mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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844 lines
23 KiB
C++
844 lines
23 KiB
C++
/* GStreamer
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* Copyright (C) 2007 Sebastien Moutte <sebastien@moutte.net>
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*
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* gstdshowaudiosrc.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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstdshowaudiosrc.h"
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GST_DEBUG_CATEGORY_STATIC (dshowaudiosrc_debug);
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#define GST_CAT_DEFAULT dshowaudiosrc_debug
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static GstStaticPadTemplate src_template = 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) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
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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, 2 ]; "
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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, 2 ]")
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);
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static void gst_dshowaudiosrc_init_interfaces (GType type);
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GST_BOILERPLATE_FULL (GstDshowAudioSrc, gst_dshowaudiosrc, GstAudioSrc,
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GST_TYPE_AUDIO_SRC, gst_dshowaudiosrc_init_interfaces);
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enum
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{
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PROP_0,
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PROP_DEVICE,
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PROP_DEVICE_NAME
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};
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static void gst_dshowaudiosrc_probe_interface_init (GstPropertyProbeInterface *
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iface);
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static const GList *gst_dshowaudiosrc_probe_get_properties (GstPropertyProbe *
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probe);
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static GValueArray *gst_dshowaudiosrc_probe_get_values (GstPropertyProbe *
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probe, guint prop_id, const GParamSpec * pspec);
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static GValueArray *gst_dshowaudiosrc_get_device_name_values (GstDshowAudioSrc *
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src);
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static void gst_dshowaudiosrc_dispose (GObject * gobject);
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static void gst_dshowaudiosrc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_dshowaudiosrc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstCaps *gst_dshowaudiosrc_get_caps (GstBaseSrc * src);
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static GstStateChangeReturn gst_dshowaudiosrc_change_state (GstElement *
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element, GstStateChange transition);
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static gboolean gst_dshowaudiosrc_open (GstAudioSrc * asrc);
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static gboolean gst_dshowaudiosrc_prepare (GstAudioSrc * asrc,
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GstRingBufferSpec * spec);
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static gboolean gst_dshowaudiosrc_unprepare (GstAudioSrc * asrc);
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static gboolean gst_dshowaudiosrc_close (GstAudioSrc * asrc);
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static guint gst_dshowaudiosrc_read (GstAudioSrc * asrc, gpointer data,
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guint length);
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static guint gst_dshowaudiosrc_delay (GstAudioSrc * asrc);
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static void gst_dshowaudiosrc_reset (GstAudioSrc * asrc);
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/* utils */
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static GstCaps *gst_dshowaudiosrc_getcaps_from_streamcaps (GstDshowAudioSrc *
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src, IPin * pin, IAMStreamConfig * streamcaps);
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static gboolean gst_dshowaudiosrc_push_buffer (guint8 * buffer, guint size,
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gpointer src_object, GstClockTime duration);
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static void
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gst_dshowaudiosrc_init_interfaces (GType type)
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{
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static const GInterfaceInfo dshowaudiosrc_info = {
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(GInterfaceInitFunc) gst_dshowaudiosrc_probe_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,
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GST_TYPE_PROPERTY_PROBE, &dshowaudiosrc_info);
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}
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static void
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gst_dshowaudiosrc_probe_interface_init (GstPropertyProbeInterface * iface)
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{
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iface->get_properties = gst_dshowaudiosrc_probe_get_properties;
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/* iface->needs_probe = gst_dshowaudiosrc_probe_needs_probe;
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iface->probe_property = gst_dshowaudiosrc_probe_probe_property;*/
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iface->get_values = gst_dshowaudiosrc_probe_get_values;
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}
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static void
