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689 lines
21 KiB
C
689 lines
21 KiB
C
/*
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* GStreamer
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* Copyright (C) 2006 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* Copyright (C) 2008 Pioneers of the Inevitable <songbird@songbirdnest.com>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a
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* copy of this software and associated documentation files (the "Software"),
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* to deal in the Software without restriction, including without limitation
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* the rights to use, copy, modify, merge, publish, distribute, sublicense,
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* and/or sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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* DEALINGS IN THE SOFTWARE.
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*
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* Alternatively, the contents of this file may be used under the
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* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
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* which case the following provisions apply instead of the ones
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* mentioned above:
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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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#include <CoreAudio/CoreAudio.h>
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#include <CoreServices/CoreServices.h>
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#include <gst/gst.h>
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#include <gst/audio/multichannel.h>
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#include "gstosxringbuffer.h"
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#include "gstosxaudiosink.h"
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#include "gstosxaudiosrc.h"
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GST_DEBUG_CATEGORY_STATIC (osx_audio_debug);
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#define GST_CAT_DEFAULT osx_audio_debug
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static void gst_osx_ring_buffer_dispose (GObject * object);
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static void gst_osx_ring_buffer_finalize (GObject * object);
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static gboolean gst_osx_ring_buffer_open_device (GstRingBuffer * buf);
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static gboolean gst_osx_ring_buffer_close_device (GstRingBuffer * buf);
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static gboolean gst_osx_ring_buffer_acquire (GstRingBuffer * buf,
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GstRingBufferSpec * spec);
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static gboolean gst_osx_ring_buffer_release (GstRingBuffer * buf);
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static gboolean gst_osx_ring_buffer_start (GstRingBuffer * buf);
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static gboolean gst_osx_ring_buffer_pause (GstRingBuffer * buf);
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static gboolean gst_osx_ring_buffer_stop (GstRingBuffer * buf);
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static guint gst_osx_ring_buffer_delay (GstRingBuffer * buf);
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static GstRingBufferClass *ring_parent_class = NULL;
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static OSStatus gst_osx_ring_buffer_render_notify (GstOsxRingBuffer * osxbuf,
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AudioUnitRenderActionFlags * ioActionFlags,
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const AudioTimeStamp * inTimeStamp, unsigned int inBusNumber,
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unsigned int inNumberFrames, AudioBufferList * ioData);
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static AudioBufferList *buffer_list_alloc (int channels, int size);
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static void buffer_list_free (AudioBufferList * list);
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static void
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gst_osx_ring_buffer_do_init (GType type)
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{
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GST_DEBUG_CATEGORY_INIT (osx_audio_debug, "osxaudio", 0,
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"OSX Audio Elements");
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}
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GST_BOILERPLATE_FULL (GstOsxRingBuffer, gst_osx_ring_buffer, GstRingBuffer,
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GST_TYPE_RING_BUFFER, gst_osx_ring_buffer_do_init);
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static void
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gst_osx_ring_buffer_base_init (gpointer g_class)
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{
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/* Nothing to do right now */
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}
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static void
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gst_osx_ring_buffer_class_init (GstOsxRingBufferClass * klass)
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{
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GObjectClass *gobject_class;
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GstObjectClass *gstobject_class;
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GstRingBufferClass *gstringbuffer_class;
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gobject_class = (GObjectClass *) klass;
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gstobject_class = (GstObjectClass *) klass;
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gstringbuffer_class = (GstRingBufferClass *) klass;
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ring_parent_class = g_type_class_peek_parent (klass);
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_dispose);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_finalize);
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gstringbuffer_class->open_device =
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GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_open_device);
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gstringbuffer_class->close_device =
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GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_close_device);
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gstringbuffer_class->acquire =
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GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_acquire);
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gstringbuffer_class->release =
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GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_release);
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gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_start);
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gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_pause);
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gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_start);
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gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_stop);
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gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_osx_ring_buffer_delay);
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GST_DEBUG ("osx ring buffer class init");
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}
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static void
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gst_osx_ring_buffer_init (GstOsxRingBuffer * ringbuffer,
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GstOsxRingBufferClass * g_class)
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{
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/* Nothing to do right now */
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}
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static void
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gst_osx_ring_buffer_dispose (GObject * object)
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{
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G_OBJECT_CLASS (ring_parent_class)->dispose (object);
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}
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static void
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gst_osx_ring_buffer_finalize (GObject * object)
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{
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G_OBJECT_CLASS (ring_parent_class)->finalize (object);
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}
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static AudioUnit
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gst_osx_ring_buffer_create_audio_unit (GstOsxRingBuffer * osxbuf,
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gboolean input, AudioDeviceID device_id)
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{
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ComponentDescription desc;
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Component comp;
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OSStatus status;
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AudioUnit unit;
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UInt32 enableIO;
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/* Create a HALOutput AudioUnit.
