mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-25 01:30:38 +00:00
a06be2e81a
no need to return boolean as it will be always TRUE. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1029>
694 lines
19 KiB
C
694 lines
19 KiB
C
/* GStreamer
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* Copyright (C) 2014 Antonio Ospite <ao2@ao2.it>
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*
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* gstalsamidisrc.c: Source element for ALSA MIDI sequencer events
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-alsamidisrc
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* @title: alsamidisrc
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* @see_also: #GstPushSrc
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*
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* The alsamidisrc element is an element that fetches ALSA MIDI sequencer
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* events and makes them available to elements understanding
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* audio/x-midi-events in their sink pads.
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*
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* It can be used to generate notes from a MIDI input device.
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*
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* ## Example launch line
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* |[
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* gst-launch -v alsamidisrc ports=129:0 ! fluiddec ! audioconvert ! autoaudiosink
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* ]|
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* This pipeline will listen for events from the sequencer device at port 129:0,
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* and generate notes using the fluiddec element.
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "gstalsaelements.h"
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#include "gstalsamidisrc.h"
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GST_DEBUG_CATEGORY_STATIC (gst_alsa_midi_src_debug);
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#define GST_CAT_DEFAULT gst_alsa_midi_src_debug
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/*
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* The MIDI specification declares some status bytes undefined:
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*
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* - 0xF4 System common - Undefined (Reserved)
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* - 0xF5 System common - Undefined (Reserved)
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* - 0xF9 System real-time - Undefined (Reserved)
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* - 0xFD System real-time - Undefined (Reserved)
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*
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* See: http://www.midi.org/techspecs/midimessages.php#2
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*
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* Some other documents define status 0xf9 as a tick message with a period of
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* 10ms:
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*
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* - http://www.blitter.com/~russtopia/MIDI/~jglatt/tech/midispec/tick.htm
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* - http://www.sequencer.de/synth/index.php/MIDI_Format#0xf9_-_MIDI_Tick
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*
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* Even if non-standard it looks like this convention is quite widespread.
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*
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* For instance Fluidsynth uses 0xF9 as a "midi tick" message:
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* http://sourceforge.net/p/fluidsynth/code-git/ci/master/tree/fluidsynth/src/midi/fluid_midi.h#l62
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*
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* And then so does the midiparse element in order to be compatible with
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* Fluidsynth and the fluiddec element.
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*
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* Do the same to behave like midiparse.
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*/
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#define MIDI_TICK 0xf9
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#define MIDI_TICK_PERIOD_MS 10
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/* Functions specific to the Alsa MIDI sequencer API */
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#define DEFAULT_BUFSIZE 65536
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#define DEFAULT_CLIENT_NAME "alsamidisrc"
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static int
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init_seq (GstAlsaMidiSrc * alsamidisrc)
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{
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int ret;
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ret = snd_seq_open (&alsamidisrc->seq, "default", SND_SEQ_OPEN_DUPLEX, 0);
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if (ret < 0) {
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GST_ERROR_OBJECT (alsamidisrc, "Cannot open sequencer - %s",
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snd_strerror (ret));
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goto error;
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}
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/*
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* Prevent Valgrind from reporting cached configuration as memory leaks, see:
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* http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=MEMORY-LEAK;hb=HEAD
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*/
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snd_config_update_free_global ();
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ret = snd_seq_set_client_name (alsamidisrc->seq, DEFAULT_CLIENT_NAME);
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if (ret < 0) {
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GST_ERROR_OBJECT (alsamidisrc, "Cannot set client name - %s",
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snd_strerror (ret));
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goto error_seq_close;
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}
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return 0;
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error_seq_close:
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snd_seq_close (alsamidisrc->seq);
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error:
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return ret;
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}
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/* Parses one or more port addresses from the string */
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static int
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parse_ports (const char *arg, GstAlsaMidiSrc * alsamidisrc)
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{
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gchar **ports_list;
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guint i;
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int ret = 0;
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GST_DEBUG_OBJECT (alsamidisrc, "ports: %s", arg);
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/*
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* Assume that ports are separated by commas.
