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Original commit message from CVS: * ext/alsa/gstalsasink.c: (gst_alsasink_reset): Ignore errors in reset, these are not fatal. They also grab the element lock which is already taking when this function is called. Fixes #405451.
866 lines
24 KiB
C
866 lines
24 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-alsasink
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* @short_description: play audio to an ALSA device
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* @see_also: alsasrc, alsamixer
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*
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* <refsect2>
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* <para>
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* This element renders raw audio samples using the ALSA api.
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* Play an Ogg/Vorbis file.
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* </para>
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* <programlisting>
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* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink
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* </programlisting>
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* </refsect2>
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*
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* Last reviewed on 2006-03-01 (0.10.4)
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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 "gstalsa.h"
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#include "gstalsasink.h"
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#include "gstalsadeviceprobe.h"
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#include <gst/gst-i18n-plugin.h>
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/* elementfactory information */
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static const GstElementDetails gst_alsasink_details =
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GST_ELEMENT_DETAILS ("Audio sink (ALSA)",
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"Sink/Audio",
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"Output to a sound card via ALSA",
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"Wim Taymans <wim@fluendo.com>");
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#define DEFAULT_DEVICE "default"
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#define DEFAULT_DEVICE_NAME ""
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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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};
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static void gst_alsasink_init_interfaces (GType type);
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GST_BOILERPLATE_FULL (GstAlsaSink, gst_alsasink, GstAudioSink,
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GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces);
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static void gst_alsasink_dispose (GObject * object);
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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);
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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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GstRingBufferSpec * 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 guint 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_reset (GstAudioSink * asink);
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static gint output_ref; /* 0 */
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static snd_output_t *output; /* NULL */
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static GStaticMutex output_mutex = G_STATIC_MUTEX_INIT;
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#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
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# define ALSA_SINK_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
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#else
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# define ALSA_SINK_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
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#endif
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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-int, "
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"endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 32, "
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"depth = (int) 32, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
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"audio/x-raw-int, "
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"endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
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"audio/x-raw-int, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
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);
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static void
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gst_alsasink_dispose (GObject * object)
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{
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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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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_free (sink->alsa_lock);
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g_static_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_static_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_init_interfaces (GType type)
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{
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gst_alsa_type_add_device_property_probe_interface (type);
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}
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static void
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gst_alsasink_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, &gst_alsasink_details);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&alsasink_sink_factory));
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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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GstBaseAudioSinkClass *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 = (GstBaseAudioSinkClass *) 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->dispose = GST_DEBUG_FUNCPTR (gst_alsasink_dispose);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasink_finalise);
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gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasink_get_property);
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasink_set_property);
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gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);
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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->reset = GST_DEBUG_FUNCPTR (gst_alsasink_reset);
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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));
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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));
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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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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, GstAlsaSinkClass * g_class)
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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->alsa_lock = g_mutex_new ();
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g_static_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_static_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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goto error; \
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} G_STMT_END;
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static GstCaps *
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gst_alsasink_getcaps (GstBaseSink * bsink)
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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;
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if (sink->handle == NULL) {
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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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GST_LOG_OBJECT (sink, "Returning cached caps");
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return gst_caps_ref (sink->cached_caps);
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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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g_return_val_if_fail (pad_template != NULL, NULL);
