mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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806 lines
22 KiB
C++
806 lines
22 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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, "
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"format = (string){ "
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GST_AUDIO_NE (S16) ", "
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GST_AUDIO_NE (U16) ", "
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GST_AUDIO_NE (S8) ", "
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GST_AUDIO_NE (U8)
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" }, "
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"rate = " GST_AUDIO_RATE_RANGE ", "
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"channels = (int) [ 1, 2 ]")
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);
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G_DEFINE_TYPE(GstDshowAudioSrc, gst_dshowaudiosrc, GST_TYPE_AUDIO_SRC);
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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_DEVICE_INDEX
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};
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#define DEFAULT_PROP_DEVICE_INDEX 0
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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, GstCaps * filter);
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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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GstAudioRingBufferSpec * 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, GstClockTime *timestamp);
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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_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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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_get_caps);
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_change_state);
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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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g_object_class_install_property
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(gobject_class, PROP_DEVICE_INDEX,
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g_param_spec_int ("device-index", "Device index",
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"Index of the enumerated audio device", 0, G_MAXINT,
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DEFAULT_PROP_DEVICE_INDEX,
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static_cast < GParamFlags > (G_PARAM_READWRITE)));
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gst_element_class_add_static_pad_template (gstelement_class, &src_template);
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gst_element_class_set_static_metadata (gstelement_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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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)
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{
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src->device = NULL;
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src->device_name = NULL;
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src->device_index = DEFAULT_PROP_DEVICE_INDEX;
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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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g_mutex_init(&src->gbarray_lock);
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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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g_mutex_clear(&src->gbarray_lock);
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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 (gst_dshowaudiosrc_parent_class)->dispose (gobject);
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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_value_dup_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_value_dup_string (value);;
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}
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break;
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}
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case PROP_DEVICE_INDEX:
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{
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src->device_index = g_value_get_int (value);
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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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GstDshowAudioSrc *src;
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g_return_if_fail (GST_IS_DSHOWAUDIOSRC (object));
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src = GST_DSHOWAUDIOSRC (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_set_string (value, src->device_name);
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break;
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case PROP_DEVICE_INDEX:
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g_value_set_int (value, src->device_index);
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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 GstCaps *
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gst_dshowaudiosrc_get_caps (GstBaseSrc * basesrc, GstCaps * filter)
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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, &src->device_index);
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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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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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if (unidevice) {
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g_free (unidevice);
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}
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if (src->caps) {
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GstCaps *caps;
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if (filter) {
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caps = gst_caps_intersect_full (filter, src->caps, GST_CAPS_INTERSECT_FIRST);
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} else {
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caps = gst_caps_ref (src->caps);
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}
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return caps;
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}
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return NULL;
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}
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static GstStateChangeReturn
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gst_dshowaudiosrc_change_state (GstElement * element, GstStateChange transition)
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{
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HRESULT hres = S_FALSE;
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GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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if (src->media_filter) {
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src->is_running = TRUE;
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hres = src->media_filter->Run (0);
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}
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if (hres != S_OK) {
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GST_ERROR ("Can't RUN the directshow capture graph (error=0x%x)", hres);
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src->is_running = FALSE;
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return GST_STATE_CHANGE_FAILURE;
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}
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break;
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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if (src->media_filter)
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hres = src->media_filter->Stop ();
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if (hres != S_OK) {
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GST_ERROR ("Can't STOP the directshow capture graph (error=0x%x)",
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hres);