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*
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* Commas are used instead of spaces because spaces are valid in client
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* names.
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*/
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ports_list = g_strsplit (arg, ",", 0);
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alsamidisrc->port_count = g_strv_length (ports_list);
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alsamidisrc->seq_ports = g_try_new (snd_seq_addr_t, alsamidisrc->port_count);
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if (!alsamidisrc->seq_ports) {
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GST_ERROR_OBJECT (alsamidisrc, "Out of memory");
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ret = -ENOMEM;
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goto out_free_ports_list;
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}
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for (i = 0; i < alsamidisrc->port_count; i++) {
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gchar *port_name = ports_list[i];
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ret = snd_seq_parse_address (alsamidisrc->seq, &alsamidisrc->seq_ports[i],
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port_name);
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if (ret < 0) {
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GST_ERROR_OBJECT (alsamidisrc, "Invalid port %s - %s", port_name,
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snd_strerror (ret));
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goto error_free_seq_ports;
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}
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}
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goto out_free_ports_list;
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error_free_seq_ports:
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g_free (alsamidisrc->seq_ports);
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out_free_ports_list:
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g_strfreev (ports_list);
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return ret;
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}
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static int
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start_queue_timer (GstAlsaMidiSrc * alsamidisrc)
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{
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int ret;
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ret = snd_seq_start_queue (alsamidisrc->seq, alsamidisrc->queue, NULL);
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if (ret < 0) {
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GST_ERROR_OBJECT (alsamidisrc, "Timer event output error: %s",
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snd_strerror (ret));
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return ret;
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}
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ret = snd_seq_drain_output (alsamidisrc->seq);
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if (ret < 0)
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GST_ERROR_OBJECT (alsamidisrc, "Drain output error: %s",
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snd_strerror (ret));
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return ret;
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}
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static void
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schedule_next_tick (GstAlsaMidiSrc * alsamidisrc)
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{
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snd_seq_event_t ev;
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snd_seq_real_time_t time;
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int ret;
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snd_seq_ev_clear (&ev);
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snd_seq_ev_set_source (&ev, 0);
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snd_seq_ev_set_dest (&ev, snd_seq_client_id (alsamidisrc->seq), 0);
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ev.type = SND_SEQ_EVENT_TICK;
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alsamidisrc->tick += 1;
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GST_TIME_TO_TIMESPEC (alsamidisrc->tick * MIDI_TICK_PERIOD_MS * GST_MSECOND,
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time);
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snd_seq_ev_schedule_real (&ev, alsamidisrc->queue, 0, &time);
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ret = snd_seq_event_output (alsamidisrc->seq, &ev);
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if (ret < 0)
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GST_ERROR_OBJECT (alsamidisrc, "Event output error: %s",
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snd_strerror (ret));
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ret = snd_seq_drain_output (alsamidisrc->seq);
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if (ret < 0)
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GST_ERROR_OBJECT (alsamidisrc, "Event drain error: %s", snd_strerror (ret));
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}
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static int
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create_port (GstAlsaMidiSrc * alsamidisrc)
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{
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snd_seq_port_info_t *pinfo;
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int ret;
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snd_seq_port_info_alloca (&pinfo);
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snd_seq_port_info_set_name (pinfo, DEFAULT_CLIENT_NAME);
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snd_seq_port_info_set_type (pinfo,
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SND_SEQ_PORT_TYPE_MIDI_GENERIC | SND_SEQ_PORT_TYPE_APPLICATION);
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snd_seq_port_info_set_capability (pinfo,
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SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE);
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ret = snd_seq_alloc_named_queue (alsamidisrc->seq, DEFAULT_CLIENT_NAME);
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if (ret < 0) {
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GST_ERROR_OBJECT (alsamidisrc, "Cannot allocate queue: %s",
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snd_strerror (ret));
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return ret;
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}
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/*
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* Sequencer queues are "per-system" entities, so it's important to remember
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* the queue id to make sure alsamidisrc refers to this very one in future
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* operations, and not to some other port created by another sequencer user.
