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GLib guarantees libintl is always present, using proxy-libintl as last resort. There is no need to mock gettex API any more. This fix static build on Windows because G_INTL_STATIC_COMPILATION must be defined before including libintl.h, and glib does it for us as part as including glib.h. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
1099 lines
32 KiB
C
1099 lines
32 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., 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-alsasrc
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* @title: alsasrc
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* @see_also: alsasink
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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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* ## Example pipelines
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* |[
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* gst-launch-1.0 -v alsasrc ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
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* ]|
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* Record from a sound card using ALSA and encode to Ogg/Vorbis.
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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 "gstalsasrc.h"
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#include <glib/gi18n-lib.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_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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#define DEFAULT_PROP_USE_DRIVER_TIMESTAMP TRUE
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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_USE_DRIVER_TIMESTAMP,
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PROP_LAST
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};
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#define gst_alsasrc_parent_class parent_class
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G_DEFINE_TYPE (GstAlsaSrc, gst_alsasrc, GST_TYPE_AUDIO_SRC);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (alsasrc, "alsasrc", GST_RANK_PRIMARY,
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GST_TYPE_ALSA_SRC, alsa_element_init (plugin));
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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 GstStateChangeReturn gst_alsasrc_change_state (GstElement * element,
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GstStateChange transition);
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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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GstAudioRingBufferSpec * 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
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(GstAudioSrc * asrc, gpointer data, guint length, GstClockTime * timestamp);
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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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/* 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, "
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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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);
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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_clear (&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_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_static_metadata (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_static_pad_template (gstelement_class,
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&alsasrc_src_factory);
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
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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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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
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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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| GST_PARAM_DOC_SHOW_DEFAULT));
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g_object_class_install_property (gobject_class, PROP_USE_DRIVER_TIMESTAMP,
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g_param_spec_boolean ("use-driver-timestamps", "Use driver timestamps",
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"Use driver timestamps or the pipeline clock timestamps",
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DEFAULT_PROP_USE_DRIVER_TIMESTAMP,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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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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case PROP_USE_DRIVER_TIMESTAMP:
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GST_OBJECT_LOCK (src);
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src->use_driver_timestamps = g_value_get_boolean (value);
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GST_OBJECT_UNLOCK (src);
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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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case PROP_USE_DRIVER_TIMESTAMP:
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GST_OBJECT_LOCK (src);
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g_value_set_boolean (value, src->use_driver_timestamps);
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GST_OBJECT_UNLOCK (src);
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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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GstAlsaSrc *alsa = 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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case GST_STATE_CHANGE_NULL_TO_NULL:
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case GST_STATE_CHANGE_READY_TO_READY:
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case GST_STATE_CHANGE_PAUSED_TO_PAUSED:
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case GST_STATE_CHANGE_PLAYING_TO_PLAYING:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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alsa->driver_timestamps = FALSE;
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clk = gst_element_get_clock (element);
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if (clk != NULL) {
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if (G_OBJECT_TYPE (clk) == GST_TYPE_SYSTEM_CLOCK) {
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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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alsa->use_driver_timestamps) {
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GST_INFO ("Using driver timestamps !");
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alsa->driver_timestamps = TRUE;
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} else {
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GST_INFO ("Not using driver timestamps !");
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alsa->driver_timestamps = FALSE;
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}
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}
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gst_object_unref (clk);
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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 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->use_driver_timestamps = DEFAULT_PROP_USE_DRIVER_TIMESTAMP;
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g_mutex_init (&alsasrc->alsa_lock);
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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 GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
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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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GST_INFO_OBJECT (src, "template caps %" GST_PTR_FORMAT, templ_caps);
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caps = gst_alsa_probe_supported_formats (GST_OBJECT (src),
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src->device, src->handle, 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 = 0;
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snd_pcm_hw_params_t *params, *params_copy;
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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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/* 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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#ifndef GST_DISABLE_GST_DEBUG
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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, NULL);
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snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);
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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, NULL);
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snd_pcm_hw_params_get_period_time_max (params, &max, NULL);
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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, NULL);
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snd_pcm_hw_params_get_periods_max (params, &max, NULL);
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GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
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}
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#endif
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/* Keep a copy of initial params struct that can be used later */
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snd_pcm_hw_params_copy (params_copy, params);
|
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/* Following pulseaudio's approach in
|
|
* https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f
|
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* we'll try various configuration to set the buffer time and period time as some
|
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* driver can be picky on the order of the calls.
