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1286 lines
38 KiB
C
1286 lines
38 KiB
C
/* GStreamer
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* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
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* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
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*
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* gstalsasink.c:
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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-alsasink
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* @title: alsasink
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* @see_also: alsasrc
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*
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* This element renders audio samples using the ALSA audio API.
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*
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* ## Example pipelines
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* |[
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* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.ogg ! audioconvert ! audioresample ! autoaudiosink
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* ]|
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*
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* Play an Ogg/Vorbis file and output audio via ALSA.
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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 <sys/ioctl.h>
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#include <fcntl.h>
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#include <errno.h>
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#include <unistd.h>
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#include <string.h>
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#include <getopt.h>
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#include <alsa/asoundlib.h>
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#include "gstalsaelements.h"
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#include "gstalsa.h"
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#include "gstalsasink.h"
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#include <gst/audio/gstaudioiec61937.h>
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#include <gst/gst-i18n-plugin.h>
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#ifndef ESTRPIPE
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#define ESTRPIPE EPIPE
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#endif
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#define DEFAULT_DEVICE "default"
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#define DEFAULT_DEVICE_NAME ""
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#define DEFAULT_CARD_NAME ""
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#define SPDIF_PERIOD_SIZE 1536
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#define SPDIF_BUFFER_SIZE 15360
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enum
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{
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PROP_0,
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PROP_DEVICE,
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PROP_DEVICE_NAME,
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PROP_CARD_NAME,
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PROP_LAST
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};
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#define gst_alsasink_parent_class parent_class
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G_DEFINE_TYPE (GstAlsaSink, gst_alsasink, GST_TYPE_AUDIO_SINK);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (alsasink, "alsasink", GST_RANK_PRIMARY,
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GST_TYPE_ALSA_SINK, alsa_element_init (plugin));
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static void gst_alsasink_finalise (GObject * object);
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static void gst_alsasink_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_alsasink_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink, GstCaps * filter);
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static gboolean gst_alsasink_query (GstBaseSink * bsink, GstQuery * query);
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static gboolean gst_alsasink_open (GstAudioSink * asink);
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static gboolean gst_alsasink_prepare (GstAudioSink * asink,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_alsasink_unprepare (GstAudioSink * asink);
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static gboolean gst_alsasink_close (GstAudioSink * asink);
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static gint gst_alsasink_write (GstAudioSink * asink, gpointer data,
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guint length);
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static guint gst_alsasink_delay (GstAudioSink * asink);
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static void gst_alsasink_pause (GstAudioSink * asink);
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static void gst_alsasink_resume (GstAudioSink * asink);
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static void gst_alsasink_stop (GstAudioSink * asink);
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static gboolean gst_alsasink_acceptcaps (GstAlsaSink * alsa, GstCaps * caps);
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static GstBuffer *gst_alsasink_payload (GstAudioBaseSink * sink,
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GstBuffer * buf);
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static gint output_ref; /* 0 */
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static snd_output_t *output; /* NULL */
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static GMutex output_mutex;
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static GstStaticPadTemplate alsasink_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-raw, "
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"format = (string) " GST_AUDIO_FORMATS_ALL ", "
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"layout = (string) interleaved, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
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PASSTHROUGH_CAPS)
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);
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static void
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gst_alsasink_finalise (GObject * object)
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{
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GstAlsaSink *sink = GST_ALSA_SINK (object);
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g_free (sink->device);
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g_mutex_clear (&sink->alsa_lock);
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g_mutex_clear (&sink->delay_lock);
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g_mutex_lock (&output_mutex);
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--output_ref;
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if (output_ref == 0) {
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snd_output_close (output);
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output = NULL;
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}
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g_mutex_unlock (&output_mutex);
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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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gst_alsasink_class_init (GstAlsaSinkClass * 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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GstAudioBaseSinkClass *gstbaseaudiosink_class;
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GstAudioSinkClass *gstaudiosink_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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gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;
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gstaudiosink_class = (GstAudioSinkClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_alsasink_finalise;
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gobject_class->get_property = gst_alsasink_get_property;
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gobject_class->set_property = gst_alsasink_set_property;
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gst_element_class_set_static_metadata (gstelement_class,
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"Audio sink (ALSA)", "Sink/Audio",
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"Output to a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
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gst_element_class_add_static_pad_template (gstelement_class,
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&alsasink_sink_factory);
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gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);
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gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_alsasink_query);
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gstbaseaudiosink_class->payload = GST_DEBUG_FUNCPTR (gst_alsasink_payload);
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gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open);
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gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare);