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*/
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alsamidisrc->queue = ret;
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snd_seq_port_info_set_timestamping (pinfo, 1);
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snd_seq_port_info_set_timestamp_real (pinfo, 1);
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snd_seq_port_info_set_timestamp_queue (pinfo, alsamidisrc->queue);
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ret = snd_seq_create_port (alsamidisrc->seq, pinfo);
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if (ret < 0) {
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GST_ERROR_OBJECT (alsamidisrc, "Cannot create port - %s",
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snd_strerror (ret));
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return ret;
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}
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/*
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* Conversely, it's not strictly necessary to remember the port id because
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* ports are per-client and alsamidisrc is only creating one port (id = 0).
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*
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* If multiple ports were to be created, the ids could be retrieved with
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* something like:
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*
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* alsamidisrc->port = snd_seq_port_info_get_port(pinfo);
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*/
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ret = start_queue_timer (alsamidisrc);
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if (ret < 0)
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GST_ERROR_OBJECT (alsamidisrc, "Cannot start timer for queue: %d - %s",
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alsamidisrc->queue, snd_strerror (ret));
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return ret;
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}
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static void
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connect_ports (GstAlsaMidiSrc * alsamidisrc)
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{
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int i;
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int ret;
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for (i = 0; i < alsamidisrc->port_count; ++i) {
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ret =
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snd_seq_connect_from (alsamidisrc->seq, 0,
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alsamidisrc->seq_ports[i].client, alsamidisrc->seq_ports[i].port);
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if (ret < 0)
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/* Issue a warning and try the other ports */
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GST_WARNING_OBJECT (alsamidisrc, "Cannot connect from port %d:%d - %s",
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alsamidisrc->seq_ports[i].client, alsamidisrc->seq_ports[i].port,
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snd_strerror (ret));
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}
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}
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/* GStreamer specific functions */
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-midi-event"));
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#define DEFAULT_PORTS NULL
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enum
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{
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PROP_0,
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PROP_PORTS,
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PROP_LAST,
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};
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#define gst_alsa_midi_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstAlsaMidiSrc, gst_alsa_midi_src, GST_TYPE_PUSH_SRC,
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GST_DEBUG_CATEGORY_INIT (gst_alsa_midi_src_debug, "alsamidisrc", 0,
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"alsamidisrc element"));
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (alsamidisrc, "alsamidisrc",
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GST_RANK_PRIMARY, GST_TYPE_ALSA_MIDI_SRC, alsa_element_init (plugin));
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static void gst_alsa_midi_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_alsa_midi_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_alsa_midi_src_start (GstBaseSrc * basesrc);
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static gboolean gst_alsa_midi_src_stop (GstBaseSrc * basesrc);
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static gboolean gst_alsa_midi_src_unlock (GstBaseSrc * basesrc);
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static gboolean gst_alsa_midi_src_unlock_stop (GstBaseSrc * basesrc);
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static void gst_alsa_midi_src_state_changed (GstElement * element,
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GstState oldstate, GstState newstate, GstState pending);
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static GstFlowReturn
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gst_alsa_midi_src_create (GstPushSrc * src, GstBuffer ** buf);
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static void
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gst_alsa_midi_src_class_init (GstAlsaMidiSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSrcClass *gstbase_src_class;
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GstPushSrcClass *gstpush_src_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
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gstbase_src_class = GST_BASE_SRC_CLASS (klass);
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gstpush_src_class = GST_PUSH_SRC_CLASS (klass);
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gobject_class->set_property = gst_alsa_midi_src_set_property;
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gobject_class->get_property = gst_alsa_midi_src_get_property;
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g_object_class_install_property (gobject_class, PROP_PORTS,
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g_param_spec_string ("ports", "Ports",
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"Comma separated list of sequencer ports (e.g. client:port,...)",
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DEFAULT_PORTS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_set_static_metadata (gstelement_class,
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"AlsaMidi Source",
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"Source",
