32, " - "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 - -enum -{ - PROP_0, - PROP_CONNECT, - PROP_SERVER, - PROP_CLIENT, - PROP_LAST -}; - -#define _do_init(bla) \ - GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element"); - -GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink, - GST_TYPE_BASE_AUDIO_SINK, _do_init); - -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); -static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * - sink); - -static void -gst_jack_audio_sink_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_set_details_simple (element_class, "Audio Sink (Jack)", - "Sink/Audio", "Output to Jack", "Wim Taymans "); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&jackaudiosink_sink_factory)); -} - -static void -gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - GstBaseSinkClass *gstbasesink_class; - GstBaseAudioSinkClass *gstbaseaudiosink_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - gstbasesink_class = (GstBaseSinkClass *) klass; - gstbaseaudiosink_class = (GstBaseAudioSinkClass *) 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)); - - 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)); - - gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps); - - gstbaseaudiosink_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, - GstJackAudioSinkClass * g_class) -{ - sink->connect = DEFAULT_PROP_CONNECT; - sink->server = g_strdup (DEFAULT_PROP_SERVER); - sink->jclient = NULL; - sink->ports = NULL; - sink->port_count = 0; -} - -static void -gst_jack_audio_sink_dispose (GObject * object) -{ - GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object); - - gst_caps_replace (&sink->caps, 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_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; - 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_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; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static GstCaps * -gst_jack_audio_sink_getcaps (GstBaseSink * bsink) -{ - 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-float", - "endianness", G_TYPE_INT, G_BYTE_ORDER, - "width", G_TYPE_INT, 32, - "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 GstRingBuffer * -gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink) -{ - GstRingBuffer *buffer; - - buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL); - GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer); - - return buffer; -} diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h deleted file mode 100644 index def423329d..0000000000 --- a/ext/jack/gstjackaudiosink.h +++ /dev/null @@ -1,78 +0,0 @@ -/* GStreamer - * Copyright (C) 2006 Wim Taymans - * - * gstjacksink.h: - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_JACK_AUDIO_SINK_H__ -#define __GST_JACK_AUDIO_SINK_H__ - -#include - -#include -#include - -#include "gstjack.h" -#include "gstjackaudioclient.h" - -G_BEGIN_DECLS - -#define GST_TYPE_JACK_AUDIO_SINK (gst_jack_audio_sink_get_type()) -#define GST_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSink)) -#define GST_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass)) -#define GST_JACK_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass)) -#define GST_IS_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SINK)) -#define GST_IS_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SINK)) - -typedef struct _GstJackAudioSink GstJackAudioSink; -typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass; - -/** - * GstJackAudioSink: - * - * Opaque #GstJackAudioSink. - */ -struct _GstJackAudioSink { - GstBaseAudioSink element; - - /*< private >*/ - /* cached caps */ - GstCaps *caps; - - /* properties */ - GstJackConnect connect; - gchar *server; - jack_client_t *jclient; - - /* our client */ - GstJackAudioClient *client; - - /* our ports */ - jack_port_t **ports; - int port_count; -}; - -struct _GstJackAudioSinkClass { - GstBaseAudioSinkClass parent_class; -}; - -GType gst_jack_audio_sink_get_type (void); - -G_END_DECLS - -#endif /* __GST_JACK_AUDIO_SINK_H__ */ diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c deleted file mode 100644 index 0ffdb23985..0000000000 --- a/ext/jack/gstjackaudiosrc.c +++ /dev/null @@ -1,874 +0,0 @@ -/* GStreamer - * Copyright (C) 2008 Tristan Matthews - * - * Permission is hereby granted, free of charge, to any person obtaining a - * copy of this software and associated documentation files (the "Software"), - * to deal in the Software without restriction, including without limitation - * the rights to