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gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare);
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gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close);
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gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write);
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gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay);
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gstaudiosink_class->stop = GST_DEBUG_FUNCPTR (gst_alsasink_stop);
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gstaudiosink_class->pause = GST_DEBUG_FUNCPTR (gst_alsasink_pause);
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gstaudiosink_class->resume = GST_DEBUG_FUNCPTR (gst_alsasink_resume);
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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_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
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g_param_spec_string ("device-name", "Device name",
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"Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_CARD_NAME,
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g_param_spec_string ("card-name", "Card name",
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"Human-readable name of the sound card", DEFAULT_CARD_NAME,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_DOC_SHOW_DEFAULT));
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}
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static void
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gst_alsasink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAlsaSink *sink;
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sink = GST_ALSA_SINK (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_free (sink->device);
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sink->device = g_value_dup_string (value);
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/* setting NULL restores the default device */
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if (sink->device == NULL) {
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sink->device = g_strdup (DEFAULT_DEVICE);
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}
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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_alsasink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAlsaSink *sink;
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sink = GST_ALSA_SINK (object);
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switch (prop_id) {
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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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case PROP_DEVICE_NAME:
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g_value_take_string (value,
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gst_alsa_find_device_name (GST_OBJECT_CAST (sink),
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sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK));
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break;
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case PROP_CARD_NAME:
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g_value_take_string (value,
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gst_alsa_find_card_name (GST_OBJECT_CAST (sink),
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sink->device, SND_PCM_STREAM_PLAYBACK));
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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_alsasink_init (GstAlsaSink * alsasink)
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{
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GST_DEBUG_OBJECT (alsasink, "initializing alsasink");
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alsasink->device = g_strdup (DEFAULT_DEVICE);
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alsasink->handle = NULL;
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alsasink->cached_caps = NULL;
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alsasink->is_paused = FALSE;
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alsasink->after_paused = FALSE;
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alsasink->hw_support_pause = FALSE;
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g_mutex_init (&alsasink->alsa_lock);
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g_mutex_init (&alsasink->delay_lock);
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g_mutex_lock (&output_mutex);
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if (output_ref == 0) {
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snd_output_stdio_attach (&output, stdout, 0);
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++output_ref;
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}
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g_mutex_unlock (&output_mutex);
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}
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#define CHECK(call, error) \
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G_STMT_START { \
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if ((err = call) < 0) { \
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GST_WARNING_OBJECT (alsa, "Error %d (%s) calling " #call, err, snd_strerror (err)); \
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goto error; \
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} \
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} G_STMT_END;
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static GstCaps *
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gst_alsasink_getcaps (GstBaseSink * bsink, GstCaps * filter)
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{
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GstElementClass *element_class;
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GstPadTemplate *pad_template;
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GstAlsaSink *sink = GST_ALSA_SINK (bsink);
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GstCaps *caps, *templ_caps;
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GST_OBJECT_LOCK (sink);
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if (sink->handle == NULL) {
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GST_OBJECT_UNLOCK (sink);
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GST_DEBUG_OBJECT (sink, "device not open, using template caps");
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return NULL; /* base class will get template caps for us */
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}
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if (sink->cached_caps) {
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if (filter) {
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caps = gst_caps_intersect_full (filter, sink->cached_caps,
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GST_CAPS_INTERSECT_FIRST);
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GST_OBJECT_UNLOCK (sink);
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GST_LOG_OBJECT (sink, "Returning cached caps %" GST_PTR_FORMAT " with "
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"filter %" GST_PTR_FORMAT " applied: %" GST_PTR_FORMAT,
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sink->cached_caps, filter, caps);
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return caps;
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} else {
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caps = gst_caps_ref (sink->cached_caps);
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GST_OBJECT_UNLOCK (sink);
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GST_LOG_OBJECT (sink, "Returning cached caps %" GST_PTR_FORMAT, caps);
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return caps;
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}
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}
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element_class = GST_ELEMENT_GET_CLASS (sink);
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pad_template = gst_element_class_get_pad_template (element_class, "sink");
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if (pad_template == NULL) {
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GST_OBJECT_UNLOCK (sink);
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g_assert_not_reached ();
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return NULL;
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}
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templ_caps = gst_pad_template_get_caps (pad_template);
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caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->device,
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sink->handle, templ_caps);
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gst_caps_unref (templ_caps);
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if (caps) {
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sink->cached_caps = gst_caps_ref (caps);
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}
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GST_OBJECT_UNLOCK (sink);
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GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);
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if (filter) {
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GstCaps *intersection;
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intersection =
