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952 lines
26 KiB
C
952 lines
26 KiB
C
/* GStreamer
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* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstalsasrc.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-alsasrc
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* @see_also: alsasink, alsamixer
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*
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* This element reads data from an audio card using the ALSA API.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
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* ]| Record from a sound card using ALSA and encode to Ogg/Vorbis.
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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 "gstalsasrc.h"
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#include "gstalsadeviceprobe.h"
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#include <gst/gst-i18n-plugin.h>
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#define DEFAULT_PROP_DEVICE "default"
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#define DEFAULT_PROP_DEVICE_NAME ""
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#define DEFAULT_PROP_CARD_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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PROP_CARD_NAME,
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PROP_LAST
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};
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static void gst_alsasrc_init_interfaces (GType type);
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#define gst_alsasrc_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstAlsaSrc, gst_alsasrc,
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GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces (g_define_type_id));
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GST_IMPLEMENT_ALSA_MIXER_METHODS (GstAlsaSrc, gst_alsasrc_mixer);
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static void gst_alsasrc_finalize (GObject * object);
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static void gst_alsasrc_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_alsasrc_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter);
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static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
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static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
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GstRingBufferSpec * spec);
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static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);
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static gboolean gst_alsasrc_close (GstAudioSrc * asrc);
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static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length);
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static guint gst_alsasrc_delay (GstAudioSrc * asrc);
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static void gst_alsasrc_reset (GstAudioSrc * asrc);
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static GstStateChangeReturn gst_alsasrc_change_state (GstElement * element,
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GstStateChange transition);
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static GstFlowReturn gst_alsasrc_create (GstBaseSrc * bsrc, guint64 offset,
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guint length, GstBuffer ** outbuf);
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static GstClockTime gst_alsasrc_get_timestamp (GstAlsaSrc * src);
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/* AlsaSrc signals and args */
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enum
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{
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LAST_SIGNAL
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};
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#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
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# define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
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#else
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# define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
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#endif
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static GstStaticPadTemplate alsasrc_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"endianness = (int) { " ALSA_SRC_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_SRC_FACTORY_ENDIANNESS " }, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 32, "
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"depth = (int) 24, "
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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_SRC_FACTORY_ENDIANNESS " }, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 24, "
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"depth = (int) 24, "
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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_SRC_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_alsasrc_finalize (GObject * object)
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{
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GstAlsaSrc *src = GST_ALSA_SRC (object);
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g_free (src->device);
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g_mutex_free (src->alsa_lock);
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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_alsasrc_init_interfaces (GType type)
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{
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static const GInterfaceInfo mixer_iface_info = {
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(GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
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NULL,
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NULL,
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};
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g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
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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_alsasrc_class_init (GstAlsaSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSrcClass *gstbasesrc_class;
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GstAudioSrcClass *gstaudiosrc_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesrc_class = (GstBaseSrcClass *) klass;