use, copy, modify, merge, publish, distribute, sublicense, - * and/or sell copies of the Software, and to permit persons to whom the - * Software is furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE - * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING - * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER - * DEALINGS IN THE SOFTWARE. - * - * Alternatively, the contents of this file may be used under the - * GNU Lesser General Public License Version 2.1 (the "LGPL"), in - * which case the following provisions apply instead of the ones - * mentioned above: - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:element-jackaudiosrc - * @see_also: #GstBaseAudioSrc, #GstRingBuffer - * - * A Src that inputs data from Jack ports. - * - * It will create N Jack ports named in_<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 #GstJackAudioSrc:connect property is set to auto, this element - * will try to connect each input port to a random physical jack output pin. - * - * When the #GstJackAudioSrc:connect property is set to none, the element will - * accept any number of output channels and will create (but not connect) an - * input 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. - * - * - * Example launch line - * |[ - * gst-launch jackaudiosrc connect=0 ! jackaudiosink connect=0 - * ]| Get audio input into gstreamer from jack. - * - * - * Last reviewed on 2008-07-22 (0.10.4) - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include -#include -#include - -#include "gstjackaudiosrc.h" -#include "gstjackringbuffer.h" -#include "gstjackutil.h" - -GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug); -#define GST_CAT_DEFAULT gst_jack_audio_src_debug - -static gboolean -gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels) -{ - jack_client_t *client; - - client = gst_jack_audio_client_get_client (src->client); - - /* remove ports we don't need */ - while (src->port_count > channels) - jack_port_unregister (client, src->ports[--src->port_count]); - - /* alloc enough input ports */ - src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels); - src->buffers = g_realloc (src->buffers, sizeof (sample_t *) * channels); - - /* create an input port for each channel */ - while (src->port_count < channels) { - gchar *name; - - /* port names start from 1 and are local to the element */ - name = - g_strdup_printf ("in_%s_%d", GST_ELEMENT_NAME (src), - src->port_count + 1); - src->ports[src->port_count] = - jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE, - JackPortIsInput, 0); - if (src->ports[src->port_count] == NULL) - return FALSE; - - src->port_count++; - - g_free (name); - } - return TRUE; -} - -static void -gst_jack_audio_src_free_channels (GstJackAudioSrc * src) -{ - gint res, i = 0; - jack_client_t *client; - - client = gst_jack_audio_client_get_client (src->client); - - /* get rid of all ports */ - while (src->port_count) { - GST_LOG_OBJECT (src, "unregister port %d", i); - if ((res = jack_port_unregister (client, src->ports[i++]))) - GST_DEBUG_OBJECT (src, "unregister of port failed (%d)", res); - - src->port_count--; - } - g_free (src->ports); - src->ports = NULL; - g_free (src->buffers); - src->buffers = NULL; -} - -/* ringbuffer abstract base class */ -static GType -gst_jack_ring_buffer_get_type (void) -{ - static GType ringbuffer_type = 0; - - if (!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 - }; - - ringbuffer_type = - g_type_register_static (GST_TYPE_RING_BUFFER, - "GstJackAudioSrcRingBuffer", &ringbuffer_info, 0); - } - return ringbuffer_type; -} - -static void -gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass) -{ - GObjectClass *gobject_class; - GstObjectClass *gstobject_class; - GstRingBufferClass *gstringbuffer_class; - - gobject_class = (GObjectClass *) klass; - gstobject_class = (GstObjectClass *) klass; - gstringbuffer_class = (GstRingBufferClass *) 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 be RT-safe. - * Writes samples from the jack input port's buffer to the gst ring buffer. - */ -static int -jack_process_cb (jack_nframes_t nframes, void *arg) -{ - GstJackAudioSrc *src; - GstRingBuffer *buf; - gint len; - guint8 *writeptr; - gint writeseg; - gint channels, i, j, flen; - sample_t *data; - - buf = GST_RING_BUFFER_CAST (arg); - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); - - channels = buf->spec.channels; - - /* get input buffers */ - for (i = 0; i < channels; i++) - src->buffers[i] = - (sample_t *) jack_port_get_buffer (src->ports[i], nframes); - - if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) { - flen = len / channels; - - /* the number of samples must be exactly the segment size */ - if (nframes * sizeof (sample_t) != flen) - goto wrong_size; - - /* the samples in the jack input buffers have to be interleaved into the - * ringbuffer */ - data = (sample_t *) writeptr; - for (i = 0; i < nframes; ++i) - for (j = 0; j < channels; ++j) - *data++ = src->buffers[j][i]; - - GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr, - len / channels, channels); - - /* we wrote one segment */ - gst_ring_buffer_advance (buf, 1); - } - return 0; - - /* ERRORS */ -wrong_size: - { - GST_ERROR_OBJECT (src, "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) -{ - GstJackAudioSrc *src; - GstJackRingBuffer *abuf; - - abuf = GST_JACK_RING_BUFFER_CAST (arg); - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg)); - - if (abuf->sample_rate != -1 && abuf->sample_rate != nframes) - goto not_supported; - - return 0; - - /* ERRORS */ -not_supported: - { - GST_ELEMENT_ERROR (src, 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) -{ - GstJackAudioSrc *src; - GstJackRingBuffer *abuf; - - abuf = GST_JACK_RING_BUFFER_CAST (arg); - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg)); - - if (abuf->buffer_size != -1 && abuf->buffer_size != nframes) - goto not_supported; - - return 0; - - /* ERRORS */ -not_supported: - { - GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, - (NULL), ("Jack changed the buffer size, which is not supported")); - return 1; - } -} - -static void -jack_shutdown_cb (void *arg) -{ - GstJackAudioSrc *src; - - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg)); - - GST_DEBUG_OBJECT (src, "shutdown"); - - GST_ELEMENT_ERROR (src, 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 (GstRingBuffer * buf) -{ - GstJackAudioSrc *src; - jack_status_t status = 0; - const gchar *name; - - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); - - GST_DEBUG_OBJECT (src, "open"); - - name = g_get_application_name (); - if (!name) - name = "GStreamer"; - - src->client = gst_jack_audio_client_new (name, src->server, - src->jclient, - GST_JACK_CLIENT_SOURCE, - jack_shutdown_cb, - jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status); - if (src->client == NULL) - goto could_not_open; - - GST_DEBUG_OBJECT (src, "opened"); - - return TRUE; - - /* ERRORS */ -could_not_open: - { - if (status & JackServerFailed) { - GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, - (_("Jack server not found")), - ("Cannot connect to the Jack server (status %d)", status)); - } else { - GST_ELEMENT_ERROR (src, 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 (GstRingBuffer * buf) -{ - GstJackAudioSrc *src; - - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); - - GST_DEBUG_OBJECT (src, "close"); - - gst_jack_audio_src_free_channels (src); - gst_jack_audio_client_free (src->client); - src->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 (GstRingBuffer * buf, GstRingBufferSpec * spec) -{ - GstJackAudioSrc *src; - GstJackRingBuffer *abuf; - const char **ports; - gint sample_rate, buffer_size; - gint i, channels, res; - jack_client_t *client; - - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); - abuf = GST_JACK_RING_BUFFER_CAST (buf); - - GST_DEBUG_OBJECT (src, "acquire"); - - client = gst_jack_audio_client_get_client (src->client); - - /* sample rate must be that of the server */ - sample_rate = jack_get_sample_rate (client); - if (sample_rate != spec->rate) - goto wrong_samplerate; - - channels = spec->channels; - - if (!gst_jack_audio_src_allocate_channels (src, channels)) - goto out_of_ports; - - gst_jack_set_layout_on_caps (&spec->caps, channels); - - 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), spec->rate * spec->bytes_per_sample); - /* 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 (src, "buffer time: %" G_GINT64_FORMAT " usec", - spec->buffer_time); - GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec", - spec->latency_time); - GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d", - buffer_size, spec->segsize, spec->segtotal); - - /* allocate the ringbuffer memory now */ - buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); - memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); - - if ((res = gst_jack_audio_client_set_active (src->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 (src->connect == GST_JACK_CONNECT_AUTO - || src->connect == GST_JACK_CONNECT_AUTO_FORCED) { - /* find all the physical output ports. A physical output port is a port - * associated with a hardware device. Someone