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caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->handle,
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gst_pad_template_get_caps (pad_template));
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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_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);
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return caps;
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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, dir;
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snd_pcm_hw_params_t *params;
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guint period_time, buffer_time;
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snd_pcm_hw_params_alloca (¶ms);
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GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s)",
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alsa->channels, alsa->rate, snd_pcm_format_name (alsa->format));
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/* start with requested values, if we cannot configure alsa for those values,
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* we set these values to -1, which will leave the default alsa values */
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buffer_time = alsa->buffer_time;
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period_time = alsa->period_time;
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retry:
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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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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);
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/* set the stream rate */
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rrate = alsa->rate;
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CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, 0),
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no_rate);
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if (rrate != alsa->rate)
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goto rate_match;
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/* get and dump some limits */
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{
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guint min, max;
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snd_pcm_hw_params_get_buffer_time_min (params, &min, &dir);
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snd_pcm_hw_params_get_buffer_time_max (params, &max, &dir);
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GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",
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alsa->buffer_time, min, max);
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snd_pcm_hw_params_get_period_time_min (params, &min, &dir);
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snd_pcm_hw_params_get_period_time_max (params, &max, &dir);
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GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",
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alsa->period_time, min, max);
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snd_pcm_hw_params_get_periods_min (params, &min, &dir);
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snd_pcm_hw_params_get_periods_max (params, &max, &dir);
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GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
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}
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/* now try to configure the buffer time and period time, if one
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* of those fail, we fall back to the defaults and emit a warning. */
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if (buffer_time != -1) {
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/* set the buffer time */
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if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
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&buffer_time, &dir)) < 0) {
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GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
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("Unable to set buffer time %i for playback: %s",
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buffer_time, snd_strerror (err)));
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/* disable buffer_time the next round */
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buffer_time = -1;
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goto retry;
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}
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}
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if (period_time != -1) {
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/* set the period time */
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if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
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&period_time, &dir)) < 0) {
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GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
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("Unable to set period time %i for playback: %s",
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period_time, snd_strerror (err)));
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/* disable period_time the next round */
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period_time = -1;
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goto retry;
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}
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}
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/* write the parameters to device */
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CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
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/* now get the configured values */
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CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
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buffer_size);
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CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, &dir),
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period_size);
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GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
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alsa->period_size);
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return 0;
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/* ERRORS */
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no_config:
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{
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GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
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("Broken configuration for playback: no configurations available: %s",
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snd_strerror (err)));
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return err;
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}
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wrong_access:
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{
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GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
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("Access type not available for playback: %s", snd_strerror (err)));
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return err;
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}
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no_sample_format:
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{
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GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
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("Sample format not available for playback: %s", snd_strerror (err)));
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return err;
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}
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no_channels:
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{
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gchar *msg = NULL;
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if ((alsa->channels) == 1)
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msg = g_strdup (_("Could not open device for playback in mono mode."));
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if ((alsa->channels) == 2)
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msg = g_strdup (_("Could not open device for playback in stereo mode."));