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gst_dshowaudiosrc_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (element_class,
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"Directshow audio capture source", "Source/Audio",
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"Receive data from a directshow audio capture graph",
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"Sebastien Moutte <sebastien@moutte.net>");
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}
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static void
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gst_dshowaudiosrc_class_init (GstDshowAudioSrcClass * 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->dispose = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_dispose);
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_set_property);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_get_property);
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_change_state);
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_get_caps);
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gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_open);
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gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_prepare);
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gstaudiosrc_class->unprepare =
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GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_unprepare);
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gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_close);
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gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_read);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_delay);
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gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_reset);
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g_object_class_install_property
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(gobject_class, PROP_DEVICE,
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g_param_spec_string ("device", "Device",
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"Directshow device reference (classID/name)", NULL,
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static_cast < GParamFlags > (G_PARAM_READWRITE)));
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g_object_class_install_property
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(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", NULL,
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static_cast < GParamFlags > (G_PARAM_READWRITE)));
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GST_DEBUG_CATEGORY_INIT (dshowaudiosrc_debug, "dshowaudiosrc", 0,
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"Directshow audio source");
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}
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static void
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gst_dshowaudiosrc_init (GstDshowAudioSrc * src, GstDshowAudioSrcClass * klass)
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{
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src->device = NULL;
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src->device_name = NULL;
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src->audio_cap_filter = NULL;
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src->dshow_fakesink = NULL;
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src->media_filter = NULL;
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src->filter_graph = NULL;
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src->caps = NULL;
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src->pins_mediatypes = NULL;
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src->gbarray = g_byte_array_new ();
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src->gbarray_lock = g_mutex_new ();
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src->is_running = FALSE;
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CoInitializeEx (NULL, COINIT_MULTITHREADED);
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}
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static void
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gst_dshowaudiosrc_dispose (GObject * gobject)
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{
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GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (gobject);
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if (src->device) {
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g_free (src->device);
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src->device = NULL;
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}
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if (src->device_name) {
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g_free (src->device_name);
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src->device_name = NULL;
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}
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if (src->caps) {
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gst_caps_unref (src->caps);
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src->caps = NULL;
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}
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if (src->pins_mediatypes) {
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gst_dshow_free_pins_mediatypes (src->pins_mediatypes);
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src->pins_mediatypes = NULL;
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}
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if (src->gbarray) {
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g_byte_array_free (src->gbarray, TRUE);
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src->gbarray = NULL;
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}
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if (src->gbarray_lock) {
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g_mutex_free (src->gbarray_lock);
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src->gbarray_lock = NULL;
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}
