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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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845 lines
23 KiB
C
845 lines
23 KiB
C
/* Based on a plugin from Martin Soto's Seamless DVD Player.
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* Copyright (C) 2003, 2004 Martin Soto <martinsoto@users.sourceforge.net>
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* 2005-6 Michael Smith <msmith@fluendo.com>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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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 <string.h>
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#include <unistd.h>
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#include <gst/gst.h>
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#include <gst/audio/gstaudioclock.h>
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#include <gst/base/gstbasesink.h>
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#include "alsaspdifsink.h"
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GST_DEBUG_CATEGORY_STATIC (alsaspdifsink_debug);
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#define GST_CAT_DEFAULT (alsaspdifsink_debug)
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/* The magic audio-type we pretend to be for AC3 output */
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#define AC3_CHANNELS 2
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#define AC3_BITS 16
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/* Define AC3 FORMAT as big endian. Fall back to swapping
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* on sound devices that don't support it */
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#define AC3_FORMAT_BE SND_PCM_FORMAT_S16_BE
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#define AC3_FORMAT_LE SND_PCM_FORMAT_S16_LE
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/* The size in bytes of an IEC958 frame. */
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#define IEC958_FRAME_SIZE 6144
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/* Size in bytes of an ALSA PCM frame (4, for this case). */
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#define ALSASPDIFSINK_BYTES_PER_FRAME ((AC3_BITS / 8) * AC3_CHANNELS)
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#define IEC958_SAMPLES_PER_FRAME (IEC958_FRAME_SIZE / ALSASPDIFSINK_BYTES_PER_FRAME)
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#if 0
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/* The duration of a single IEC958 frame. */
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#define IEC958_FRAME_DURATION (32 * GST_MSECOND)
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/* Maximal synchronization difference. Measures will be taken if
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block timestamps differ from actual playing time in more than this
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value. */
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#define MAX_SYNC_DIFF (IEC958_FRAME_DURATION * 0.8)
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/* Playing time for the given number of ALSA PCM frames. */
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#define ALSASPDIFSINK_TIME_PER_FRAMES(sink, frames) \
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(((GstClockTime) (frames) * GST_SECOND) / AC3_RATE)
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/* Number of ALSA PCM frames for the given playing time. */
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#define ALSASPDIFSINK_FRAMES_PER_TIME(sink, time) \
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(((GstClockTime) AC3_RATE * (time)) / GST_SECOND)
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#endif
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/* ElementFactory information. */
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static GstElementDetails alsaspdifsink_details = {
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"S/PDIF ALSA audiosink",
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"Sink/Audio",
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"Feeds audio to S/PDIF interfaces through the ALSA sound driver",
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"Martin Soto <martinsoto@users.sourceforge.net>\n"
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"Michael Smith <msmith@fluendo.com>"
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};
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/* AlsaSPDIFSink signals and args */
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enum
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{
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_CARD,
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PROP_DEVICE
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};
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static GstStaticPadTemplate alsaspdifsink_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-iec958")
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);
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (alsaspdifsink_debug, "alsaspdifsink", 0, \
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"ALSA S/PDIF audio sink element");
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GST_BOILERPLATE_FULL (AlsaSPDIFSink, alsaspdifsink, GstBaseSink,
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GST_TYPE_BASE_SINK, _do_init);
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static void alsaspdifsink_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void alsaspdifsink_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static gboolean alsaspdifsink_event (GstBaseSink * bsink, GstEvent * event);