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"Push ALSA MIDI sequencer events around", "Antonio Ospite <ao2@ao2.it>");
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gst_element_class_add_static_pad_template (gstelement_class, &srctemplate);
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gstbase_src_class->start = GST_DEBUG_FUNCPTR (gst_alsa_midi_src_start);
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gstbase_src_class->stop = GST_DEBUG_FUNCPTR (gst_alsa_midi_src_stop);
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gstbase_src_class->unlock = GST_DEBUG_FUNCPTR (gst_alsa_midi_src_unlock);
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gstbase_src_class->unlock_stop =
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GST_DEBUG_FUNCPTR (gst_alsa_midi_src_unlock_stop);
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gstpush_src_class->create = GST_DEBUG_FUNCPTR (gst_alsa_midi_src_create);
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gstelement_class->state_changed =
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GST_DEBUG_FUNCPTR (gst_alsa_midi_src_state_changed);
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}
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static void
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gst_alsa_midi_src_init (GstAlsaMidiSrc * alsamidisrc)
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{
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alsamidisrc->ports = DEFAULT_PORTS;
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gst_base_src_set_format (GST_BASE_SRC (alsamidisrc), GST_FORMAT_TIME);
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gst_base_src_set_live (GST_BASE_SRC (alsamidisrc), TRUE);
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}
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static void
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gst_alsa_midi_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAlsaMidiSrc *src;
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src = GST_ALSA_MIDI_SRC (object);
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switch (prop_id) {
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case PROP_PORTS:
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g_free (src->ports);
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src->ports = g_value_dup_string (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_alsa_midi_src_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstAlsaMidiSrc *src;
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g_return_if_fail (GST_IS_ALSA_MIDI_SRC (object));
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src = GST_ALSA_MIDI_SRC (object);
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switch (prop_id) {
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case PROP_PORTS:
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g_value_set_string (value, src->ports);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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push_buffer (GstAlsaMidiSrc * alsamidisrc, gpointer data, guint size,
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GstClockTime time, GstBufferList * buffer_list)
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{
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gpointer local_data;
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GstBuffer *buffer;
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buffer = gst_buffer_new ();
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GST_BUFFER_DTS (buffer) = time;
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GST_BUFFER_PTS (buffer) = time;
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local_data = g_memdup (data, size);
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gst_buffer_append_memory (buffer,
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gst_memory_new_wrapped (0, local_data, size, 0, size, local_data,
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g_free));
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GST_MEMDUMP_OBJECT (alsamidisrc, "MIDI data:", local_data, size);
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gst_buffer_list_add (buffer_list, buffer);
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}
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static void
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push_tick_buffer (GstAlsaMidiSrc * alsamidisrc, GstClockTime time,
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GstBufferList * buffer_list)
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{
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alsamidisrc->buffer[0] = MIDI_TICK;
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push_buffer (alsamidisrc, alsamidisrc->buffer, 1, time, buffer_list);
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}
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static GstFlowReturn
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gst_alsa_midi_src_create (GstPushSrc * src, GstBuffer ** buf)
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{
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GstAlsaMidiSrc *alsamidisrc;
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GstBufferList *buffer_list;
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GstClockTime time;
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long size_ev = 0;
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int err;
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int ret;
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guint len;
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alsamidisrc = GST_ALSA_MIDI_SRC (src);
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buffer_list = gst_buffer_list_new ();
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poll:
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ret = gst_poll_wait (alsamidisrc->poll, GST_CLOCK_TIME_NONE);
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if (ret <= 0) {
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if (ret < 0 && errno == EBUSY) {
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GST_INFO_OBJECT (alsamidisrc, "flushing");
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gst_buffer_list_unref (buffer_list);
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return GST_FLOW_FLUSHING;
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}
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GST_ERROR_OBJECT (alsamidisrc, "ERROR in poll: %s", strerror (errno));
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} else {
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/* There are events available */
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do {
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snd_seq_event_t *event;
|
|
err = snd_seq_event_input (alsamidisrc->seq, &event);
|
|
if (err < 0)
|
|
break; /* Processed all events */
|
|
|
|
if (event) {
|
|
time = GST_TIMESPEC_TO_TIME (event->time.time) - alsamidisrc->delay;
|
|
|
|
/*
|
|
* Special handling is needed because decoding SND_SEQ_EVENT_TICK is
|
|
* not supported by alsa-lib.