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/* clean dshow */
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if (src->audio_cap_filter)
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src->audio_cap_filter->Release ();
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CoUninitialize ();
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G_OBJECT_CLASS (parent_class)->dispose (gobject);
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}
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static const GList *
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gst_dshowaudiosrc_probe_get_properties (GstPropertyProbe * probe)
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{
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GObjectClass *klass = G_OBJECT_GET_CLASS (probe);
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static GList *props = NULL;
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if (!props) {
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GParamSpec *pspec;
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pspec = g_object_class_find_property (klass, "device-name");
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props = g_list_append (props, pspec);
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}
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return props;
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}
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static GValueArray *
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gst_dshowaudiosrc_get_device_name_values (GstDshowAudioSrc * src)
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{
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GValueArray *array = g_value_array_new (0);
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ICreateDevEnum *devices_enum = NULL;
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IEnumMoniker *moniker_enum = NULL;
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IMoniker *moniker = NULL;
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HRESULT hres = S_FALSE;
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ULONG fetched;
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hres = CoCreateInstance (CLSID_SystemDeviceEnum, NULL, CLSCTX_INPROC_SERVER,
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IID_ICreateDevEnum, (LPVOID *) & devices_enum);
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if (hres != S_OK) {
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GST_ERROR
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("Can't create an instance of the system device enumerator (error=0x%x)",
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hres);
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array = NULL;
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goto clean;
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}
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hres = devices_enum->CreateClassEnumerator (CLSID_AudioInputDeviceCategory,
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&moniker_enum, 0);
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if (hres != S_OK || !moniker_enum) {
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GST_ERROR ("Can't get enumeration of audio devices (error=0x%x)", hres);
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array = NULL;
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goto clean;
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}
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moniker_enum->Reset ();
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while (hres = moniker_enum->Next (1, &moniker, &fetched), hres == S_OK) {
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IPropertyBag *property_bag = NULL;
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hres = moniker->BindToStorage (NULL, NULL, IID_IPropertyBag,
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(LPVOID *) & property_bag);
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if (SUCCEEDED (hres) && property_bag) {
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VARIANT varFriendlyName;
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VariantInit (&varFriendlyName);
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hres = property_bag->Read (L"FriendlyName", &varFriendlyName, NULL);
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if (hres == S_OK && varFriendlyName.bstrVal) {
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gchar *friendly_name =
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g_utf16_to_utf8 ((const gunichar2 *) varFriendlyName.bstrVal,
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wcslen (varFriendlyName.bstrVal), NULL, NULL, NULL);
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GValue value = { 0 };
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g_value_init (&value, G_TYPE_STRING);
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g_value_set_string (&value, friendly_name);
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g_value_array_append (array, &value);
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g_value_unset (&value);
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g_free (friendly_name);
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SysFreeString (varFriendlyName.bstrVal);
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}
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property_bag->Release ();
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}
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moniker->Release ();
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}
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clean:
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if (moniker_enum)
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moniker_enum->Release ();
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if (devices_enum)
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devices_enum->Release ();
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return array;
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}
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static GValueArray *
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gst_dshowaudiosrc_probe_get_values (GstPropertyProbe * probe,