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static GstFlowReturn alsaspdifsink_render (GstBaseSink * bsink,
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GstBuffer * buf);
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static void alsaspdifsink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
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GstClockTime * start, GstClockTime * end);
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static gboolean alsaspdifsink_set_caps (GstBaseSink * bsink, GstCaps * caps);
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static gboolean alsaspdifsink_open (AlsaSPDIFSink * sink);
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static void alsaspdifsink_close (AlsaSPDIFSink * sink);
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static GstClock *alsaspdifsink_provide_clock (GstElement * elem);
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static GstClockTime alsaspdifsink_get_time (GstClock * clock,
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gpointer user_data);
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static void alsaspdifsink_dispose (GObject * object);
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static void alsaspdifsink_finalize (GObject * object);
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static GstStateChangeReturn alsaspdifsink_change_state (GstElement * element,
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GstStateChange transition);
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static int alsaspdifsink_find_pcm_device (AlsaSPDIFSink * sink);
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static gboolean alsaspdifsink_set_params (AlsaSPDIFSink * sink);
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static snd_pcm_sframes_t alsaspdifsink_delay (AlsaSPDIFSink * sink);
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/* Alsa error handler to suppress messages from within the ALSA library */
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static void ignore_alsa_err (const char *file, int line, const char *function,
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int err, const char *fmt, ...);
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static void
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alsaspdifsink_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_set_details (element_class, &alsaspdifsink_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&alsaspdifsink_sink_factory));
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}
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static void
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alsaspdifsink_class_init (AlsaSPDIFSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSinkClass *gstbasesink_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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gobject_class->set_property = alsaspdifsink_set_property;
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gobject_class->get_property = alsaspdifsink_get_property;
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gobject_class->dispose = alsaspdifsink_dispose;
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gobject_class->finalize = alsaspdifsink_finalize;
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gstelement_class->change_state = alsaspdifsink_change_state;
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gstelement_class->provide_clock = alsaspdifsink_provide_clock;
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gstbasesink_class->event = alsaspdifsink_event;
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gstbasesink_class->render = alsaspdifsink_render;
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gstbasesink_class->get_times = alsaspdifsink_get_times;
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gstbasesink_class->set_caps = alsaspdifsink_set_caps;
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#if 0
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/* We ignore the device property anyway, so don't install it
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* we don't want the user supplying just any device string for us.
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* At most we might want a card number and an iec958.%d device name
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* to attempt */
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g_object_class_install_property (gobject_class, PROP_DEVICE,
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g_param_spec_string ("device", "Device",
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"ALSA device, as defined in an asound configuration file",
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"default", G_PARAM_READWRITE));
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#endif
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g_object_class_install_property (gobject_class, PROP_CARD,
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g_param_spec_int ("card", "Card",
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"ALSA card number for the SPDIF device to use",
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0, G_MAXINT, 0, G_PARAM_READWRITE));
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snd_lib_error_set_handler (ignore_alsa_err);
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}
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static void
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alsaspdifsink_init (AlsaSPDIFSink * sink, AlsaSPDIFSinkClass * g_class)
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{
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/* Create the provided clock. */
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sink->clock = gst_audio_clock_new ("clock", alsaspdifsink_get_time, sink);