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gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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return intersection;
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} else {
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return caps;
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}
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}
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static gboolean
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gst_alsasink_acceptcaps (GstAlsaSink * alsa, GstCaps * caps)
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{
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GstPad *pad = GST_BASE_SINK (alsa)->sinkpad;
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GstCaps *pad_caps;
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GstStructure *st;
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gboolean ret = FALSE;
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GstAudioRingBufferSpec spec = { 0 };
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pad_caps = gst_pad_query_caps (pad, caps);
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if (!pad_caps || gst_caps_is_empty (pad_caps)) {
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if (pad_caps)
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gst_caps_unref (pad_caps);
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ret = FALSE;
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goto done;
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}
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gst_caps_unref (pad_caps);
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/* If we've not got fixed caps, creating a stream might fail, so let's just
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* return from here with default acceptcaps behaviour */
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if (!gst_caps_is_fixed (caps))
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goto done;
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/* parse helper expects this set, so avoid nasty warning
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* will be set properly later on anyway */
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spec.latency_time = GST_SECOND;
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if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
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goto done;
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/* Make sure input is framed (one frame per buffer) and can be payloaded */
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switch (spec.type) {
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
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{
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gboolean framed = FALSE, parsed = FALSE;
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st = gst_caps_get_structure (caps, 0);
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gst_structure_get_boolean (st, "framed", &framed);
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gst_structure_get_boolean (st, "parsed", &parsed);
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if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
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goto done;
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}
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default:{
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}
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}
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ret = TRUE;
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done:
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gst_caps_replace (&spec.caps, NULL);
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return ret;
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}
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|
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static gboolean
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gst_alsasink_query (GstBaseSink * sink, GstQuery * query)
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{
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GstAlsaSink *alsa = GST_ALSA_SINK (sink);
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gboolean ret;
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_ACCEPT_CAPS:
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{
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GstCaps *caps;
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gst_query_parse_accept_caps (query, &caps);
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ret = gst_alsasink_acceptcaps (alsa, caps);
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gst_query_set_accept_caps_result (query, ret);
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ret = TRUE;
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break;
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}
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default:
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ret = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
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break;
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}
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return ret;
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}
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|
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static int
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set_hwparams (GstAlsaSink * alsa)
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{
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guint rrate;
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gint err = 0;
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snd_pcm_hw_params_t *params, *params_copy;
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|
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snd_pcm_hw_params_malloc (¶ms);
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snd_pcm_hw_params_malloc (¶ms_copy);
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GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "
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"SPDIF (%d)", alsa->channels, alsa->rate,
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snd_pcm_format_name (alsa->format), alsa->iec958);
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/* choose all parameters */
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CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
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/* set the interleaved read/write format */
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CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
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wrong_access);
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/* set the sample format */
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if (alsa->iec958) {
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/* Try to use big endian first else fallback to le and swap bytes */
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if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {
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alsa->format = SND_PCM_FORMAT_S16_LE;
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alsa->need_swap = TRUE;
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GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");
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} else {
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alsa->need_swap = FALSE;
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}
|
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}
|
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CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
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no_sample_format);
|
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/* set the count of channels */
|
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CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
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no_channels);
|
|
/* set the stream rate */
|
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rrate = alsa->rate;
|
|
CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
|
|
no_rate);
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
/* get and dump some limits */
|
|
{
|
|
guint min, max;
|
|
|
|
snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL);
|
|
snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);
|
|
|
|
GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",
|
|
alsa->buffer_time, min, max);
|
|
|
|
snd_pcm_hw_params_get_period_time_min (params, &min, NULL);
|
|
snd_pcm_hw_params_get_period_time_max (params, &max, NULL);
|
|
|
|
GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",
|
|
alsa->period_time, min, max);
|
|
|
|
snd_pcm_hw_params_get_periods_min (params, &min, NULL);
|
|
snd_pcm_hw_params_get_periods_max (params, &max, NULL);
|
|
|
|
GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
|
|
}
|
|
#endif
|
|
/* Keep a copy of initial params struct that can be used later */
|
|
snd_pcm_hw_params_copy (params_copy, params);
|
|
if (!alsa->iec958) {
|
|
/* Following pulseaudio's approach in
|
|
* https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f
|
|
* we'll try various configuration to set the period time and buffer time as some
|
|
* driver can be picky on the order of the calls.