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guint prop_id, const GParamSpec * pspec)
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{
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GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (probe);
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GValueArray *array = NULL;
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switch (prop_id) {
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case PROP_DEVICE_NAME:
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array = gst_dshowaudiosrc_get_device_name_values (src);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (probe, prop_id, pspec);
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break;
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}
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return array;
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}
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static void
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gst_dshowaudiosrc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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{
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if (src->device) {
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g_free (src->device);
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src->device = NULL;
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}
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if (g_value_get_string (value)) {
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src->device = g_strdup (g_value_get_string (value));
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}
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break;
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}
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case PROP_DEVICE_NAME:
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{
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if (src->device_name) {
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g_free (src->device_name);
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src->device_name = NULL;
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}
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if (g_value_get_string (value)) {
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src->device_name = g_strdup (g_value_get_string (value));
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}
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break;
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}
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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_dshowaudiosrc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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}
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static GstCaps *
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gst_dshowaudiosrc_get_caps (GstBaseSrc * basesrc)
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{
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HRESULT hres = S_OK;
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IBindCtx *lpbc = NULL;
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IMoniker *audiom = NULL;
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DWORD dwEaten;
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GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (basesrc);
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gunichar2 *unidevice = NULL;
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if (src->device) {
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g_free (src->device);
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src->device = NULL;
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}
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src->device =
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gst_dshow_getdevice_from_devicename (&CLSID_AudioInputDeviceCategory,
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&src->device_name);
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if (!src->device) {
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GST_ERROR ("No audio device found.");
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return NULL;
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}
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unidevice =
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g_utf8_to_utf16 (src->device, strlen (src->device), NULL, NULL, NULL);
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if (!src->audio_cap_filter) {
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hres = CreateBindCtx (0, &lpbc);
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if (SUCCEEDED (hres)) {
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hres =
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MkParseDisplayName (lpbc, (LPCOLESTR) unidevice, &dwEaten, &audiom);
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if (SUCCEEDED (hres)) {
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hres = audiom->BindToObject (lpbc, NULL, IID_IBaseFilter,
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(LPVOID *) & src->audio_cap_filter);
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audiom->Release ();
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}
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lpbc->Release ();
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}
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}
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if (src->audio_cap_filter && !src->caps) {
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/* get the capture pins supported types */
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IPin *capture_pin = NULL;
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IEnumPins *enumpins = NULL;
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HRESULT hres;
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hres = src->audio_cap_filter->EnumPins (&enumpins);
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if (SUCCEEDED (hres)) {
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while (enumpins->Next (1, &capture_pin, NULL) == S_OK) {