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sink->card = 0;
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sink->device = g_strdup ("default");
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}
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static void
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alsaspdifsink_dispose (GObject * object)
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{
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AlsaSPDIFSink *sink = ALSASPDIFSINK (object);
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if (sink->clock)
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gst_object_unref (sink->clock);
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sink->clock = NULL;
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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alsaspdifsink_finalize (GObject * object)
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{
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AlsaSPDIFSink *sink = ALSASPDIFSINK (object);
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g_free (sink->device);
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sink->device = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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alsaspdifsink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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AlsaSPDIFSink *sink;
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sink = ALSASPDIFSINK (object);
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switch (prop_id) {
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/*
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case PROP_DEVICE:
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if(sink->device)
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g_free(sink->device);
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sink->device = g_strdup(g_value_get_string(value));
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break;
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*/
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case PROP_CARD:
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sink->card = g_value_get_int (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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alsaspdifsink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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AlsaSPDIFSink *sink;
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sink = ALSASPDIFSINK (object);
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switch (prop_id) {
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/*
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case PROP_DEVICE:
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g_value_set_string(value, sink->device);
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break;
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*/
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case PROP_CARD:
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g_value_set_int (value, sink->card);
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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 gboolean
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alsaspdifsink_set_caps (GstBaseSink * bsink, GstCaps * caps)
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{
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AlsaSPDIFSink *sink = ALSASPDIFSINK (bsink);
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if (!gst_structure_get_int (gst_caps_get_structure (caps, 0), "rate",
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&sink->rate))
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sink->rate = 48000;
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if (!alsaspdifsink_set_params (sink)) {
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
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("Cannot set ALSA hardware parameters"), GST_ERROR_SYSTEM);
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return FALSE;
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}
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return TRUE;
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}
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static GstClock *
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alsaspdifsink_provide_clock (GstElement * elem)
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{
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AlsaSPDIFSink *sink = ALSASPDIFSINK (elem);
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return GST_CLOCK (gst_object_ref (sink->clock));
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}
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static GstClockTime
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alsaspdifsink_get_time (GstClock * clock, gpointer user_data)
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{
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GstClockTime result;
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snd_pcm_sframes_t raw, delay, samples;
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AlsaSPDIFSink *sink = ALSASPDIFSINK (user_data);
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raw = samples = sink->frames * IEC958_SAMPLES_PER_FRAME;
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delay = alsaspdifsink_delay (sink);
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if (samples > delay)
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samples -= delay;
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else
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samples = 0;
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result = gst_util_uint64_scale_int (samples, GST_SECOND, sink->rate);