|
|
*/
|
|
if (event->type == SND_SEQ_EVENT_TICK) {
|
|
push_tick_buffer (alsamidisrc, time, buffer_list);
|
|
schedule_next_tick (alsamidisrc);
|
|
continue;
|
|
}
|
|
|
|
size_ev =
|
|
snd_midi_event_decode (alsamidisrc->parser, alsamidisrc->buffer,
|
|
DEFAULT_BUFSIZE, event);
|
|
if (size_ev < 0) {
|
|
/* ENOENT indicates an event that is not a MIDI message, silently skip it */
|
|
if (-ENOENT == size_ev) {
|
|
GST_WARNING_OBJECT (alsamidisrc,
|
|
"Warning: Received non-MIDI message");
|
|
goto poll;
|
|
} else {
|
|
GST_ERROR_OBJECT (alsamidisrc,
|
|
"Error decoding event from ALSA to output: %s",
|
|
strerror (-size_ev));
|
|
goto error;
|
|
}
|
|
} else {
|
|
push_buffer (alsamidisrc, alsamidisrc->buffer, size_ev, time,
|
|
buffer_list);
|
|
}
|
|
}
|
|
} while (err > 0);
|
|
}
|
|
|
|
len = gst_buffer_list_length (buffer_list);
|
|
if (len == 0)
|
|
goto error;
|
|
|
|
/* Pop the _last_ buffer in the list */
|
|
*buf = gst_buffer_copy (gst_buffer_list_get (buffer_list, len - 1));
|
|
gst_buffer_list_remove (buffer_list, len - 1, 1);
|
|
--len;
|
|
|
|
/*
|
|
* If there are no more buffers left, free the list, otherwise push all the
|
|
* _previous_ buffers left in the list.
|
|
*
|
|
* The one popped above will be pushed last when this function returns.
|
|
*/
|
|
if (len == 0)
|
|
gst_buffer_list_unref (buffer_list);
|
|
else
|
|
gst_pad_push_list (GST_BASE_SRC (src)->srcpad, buffer_list);
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
error:
|
|
gst_buffer_list_unref (buffer_list);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsa_midi_src_start (GstBaseSrc * basesrc)
|
|
{
|
|
GstAlsaMidiSrc *alsamidisrc;
|
|
int ret;
|
|
|
|
alsamidisrc = GST_ALSA_MIDI_SRC (basesrc);
|
|
|
|
alsamidisrc->tick = 0;
|
|
alsamidisrc->port_count = 0;
|
|
|
|
ret = init_seq (alsamidisrc);
|
|
if (ret < 0)
|
|
goto err;
|
|
|
|
if (alsamidisrc->ports) {
|
|
ret = parse_ports (alsamidisrc->ports, alsamidisrc);
|
|
if (ret < 0)
|
|
goto error_seq_close;
|
|
}
|
|
|
|
ret = create_port (alsamidisrc);
|
|
if (ret < 0)
|
|
goto error_free_seq_ports;
|
|
|
|
connect_ports (alsamidisrc);
|
|
|
|
ret = snd_seq_nonblock (alsamidisrc->seq, 1);
|
|
if (ret < 0) {
|
|
GST_ERROR_OBJECT (alsamidisrc, "Cannot set nonblock mode - %s",
|
|
snd_strerror (ret));
|
|
goto error_free_seq_ports;
|
|
}
|
|
|
|
snd_midi_event_new (DEFAULT_BUFSIZE, &alsamidisrc->parser);
|
|
snd_midi_event_init (alsamidisrc->parser);
|
|
snd_midi_event_reset_decode (alsamidisrc->parser);
|
|
|
|
snd_midi_event_no_status (alsamidisrc->parser, 1);
|
|
|
|
alsamidisrc->buffer = g_try_malloc (DEFAULT_BUFSIZE);
|
|
if (alsamidisrc->buffer == NULL)
|
|
goto error_free_parser;
|
|
|
|
{
|
|
struct pollfd *pfds;
|
|
int npfds, i;
|
|
|
|
npfds = snd_seq_poll_descriptors_count (alsamidisrc->seq, POLLIN);
|
|
pfds = g_newa (struct pollfd, npfds);
|
|
|
|
snd_seq_poll_descriptors (alsamidisrc->seq, pfds, npfds, POLLIN);