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IKsPropertySet *pKs = NULL;
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hres =
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capture_pin->QueryInterface (IID_IKsPropertySet, (LPVOID *) & pKs);
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if (SUCCEEDED (hres) && pKs) {
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DWORD cbReturned;
|
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GUID pin_category;
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RPC_STATUS rpcstatus;
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hres =
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pKs->Get (AMPROPSETID_Pin,
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AMPROPERTY_PIN_CATEGORY, NULL, 0, &pin_category, sizeof (GUID),
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&cbReturned);
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/* we only want capture pins */
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if (UuidCompare (&pin_category, (UUID *) & PIN_CATEGORY_CAPTURE,
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&rpcstatus) == 0) {
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IAMStreamConfig *streamcaps = NULL;
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|
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if (SUCCEEDED (capture_pin->QueryInterface (IID_IAMStreamConfig,
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(LPVOID *) & streamcaps))) {
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src->caps =
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gst_dshowaudiosrc_getcaps_from_streamcaps (src, capture_pin,
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streamcaps);
|
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streamcaps->Release ();
|
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}
|
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}
|
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pKs->Release ();
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}
|
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capture_pin->Release ();
|
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}
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enumpins->Release ();
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}
|
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}
|
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|
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if (unidevice) {
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g_free (unidevice);
|
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}
|
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|
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if (src->caps) {
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return gst_caps_ref (src->caps);
|
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}
|
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|
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return NULL;
|
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}
|
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|
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static GstStateChangeReturn
|
|
gst_dshowaudiosrc_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
HRESULT hres = S_FALSE;
|
|
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
if (src->media_filter)
|
|
hres = src->media_filter->Run (0);
|
|
if (hres != S_OK) {
|
|
GST_ERROR ("Can't RUN the directshow capture graph (error=0x%x)", hres);
|
|
src->is_running = FALSE;
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
} else {
|
|
src->is_running = TRUE;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
if (src->media_filter)
|
|
hres = src->media_filter->Stop ();
|
|
if (hres != S_OK) {
|
|
GST_ERROR ("Can't STOP the directshow capture graph (error=0x%x)",
|
|
hres);
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
}
|
|
src->is_running = FALSE;
|
|
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
}
|
|
|
|
static gboolean
|
|
gst_dshowaudiosrc_open (GstAudioSrc * asrc)
|
|
{
|
|
HRESULT hres = S_FALSE;
|
|
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
|
|
|
|
hres = CoCreateInstance (CLSID_FilterGraph, NULL, CLSCTX_INPROC,
|
|
IID_IFilterGraph, (LPVOID *) & src->filter_graph);
|
|
if (hres != S_OK || !src->filter_graph) {
|
|
GST_ERROR
|
|
("Can't create an instance of the directshow graph manager (error=0x%x)",
|
|
hres);
|
|
goto error;
|
|
}
|
|
|
|
hres =
|
|
src->filter_graph->QueryInterface (IID_IMediaFilter,
|
|
(LPVOID *) & src->media_filter);
|
|
if (hres != S_OK || !src->media_filter) {
|
|
GST_ERROR
|
|
("Can't get IMediacontrol interface from the graph manager (error=0x%x)",
|
|
hres);
|
|
goto error;
|
|
}
|
|
|
|
src->dshow_fakesink = new CDshowFakeSink;
|
|
src->dshow_fakesink->AddRef ();
|
|
|
|
hres = src->filter_graph->AddFilter (src->audio_cap_filter, L"capture");
|
|
if (hres != S_OK) {
|
|
GST_ERROR
|
|
("Can't add the directshow capture filter to the graph (error=0x%x)",
|
|
hres);
|
|
goto error;
|
|
}
|
|
|
|
hres = src->filter_graph->AddFilter (src->dshow_fakesink, L"fakesink");
|
|
if (hres != S_OK) {
|
|
GST_ERROR ("Can't add our fakesink filter to the graph (error=0x%x)", hres);
|
|
goto error;
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
error:
|
|
if (src->dshow_fakesink) {
|
|
src->dshow_fakesink->Release ();
|
|
src->dshow_fakesink = NULL;
|
|
}
|
|
|
|
if (src->media_filter) {
|
|
src->media_filter->Release ();
|
|
src->media_filter = NULL;
|
|
}
|
|
if (src->filter_graph) {
|
|
src->filter_graph->Release ();
|
|
src->filter_graph = NULL;
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_dshowaudiosrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
|
|
{
|
|
HRESULT hres;
|
|
IPin *input_pin = NULL;
|
|
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
|
|
|
|
/* search the negociated caps in our caps list to get its index and the corresponding mediatype */
|
|
if (gst_caps_is_subset (spec->caps, src->caps)) {
|
|
guint i = 0;
|
|
gint res = -1;
|
|
|
|
for (; i < gst_caps_get_size (src->caps) && res == -1; i++) {
|
|
GstCaps *capstmp = gst_caps_copy_nth (src->caps, i);
|
|
|
|
if (gst_caps_is_subset (spec->caps, capstmp)) {
|
|
res = i;
|
|
}
|
|
gst_caps_unref (capstmp);