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GST_LOG_OBJECT (sink, "Samples raw: %d, delay: %d, real: %d, "
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"Time: %" GST_TIME_FORMAT, (int) raw, (int) delay, (int) samples,
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GST_TIME_ARGS (result));
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return result;
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}
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static gboolean
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alsaspdifsink_open (AlsaSPDIFSink * sink)
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{
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char *pcm_name = sink->device;
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int err;
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char devstr[256]; /* Storage for local 'default' device string */
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/*
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* Try and open our default iec958 device. Fall back to searching on card x
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* if this fails, which should only happen on older alsa setups
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*/
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/* The string will be one of these:
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* SPDIF_CON: Non-audio flag not set:
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* spdif:{AES0 0x0 AES1 0x82 AES2 0x0 AES3 0x2}
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* SPDIF_CON: Non-audio flag set:
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* spdif:{AES0 0x2 AES1 0x82 AES2 0x0 AES3 0x2}
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*/
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sprintf (devstr,
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"iec958:{CARD %d AES0 0x%02x AES1 0x%02x AES2 0x%02x AES3 0x%02x}",
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sink->card,
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IEC958_AES0_NONAUDIO,
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IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
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0, IEC958_AES3_CON_FS_48000);
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GST_DEBUG_OBJECT (sink, "Generated device string \"%s\"", devstr);
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pcm_name = devstr;
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err = snd_pcm_open (&(sink->pcm), pcm_name, SND_PCM_STREAM_PLAYBACK, 0);
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if (err < 0) {
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GST_DEBUG_OBJECT ("Open failed for %s - searching for IEC958 manually\n",
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pcm_name);
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err = alsaspdifsink_find_pcm_device (sink);
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if (err == 0 && sink->pcm == NULL)
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goto open_failed;
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}
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if (err < 0)
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goto failed;
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return TRUE;
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/* ERRORS */
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open_failed:
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{
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
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("Could not open IEC958/SPDIF output device"), GST_ERROR_SYSTEM);
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return FALSE;
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}
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failed:
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{
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
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("snd_pcm_open: %s", snd_strerror (err)), GST_ERROR_SYSTEM);
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return FALSE;
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}
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}
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static gboolean
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alsaspdifsink_set_params (AlsaSPDIFSink * sink)
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{
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snd_pcm_hw_params_t *params;
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unsigned int rate;
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int err;
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snd_pcm_hw_params_malloc (¶ms);
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err = snd_pcm_hw_params_any (sink->pcm, params);
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if (err < 0) {
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
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("Broken configuration for this PCM: "
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"no configurations available"), GST_ERROR_SYSTEM);
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goto __error;
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}
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/* Set interleaved access. */
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err = snd_pcm_hw_params_set_access (sink->pcm, params,
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SND_PCM_ACCESS_RW_INTERLEAVED);
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if (err < 0) {
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