|
|
|
|
alsamidisrc->poll = gst_poll_new (TRUE);
|
|
for (i = 0; i < npfds; ++i) {
|
|
GstPollFD fd = GST_POLL_FD_INIT;
|
|
|
|
fd.fd = pfds[i].fd;
|
|
gst_poll_add_fd (alsamidisrc->poll, &fd);
|
|
gst_poll_fd_ctl_read (alsamidisrc->poll, &fd, TRUE);
|
|
gst_poll_fd_ctl_write (alsamidisrc->poll, &fd, FALSE);
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
error_free_parser:
|
|
snd_midi_event_free (alsamidisrc->parser);
|
|
error_free_seq_ports:
|
|
g_free (alsamidisrc->seq_ports);
|
|
error_seq_close:
|
|
snd_seq_close (alsamidisrc->seq);
|
|
err:
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsa_midi_src_stop (GstBaseSrc * basesrc)
|
|
{
|
|
GstAlsaMidiSrc *alsamidisrc;
|
|
|
|
alsamidisrc = GST_ALSA_MIDI_SRC (basesrc);
|
|
|
|
if (alsamidisrc->poll != NULL) {
|
|
gst_poll_free (alsamidisrc->poll);
|
|
alsamidisrc->poll = NULL;
|
|
}
|
|
g_free (alsamidisrc->ports);
|
|
g_free (alsamidisrc->buffer);
|
|
snd_midi_event_free (alsamidisrc->parser);
|
|
g_free (alsamidisrc->seq_ports);
|
|
snd_seq_close (alsamidisrc->seq);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsa_midi_src_unlock (GstBaseSrc * basesrc)
|
|
{
|
|
GstAlsaMidiSrc *alsamidisrc = GST_ALSA_MIDI_SRC (basesrc);
|
|
|
|
gst_poll_set_flushing (alsamidisrc->poll, TRUE);
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsa_midi_src_unlock_stop (GstBaseSrc * basesrc)
|
|
{
|
|
GstAlsaMidiSrc *alsamidisrc = GST_ALSA_MIDI_SRC (basesrc);
|
|
|
|
gst_poll_set_flushing (alsamidisrc->poll, FALSE);
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_alsa_midi_src_state_changed (GstElement * element, GstState oldstate,
|
|
GstState newstate, GstState pending)
|
|
{
|
|
GstAlsaMidiSrc *alsamidisrc;
|
|
|
|
alsamidisrc = GST_ALSA_MIDI_SRC (element);
|
|
|
|
if (newstate == GST_STATE_PLAYING) {
|
|
GstClockTime gst_time;
|
|
GstClockTime base_time;
|
|
GstClockTime running_time;
|
|
GstClockTime queue_time;
|
|
GstClock *clock;
|
|
snd_seq_queue_status_t *status;
|
|
|
|
clock = gst_element_get_clock (element);
|
|
if (clock == NULL) {
|
|
GST_WARNING_OBJECT (element, "No clock present");
|
|
return;
|
|
}
|
|
gst_time = gst_clock_get_time (clock);
|
|
gst_object_unref (clock);
|
|
base_time = gst_element_get_base_time (element);
|
|
running_time = gst_time - base_time;
|
|
|
|
snd_seq_queue_status_malloc (&status);
|
|
snd_seq_get_queue_status (alsamidisrc->seq, alsamidisrc->queue, status);
|
|
queue_time =
|
|
GST_TIMESPEC_TO_TIME (*snd_seq_queue_status_get_real_time (status));
|
|
snd_seq_queue_status_free (status);
|
|
|
|
/*
|
|
* The fact that the ALSA sequencer queue started before the pipeline
|
|
* transition to the PLAYING state ensures that the pipeline delay is
|
|
* always positive.
|
|
*/
|
|
alsamidisrc->delay = queue_time - running_time;
|
|
|
|
if (alsamidisrc->tick == 0) {
|
|
schedule_next_tick (alsamidisrc);
|
|
}
|
|
}
|
|
}
|