|
|
*/
|
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if (alsa->period_time != -1 && alsa->buffer_time != -1) {
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if ((snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
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&alsa->period_time, NULL) >= 0)
|
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&& (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
|
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&alsa->buffer_time, NULL) >= 0)) {
|
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GST_DEBUG_OBJECT (alsa, "period time %u buffer time %u set correctly",
|
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alsa->period_time, alsa->buffer_time);
|
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goto success;
|
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}
|
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/* Try the new order with previous params struct as current one might
|
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have partial settings from the order that was tried unsuccessfully */
|
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snd_pcm_hw_params_copy (params, params_copy);
|
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if ((snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
|
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&alsa->buffer_time, NULL) >= 0)
|
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&& (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
|
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&alsa->period_time, NULL) >= 0)) {
|
|
GST_DEBUG_OBJECT (alsa, "buffer time %u period time %u set correctly",
|
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alsa->buffer_time, alsa->period_time);
|
|
goto success;
|
|
}
|
|
}
|
|
if (alsa->period_time != -1) {
|
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snd_pcm_hw_params_copy (params, params_copy);
|
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/* set the period time only */
|
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if ((snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
|
|
&alsa->period_time, NULL) >= 0)) {
|
|
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 only */
|
|
if ((snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
|
|
&alsa->buffer_time, NULL) >= 0)) {
|
|
GST_DEBUG_OBJECT (alsa, "buffer time %u set correctly",
|
|
alsa->buffer_time);
|
|
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);
|
|
CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
|
|
buffer_size);
|
|
GST_DEBUG_OBJECT (alsa, "buffer size : %lu", alsa->buffer_size);
|
|
CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size,
|
|
NULL), period_size);
|
|
GST_DEBUG_OBJECT (alsa, "period size : %lu", alsa->period_size);
|
|
|
|
goto exit;
|
|
/* ERRORS */
|
|
no_config:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Broken configuration for recording: no configurations available: %s",
|
|
snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
wrong_access:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Access type not available for recording: %s", snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
no_sample_format:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Sample format not available for recording: %s", snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
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);
|
|
goto exit;
|
|
}
|
|
no_rate:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Rate %iHz not available for recording: %s",
|
|
alsa->rate, snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
rate_match:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
|
|
err = -EINVAL;
|
|
goto exit;
|
|
}
|
|
buffer_size:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get buffer size for recording: %s", snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
period_size:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to get period size for recording: %s", snd_strerror (err)));
|
|
goto exit;
|
|
}
|
|
set_hw_params:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set hw params for recording: %s", snd_strerror (err)));
|
|
}
|
|
exit:
|
|
{
|
|
snd_pcm_hw_params_free (params);
|
|
snd_pcm_hw_params_free (params_copy);
|
|
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);
|
|
/* use monotonic timestamping */
|
|
CHECK (snd_pcm_sw_params_set_tstamp_mode (alsa->handle, params,
|
|