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("Access type not available"), GST_ERROR_SYSTEM);
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goto __error;
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}
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err = snd_pcm_hw_params_set_format (sink->pcm, params, AC3_FORMAT_BE);
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if (err < 0) {
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/* Try LE output and swap data */
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GST_DEBUG_OBJECT (sink, "PCM format S16_BE not supported, trying S16_LE");
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err = snd_pcm_hw_params_set_format (sink->pcm, params, AC3_FORMAT_LE);
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sink->need_swap = TRUE;
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} else
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sink->need_swap = FALSE;
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if (err < 0) {
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
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("Sample format not available"), GST_ERROR_SYSTEM);
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goto __error;
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}
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err = snd_pcm_hw_params_set_channels (sink->pcm, params, AC3_CHANNELS);
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if (err < 0) {
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
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("Channels count not available"), GST_ERROR_SYSTEM);
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goto __error;
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}
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rate = sink->rate;
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GST_DEBUG_OBJECT (sink, "Setting S/PDIF sample rate: %d", rate);
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err = snd_pcm_hw_params_set_rate_near (sink->pcm, params, &rate, 0);
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if (err != 0) {
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
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("Rate not available"), GST_ERROR_SYSTEM);
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goto __error;
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}
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err = snd_pcm_hw_params (sink->pcm, params);
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if (err < 0) {
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GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
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("PCM hw_params failed: %s", snd_strerror (err)), GST_ERROR_SYSTEM);
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goto __error;
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}
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snd_pcm_hw_params_free (params);
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return TRUE;
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|
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/* ERRORS */
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__error:
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{
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snd_pcm_hw_params_free (params);
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return FALSE;
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}
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}
|
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|
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static void
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alsaspdifsink_close (AlsaSPDIFSink * sink)
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{
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if (sink->pcm) {
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snd_pcm_close (sink->pcm);
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sink->pcm = NULL;
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}
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}
|
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|
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/* Try and find an IEC958 PCM device and mixer on card 0 and open it
|
|
* This function is only used on older ALSA installs that don't have the
|
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* correct iec958 alias stuff set up, and relies on there being only
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* one IEC958 PCM device (relies IEC958 in the device name) and one IEC958
|
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* mixer control for doing the settings.
|
|
*/
|
|
static int
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alsaspdifsink_find_pcm_device (AlsaSPDIFSink * sink)
|
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{
|
|
int err = -1, dev, idx, count;
|
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const gchar *ctl_name = "hw:0";
|
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const gchar *spdif_name = SND_CTL_NAME_IEC958 ("", PLAYBACK, NONE);
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int card = sink->card;
|
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gchar pcm_name[24];
|
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snd_pcm_t *pcm = NULL;
|
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snd_ctl_t *ctl = NULL;
|
|
snd_ctl_card_info_t *info = NULL;
|
|
snd_ctl_elem_list_t *clist = NULL;
|
|
snd_ctl_elem_id_t *cid = NULL;
|
|
snd_pcm_info_t *pinfo = NULL;
|
|
|
|
GST_WARNING ("Opening IEC958 named device failed. Trying to autodetect");
|
|
|
|
if ((err = snd_ctl_open (&ctl, ctl_name, card)) < 0)
|
|
return err;