|
|
}
|
|
|
|
if (res != -1 && src->pins_mediatypes) {
|
|
/*get the corresponding media type and build the dshow graph */
|
|
GstCapturePinMediaType *pin_mediatype = NULL;
|
|
GList *type = g_list_nth (src->pins_mediatypes, res);
|
|
|
|
if (type) {
|
|
pin_mediatype = (GstCapturePinMediaType *) type->data;
|
|
|
|
src->dshow_fakesink->gst_set_media_type (pin_mediatype->mediatype);
|
|
src->dshow_fakesink->gst_set_buffer_callback (
|
|
(push_buffer_func) gst_dshowaudiosrc_push_buffer, src);
|
|
|
|
gst_dshow_get_pin_from_filter (src->dshow_fakesink, PINDIR_INPUT,
|
|
&input_pin);
|
|
if (!input_pin) {
|
|
GST_ERROR ("Can't get input pin from our directshow fakesink filter");
|
|
goto error;
|
|
}
|
|
|
|
hres = src->filter_graph->ConnectDirect (pin_mediatype->capture_pin,
|
|
input_pin, NULL);
|
|
input_pin->Release ();
|
|
|
|
if (hres != S_OK) {
|
|
GST_ERROR
|
|
("Can't connect capture filter with fakesink filter (error=0x%x)",
|
|
hres);
|
|
goto error;
|
|
}
|
|
|
|
spec->segsize = spec->rate * spec->channels;
|
|
spec->segtotal = 1;
|
|
}
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
error:
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_dshowaudiosrc_unprepare (GstAudioSrc * asrc)
|
|
{
|
|
IPin *input_pin = NULL, *output_pin = NULL;
|
|
HRESULT hres = S_FALSE;
|
|
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
|
|
|
|
/* disconnect filters */
|
|
gst_dshow_get_pin_from_filter (src->audio_cap_filter, PINDIR_OUTPUT,
|
|
&output_pin);
|
|
if (output_pin) {
|
|
hres = src->filter_graph->Disconnect (output_pin);
|
|
output_pin->Release ();
|
|
}
|
|
|
|
gst_dshow_get_pin_from_filter (src->dshow_fakesink, PINDIR_INPUT, &input_pin);
|
|
if (input_pin) {
|
|
hres = src->filter_graph->Disconnect (input_pin);
|
|
input_pin->Release ();
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_dshowaudiosrc_close (GstAudioSrc * asrc)
|
|
{
|
|
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
|
|
|
|
if (!src->filter_graph)
|
|
return TRUE;
|
|
|
|
/*remove filters from the graph */
|
|
src->filter_graph->RemoveFilter (src->audio_cap_filter);
|
|
src->filter_graph->RemoveFilter (src->dshow_fakesink);
|
|
|
|
/*release our gstreamer dshow sink */
|
|
src->dshow_fakesink->Release ();
|
|
src->dshow_fakesink = NULL;
|
|
|
|
/*release media filter interface */
|
|
src->media_filter->Release ();
|
|
src->media_filter = NULL;
|
|
|
|
/*release the filter graph manager */
|
|
src->filter_graph->Release ();
|
|
src->filter_graph = NULL;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static guint
|
|
gst_dshowaudiosrc_read (GstAudioSrc * asrc, gpointer data, guint length)
|
|
{
|
|
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
|
|
guint ret = 0;
|
|
|
|
if (!src->is_running)
|
|
return -1;
|
|
|
|
if (src->gbarray) {
|
|
test:
|
|
if (src->gbarray->len >= length) {
|
|
g_mutex_lock (src->gbarray_lock);
|
|
memcpy (data, src->gbarray->data + (src->gbarray->len - length), length);
|
|
g_byte_array_remove_range (src->gbarray, src->gbarray->len - length,
|
|
length);
|
|
ret = length;
|
|
g_mutex_unlock (src->gbarray_lock);
|
|
} else {
|
|
if (src->is_running) {
|
|
Sleep (100);
|
|
goto test;
|
|
}
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static guint
|
|
gst_dshowaudiosrc_delay (GstAudioSrc * asrc)
|
|
{
|
|
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
|
|
guint ret = 0;
|
|
|
|
if (src->gbarray) {
|
|
g_mutex_lock (src->gbarray_lock);
|
|
if (src->gbarray->len) {
|
|
ret = src->gbarray->len / 4;
|
|
}
|
|
g_mutex_unlock (src->gbarray_lock);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_dshowaudiosrc_reset (GstAudioSrc * asrc)
|
|
{
|
|
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
|
|
|
|
g_mutex_lock (src->gbarray_lock);
|
|
GST_DEBUG ("byte array size= %d", src->gbarray->len);
|
|
if (src->gbarray->len > 0)
|
|
g_byte_array_remove_range (src->gbarray, 0, src->gbarray->len);
|
|
g_mutex_unlock (src->gbarray_lock);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_dshowaudiosrc_getcaps_from_streamcaps (GstDshowAudioSrc * src, IPin * pin,
|
|
IAMStreamConfig * streamcaps)
|
|
{
|
|
GstCaps *caps = NULL;
|
|
HRESULT hres = S_OK;
|
|
int icount = 0;
|
|
int isize = 0;
|
|
AUDIO_STREAM_CONFIG_CAPS ascc;
|
|
int i = 0;
|
|
|
|
if (!streamcaps)
|
|
return NULL;
|
|
|
|
streamcaps->GetNumberOfCapabilities (&icount, &isize);
|
|
|
|
if (isize != sizeof (ascc))
|
|
return NULL;
|
|
|
|
for (; i < icount; i++) {
|
|
GstCapturePinMediaType *pin_mediatype = g_new0 (GstCapturePinMediaType, 1);
|
|
|
|
pin->AddRef ();
|
|
pin_mediatype->capture_pin = pin;
|
|
|
|
hres = streamcaps->GetStreamCaps (i, &pin_mediatype->mediatype,
|
|
(BYTE *) & ascc);
|
|
if (hres == S_OK && pin_mediatype->mediatype) {
|
|
GstCaps *mediacaps = NULL;
|
|
|
|
if (!caps)
|
|
caps = gst_caps_new_empty ();
|
|
|
|
if (gst_dshow_check_mediatype (pin_mediatype->mediatype, MEDIASUBTYPE_PCM,
|
|
FORMAT_WaveFormatEx)) {
|
|
WAVEFORMATEX *wavformat =
|
|
(WAVEFORMATEX *) pin_mediatype->mediatype->pbFormat;
|
|
mediacaps =
|
|
gst_caps_new_simple ("audio/x-raw-int", "width", G_TYPE_INT,
|
|
wavformat->wBitsPerSample, "depth", G_TYPE_INT,
|
|
wavformat->wBitsPerSample, "endianness", G_TYPE_INT, G_BYTE_ORDER,
|
|
"signed", G_TYPE_BOOLEAN, TRUE, "channels", G_TYPE_INT,
|
|
wavformat->nChannels, "rate", G_TYPE_INT, wavformat->nSamplesPerSec,
|
|
NULL);
|
|
|
|
if (mediacaps) {
|
|
src->pins_mediatypes =
|
|
g_list_append (src->pins_mediatypes, pin_mediatype);
|
|
gst_caps_append (caps, mediacaps);
|
|
} else {
|
|
gst_dshow_free_pin_mediatype (pin_mediatype);
|
|
}
|
|
} else {
|
|
gst_dshow_free_pin_mediatype (pin_mediatype);
|
|
}
|
|
} else {
|
|
gst_dshow_free_pin_mediatype (pin_mediatype);
|
|
}
|
|
}
|
|
|
|
if (caps && gst_caps_is_empty (caps)) {
|
|
gst_caps_unref (caps);
|
|
caps = NULL;
|
|
}
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gboolean
|
|
gst_dshowaudiosrc_push_buffer (guint8 * buffer, guint size, gpointer src_object,
|
|
GstClockTime duration)
|
|
{
|
|
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (src_object);
|
|
|
|
if (!buffer || size == 0 || !src) {
|
|
return FALSE;
|
|
}
|
|
|
|
g_mutex_lock (src->gbarray_lock);
|
|
g_byte_array_prepend (src->gbarray, buffer, size);
|
|
g_mutex_unlock (src->gbarray_lock);
|
|
|
|
return TRUE;
|
|
}
|