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return GST_STATE_CHANGE_FAILURE;
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}
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src->is_running = FALSE;
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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default:
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break;
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}
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return GST_ELEMENT_CLASS(gst_dshowaudiosrc_parent_class)->change_state(element, transition);
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}
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static gboolean
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gst_dshowaudiosrc_open (GstAudioSrc * asrc)
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{
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HRESULT hres = S_FALSE;
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GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
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hres = CoCreateInstance (CLSID_FilterGraph, NULL, CLSCTX_INPROC,
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IID_IFilterGraph, (LPVOID *) & src->filter_graph);
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if (hres != S_OK || !src->filter_graph) {
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GST_ERROR
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("Can't create an instance of the directshow graph manager (error=0x%x)",
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hres);
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goto error;
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}
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hres =
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src->filter_graph->QueryInterface (IID_IMediaFilter,
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(LPVOID *) & src->media_filter);
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if (hres != S_OK || !src->media_filter) {
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GST_ERROR
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("Can't get IMediacontrol interface from the graph manager (error=0x%x)",
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hres);
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goto error;
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}
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src->dshow_fakesink = new CDshowFakeSink;
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src->dshow_fakesink->AddRef ();
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hres = src->filter_graph->AddFilter (src->audio_cap_filter, L"capture");
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if (hres != S_OK) {
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GST_ERROR
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("Can't add the directshow capture filter to the graph (error=0x%x)",
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hres);
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goto error;
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}
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hres = src->filter_graph->AddFilter (src->dshow_fakesink, L"fakesink");
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if (hres != S_OK) {
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GST_ERROR ("Can't add our fakesink filter to the graph (error=0x%x)", hres);
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goto error;
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}
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return TRUE;
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error:
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if (src->dshow_fakesink) {
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src->dshow_fakesink->Release ();
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src->dshow_fakesink = NULL;
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}
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if (src->media_filter) {
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src->media_filter->Release ();
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src->media_filter = NULL;
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}
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if (src->filter_graph) {
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src->filter_graph->Release ();
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src->filter_graph = NULL;
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}
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return FALSE;
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}
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static gboolean
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gst_dshowaudiosrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
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{
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HRESULT hres;
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IPin *input_pin = NULL;
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GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
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GstCaps *current_caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (asrc));
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if (current_caps) {
|
|
if (gst_caps_is_equal (spec->caps, current_caps)) {
|
|
gst_caps_unref (current_caps);
|
|
return TRUE;
|
|
}
|
|
gst_caps_unref (current_caps);
|
|
}
|
|
/* In 1.0, prepare() seems to be called in the PLAYING state. Most
|
|
of the time you can't do much on a running graph. */
|
|
|
|
gboolean was_running = src->is_running;
|
|
if (was_running) {
|
|
HRESULT hres = src->media_filter->Stop ();
|
|
if (hres != S_OK) {
|
|
GST_ERROR("Can't STOP the directshow capture graph for preparing (error=0x%x)", hres);
|
|
return FALSE;
|
|
}
|
|
src->is_running = FALSE;
|
|
}
|
|
|
|
/* search the negotiated 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;
|
|
}
|
|
|
|
spec->segsize = (gint) (spec->info.bpf * spec->info.rate * spec->latency_time /
|
|
GST_MSECOND);
|
|
spec->segtotal = (gint) ((gfloat) spec->buffer_time /
|
|
(gfloat) spec->latency_time + 0.5);
|
|
if (!gst_dshow_configure_latency (pin_mediatype->capture_pin,
|
|
spec->segsize))
|
|
{
|
|
GST_WARNING ("Could not change capture latency");
|
|
spec->segsize = spec->info.rate * spec->info.channels;
|
|
spec->segtotal = 2;
|
|
};
|
|
GST_INFO ("Configuring with segsize:%d segtotal:%d", spec->segsize, spec->segtotal);
|
|
|
|
if (gst_dshow_is_pin_connected (pin_mediatype->capture_pin)) {
|
|
GST_DEBUG_OBJECT (src,
|
|
"capture_pin already connected, disconnecting");
|
|
src->filter_graph->Disconnect (pin_mediatype->capture_pin);
|
|
}
|
|
|
|
if (gst_dshow_is_pin_connected (input_pin)) {
|
|
GST_DEBUG_OBJECT (src, "input_pin already connected, disconnecting");
|
|
src->filter_graph->Disconnect (input_pin);
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
}
|
|
}
|
|
}
|
|
|
|
if (was_running) {
|
|
HRESULT hres = src->media_filter->Run (0);
|
|
if (hres != S_OK) {
|
|
GST_ERROR("Can't RUN the directshow capture graph after prepare (error=0x%x)", hres);
|
|
return FALSE;
|
|
}
|
|
|
|
src->is_running = TRUE;
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
error:
|
|
/* Don't restart the graph, we're out anyway. */
|
|
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, GstClockTime *timestamp)
|
|
{
|
|
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 (GST_AUDIO_BASE_SRC(src)->ringbuffer->spec.latency_time /
|
|
GST_MSECOND / 10);
|
|
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)) {
|
|
GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
|
|
WAVEFORMATEX *wavformat =
|
|
(WAVEFORMATEX *) pin_mediatype->mediatype->pbFormat;
|
|
|
|
switch (wavformat->wFormatTag) {
|
|
case WAVE_FORMAT_PCM:
|
|
format = gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, wavformat->wBitsPerSample, wavformat->wBitsPerSample);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (format != GST_AUDIO_FORMAT_UNKNOWN) {
|
|
GstAudioInfo info;
|
|
|
|
gst_audio_info_init(&info);
|
|
gst_audio_info_set_format(&info,
|
|
format,
|
|
wavformat->nSamplesPerSec,
|
|
wavformat->nChannels,
|
|
NULL);
|
|
mediacaps = gst_audio_info_to_caps(&info);
|
|
}
|
|
|
|
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;
|
|
}
|