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gstaudiosrc_class = (GstAudioSrcClass *) klass;
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gobject_class->finalize = gst_alsasrc_finalize;
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gobject_class->get_property = gst_alsasrc_get_property;
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gobject_class->set_property = gst_alsasrc_set_property;
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gst_element_class_set_details_simple (gstelement_class,
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"Audio source (ALSA)", "Source/Audio",
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"Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&alsasrc_src_factory));
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
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gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_alsasrc_create);
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gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
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gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
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gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
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gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
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gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
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gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_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_PROP_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",
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DEFAULT_PROP_DEVICE_NAME, 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",
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DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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}
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static GstClockTime
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gst_alsasrc_get_timestamp (GstAlsaSrc * src)
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{
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snd_pcm_status_t *status;
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snd_htimestamp_t tstamp;
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GstClockTime timestamp;
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snd_pcm_uframes_t availmax;
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GST_DEBUG_OBJECT (src, "Getting alsa timestamp!");
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if (!src) {
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GST_ERROR_OBJECT (src, "No alsa handle created yet !");
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return 0;
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}
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if (snd_pcm_status_malloc (&status) != 0) {
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GST_ERROR_OBJECT (src, "snd_pcm_status_malloc failed");
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}
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if (snd_pcm_status (src->handle, status) != 0) {
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GST_ERROR_OBJECT (src, "snd_pcm_status failed");
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}
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/* get high resolution time stamp from driver */
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snd_pcm_status_get_htstamp (status, &tstamp);
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timestamp = GST_TIMESPEC_TO_TIME (tstamp);
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/* Max available frames sets the depth of the buffer */
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availmax = snd_pcm_status_get_avail_max (status);
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/* Compensate the fact that the timestamp references the last sample */
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timestamp -= gst_util_uint64_scale_int (availmax * 2, GST_SECOND, src->rate);
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/* Compensate for the delay until the package is available */
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timestamp += gst_util_uint64_scale_int (snd_pcm_status_get_delay (status),
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GST_SECOND, src->rate);
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snd_pcm_status_free (status);
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GST_DEBUG_OBJECT (src, "ALSA timestamp : %" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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return timestamp;
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}
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static void
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gst_alsasrc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAlsaSrc *src;
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src = GST_ALSA_SRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_free (src->device);
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src->device = g_value_dup_string (value);
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if (src->device == NULL) {
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src->device = g_strdup (DEFAULT_PROP_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_alsasrc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAlsaSrc *src;
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src = GST_ALSA_SRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_value_set_string (value, src->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 (src),
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src->device, src->handle, SND_PCM_STREAM_CAPTURE));
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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 (src),
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src->device, SND_PCM_STREAM_CAPTURE));
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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 GstStateChangeReturn
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gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
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GstAlsaSrc *asrc = GST_ALSA_SRC (element);
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GstClock *clk;
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switch (transition) {
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/* Show the compiler that we care */
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case GST_STATE_CHANGE_NULL_TO_READY:
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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clk = src->clock;
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asrc->driver_timestamps = FALSE;
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if (GST_IS_SYSTEM_CLOCK (clk)) {
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gint clocktype;
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g_object_get (clk, "clock-type", &clocktype, NULL);
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if (clocktype == GST_CLOCK_TYPE_MONOTONIC) {
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asrc->driver_timestamps = TRUE;
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}
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}
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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return ret;
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}
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static GstFlowReturn
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gst_alsasrc_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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GstBuffer ** outbuf)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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GstAlsaSrc *asrc = GST_ALSA_SRC (bsrc);
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ret =
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GST_BASE_SRC_CLASS (parent_class)->create (bsrc, offset, length, outbuf);
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if (asrc->driver_timestamps == TRUE && *outbuf) {
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GST_BUFFER_TIMESTAMP (*outbuf) =
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gst_alsasrc_get_timestamp ((GstAlsaSrc *) bsrc);
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}
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return ret;
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}
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static void
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gst_alsasrc_init (GstAlsaSrc * alsasrc)
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{
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GST_DEBUG_OBJECT (alsasrc, "initializing");
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alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
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alsasrc->cached_caps = NULL;
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alsasrc->driver_timestamps = FALSE;
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alsasrc->alsa_lock = g_mutex_new ();
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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_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
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{
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GstElementClass *element_class;
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GstPadTemplate *pad_template;
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GstAlsaSrc *src;
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GstCaps *caps, *templ_caps;
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src = GST_ALSA_SRC (bsrc);
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if (src->handle == NULL) {
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GST_DEBUG_OBJECT (src, "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 (src->cached_caps) {
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GST_LOG_OBJECT (src, "Returning cached caps");
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if (filter)
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return gst_caps_intersect_full (filter, src->cached_caps,
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GST_CAPS_INTERSECT_FIRST);
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else
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return gst_caps_ref (src->cached_caps);
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}
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element_class = GST_ELEMENT_GET_CLASS (src);
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pad_template = gst_element_class_get_pad_template (element_class, "src");
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g_return_val_if_fail (pad_template != NULL, NULL);
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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 (src), src->handle,
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templ_caps);
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gst_caps_unref (templ_caps);
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if (caps) {
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src->cached_caps = gst_caps_ref (caps);
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}
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GST_INFO_OBJECT (src, "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 int
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set_hwparams (GstAlsaSrc * alsa)
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{
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guint rrate;
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gint err;
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snd_pcm_hw_params_t *params;
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snd_pcm_hw_params_malloc (¶ms);
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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, NULL),
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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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if (alsa->buffer_time != -1) {
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/* set the buffer time */
|
|
CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
|
|
&alsa->buffer_time, NULL), buffer_time);
|
|
}
|
|
if (alsa->period_time != -1) {
|
|
/* set the period time */
|
|
CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
|
|
&alsa->period_time, NULL), period_time);
|
|
}
|
|
|
|
/* write the parameters to device */
|
|
CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
|
|
|
|
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);
|
|
|
|
snd_pcm_hw_params_free (params);
|
|
return 0;
|
|
|
|
/* ERRORS */
|
|
no_config:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Broken configuration for recording: no configurations available: %s",
|
|
snd_strerror (err)));
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
wrong_access:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Access type not available for recording: %s", snd_strerror (err)));