|
|
|
|
snd_ctl_card_info_malloc (&info);
|
|
snd_pcm_info_malloc (&pinfo);
|
|
|
|
/* Find a mixer for IEC958 settings */
|
|
snd_ctl_elem_list_malloc (&clist);
|
|
if ((err = snd_ctl_elem_list (ctl, clist)) < 0)
|
|
goto beach;
|
|
|
|
if ((err =
|
|
snd_ctl_elem_list_alloc_space (clist,
|
|
snd_ctl_elem_list_get_count (clist))) < 0)
|
|
goto beach;
|
|
if ((err = snd_ctl_elem_list (ctl, clist)) < 0)
|
|
goto beach;
|
|
|
|
count = snd_ctl_elem_list_get_used (clist);
|
|
for (idx = 0; idx < count; idx++) {
|
|
if (strstr (snd_ctl_elem_list_get_name (clist, idx), spdif_name) != NULL)
|
|
break;
|
|
}
|
|
if (idx == count) {
|
|
/* No SPDIF mixer availble */
|
|
err = 0;
|
|
goto beach;
|
|
}
|
|
snd_ctl_elem_id_malloc (&cid);
|
|
snd_ctl_elem_list_get_id (clist, idx, cid);
|
|
|
|
/* Now find a PCM device for IEC 958 */
|
|
if ((err = snd_ctl_card_info (ctl, info)) < 0)
|
|
goto beach;
|
|
dev = -1;
|
|
do {
|
|
if (snd_ctl_pcm_next_device (ctl, &dev) < 0)
|
|
goto beach;
|
|
if (dev < 0)
|
|
break; /* No more devices */
|
|
|
|
/* Filter for playback devices */
|
|
snd_pcm_info_set_device (pinfo, dev);
|
|
snd_pcm_info_set_subdevice (pinfo, 0);
|
|
snd_pcm_info_set_stream (pinfo, SND_PCM_STREAM_PLAYBACK);
|
|
if ((err = snd_ctl_pcm_info (ctl, pinfo)) < 0) {
|
|
if (err != -ENOENT)
|
|
goto beach; /* Genuine error */
|
|
|
|
/* Device has no playback streams */
|
|
continue;
|
|
}
|
|
if (strstr (snd_pcm_info_get_name (pinfo), "IEC958") == NULL)
|
|
continue; /* Not the device we are looking for */
|
|
|
|
count = snd_pcm_info_get_subdevices_count (pinfo);
|
|
GST_LOG_OBJECT (sink, "Device %d has %d subdevices\n", dev,
|
|
snd_pcm_info_get_subdevices_count (pinfo));
|
|
for (idx = 0; idx < count; idx++) {
|
|
snd_pcm_info_set_subdevice (pinfo, idx);
|
|
|
|
if ((err = snd_ctl_pcm_info (ctl, pinfo)) < 0)
|
|
goto beach;
|
|
|
|
g_assert (snd_pcm_info_get_stream (pinfo) == SND_PCM_STREAM_PLAYBACK);
|
|
|
|
GST_LOG_OBJECT (sink, "Found playback stream on dev %d sub-d %d\n", dev,
|
|
idx);
|
|
|
|
/* Found a suitable PCM device, let's open it */
|
|
g_snprintf (pcm_name, 24, "hw:%d,%d", card, dev);
|
|
if ((err =
|
|
snd_pcm_open (&(pcm), pcm_name, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
|
|
goto beach;
|
|
|
|
break;
|
|
}
|
|
} while (pcm == NULL);
|
|
|
|
if (pcm != NULL) {
|
|
snd_ctl_elem_value_t *cval;
|
|
snd_aes_iec958_t iec958;
|
|
|
|
/* Have a PCM device and a mixer, set things up */
|
|
snd_ctl_elem_value_malloc (&cval);
|
|
snd_ctl_elem_value_set_id (cval, cid);
|
|
snd_ctl_elem_value_get_iec958 (cval, &iec958);
|
|
iec958.status[0] = IEC958_AES0_NONAUDIO;
|
|
iec958.status[1] = IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER;
|
|
iec958.status[2] = 0;
|
|
iec958.status[3] = IEC958_AES3_CON_FS_48000;
|
|
snd_ctl_elem_value_set_iec958 (cval, &iec958);
|
|
snd_ctl_elem_value_free (cval);
|
|
|
|
sink->pcm = pcm;
|
|
pcm = NULL;
|
|
err = 0;
|
|
}
|
|
|
|
beach:
|
|
if (pcm)
|
|
snd_pcm_close (pcm);
|
|
if (clist)
|
|
snd_ctl_elem_list_clear (clist);
|
|
if (ctl)
|
|
snd_ctl_close (ctl);
|
|
if (clist)
|
|
snd_ctl_elem_list_free (clist);
|
|
if (cid)
|
|
snd_ctl_elem_id_free (cid);
|
|
if (info)
|
|
snd_ctl_card_info_free (info);
|
|
if (pinfo)
|
|
snd_pcm_info_free (pinfo);
|
|
return err;
|
|
}
|
|
|
|
static void
|
|
alsaspdifsink_write_frame (AlsaSPDIFSink * sink, guchar * buf)
|
|
{
|
|
snd_pcm_sframes_t res;
|
|
int num_frames = IEC958_FRAME_SIZE / ALSASPDIFSINK_BYTES_PER_FRAME;
|
|
|
|
/* If we couldn't output big endian when we opened the devic, then
|
|
* we need to swap here */
|
|
if (sink->need_swap) {
|
|
int i;
|
|
guchar tmp;
|
|
|
|
for (i = 0; i < IEC958_FRAME_SIZE; i += 2) {
|
|
tmp = buf[i];
|
|
buf[i] = buf[i + 1];
|
|
buf[i + 1] = tmp;
|
|
}
|
|
}
|
|
|
|
res = 0;
|
|
do {
|
|
if (res == -EPIPE) {
|
|
/* Underrun. */
|
|
GST_INFO_OBJECT (sink, "buffer underrun");
|
|
res = snd_pcm_prepare (sink->pcm);
|
|
} else if (res == -ESTRPIPE) {
|
|
/* Suspend. */
|
|
while ((res = snd_pcm_resume (sink->pcm)) == -EAGAIN) {
|
|
GST_DEBUG_OBJECT (sink, "sleeping for suspend");
|
|
g_usleep (100000);
|
|
}
|
|
|
|
if (res < 0) {
|
|
res = snd_pcm_prepare (sink->pcm);
|
|
}
|
|
}
|
|
|
|
if (res >= 0) {
|
|
res = snd_pcm_writei (sink->pcm, (void *) buf, num_frames);
|
|
}
|
|
|
|
if (res > 0) {
|
|
num_frames -= res;
|
|
}
|
|
|
|
} while (res == -EPIPE || num_frames > 0);
|
|
|
|
sink->frames++;
|
|
|
|
if (res < 0) {
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
|
|
("writei returned error: %s", snd_strerror (res)), GST_ERROR_SYSTEM);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
alsaspdifsink_event (GstBaseSink * bsink, GstEvent * event)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (bsink);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_START:
|
|
snd_pcm_drop (sink->pcm);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
snd_pcm_start (sink->pcm);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
alsaspdifsink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
|
|
GstClockTime * start, GstClockTime * end)
|
|
{
|
|
/* Like GstBaseAudioSink, we set these to NONE */
|
|
*start = GST_CLOCK_TIME_NONE;
|
|
*end = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
static snd_pcm_sframes_t
|
|
alsaspdifsink_delay (AlsaSPDIFSink * sink)
|
|
{
|
|
snd_pcm_sframes_t delay;
|
|
int err;
|
|
|
|
err = snd_pcm_delay (sink->pcm, &delay);
|
|
if (err < 0 || delay < 0) {
|
|
return 0;
|
|
}
|
|
|
|
return delay;
|
|
}
|
|
|
|
#if 0
|
|
static void
|
|
generate_iec958_zero_frame (guchar * buffer)
|
|
{
|
|
/* 2 sync words, 16 bits each */
|
|
buffer[0] = 0xF8;
|
|
buffer[1] = 0x72;
|
|
buffer[2] = 0x4E;
|
|
buffer[3] = 0x1F;
|
|
|
|
/* 16-bit burst-info. Contains data type (zero here, for 'null data'),
|
|
stream number (we output '0' for this always), and a few other bits.