needs connect to a physical - * port in order to capture something. */ - ports = - jack_get_ports (client, NULL, NULL, - JackPortIsPhysical | JackPortIsOutput); - if (ports == NULL) { - /* no ports? fine then we don't do anything except for posting a warning - * message. */ - GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL), - ("No physical output ports found, leaving ports unconnected")); - goto done; - } - - for (i = 0; i < channels; i++) { - /* stop when all output ports are exhausted */ - if (ports[i] == NULL) { - /* post a warning that we could not connect all ports */ - GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL), - ("No more physical ports, leaving some ports unconnected")); - break; - } - 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; - } - free (ports); - } -done: - - abuf->sample_rate = sample_rate; - abuf->buffer_size = buffer_size; - abuf->channels = spec->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, spec->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))); - free (ports); - return FALSE; - } -} - -/* function is called with LOCK */ -static gboolean -gst_jack_ring_buffer_release (GstRingBuffer * 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 */ - gst_buffer_unref (buf->data); - buf->data = NULL; - - return TRUE; -} - -static gboolean -gst_jack_ring_buffer_start (GstRingBuffer * buf) -{ - GstJackAudioSrc *src; - - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); - - GST_DEBUG_OBJECT (src, "start"); - - return TRUE; -} - -static gboolean -gst_jack_ring_buffer_pause (GstRingBuffer * buf) -{ - GstJackAudioSrc *src; - - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); - - GST_DEBUG_OBJECT (src, "pause"); - - return TRUE; -} - -static gboolean -gst_jack_ring_buffer_stop (GstRingBuffer * buf) -{ - GstJackAudioSrc *src; - - src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf)); - - GST_DEBUG_OBJECT (src, "stop"); - - return TRUE; -} - -static guint -gst_jack_ring_buffer_delay (GstRingBuffer * buf) -{ - GstJackAudioSrc *src; - guint i, res = 0, 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; -} - -/* Audiosrc signals and args */ -enum -{ - /* FILL ME */ - LAST_SIGNAL -}; - -#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO -#define DEFAULT_PROP_SERVER NULL - -enum -{ - PROP_0, - PROP_CONNECT, - PROP_SERVER, - PROP_CLIENT, - 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-float, " - "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, " - "width = (int) 32, " - "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") - ); - -#define _do_init(bla) \ - GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element"); - -GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc, - GST_TYPE_BASE_AUDIO_SRC, _do_init); - -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); -static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * - src); - -/* GObject vmethod implementations */ - -static void -gst_jack_audio_src_base_init (gpointer gclass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (gclass); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_factory)); - gst_element_class_set_details_simple (element_class, "Audio Source (Jack)", - "Source/Audio", - "Input from Jack", "Tristan Matthews "); -} - -/* 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; - GstBaseAudioSrcClass *gstbaseaudiosrc_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - - gstbasesrc_class = (GstBaseSrcClass *) klass; - gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) 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)); - - 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)); - - gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps); - gstbaseaudiosrc_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 (); -} - -/* initialize the new element - * instantiate pads and add them to element - * set pad calback functions - * initialize instance structure - */ -static void -gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) -{ - //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; -} - -static void -gst_jack_audio_src_dispose (GObject * object) -{ - GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); - - gst_caps_replace (&src->caps, 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_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; - 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_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; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static GstCaps * -gst_jack_audio_src_getcaps (GstBaseSrc * bsrc) -{ - 