|
|
*/
|
|
if (alsa->buffer_time != -1 && alsa->period_time != -1) {
|
|
if (((err = snd_pcm_hw_params_set_period_time_near (alsa->handle,
|
|
params, &alsa->period_time, NULL)) >= 0)
|
|
&& ((err =
|
|
snd_pcm_hw_params_set_buffer_time_near (alsa->handle,
|
|
params, &alsa->buffer_time, NULL)) >= 0)) {
|
|
GST_DEBUG_OBJECT (alsa, "period time %u buffer time %u set correctly",
|
|
alsa->period_time, alsa->buffer_time);
|
|
goto success;
|
|
}
|
|
/* Try the new order with previous params struct as current one might
|
|
have partial settings from the order that was tried unsuccessfully */
|
|
snd_pcm_hw_params_copy (params, params_copy);
|
|
if (((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle,
|
|
params, &alsa->buffer_time, NULL)) >= 0)
|
|
&& ((err =
|
|
snd_pcm_hw_params_set_period_time_near (alsa->handle,
|
|
params, &alsa->period_time, NULL)) >= 0)) {
|
|
GST_DEBUG_OBJECT (alsa, "buffer time %u period time %u set correctly",
|
|
alsa->buffer_time, alsa->period_time);
|
|
goto success;
|
|
}
|
|
}
|
|
/* now try to configure the period time and buffer time exclusively
|
|
* if both fail we fall back to the defaults */
|
|
if (alsa->period_time != -1) {
|
|
snd_pcm_hw_params_copy (params, params_copy);
|
|
/* set the period time */
|
|
if ((err =
|
|
snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
|
|
&alsa->period_time, NULL)) < 0) {
|
|
GST_DEBUG_OBJECT (alsa, "Unable to set period time %i for playback: %s",
|
|
alsa->period_time, snd_strerror (err));
|
|
} else {
|
|
GST_DEBUG_OBJECT (alsa, "period time %u set correctly",
|
|
alsa->period_time);
|
|
goto success;
|
|
}
|
|
}
|
|
if (alsa->buffer_time != -1) {
|
|
snd_pcm_hw_params_copy (params, params_copy);
|
|
/* set the buffer time */
|
|
if ((err =
|
|
snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
|
|
&alsa->buffer_time, NULL)) < 0) {
|
|
GST_DEBUG_OBJECT (alsa, "Unable to set buffer time %i for playback: %s",
|
|
alsa->buffer_time, snd_strerror (err));
|
|
} else {
|
|
GST_DEBUG_OBJECT (alsa, "buffer time %u set correctly",
|
|
alsa->buffer_time);
|
|
goto success;
|
|
}
|
|
}
|
|
} else {
|
|
/* Set buffer size and period size manually for SPDIF */
|
|
snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;
|
|
snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;
|
|
|
|
CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,
|
|
&buffer_size), buffer_size);
|
|
CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,
|
|
&period_size, NULL), period_size);
|
|
goto success;
|
|
}
|
|
/* Set nothing if all above failed */
|
|
snd_pcm_hw_params_copy (params, params_copy);
|
|
GST_DEBUG_OBJECT (alsa, "Not setting period time and buffer time");
|
|
|
|
success:
|
|
/* write the parameters to device */
|
|
CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
|
|
/* now get the configured values */
|
|
CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
|
|
buffer_size);
|
|
CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size,
|
|
NULL), period_size);
|
|
|
|
GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
|
|
alsa->period_size);
|
|
|
|
/* Check if hardware supports pause */
|
|
alsa->hw_support_pause = snd_pcm_hw_params_can_pause (params);
|
|
GST_DEBUG_OBJECT (alsa, "Hw support pause: %s",
|
|
alsa->hw_support_pause ? "yes" : "no");
|
|
|
|
goto exit;
|
|
/* ERRORS */
|
|
no_config:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Broken configuration for playback: no configurations available: %s",
|
|
snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
wrong_access:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Access type not available for playback: %s", snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
no_sample_format:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Sample format not available for playback: %s", snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
no_channels:
|
|
{
|
|
gchar *msg = NULL;
|
|
|
|
if ((alsa->channels) == 1)
|
|
msg = g_strdup (_("Could not open device for playback in mono mode."));
|
|
if ((alsa->channels) == 2)
|
|
msg = g_strdup (_("Could not open device for playback in stereo mode."));
|
|
if ((alsa->channels) > 2)
|
|
msg =
|
|
g_strdup_printf (_
|
|
("Could not open device for playback in %d-channel mode."),
|
|
alsa->channels);
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
|
|
("%s", snd_strerror (err)));
|
|
g_free (msg);
|
|
goto exit;