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* This is the lowest-level output API that is actually sensibly usable
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* (the lower level ones require that you do channel-remapping yourself,
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* and the CoreAudio channel mapping is sufficiently complex that doing
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* so would be very difficult)
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*
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* Note that for input we request an output unit even though we will do
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* input with it: http://developer.apple.com/technotes/tn2002/tn2091.html
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*/
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desc.componentType = kAudioUnitType_Output;
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desc.componentSubType = kAudioUnitSubType_HALOutput;
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desc.componentManufacturer = kAudioUnitManufacturer_Apple;
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desc.componentFlags = 0;
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desc.componentFlagsMask = 0;
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comp = FindNextComponent (NULL, &desc);
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if (comp == NULL) {
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GST_WARNING_OBJECT (osxbuf, "Couldn't find HALOutput component");
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return NULL;
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}
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status = OpenAComponent (comp, &unit);
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if (status) {
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GST_WARNING_OBJECT (osxbuf, "Couldn't open HALOutput component");
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return NULL;
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}
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if (input) {
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/* enable input */
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enableIO = 1;
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status = AudioUnitSetProperty (unit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, /* 1 = input element */
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&enableIO, sizeof (enableIO));
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if (status) {
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CloseComponent (unit);
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GST_WARNING_OBJECT (osxbuf, "Failed to enable input: %lx",
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(gulong) status);
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return NULL;
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}
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/* disable output */
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enableIO = 0;
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status = AudioUnitSetProperty (unit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, /* 0 = output element */
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&enableIO, sizeof (enableIO));
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if (status) {
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CloseComponent (unit);
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GST_WARNING_OBJECT (osxbuf, "Failed to disable output: %lx",
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(gulong) status);
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return NULL;
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}
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}
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/* Specify which device we're using. */
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GST_DEBUG_OBJECT (osxbuf, "Setting device to %d", (int) device_id);
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status = AudioUnitSetProperty (unit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, /* N/A for global */
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&device_id, sizeof (AudioDeviceID));
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if (status) {
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CloseComponent (unit);
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GST_WARNING_OBJECT (osxbuf, "Failed to set device: %lx", (gulong) status);
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return NULL;
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}
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GST_DEBUG_OBJECT (osxbuf, "Create HALOutput AudioUnit: %p", unit);
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return unit;
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}
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static gboolean
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gst_osx_ring_buffer_open_device (GstRingBuffer * buf)
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{
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GstOsxRingBuffer *osxbuf;
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GstOsxAudioSink *sink;
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GstOsxAudioSrc *src;
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AudioStreamBasicDescription asbd_in;
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OSStatus status;
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UInt32 propertySize;
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osxbuf = GST_OSX_RING_BUFFER (buf);
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sink = NULL;
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src = NULL;
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osxbuf->audiounit = gst_osx_ring_buffer_create_audio_unit (osxbuf,
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osxbuf->is_src, osxbuf->device_id);
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if (osxbuf->is_src) {
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src = GST_OSX_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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propertySize = sizeof (asbd_in);
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status = AudioUnitGetProperty (osxbuf->audiounit,
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kAudioUnitProperty_StreamFormat,
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kAudioUnitScope_Input, 1, &asbd_in, &propertySize);
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if (status) {
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CloseComponent (osxbuf->audiounit);
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osxbuf->audiounit = NULL;
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GST_WARNING_OBJECT (osxbuf, "Unable to obtain device properties: %lx",
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(gulong) status);
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return FALSE;
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}
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src->deviceChannels = asbd_in.mChannelsPerFrame;
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} else {