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-float", - "endianness", G_TYPE_INT, G_BYTE_ORDER, - "width", G_TYPE_INT, 32, - "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 GstRingBuffer * -gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src) -{ - GstRingBuffer *buffer; - - buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL); - GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer); - - return buffer; -} diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h deleted file mode 100644 index 7e99b69df9..0000000000 --- a/ext/jack/gstjackaudiosrc.h +++ /dev/null @@ -1,97 +0,0 @@ -/* GStreamer - * Copyright (C) 2008 Tristan Matthews - * - * Permission is hereby granted, free of charge, to any person obtaining a - * copy of this software and associated documentation files (the "Software"), - * to deal in the Software without restriction, including without limitation - * the rights to use, copy, modify, merge, publish, distribute, sublicense, - * and/or sell copies of the Software, and to permit persons to whom the - * Software is furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE - * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING - * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER - * DEALINGS IN THE SOFTWARE. - * - * Alternatively, the contents of this file may be used under the - * GNU Lesser General Public License Version 2.1 (the "LGPL"), in - * which case the following provisions apply instead of the ones - * mentioned above: - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_JACK_AUDIO_SRC_H__ -#define __GST_JACK_AUDIO_SRC_H__ - -#include - -#include -#include - -#include "gstjackaudioclient.h" -#include "gstjack.h" - -G_BEGIN_DECLS - -#define GST_TYPE_JACK_AUDIO_SRC (gst_jack_audio_src_get_type()) -#define GST_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc)) -#define GST_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass)) -#define GST_JACK_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass)) -#define GST_IS_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC)) -#define GST_IS_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC)) - -typedef struct _GstJackAudioSrc GstJackAudioSrc; -typedef struct _GstJackAudioSrcClass GstJackAudioSrcClass; - -struct _GstJackAudioSrc -{ - GstBaseAudioSrc src; - - /*< private >*/ - /* cached caps */ - GstCaps *caps; - - /* properties */ - GstJackConnect connect; - gchar *server; - jack_client_t *jclient; - - /* our client */ - GstJackAudioClient *client; - - /* our ports */ - jack_port_t **ports; - int port_count; - sample_t **buffers; -}; - -struct _GstJackAudioSrcClass -{ - GstBaseAudioSrcClass parent_class; -}; - -GType gst_jack_audio_src_get_type (void); - -G_END_DECLS - -#endif /* __GST_JACK_AUDIO_SRC_H__ */ diff --git a/ext/jack/gstjackringbuffer.h b/ext/jack/gstjackringbuffer.h deleted file mode 100644 index 266fdfa31d..0000000000 --- a/ext/jack/gstjackringbuffer.h +++ /dev/null @@ -1,88 +0,0 @@ -/* - * GStreamer - * Copyright (C) 2006 Wim Taymans - * Copyright (C) 2008 Tristan Matthews - * - * Permission is hereby granted, free of charge, to any person obtaining a - * copy of this software and associated documentation files (the "Software"), - * to deal in the Software without restriction, including without limitation - * the rights to use, copy, modify, merge, publish, distribute, sublicense, - * and/or sell copies of the Software, and to permit persons to whom the - * Software is furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE - * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING - * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER - * DEALINGS IN THE SOFTWARE. - * - * Alternatively, the contents of this file may be used under the - * GNU Lesser General Public License Version 2.1 (the "LGPL"), in - * which case the following provisions apply instead of the ones - * mentioned above: - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __GST_JACK_RING_BUFFER_H__ -#define __GST_JACK_RING_BUFFER_H__ - -#define GST_TYPE_JACK_RING_BUFFER (gst_jack_ring_buffer_get_type()) -#define GST_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer)) -#define GST_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass)) -#define GST_JACK_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass)) -#define GST_JACK_RING_BUFFER_CAST(obj) ((GstJackRingBuffer *)obj) -#define GST_IS_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER)) -#define GST_IS_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER)) - -typedef struct _GstJackRingBuffer