|
|
}
|
|
no_rate:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Rate %iHz not available for playback: %s",
|
|
alsa->rate, snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
buffer_size:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get buffer size for playback: %s", snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
period_size:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get period size for playback: %s", snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
set_hw_params:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set hw params for playback: %s", snd_strerror (err)));
|
|
}
|
|
exit:
|
|
{
|
|
snd_pcm_hw_params_free (params);
|
|
snd_pcm_hw_params_free (params_copy);
|
|
return err;
|
|
}
|
|
}
|
|
|
|
static int
|
|
set_swparams (GstAlsaSink * alsa)
|
|
{
|
|
int err;
|
|
snd_pcm_sw_params_t *params;
|
|
|
|
snd_pcm_sw_params_malloc (¶ms);
|
|
|
|
/* get the current swparams */
|
|
CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
|
|
/* start the transfer when the buffer is almost full: */
|
|
/* (buffer_size / avail_min) * avail_min */
|
|
CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
|
|
(alsa->buffer_size / alsa->period_size) * alsa->period_size),
|
|
start_threshold);
|
|
|
|
/* allow the transfer when at least period_size samples can be processed */
|
|
CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
|
|
alsa->period_size), set_avail);
|
|
|
|
#if GST_CHECK_ALSA_VERSION(1,0,16)
|
|
/* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
|
|
#else
|
|
/* align all transfers to 1 sample */
|
|
CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
|
|
#endif
|
|
|
|
/* write the parameters to the playback device */
|
|
CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
|
|
|
|
snd_pcm_sw_params_free (params);
|
|
return 0;
|
|
|
|
/* ERRORS */
|
|
no_config:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to determine current swparams for playback: %s",
|
|
snd_strerror (err)));
|
|
snd_pcm_sw_params_free (params);
|
|
return err;
|
|
}
|
|
start_threshold:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set start threshold mode for playback: %s",
|
|
snd_strerror (err)));
|
|
snd_pcm_sw_params_free (params);
|
|
return err;
|
|
}
|
|
set_avail:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set avail min for playback: %s", snd_strerror (err)));
|
|
snd_pcm_sw_params_free (params);
|
|
return err;
|
|
}
|
|
#if !GST_CHECK_ALSA_VERSION(1,0,16)
|
|
set_align:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set transfer align for playback: %s", snd_strerror (err)));
|
|
snd_pcm_sw_params_free (params);
|
|
return err;
|
|
}
|
|
#endif
|
|
set_sw_params:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set sw params for playback: %s", snd_strerror (err)));
|
|
snd_pcm_sw_params_free (params);
|
|
return err;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
alsasink_parse_spec (GstAlsaSink * alsa, GstAudioRingBufferSpec * spec)
|
|
{
|
|
/* Initialize our boolean */
|
|
alsa->iec958 = FALSE;
|
|
|
|
switch (spec->type) {
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
|
|
switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
|
|
case GST_AUDIO_FORMAT_U8:
|
|
alsa->format = SND_PCM_FORMAT_U8;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S8:
|
|
alsa->format = SND_PCM_FORMAT_S8;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S16LE:
|
|
alsa->format = SND_PCM_FORMAT_S16_LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S16BE:
|
|
alsa->format = SND_PCM_FORMAT_S16_BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U16LE:
|
|
alsa->format = SND_PCM_FORMAT_U16_LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U16BE:
|
|
alsa->format = SND_PCM_FORMAT_U16_BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S24_32LE:
|
|
alsa->format = SND_PCM_FORMAT_S24_LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S24_32BE:
|
|
alsa->format = SND_PCM_FORMAT_S24_BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U24_32LE:
|
|
alsa->format = SND_PCM_FORMAT_U24_LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U24_32BE:
|
|
alsa->format = SND_PCM_FORMAT_U24_BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32LE:
|
|
alsa->format = SND_PCM_FORMAT_S32_LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32BE:
|
|
alsa->format = SND_PCM_FORMAT_S32_BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U32LE:
|
|
alsa->format = SND_PCM_FORMAT_U32_LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U32BE:
|
|
alsa->format = SND_PCM_FORMAT_U32_BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S24LE:
|
|
alsa->format = SND_PCM_FORMAT_S24_3LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S24BE:
|
|
alsa->format = SND_PCM_FORMAT_S24_3BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U24LE:
|
|
alsa->format = SND_PCM_FORMAT_U24_3LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U24BE:
|
|
alsa->format = SND_PCM_FORMAT_U24_3BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S20LE:
|
|
alsa->format = SND_PCM_FORMAT_S20_3LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S20BE:
|
|
alsa->format = SND_PCM_FORMAT_S20_3BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U20LE:
|
|
alsa->format = SND_PCM_FORMAT_U20_3LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U20BE:
|
|
alsa->format = SND_PCM_FORMAT_U20_3BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S18LE:
|
|
alsa->format = SND_PCM_FORMAT_S18_3LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_S18BE:
|
|
alsa->format = SND_PCM_FORMAT_S18_3BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U18LE:
|
|
alsa->format = SND_PCM_FORMAT_U18_3LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_U18BE:
|
|
alsa->format = SND_PCM_FORMAT_U18_3BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32LE:
|
|
alsa->format = SND_PCM_FORMAT_FLOAT_LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32BE:
|
|
alsa->format = SND_PCM_FORMAT_FLOAT_BE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F64LE:
|
|
alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
|
|
break;
|
|
case GST_AUDIO_FORMAT_F64BE:
|
|
alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
|
|
break;
|
|
default:
|
|
goto error;
|
|
}
|
|
break;
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
|
|
alsa->format = SND_PCM_FORMAT_A_LAW;
|
|
break;
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
|
|
alsa->format = SND_PCM_FORMAT_MU_LAW;
|
|
break;
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
|
|
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
|
|
alsa->format = SND_PCM_FORMAT_S16_BE;
|
|
alsa->iec958 = TRUE;
|
|
break;
|
|
default:
|
|
goto error;
|
|
|
|
}
|
|
alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
|
|
alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
|
|
alsa->buffer_time = spec->buffer_time;
|
|
alsa->period_time = spec->latency_time;
|
|
alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
|
|
|
|
if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9)
|
|
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK
|
|
(alsa)->ringbuffer, alsa_position[alsa->channels - 1]);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasink_open (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
/* open in non-blocking mode, we'll use snd_pcm_wait() for space to become
|
|
* available. */
|
|
CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,
|
|
SND_PCM_NONBLOCK), open_error);
|
|
GST_LOG_OBJECT (alsa, "Opened device %s", alsa->device);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
open_error:
|
|
{
|
|
if (err == -EBUSY) {
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
|
|
(_("Could not open audio device for playback. "
|
|
"Device is being used by another application.")),
|
|
("Device '%s' is busy", alsa->device));
|
|
} else {
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,
|
|
(_("Could not open audio device for playback.")),
|
|
("Playback open error on device '%s': %s", alsa->device,
|
|
snd_strerror (err)));
|
|
}
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
if (alsa->iec958) {
|
|
snd_pcm_close (alsa->handle);
|
|
alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa), alsa->device);
|
|
if (G_UNLIKELY (!alsa->handle)) {
|
|
goto no_iec958;
|
|
}
|
|
}
|
|
|
|
if (!alsasink_parse_spec (alsa, spec))
|
|
goto spec_parse;
|
|
|
|
CHECK (set_hwparams (alsa), hw_params_failed);
|
|
CHECK (set_swparams (alsa), sw_params_failed);
|
|
|
|
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
|
|
spec->segsize = alsa->period_size * alsa->bpf;
|
|
spec->segtotal = alsa->buffer_size / alsa->period_size;
|
|
|
|
{
|
|
snd_output_t *out_buf = NULL;
|
|
char *msg = NULL;
|
|
|
|
snd_output_buffer_open (&out_buf);
|
|
snd_pcm_dump_hw_setup (alsa->handle, out_buf);
|
|
snd_output_buffer_string (out_buf, &msg);