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
no_sample_format:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Sample format not available for recording: %s", snd_strerror (err)));
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
no_channels:
|
|
{
|
|
gchar *msg = NULL;
|
|
|
|
if ((alsa->channels) == 1)
|
|
msg = g_strdup (_("Could not open device for recording in mono mode."));
|
|
if ((alsa->channels) == 2)
|
|
msg = g_strdup (_("Could not open device for recording in stereo mode."));
|
|
if ((alsa->channels) > 2)
|
|
msg =
|
|
g_strdup_printf (_
|
|
("Could not open device for recording in %d-channel mode"),
|
|
alsa->channels);
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
|
|
("%s", snd_strerror (err)));
|
|
g_free (msg);
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
no_rate:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Rate %iHz not available for recording: %s",
|
|
alsa->rate, snd_strerror (err)));
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
rate_match:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
|
|
snd_pcm_hw_params_free (params);
|
|
return -EINVAL;
|
|
}
|
|
buffer_time:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set buffer time %i for recording: %s",
|
|
alsa->buffer_time, snd_strerror (err)));
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
buffer_size:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get buffer size for recording: %s", snd_strerror (err)));
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
period_time:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set period time %i for recording: %s", alsa->period_time,
|
|
snd_strerror (err)));
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
period_size:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get period size for recording: %s", snd_strerror (err)));
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
set_hw_params:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set hw params for recording: %s", snd_strerror (err)));
|
|
snd_pcm_hw_params_free (params);
|
|
return err;
|
|
}
|
|
}
|
|
|
|
static int
|
|
set_swparams (GstAlsaSrc * 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);
|
|
/* 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);
|
|
/* start the transfer on first read */
|
|
CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
|
|
0), start_threshold);
|
|
|
|
#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 recording 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
|
|
alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec)
|
|
{
|
|
switch (spec->type) {
|
|
case GST_BUFTYPE_LINEAR:
|
|
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_alsasrc_open (GstAudioSrc * asrc)
|
|
{
|
|
GstAlsaSrc *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SRC (asrc);
|
|
|
|
CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
|
|
SND_PCM_NONBLOCK), open_error);
|
|
|
|
if (!alsa->mixer)
|
|
alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
open_error:
|
|
{
|
|
if (err == -EBUSY) {
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
|
|
(_("Could not open audio device for recording. "
|
|
"Device is being used by another application.")),
|
|
("Device '%s' is busy", alsa->device));
|
|
} else {
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
|
|
(_("Could not open audio device for recording.")),
|
|
("Recording open error on device '%s': %s", alsa->device,
|
|
snd_strerror (err)));
|
|
}
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
|
|
{
|
|
GstAlsaSrc *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SRC (asrc);
|
|
|
|
if (!alsasrc_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);
|
|
CHECK (snd_pcm_prepare (alsa->handle), prepare_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;
|
|
spec->silence_sample[0] = 0;
|
|
spec->silence_sample[1] = 0;
|
|
spec->silence_sample[2] = 0;
|
|
spec->silence_sample[3] = 0;
|
|
|
|
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;
|
|
}
|
|
prepare_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Prepare failed: %s", snd_strerror (err)));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasrc_unprepare (GstAudioSrc * asrc)
|
|
{
|
|
GstAlsaSrc *alsa;
|
|
|
|
alsa = GST_ALSA_SRC (asrc);
|
|
|
|
snd_pcm_drop (alsa->handle);
|
|
snd_pcm_hw_free (alsa->handle);
|
|
snd_pcm_nonblock (alsa->handle, 1);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_alsasrc_close (GstAudioSrc * asrc)
|
|
{
|
|
GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
|
|
|
|
snd_pcm_close (alsa->handle);
|
|
alsa->handle = NULL;
|
|
|
|
if (alsa->mixer) {
|
|
gst_alsa_mixer_free (alsa->mixer);
|
|
alsa->mixer = NULL;
|
|
}
|
|
|
|
gst_caps_replace (&alsa->cached_caps, NULL);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/*
|
|
* Underrun and suspend recovery
|
|
*/
|
|
static gint
|
|
xrun_recovery (GstAlsaSrc * 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_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
|
|
{
|
|
GstAlsaSrc *alsa;
|
|
gint err;
|
|
gint cptr;
|
|
gint16 *ptr;
|
|
|
|
alsa = GST_ALSA_SRC (asrc);
|
|
|
|
cptr = length / alsa->bytes_per_sample;
|
|
ptr = data;
|
|
|
|
GST_ALSA_SRC_LOCK (asrc);
|
|
while (cptr > 0) {
|
|
if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
|
|
if (err == -EAGAIN) {
|
|
GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
|
|
continue;
|
|
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
|
|
goto read_error;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
ptr += err * alsa->channels;
|
|
cptr -= err;
|
|
}
|
|
GST_ALSA_SRC_UNLOCK (asrc);
|
|
|
|
return length - (cptr * alsa->bytes_per_sample);
|
|
|
|
read_error:
|
|
{
|
|
GST_ALSA_SRC_UNLOCK (asrc);
|
|
return length; /* skip one period */
|
|
}
|
|
}
|
|
|
|
static guint
|
|
gst_alsasrc_delay (GstAudioSrc * asrc)
|
|
{
|
|
GstAlsaSrc *alsa;
|
|
snd_pcm_sframes_t delay;
|
|
int res;
|
|
|
|
alsa = GST_ALSA_SRC (asrc);
|
|
|
|
res = snd_pcm_delay (alsa->handle, &delay);
|
|
if (G_UNLIKELY (res < 0)) {
|
|
GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
|
|
delay = 0;
|
|
}
|
|
|
|
return CLAMP (delay, 0, alsa->buffer_size);
|
|
}
|
|
|
|
static void
|
|
gst_alsasrc_reset (GstAudioSrc * asrc)
|
|
{
|
|
GstAlsaSrc *alsa;
|
|
gint err;
|
|
|
|
alsa = GST_ALSA_SRC (asrc);
|
|
|
|
GST_ALSA_SRC_LOCK (asrc);
|
|
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_SRC_UNLOCK (asrc);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
drop_error:
|
|
{
|
|
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
|
|
snd_strerror (err));
|
|
GST_ALSA_SRC_UNLOCK (asrc);
|
|
return;
|
|
}
|
|
prepare_error:
|
|
{
|
|
GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
|
|
snd_strerror (err));
|
|
GST_ALSA_SRC_UNLOCK (asrc);
|
|
return;
|
|
}
|
|
}
|