|
|
As it happens, all-zero is the correct value.
|
|
*/
|
|
buffer[4] = 0;
|
|
buffer[5] = 0;
|
|
|
|
/* 16-bit frame size. Also zero */
|
|
buffer[6] = 0;
|
|
buffer[7] = 0;
|
|
|
|
memset (buffer + 8, 0, IEC958_FRAME_SIZE - 8);
|
|
}
|
|
#endif
|
|
|
|
static GstFlowReturn
|
|
alsaspdifsink_render (GstBaseSink * bsink, GstBuffer * buf)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (bsink);
|
|
|
|
#if 0
|
|
GstClockTime next_write;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
|
|
sink->cur_ts = GST_BUFFER_TIMESTAMP (buf);
|
|
|
|
next_write = gst_element_get_time (GST_ELEMENT (sink)) +
|
|
alsaspdifsink_current_delay (sink);
|
|
|
|
/*
|
|
fprintf (stderr, "Drift: % 0.6fs, delay: % 0.6fs\r",
|
|
GST_TIME_ARGS (GST_CLOCK_DIFF (sink->cur_ts, next_write)),
|
|
GST_TIME_ARGS (alsaspdifsink_current_delay (sink)));
|
|
*/
|
|
|
|
/* If we're too far behind, send empty IEC958 frames. */
|
|
if (sink->cur_ts > next_write + MAX_SYNC_DIFF) {
|
|
int frames = (int) (
|
|
((double) (sink->cur_ts - next_write)) /
|
|
(double) IEC958_FRAME_DURATION + 0.5);
|
|
int i;
|
|
|
|
for (i = 0; i < frames; i++) {
|
|
static guchar frame[IEC958_FRAME_SIZE];
|
|
|
|
generate_iec958_zero_frame (frame);
|
|
|
|
alsaspdifsink_write_frame (sink, frame);
|
|
}
|
|
}
|
|
/* If we're too far ahead, just drop this buffer */
|
|
else if (sink->cur_ts + MAX_SYNC_DIFF < next_write) {
|
|
goto end;
|
|
}
|
|
#endif
|
|
|
|
GST_LOG_OBJECT (sink, "Writing %d bytes to spdif out", GST_BUFFER_SIZE (buf));
|
|
if (GST_BUFFER_SIZE (buf) == IEC958_FRAME_SIZE)
|
|
alsaspdifsink_write_frame (sink, GST_BUFFER_DATA (buf));
|
|
else
|
|
GST_WARNING_OBJECT (sink, "Ignoring buffer of incorrect size");
|
|
|
|
#if 0
|
|
end:
|
|
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_DURATION (buf)))
|
|
sink->cur_ts = GST_BUFFER_DURATION (buf);
|
|
#endif
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
/* Drop error output from within alsalib on the floor */
|
|
static void
|
|
ignore_alsa_err (const char *file, int line, const char *function,
|
|
int err, const char *fmt, ...)
|
|
{
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
alsaspdifsink_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
AlsaSPDIFSink *sink = ALSASPDIFSINK (element);
|
|
GstStateChangeReturn ret;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
sink->frames = 0;
|
|
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->clock), 0);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
if (!alsaspdifsink_open (sink)) {
|
|
GST_WARNING_OBJECT (sink, "Failed to open alsa device");
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
GST_INFO_OBJECT (sink, "Parent change_state returned %d", ret);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
alsaspdifsink_close (sink);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
/* no rank so it doesn't get autoplugged by autoaudiosink */
|
|
if (!gst_element_register (plugin, "alsaspdifsink", GST_RANK_NONE,
|
|
GST_TYPE_ALSASPDIFSINK)) {
|
|
return FALSE;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"alsaspdif",
|
|
"Alsa plugin for S/PDIF output",
|
|
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
|