|
|
GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
|
|
snd_output_close (out_buf);
|
|
snd_output_buffer_open (&out_buf);
|
|
snd_pcm_dump_sw_setup (alsa->handle, out_buf);
|
|
snd_output_buffer_string (out_buf, &msg);
|
|
GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
|
|
snd_output_close (out_buf);
|
|
}
|
|
|
|
#ifdef SND_CHMAP_API_VERSION
|
|
alsa_detect_channels_mapping (GST_OBJECT (alsa), alsa->handle, spec,
|
|
alsa->channels, GST_AUDIO_BASE_SINK (alsa)->ringbuffer);
|
|
#endif /* SND_CHMAP_API_VERSION */
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_iec958:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL),
|
|
("Could not open IEC958 (SPDIF) device for playback"));
|
|
return FALSE;
|
|
}
|
|
spec_parse:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Error parsing spec"));
|
|
return FALSE;
|
|
}
|
|
hw_params_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Setting of hwparams failed: %s", snd_strerror (err)));
|
|
return FALSE;
|
|
}
|
|
sw_params_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Setting of swparams failed: %s", snd_strerror (err)));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasink_unprepare (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
snd_pcm_drop (alsa->handle);
|
|
snd_pcm_hw_free (alsa->handle);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasink_close (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa = GST_ALSA_SINK (asink);
|
|
|
|
GST_OBJECT_LOCK (asink);
|
|
if (alsa->handle) {
|
|
snd_pcm_close (alsa->handle);
|
|
alsa->handle = NULL;
|
|
}
|
|
gst_caps_replace (&alsa->cached_caps, NULL);
|
|
GST_OBJECT_UNLOCK (asink);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
/*
|
|
* Underrun and suspend recovery
|
|
*/
|
|
static gint
|
|
xrun_recovery (GstAlsaSink * alsa, snd_pcm_t * handle, gint err)
|
|
{
|
|
GST_WARNING_OBJECT (alsa, "xrun recovery %d: %s", err, g_strerror (-err));
|
|
|
|
if (err == -EPIPE) { /* under-run */
|
|
err = snd_pcm_prepare (handle);
|
|
if (err < 0)
|
|
GST_WARNING_OBJECT (alsa,
|
|
"Can't recover from underrun, prepare failed: %s",
|
|
snd_strerror (err));
|
|
gst_audio_base_sink_report_device_failure (GST_AUDIO_BASE_SINK (alsa));
|
|
return 0;
|
|
} else if (err == -ESTRPIPE) {
|
|
while ((err = snd_pcm_resume (handle)) == -EAGAIN)
|
|
g_usleep (100); /* wait until the suspend flag is released */
|
|
|
|
if (err < 0) {
|
|
err = snd_pcm_prepare (handle);
|
|
if (err < 0)
|
|
GST_WARNING_OBJECT (alsa,
|
|
"Can't recover from suspend, prepare failed: %s",
|
|
snd_strerror (err));
|
|
}
|
|
if (err == 0)
|
|
gst_audio_base_sink_report_device_failure (GST_AUDIO_BASE_SINK (alsa));
|
|
return 0;
|
|
}
|
|
return err;
|
|
}
|
|
|
|
static gint
|
|
gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
gint cptr;
|
|
guint8 *ptr = data;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
if (alsa->iec958 && alsa->need_swap) {
|
|
guint i;
|
|
guint16 *ptr_tmp = (guint16 *) ptr;
|
|
|
|
GST_DEBUG_OBJECT (asink, "swapping bytes");
|
|
for (i = 0; i < length / 2; i++) {
|
|
ptr_tmp[i] = GUINT16_SWAP_LE_BE (ptr_tmp[i]);
|
|
}
|
|
}
|
|
|
|
GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length);
|
|
|
|
cptr = length / alsa->bpf;
|
|
|
|
GST_ALSA_SINK_LOCK (asink);
|
|
while (cptr > 0) {
|
|
/* start by doing a blocking wait for free space. Set the timeout
|
|
* to 4 times the period time */
|
|
err = snd_pcm_wait (alsa->handle, (4 * alsa->period_time / 1000));
|
|
if (err < 0) {
|
|
GST_DEBUG_OBJECT (asink, "wait error, %d", err);
|
|
} else {
|
|
GST_DELAY_SINK_LOCK (asink);
|
|
err = snd_pcm_writei (alsa->handle, ptr, cptr);
|
|
GST_DELAY_SINK_UNLOCK (asink);
|
|
}
|
|
|
|
if (err < 0) {
|
|
GST_DEBUG_OBJECT (asink, "Write error: %s (%d)", snd_strerror (err), err);
|
|
if (err == -EAGAIN) {
|
|
/* will continue out of the if/else group */
|
|
} else if (err == -ENODEV) {
|
|
goto device_disappeared;
|
|
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
|
|
goto write_error;
|
|
}
|
|
|
|
/* Unlock so that _reset() can run and break an otherwise infinit loop
|
|
* here */
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
g_thread_yield ();
|
|
GST_ALSA_SINK_LOCK (asink);
|
|
continue;
|
|
} else if (err == 0 && alsa->hw_support_pause) {
|
|
/* We might be already paused, if so, just bail */
|
|
if (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PAUSED)
|
|