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sink = GST_OSX_AUDIO_SINK (GST_OBJECT_PARENT (buf));
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/* needed for the sink's volume control */
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sink->audiounit = osxbuf->audiounit;
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}
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return TRUE;
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}
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static gboolean
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gst_osx_ring_buffer_close_device (GstRingBuffer * buf)
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{
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GstOsxRingBuffer *osxbuf;
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osxbuf = GST_OSX_RING_BUFFER (buf);
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CloseComponent (osxbuf->audiounit);
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osxbuf->audiounit = NULL;
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return TRUE;
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}
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static AudioChannelLabel
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gst_audio_channel_position_to_coreaudio_channel_label (GstAudioChannelPosition
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position, int channel)
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{
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switch (position) {
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case GST_AUDIO_CHANNEL_POSITION_NONE:
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return kAudioChannelLabel_Discrete_0 | channel;
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case GST_AUDIO_CHANNEL_POSITION_FRONT_MONO:
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return kAudioChannelLabel_Mono;
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case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
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return kAudioChannelLabel_Left;
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case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
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return kAudioChannelLabel_Right;
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case GST_AUDIO_CHANNEL_POSITION_REAR_CENTER:
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return kAudioChannelLabel_CenterSurround;
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case GST_AUDIO_CHANNEL_POSITION_REAR_LEFT:
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return kAudioChannelLabel_LeftSurround;
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case GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT:
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return kAudioChannelLabel_RightSurround;
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case GST_AUDIO_CHANNEL_POSITION_LFE:
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return kAudioChannelLabel_LFEScreen;
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case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER:
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return kAudioChannelLabel_Center;
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case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER:
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return kAudioChannelLabel_Center; // ???
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case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER:
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return kAudioChannelLabel_Center; // ???
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case GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT:
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return kAudioChannelLabel_LeftSurroundDirect;
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case GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT:
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return kAudioChannelLabel_RightSurroundDirect;
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default:
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return kAudioChannelLabel_Unknown;
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}
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}
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static gboolean
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gst_osx_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
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{
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/* Configure the output stream and allocate ringbuffer memory */
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GstOsxRingBuffer *osxbuf;
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AudioStreamBasicDescription format;
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AudioChannelLayout *layout = NULL;
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OSStatus status;
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UInt32 propertySize;
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int layoutSize;
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int element;
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int i;
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AudioUnitScope scope;
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gboolean ret = FALSE;
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GstStructure *structure;
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GstAudioChannelPosition *positions;
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UInt32 frameSize;
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osxbuf = GST_OSX_RING_BUFFER (buf);
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/* Fill out the audio description we're going to be using */
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format.mFormatID = kAudioFormatLinearPCM;
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format.mSampleRate = (double) spec->rate;
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format.mChannelsPerFrame = spec->channels;
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format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
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format.mBytesPerFrame = spec->channels * sizeof (float);
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format.mBitsPerChannel = sizeof (float) * 8;
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format.mBytesPerPacket = spec->channels * sizeof (float);
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format.mFramesPerPacket = 1;
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format.mReserved = 0;
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/* Describe channels */
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layoutSize = sizeof (AudioChannelLayout) +
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spec->channels * sizeof (AudioChannelDescription);
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layout = g_malloc (layoutSize);
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structure = gst_caps_get_structure (spec->caps, 0);
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positions = gst_audio_get_channel_positions (structure);