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if ((alsa->channels) > 2)
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msg =
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g_strdup_printf (_
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("Could not open device for playback in %d-channel mode."),
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alsa->channels);
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GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (msg), (snd_strerror (err)));
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g_free (msg);
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return err;
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}
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no_rate:
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{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Rate %iHz not available for playback: %s",
|
|
alsa->rate, snd_strerror (err)));
|
|
return err;
|
|
}
|
|
rate_match:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
|
|
return -EINVAL;
|
|
}
|
|
buffer_size:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get buffer size for playback: %s", snd_strerror (err)));
|
|
return err;
|
|
}
|
|
period_size:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get period size for playback: %s", snd_strerror (err)));
|
|
return err;
|
|
}
|
|
set_hw_params:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set hw params for playback: %s", snd_strerror (err)));
|
|
return err;
|
|
}
|
|
}
|
|
|
|
static int
|
|
set_swparams (GstAlsaSink * alsa)
|
|
{
|
|
int err;
|
|
snd_pcm_sw_params_t *params;
|
|
|
|
snd_pcm_sw_params_alloca (¶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);
|
|
/* align all transfers to 1 sample */
|
|
CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
|
|
|
|
/* write the parameters to the playback device */
|
|
CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
|
|
|
|
return 0;
|
|
|
|
/* ERRORS */
|
|
no_config:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to determine current swparams for playback: %s",
|
|
snd_strerror (err)));
|
|
return err;
|
|
}
|
|
start_threshold:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set start threshold mode for playback: %s",
|
|
snd_strerror (err)));
|
|
return err;
|
|
}
|
|
set_avail:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set avail min for playback: %s", snd_strerror (err)));
|
|
return err;
|
|
}
|
|
set_align:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set transfer align for playback: %s", snd_strerror (err)));
|
|
return err;
|
|
}
|
|
set_sw_params:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set sw params for playback: %s", snd_strerror (err)));
|
|
return err;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec)
|
|
{
|
|
switch (spec->type) {
|
|
case GST_BUFTYPE_LINEAR:
|
|
GST_DEBUG_OBJECT (alsa,
|
|
"Linear format : depth=%d, width=%d, sign=%d, bigend=%d", spec->depth,
|
|
spec->width, spec->sign, spec->bigend);
|
|
|
|
alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
|
|
spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
|
|
break;
|
|
case GST_BUFTYPE_FLOAT:
|
|
switch (spec->format) {
|
|
case GST_FLOAT32_LE:
|
|
alsa->format = SND_PCM_FORMAT_FLOAT_LE;
|
|
break;
|
|
case GST_FLOAT32_BE:
|
|
alsa->format = SND_PCM_FORMAT_FLOAT_BE;
|
|
break;
|
|
case GST_FLOAT64_LE:
|
|
alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
|
|
break;
|
|
case GST_FLOAT64_BE:
|
|
alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
|
|
break;
|
|
default:
|
|
goto error;
|
|
}
|
|
break;
|
|
case GST_BUFTYPE_A_LAW:
|
|
alsa->format = SND_PCM_FORMAT_A_LAW;
|
|
break;
|
|
case GST_BUFTYPE_MU_LAW:
|
|
alsa->format = SND_PCM_FORMAT_MU_LAW;
|
|
break;
|
|
default:
|
|
goto error;
|
|
|
|
}
|
|
alsa->rate = spec->rate;
|
|
alsa->channels = spec->channels;
|
|
alsa->buffer_time = spec->buffer_time;
|
|
alsa->period_time = spec->latency_time;
|
|
alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasink_open (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,
|
|
SND_PCM_NONBLOCK), open_error);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
open_error:
|
|
{
|
|
if (err == -EBUSY) {
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY, (NULL), ("Device '%s' is busy",
|
|
alsa->device));
|
|
} else {
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,
|
|
(NULL), ("Playback open error on device '%s': %s", alsa->device,
|
|
snd_strerror (err)));
|
|
}
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
if (!alsasink_parse_spec (alsa, spec))
|
|
goto spec_parse;
|
|
|
|
CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
|
|
|
|
CHECK (set_hwparams (alsa), hw_params_failed);
|
|
CHECK (set_swparams (alsa), sw_params_failed);
|
|
|
|
alsa->bytes_per_sample = spec->bytes_per_sample;
|
|
spec->segsize = alsa->period_size * spec->bytes_per_sample;
|
|
spec->segtotal = alsa->buffer_size / alsa->period_size;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
spec_parse:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Error parsing spec"));
|
|
return FALSE;
|
|
}
|
|
non_block:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Could not set device to blocking: %s", snd_strerror (err)));
|
|
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;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
CHECK (snd_pcm_drop (alsa->handle), drop);
|
|
|
|
CHECK (snd_pcm_hw_free (alsa->handle), hw_free);
|
|
|
|
CHECK (snd_pcm_nonblock (alsa->handle, 1), non_block);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
drop:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Could not drop samples: %s", snd_strerror (err)));
|
|
return FALSE;
|
|
}
|
|
hw_free:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Could not free hw params: %s", snd_strerror (err)));
|
|
return FALSE;
|
|
}
|
|
non_block:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Could not set device to nonblocking: %s", snd_strerror (err)));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasink_close (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa = GST_ALSA_SINK (asink);
|
|
gint err;
|
|
|
|
CHECK (snd_pcm_close (alsa->handle), close_error);
|
|
alsa->handle = NULL;
|
|
gst_caps_replace (&alsa->cached_caps, NULL);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
close_error:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, CLOSE, (NULL),
|
|
("Playback close error: %s", snd_strerror (err)));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
|
|
/*
|
|
* Underrun and suspend recovery
|
|
*/
|
|
static gint
|
|
xrun_recovery (GstAlsaSink * alsa, snd_pcm_t * handle, gint err)
|
|
{
|
|
GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
|
|
|
|
if (err == -EPIPE) { /* under-run */
|
|
err = snd_pcm_prepare (handle);
|
|
if (err < 0)
|
|
GST_WARNING_OBJECT (alsa,
|
|
"Can't recovery from underrun, prepare failed: %s",
|
|
snd_strerror (err));
|
|
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 recovery from suspend, prepare failed: %s",
|
|
snd_strerror (err));
|
|
}
|
|
return 0;
|
|
}
|
|
return err;
|
|
}
|
|
|
|
static guint
|
|
gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
gint cptr;
|
|
gint16 *ptr;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
cptr = length / alsa->bytes_per_sample;
|
|
ptr = data;
|
|
|
|
GST_ALSA_LOCK (asink);
|
|
while (cptr > 0) {
|
|
err = snd_pcm_writei (alsa->handle, ptr, cptr);
|
|
|
|
GST_DEBUG_OBJECT (asink, "written %d result %d", cptr, err);
|
|
if (err < 0) {
|
|
GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err));
|
|
if (err == -EAGAIN) {
|
|
continue;
|
|
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
|
|
goto write_error;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
ptr += err * alsa->channels;
|
|
cptr -= err;
|
|
}
|
|
GST_ALSA_UNLOCK (asink);
|
|
|
|
return length - cptr;
|
|
|
|
write_error:
|
|
{
|
|
GST_ALSA_UNLOCK (asink);
|
|
return length; /* skip one period */
|
|
}
|
|
}
|
|
|
|
static guint
|
|
gst_alsasink_delay (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
snd_pcm_sframes_t delay;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
snd_pcm_delay (alsa->handle, &delay);
|
|
|
|
return delay;
|
|
}
|
|
|
|
static void
|
|
gst_alsasink_reset (GstAudioSink * asink)
|
|
{
|
|
GstAlsaSink *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SINK (asink);
|
|
|
|
GST_ALSA_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, "reset done");
|
|
GST_ALSA_UNLOCK (asink);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
drop_error:
|
|
{
|
|
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
|
|
snd_strerror (err));
|
|
GST_ALSA_UNLOCK (asink);
|
|
return;
|
|
}
|
|
prepare_error:
|
|
{
|
|
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
|
|
snd_strerror (err));
|
|
GST_ALSA_UNLOCK (asink);
|
|
return;
|
|
}
|
|
}
|