/* * GStreamer * Copyright (C) 2005,2006 Zaheer Abbas Merali * Copyright (C) 2008 Pioneers of the Inevitable * * Permission is hereby granted, free of charge, to any person obtaining a * copy of this software and associated documentation files (the "Software"), * to deal in the Software without restriction, including without limitation * the rights to use, copy, modify, merge, publish, distribute, sublicense, * and/or sell copies of the Software, and to permit persons to whom the * Software is furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER * DEALINGS IN THE SOFTWARE. * * Alternatively, the contents of this file may be used under the * GNU Lesser General Public License Version 2.1 (the "LGPL"), in * which case the following provisions apply instead of the ones * mentioned above: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-osxaudiosrc * * This element captures raw audio samples using the CoreAudio api. * * * Example launch line * |[ * gst-launch-1.0 osxaudiosrc ! wavenc ! filesink location=audio.wav * ]| * */ #ifdef HAVE_CONFIG_H # include #endif #include #include "gstosxaudiosrc.h" #include "gstosxaudioelement.h" GST_DEBUG_CATEGORY_STATIC (osx_audiosrc_debug); #define GST_CAT_DEFAULT osx_audiosrc_debug /* Filter signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0, ARG_DEVICE }; static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_OSX_AUDIO_SRC_CAPS) ); static void gst_osx_audio_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_osx_audio_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstStateChangeReturn gst_osx_audio_src_change_state (GstElement * element, GstStateChange transition); static GstCaps *gst_osx_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter); static GstAudioRingBuffer *gst_osx_audio_src_create_ringbuffer (GstAudioBaseSrc * src); static void gst_osx_audio_src_osxelement_init (gpointer g_iface, gpointer iface_data); static OSStatus gst_osx_audio_src_io_proc (GstOsxAudioRingBuffer * buf, AudioUnitRenderActionFlags * ioActionFlags, const AudioTimeStamp * inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList); static void gst_osx_audio_src_do_init (GType type) { static const GInterfaceInfo osxelement_info = { gst_osx_audio_src_osxelement_init, NULL, NULL }; GST_DEBUG_CATEGORY_INIT (osx_audiosrc_debug, "osxaudiosrc", 0, "OSX Audio Src"); g_type_add_interface_static (type, GST_OSX_AUDIO_ELEMENT_TYPE, &osxelement_info); } #define gst_osx_audio_src_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstOsxAudioSrc, gst_osx_audio_src, GST_TYPE_AUDIO_BASE_SRC, gst_osx_audio_src_do_init (g_define_type_id)); static void gst_osx_audio_src_class_init (GstOsxAudioSrcClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseSrcClass *gstbasesrc_class; GstAudioBaseSrcClass *gstaudiobasesrc_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasesrc_class = (GstBaseSrcClass *) klass; gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass; gobject_class->set_property = gst_osx_audio_src_set_property; gobject_class->get_property = gst_osx_audio_src_get_property; gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_osx_audio_src_change_state); gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_osx_audio_src_get_caps); g_object_class_install_property (gobject_class, ARG_DEVICE, g_param_spec_int ("device", "Device ID", "Device ID of input device", 0, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstaudiobasesrc_class->create_ringbuffer = GST_DEBUG_FUNCPTR (gst_osx_audio_src_create_ringbuffer); gst_element_class_add_static_pad_template (gstelement_class, &src_factory); gst_element_class_set_static_metadata (gstelement_class, "Audio Source (OSX)", "Source/Audio", "Input from a sound card in OS X", "Zaheer Abbas Merali "); } static void gst_osx_audio_src_init (GstOsxAudioSrc * src) { gst_base_src_set_live (GST_BASE_SRC (src), TRUE); src->device_id = kAudioDeviceUnknown; } static void gst_osx_audio_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstOsxAudioSrc *src = GST_OSX_AUDIO_SRC (object); switch (prop_id) { case ARG_DEVICE: src->device_id = g_value_get_int (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_osx_audio_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstOsxAudioSrc *src = GST_OSX_AUDIO_SRC (object); switch (prop_id) { case ARG_DEVICE: g_value_set_int (value, src->device_id); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstStateChangeReturn gst_osx_audio_src_change_state (GstElement * element, GstStateChange transition) { GstOsxAudioSrc *osxsrc = GST_OSX_AUDIO_SRC (element); GstOsxAudioRingBuffer *ringbuffer; GstStateChangeReturn ret; switch (transition) { default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); if (ret == GST_STATE_CHANGE_FAILURE) goto out; switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: /* The device is open now, so fix our device_id if it changed */ ringbuffer = GST_OSX_AUDIO_RING_BUFFER (GST_AUDIO_BASE_SRC (osxsrc)->ringbuffer); if (ringbuffer->core_audio->device_id != osxsrc->device_id) { osxsrc->device_id = ringbuffer->core_audio->device_id; g_object_notify (G_OBJECT (osxsrc), "device"); } break; default: break; } out: return ret; } static GstCaps * gst_osx_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter) { GstOsxAudioSrc *osxsrc; GstAudioRingBuffer *buf; GstOsxAudioRingBuffer *osxbuf; GstCaps *caps, *filtered_caps; osxsrc = GST_OSX_AUDIO_SRC (src); GST_OBJECT_LOCK (osxsrc); buf = GST_AUDIO_BASE_SRC (src)->ringbuffer; if (buf) gst_object_ref (buf); GST_OBJECT_UNLOCK (osxsrc); if (!buf) { GST_DEBUG_OBJECT (src, "no ring buffer, using template caps"); return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter); } osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf); /* protect against cached_caps going away */ GST_OBJECT_LOCK (buf); if (osxbuf->core_audio->cached_caps_valid) { GST_LOG_OBJECT (src, "Returning cached caps"); caps = gst_caps_ref (osxbuf->core_audio->cached_caps); } else if (buf->open) { GstCaps *template_caps; /* Get template caps */ template_caps = gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SRC_PAD (osxsrc)); /* Device is open, let's probe its caps */ caps = gst_core_audio_probe_caps (osxbuf->core_audio, template_caps); gst_caps_replace (&osxbuf->core_audio->cached_caps, caps); gst_caps_unref (template_caps); } else { GST_DEBUG_OBJECT (src, "ring buffer not open, using template caps"); caps = GST_BASE_SRC_CLASS (parent_class)->get_caps (src, NULL); } GST_OBJECT_UNLOCK (buf); gst_object_unref (buf); if (!caps) return NULL; if (!filter) return caps; /* Take care of filtered caps */ filtered_caps = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); return filtered_caps; } static GstAudioRingBuffer * gst_osx_audio_src_create_ringbuffer (GstAudioBaseSrc * src) { GstOsxAudioSrc *osxsrc; GstOsxAudioRingBuffer *ringbuffer; osxsrc = GST_OSX_AUDIO_SRC (src); GST_DEBUG_OBJECT (osxsrc, "Creating ringbuffer"); ringbuffer = g_object_new (GST_TYPE_OSX_AUDIO_RING_BUFFER, NULL); GST_DEBUG_OBJECT (osxsrc, "osx src 0x%p element 0x%p ioproc 0x%p", osxsrc, GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsrc), (void *) gst_osx_audio_src_io_proc); ringbuffer->core_audio->element = GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsrc); ringbuffer->core_audio->is_src = TRUE; /* By default the coreaudio instance created by the ringbuffer * has device_id==kAudioDeviceUnknown. The user might have * selected a different one here */ if (ringbuffer->core_audio->device_id != osxsrc->device_id) ringbuffer->core_audio->device_id = osxsrc->device_id; return GST_AUDIO_RING_BUFFER (ringbuffer); } static OSStatus gst_osx_audio_src_io_proc (GstOsxAudioRingBuffer * buf, AudioUnitRenderActionFlags * ioActionFlags, const AudioTimeStamp * inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList) { OSStatus status; guint8 *writeptr; gint writeseg; gint len; gint remaining; UInt32 n; gint offset = 0; /* Previous invoke of AudioUnitRender changed mDataByteSize into * number of bytes actually read. Reset the members. */ for (n = 0; n < buf->core_audio->recBufferList->mNumberBuffers; ++n) { buf->core_audio->recBufferList->mBuffers[n].mDataByteSize = buf->core_audio->recBufferSize; } status = AudioUnitRender (buf->core_audio->audiounit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, buf->core_audio->recBufferList); if (status) { GST_WARNING_OBJECT (buf, "AudioUnitRender returned %d", (int) status); return status; } /* TODO: To support non-interleaved audio, go over all mBuffers, * not just the first one. */ remaining = buf->core_audio->recBufferList->mBuffers[0].mDataByteSize; while (remaining) { if (!gst_audio_ring_buffer_prepare_read (GST_AUDIO_RING_BUFFER (buf), &writeseg, &writeptr, &len)) return 0; len -= buf->segoffset; if (len > remaining) len = remaining; memcpy (writeptr + buf->segoffset, (char *) buf->core_audio->recBufferList->mBuffers[0].mData + offset, len); buf->segoffset += len; offset += len; remaining -= len; if ((gint) buf->segoffset == GST_AUDIO_RING_BUFFER (buf)->spec.segsize) { /* we wrote one segment */ gst_audio_ring_buffer_advance (GST_AUDIO_RING_BUFFER (buf), 1); buf->segoffset = 0; } } return 0; } static void gst_osx_audio_src_osxelement_init (gpointer g_iface, gpointer iface_data) { GstOsxAudioElementInterface *iface = (GstOsxAudioElementInterface *) g_iface; iface->io_proc = (AURenderCallback) gst_osx_audio_src_io_proc; }