GstJackRingBuffer; -typedef struct _GstJackRingBufferClass GstJackRingBufferClass; - -struct _GstJackRingBuffer -{ - GstRingBuffer object; - - gint sample_rate; - gint buffer_size; - gint channels; -}; - -struct _GstJackRingBufferClass -{ - GstRingBufferClass parent_class; -}; - -static void gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass); -static void gst_jack_ring_buffer_init(GstJackRingBuffer * ringbuffer, - GstJackRingBufferClass * klass); - -static GstRingBufferClass *ring_parent_class = NULL; - -static gboolean gst_jack_ring_buffer_open_device(GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_close_device(GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_acquire(GstRingBuffer * buf,GstRingBufferSpec * spec); -static gboolean gst_jack_ring_buffer_release(GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_start(GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_pause(GstRingBuffer * buf); -static gboolean gst_jack_ring_buffer_stop(GstRingBuffer * buf); -static guint gst_jack_ring_buffer_delay(GstRingBuffer * buf); - -#endif diff --git a/ext/jack/gstjackutil.c b/ext/jack/gstjackutil.c deleted file mode 100644 index cde84d8e85..0000000000 --- a/ext/jack/gstjackutil.c +++ /dev/null @@ -1,114 +0,0 @@ -/* GStreamer Jack utility functions - * Copyright (C) 2010 Tristan Matthews - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#include "gstjackutil.h" -#include - -static const GstAudioChannelPosition default_positions[8][8] = { - /* 1 channel */ - { - GST_AUDIO_CHANNEL_POSITION_FRONT_MONO, - }, - /* 2 channels */ - { - GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, - GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, - }, - /* 3 channels (2.1) */ - { - GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, - GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, - GST_AUDIO_CHANNEL_POSITION_LFE, /* or FRONT_CENTER for 3.0? */ - }, - /* 4 channels (4.0 or 3.1?) */ - { - GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, - GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, - GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, - GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, - }, - /* 5 channels */ - { - GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, - GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, - GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, - GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, - GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, - }, - /* 6 channels */ - { - GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, - GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, - GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, - GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, - GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, - GST_AUDIO_CHANNEL_POSITION_LFE, - }, - /* 7 channels */ - { - GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, - GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, - GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, - GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, - GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, - GST_AUDIO_CHANNEL_POSITION_LFE, - GST_AUDIO_CHANNEL_POSITION_REAR_CENTER, - }, - /* 8 channels */ - { - GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, - GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, - GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, - GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, - GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, - GST_AUDIO_CHANNEL_POSITION_LFE, - GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, - GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT, - } -}; - - -/* if channels are less than or equal to 8, we set a default layout, - * otherwise set layout to an array of GST_AUDIO_CHANNEL_POSITION_NONE */ -void -gst_jack_set_layout_on_caps (GstCaps ** caps, gint channels) -{ - int c; - GValue pos = { 0 }; - GValue chanpos = { 0 }; - gst_caps_unref (*caps); - - if (channels <= 8) { - g_assert (channels >= 1); - gst_audio_set_channel_positions (gst_caps_get_structure (*caps, 0), - default_positions[channels - 1]); - } else { - g_value_init (&chanpos, GST_TYPE_ARRAY); - g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION); - for (c = 0; c < channels; c++) { - g_value_set_enum (&pos, GST_AUDIO_CHANNEL_POSITION_NONE); - gst_value_array_append_value (&chanpos, &pos); - } - g_value_unset (&pos); - gst_structure_set_value (gst_caps_get_structure (*caps, 0), - "channel-positions", &chanpos); - g_value_unset (&chanpos); - } - gst_caps_ref (*caps); -} diff --git a/ext/jack/gstjackutil.h b/ext/jack/gstjackutil.h deleted file mode 100644 index e330afd5e5..0000000000 --- a/ext/jack/gstjackutil.h +++ /dev/null @@ -1,30 +0,0 @@ -/* GStreamer - * Copyright (C) 2010 Tristan Matthews - * - * gstjackutil.h: - * - * 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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef _GST_JACK_UTIL_H_ -#define _GST_JACK_UTIL_H_ - -#include - -void -gst_jack_set_layout_on_caps (GstCaps **caps, gint channels); - -#endif // _GST_JACK_UTIL_H_ diff --git a/gst-plugins-bad.spec.in b/gst-plugins-bad.spec.in index 6ad7d763cb..33c8bd887d 100644 --- a/gst-plugins-bad.spec.in +++ b/gst-plugins-bad.spec.in @@ -165,7 +165,6 @@ rm -rf $RPM_BUILD_ROOT @USE_DC1394_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstdc1394.so @USE_TIMIDITY_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgsttimidity.so @USE_WILDMIDI_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstwildmidi.so -@USE_JACK_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstjack.so @USE_SNDFILE_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstsndfile.so @USE_CELT_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstcelt.so @USE_MPEG2ENC_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstmpeg2enc.so diff --git a/tests/examples/Makefile.am b/tests/examples/Makefile.am index 4a158d951a..f085c8b423 100644 --- a/tests/examples/Makefile.am +++ b/tests/examples/Makefile.am @@ -1,9 +1,4 @@ if HAVE_GTK -if USE_JACK -JACK_EXAMPLES=jack -else -JACK_EXAMPLES= -endif GTK_EXAMPLES=camerabin mxf scaletempo camerabin2 else GTK_EXAMPLES= @@ -21,7 +16,7 @@ else CAMERABIN2= endif -SUBDIRS= $(DIRECTFB_DIR) $(GTK_EXAMPLES) $(JACK_EXAMPLES) -DIST_SUBDIRS= camerabin $(CAMERABIN2) directfb jack mxf scaletempo +SUBDIRS= $(DIRECTFB_DIR) $(GTK_EXAMPLES) +DIST_SUBDIRS= camerabin $(CAMERABIN2) directfb mxf scaletempo include $(top_srcdir)/common/parallel-subdirs.mak diff --git a/tests/examples/jack/Makefile.am b/tests/examples/jack/Makefile.am deleted file mode 100644 index 4adfd13142..0000000000 --- a/tests/examples/jack/Makefile.am +++ /dev/null @@ -1,6 +0,0 @@ -noinst_PROGRAMS = jack_client - -jack_client_SOURCES = jack_client.c -jack_client_CFLAGS = $(GST_CFLAGS) $(GTK_CFLAGS) $(JACK_CFLAGS) -jack_client_LDFLAGS = $(GST_LIBS) $(GTK_LIBS) $(JACK_LIBS) - diff --git a/tests/examples/jack/jack_client.c b/tests/examples/jack/jack_client.c deleted file mode 100644 index 99599ab5cb..0000000000 --- a/tests/examples/jack/jack_client.c +++ /dev/null @@ -1,79 +0,0 @@ -/* This app demonstrates the creation and use of a jack client in conjunction - * with the jack plugins. This way, an application can control the jack client - * directly. - */ - -#include -#include -#include - -static gboolean -quit_cb (gpointer data) -{ - gtk_main_quit (); - return FALSE; -} - -int -main (int argc, char **argv) -{ - jack_client_t *src_client, *sink_client; - jack_status_t status; - GstElement *pipeline, *src, *sink; - GstStateChangeReturn ret; - - gst_init (&argc, &argv); - - /* create jack clients */ - src_client = jack_client_open ("src_client", JackNoStartServer, &status); - if (src_client == NULL) { - if (status & JackServerFailed) - g_print ("JACK server not running\n"); - else - g_print ("jack_client_open() failed, status = 0x%2.0x\n", status); - return 1; - } - - sink_client = jack_client_open ("sink_client", JackNoStartServer, &status); - if (sink_client == NULL) { - if (status & JackServerFailed) - g_print ("JACK server not running\n"); - else - g_print ("jack_client_open() failed, status = 0x%2.0x\n", status); - return 1; - } - - /* create gst elements */ - pipeline = gst_pipeline_new ("my_pipeline"); - - src = gst_element_factory_make ("jackaudiosrc", NULL); - sink = gst_element_factory_make ("jackaudiosink", NULL); - - g_object_set (src, "client", src_client, NULL); - g_object_set (sink, "client", sink_client, NULL); - - gst_bin_add_many (GST_BIN (pipeline), src, sink, NULL); - - /* link everything together */ - if (!gst_element_link (src, sink)) { - g_print ("Failed to link elements!\n"); - return 1; - } - - /* run */ - ret = gst_element_set_state (pipeline, GST_STATE_PLAYING); - if (ret == GST_STATE_CHANGE_FAILURE) { - g_print ("Failed to start up pipeline!\n"); - return 1; - } - - /* quit after 5 seconds */ - g_timeout_add (5000, (GSourceFunc) quit_cb, NULL); - gtk_main (); - - /* clean up */ - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_object_unref (pipeline); - - return 0; -}