SND_PCM_TSTAMP_MMAP), tstamp_mode);
|
|
|
|
#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;
|
|
}
|
|
tstamp_mode:
|
|
{
|
|
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
|
|
("Unable to set tstamp mode 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, GstAudioRingBufferSpec * spec)
|
|
{
|
|
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;
|
|
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_SRC
|
|
(alsa)->ringbuffer, alsa_position[alsa->channels - 1]);
|
|
|
|
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,
|
|
(alsa->driver_timestamps) ? 0 : SND_PCM_NONBLOCK), open_error);
|
|
|
|
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, GstAudioRingBufferSpec * 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->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_SRC (alsa)->ringbuffer);
|
|
#endif /* SND_CHMAP_API_VERSION */
|
|
|
|
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;
|
|
|
|
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_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));
|
|
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));
|
|
}
|
|
return 0;
|
|
}
|
|
return err;
|
|
}
|
|
|
|
static GstClockTime
|
|
gst_alsasrc_get_timestamp (GstAlsaSrc * asrc)
|
|
{
|
|
snd_pcm_status_t *status;
|
|
snd_htimestamp_t tstamp;
|
|
GstClockTime timestamp;
|
|
snd_pcm_uframes_t avail;
|
|
gint err = -EPIPE;
|
|
|
|
if (G_UNLIKELY (!asrc)) {
|
|
GST_ERROR_OBJECT (asrc, "No alsa handle created yet !");
|
|
return GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
if (G_UNLIKELY (snd_pcm_status_malloc (&status) != 0)) {
|
|
GST_ERROR_OBJECT (asrc, "snd_pcm_status_malloc failed");
|
|
return GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
if (G_UNLIKELY (snd_pcm_status (asrc->handle, status) != 0)) {
|
|
GST_ERROR_OBJECT (asrc, "snd_pcm_status failed");
|
|
return GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
/* in case an xrun condition has occurred we need to handle this */
|
|
if (snd_pcm_status_get_state (status) != SND_PCM_STATE_RUNNING) {
|
|
if (xrun_recovery (asrc, asrc->handle, err) < 0) {
|
|
GST_WARNING_OBJECT (asrc, "Could not recover from xrun condition !");
|
|
}
|
|
/* reload the status alsa status object, since recovery made it invalid */
|
|
if (G_UNLIKELY (snd_pcm_status (asrc->handle, status) != 0)) {
|
|
GST_ERROR_OBJECT (asrc, "snd_pcm_status failed");
|
|
}
|
|
}
|
|
|
|
/* get high resolution time stamp from driver */
|
|
snd_pcm_status_get_htstamp (status, &tstamp);
|
|
|
|
if (tstamp.tv_sec == 0 && tstamp.tv_nsec == 0)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
timestamp = GST_TIMESPEC_TO_TIME (tstamp);
|
|
|
|
/* max available frames sets the depth of the buffer */
|
|
avail = snd_pcm_status_get_avail (status);
|
|
|
|
/* calculate the timestamp of the next sample to be read */
|
|
timestamp -= gst_util_uint64_scale_int (avail, GST_SECOND, asrc->rate);
|
|
|
|
/* compensate for the fact that we really need the timestamp of the
|
|
* previously read data segment */
|
|
timestamp -= asrc->period_time * 1000;
|
|
|
|
snd_pcm_status_free (status);
|
|
|
|
GST_LOG_OBJECT (asrc, "ALSA timestamp : %" GST_TIME_FORMAT
|
|
", delay %lu", GST_TIME_ARGS (timestamp), avail);
|
|
|
|
return timestamp;
|
|
}
|
|
|
|
static guint
|
|
gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length,
|
|
GstClockTime * timestamp)
|
|
{
|
|
GstAlsaSrc *alsa;
|
|
gint err;
|
|
gint cptr;
|
|
guint8 *ptr = data;
|
|
|
|
alsa = GST_ALSA_SRC (asrc);
|
|
|
|
cptr = length / alsa->bpf;
|
|
|
|
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 (err == -ENODEV) {
|
|
goto device_disappeared;
|
|
} else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
|
|
goto read_error;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
|
|
cptr -= err;
|
|
}
|
|
GST_ALSA_SRC_UNLOCK (asrc);
|
|
|
|
/* if driver timestamps are enabled we need to return this here */
|
|
if (alsa->driver_timestamps && timestamp)
|
|
*timestamp = gst_alsasrc_get_timestamp (alsa);
|
|
|
|
return length - (cptr * alsa->bpf);
|
|
|
|
read_error:
|
|
{
|
|
GST_ALSA_SRC_UNLOCK (asrc);
|
|
return length; /* skip one period */
|
|
}
|
|
device_disappeared:
|
|
{
|
|
GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
|
|
(_("Error recording from audio device. "
|
|
"The device has been disconnected.")), (NULL));
|
|
GST_ALSA_SRC_UNLOCK (asrc);
|
|
return (guint) - 1;
|
|
}
|
|
}
|
|
|
|
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;
|
|
}
|
|
}
|