break;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr);
|
|
ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
|
|
cptr -= err;
|
|
}
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
|
|
return length - (cptr * alsa->bpf);
|
|
|
|
write_error:
|
|
{
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
return length; /* skip one period */
|
|
}
|
|
device_disappeared:
|
|
{
|
|
GST_ELEMENT_ERROR (asink, RESOURCE, WRITE,
|
|
(_("Error outputting to audio device. "
|
|
"The device has been disconnected.")), (NULL));
|
|
goto write_error;
|
|
}
|
|
}
|
|
|
|
static guint
|
|
gst_alsasink_delay (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
snd_pcm_sframes_t delay;
|
|
int res = 0;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
GST_DELAY_SINK_LOCK (asink);
|
|
if (alsa->is_paused == TRUE) {
|
|
delay = alsa->pos_in_buffer;
|
|
alsa->is_paused = FALSE;
|
|
alsa->after_paused = TRUE;
|
|
} else {
|
|
if (alsa->after_paused == TRUE) {
|
|
delay = alsa->pos_in_buffer;
|
|
alsa->after_paused = FALSE;
|
|
} else {
|
|
res = snd_pcm_delay (alsa->handle, &delay);
|
|
}
|
|
}
|
|
GST_DELAY_SINK_UNLOCK (asink);
|
|
if (G_UNLIKELY (res < 0)) {
|
|
/* on errors, report 0 delay */
|
|
GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
|
|
delay = 0;
|
|
}
|
|
if (G_UNLIKELY (delay < 0)) {
|
|
/* make sure we never return a negative delay */
|
|
GST_WARNING_OBJECT (alsa, "snd_pcm_delay returned negative delay");
|
|
delay = 0;
|
|
}
|
|
|
|
return delay;
|
|
}
|
|
|
|
static void
|
|
gst_alsasink_pause (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
snd_pcm_sframes_t delay;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
if (alsa->hw_support_pause == TRUE) {
|
|
GST_ALSA_SINK_LOCK (asink);
|
|
snd_pcm_delay (alsa->handle, &delay);
|
|
alsa->pos_in_buffer = delay;
|
|
CHECK (snd_pcm_pause (alsa->handle, 1), pause_error);
|
|
GST_DEBUG_OBJECT (alsa, "pause done");
|
|
alsa->is_paused = TRUE;
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
} else {
|
|
gst_alsasink_stop (asink);
|
|
}
|
|
|
|
return;
|
|
|
|
pause_error:
|
|
{
|
|
GST_ERROR_OBJECT (alsa, "alsa-pause: pcm pause error: %s",
|
|
snd_strerror (err));
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_alsasink_resume (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
if (alsa->hw_support_pause == TRUE) {
|
|
GST_ALSA_SINK_LOCK (asink);
|
|
CHECK (snd_pcm_pause (alsa->handle, 0), resume_error);
|
|
GST_DEBUG_OBJECT (alsa, "resume done");
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
}
|
|
|
|
return;
|
|
|
|
resume_error:
|
|
{
|
|
GST_ERROR_OBJECT (alsa, "alsa-resume: pcm resume error: %s",
|
|
snd_strerror (err));
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_alsasink_stop (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
GST_ALSA_SINK_LOCK (asink);
|
|
GST_DEBUG_OBJECT (alsa, "drop");
|
|
CHECK (snd_pcm_drop (alsa->handle), drop_error);
|
|
GST_DEBUG_OBJECT (alsa, "prepare");
|
|
CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
|
|
GST_DEBUG_OBJECT (alsa, "stop done");
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
drop_error:
|
|
{
|
|
GST_ERROR_OBJECT (alsa, "alsa-stop: pcm drop error: %s",
|
|
snd_strerror (err));
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
return;
|
|
}
|
|
prepare_error:
|
|
{
|
|
GST_ERROR_OBJECT (alsa, "alsa-stop: pcm prepare error: %s",
|
|
snd_strerror (err));
|
|
GST_ALSA_SINK_UNLOCK (asink);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_alsasink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
|
|
alsa = GST_ALSA_SINK (sink);
|
|
|
|
if (alsa->iec958) {
|
|
GstBuffer *out;
|
|
gint framesize;
|
|
GstMapInfo iinfo, oinfo;
|
|
|
|
framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
|
|
if (framesize <= 0)
|
|
return NULL;
|
|
|
|
out = gst_buffer_new_and_alloc (framesize);
|
|
|
|
gst_buffer_map (buf, &iinfo, GST_MAP_READ);
|
|
gst_buffer_map (out, &oinfo, GST_MAP_WRITE);
|
|
|
|
if (!gst_audio_iec61937_payload (iinfo.data, iinfo.size,
|
|
oinfo.data, oinfo.size, &sink->ringbuffer->spec, G_BIG_ENDIAN)) {
|
|
gst_buffer_unmap (buf, &iinfo);
|
|
gst_buffer_unmap (out, &oinfo);
|
|
gst_buffer_unref (out);
|
|
return NULL;
|
|
}
|
|
|
|
gst_buffer_unmap (buf, &iinfo);
|
|
gst_buffer_unmap (out, &oinfo);
|
|
|
|
gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_METADATA, 0, -1);
|
|
return out;
|
|
}
|
|
|
|
return gst_buffer_ref (buf);
|
|
}
|