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layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
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layout->mChannelBitmap = 0; /* Not used */
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layout->mNumberChannelDescriptions = spec->channels;
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for (i = 0; i < spec->channels; i++) {
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if (positions) {
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layout->mChannelDescriptions[i].mChannelLabel =
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gst_audio_channel_position_to_coreaudio_channel_label (positions[i],
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i);
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} else {
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/* Discrete channel numbers are ORed into this */
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layout->mChannelDescriptions[i].mChannelLabel =
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kAudioChannelLabel_Discrete_0 | i;
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}
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/* Others unused */
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layout->mChannelDescriptions[i].mChannelFlags = 0;
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layout->mChannelDescriptions[i].mCoordinates[0] = 0.f;
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layout->mChannelDescriptions[i].mCoordinates[1] = 0.f;
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layout->mChannelDescriptions[i].mCoordinates[2] = 0.f;
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}
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if (positions) {
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g_free (positions);
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positions = NULL;
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}
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GST_LOG_OBJECT (osxbuf, "Format: %x, %f, %u, %x, %d, %d, %d, %d, %d",
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(unsigned int) format.mFormatID,
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format.mSampleRate,
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(unsigned int) format.mChannelsPerFrame,
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(unsigned int) format.mFormatFlags,
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(unsigned int) format.mBytesPerFrame,
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(unsigned int) format.mBitsPerChannel,
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(unsigned int) format.mBytesPerPacket,
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(unsigned int) format.mFramesPerPacket, (unsigned int) format.mReserved);
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GST_DEBUG_OBJECT (osxbuf, "Setting format for AudioUnit");
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scope = osxbuf->is_src ? kAudioUnitScope_Output : kAudioUnitScope_Input;
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element = osxbuf->is_src ? 1 : 0;
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propertySize = sizeof (format);
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status = AudioUnitSetProperty (osxbuf->audiounit,
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kAudioUnitProperty_StreamFormat, scope, element, &format, propertySize);
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if (status) {
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GST_WARNING_OBJECT (osxbuf, "Failed to set audio description: %lx",
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(gulong) status);
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goto done;
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}
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status = AudioUnitSetProperty (osxbuf->audiounit,
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kAudioUnitProperty_AudioChannelLayout,
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scope, element, layout, layoutSize);
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if (status) {
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GST_WARNING_OBJECT (osxbuf, "Failed to set output channel layout: %lx",
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(gulong) status);
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goto done;
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}
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spec->segsize =
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(spec->latency_time * spec->rate / G_USEC_PER_SEC) *
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spec->bytes_per_sample;
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spec->segtotal = spec->buffer_time / spec->latency_time;
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/* create AudioBufferList needed for recording */
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if (osxbuf->is_src) {
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propertySize = sizeof (frameSize);
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status = AudioUnitGetProperty (osxbuf->audiounit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, /* N/A for global */
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&frameSize, &propertySize);
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if (status) {
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GST_WARNING_OBJECT (osxbuf, "Failed to get frame size: %lx",
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(gulong) status);
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goto done;
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}
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osxbuf->recBufferList = buffer_list_alloc (format.mChannelsPerFrame,
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frameSize * format.mBytesPerFrame);
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}
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buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
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memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
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osxbuf->segoffset = 0;
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status = AudioUnitInitialize (osxbuf->audiounit);
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if (status) {
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gst_buffer_unref (buf->data);
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buf->data = NULL;
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if (osxbuf->recBufferList) {
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buffer_list_free (osxbuf->recBufferList);
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osxbuf->recBufferList = NULL;
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}
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GST_WARNING_OBJECT (osxbuf,
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"Failed to initialise AudioUnit: %d", (int) status);
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goto done;
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}
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GST_DEBUG_OBJECT (osxbuf, "osx ring buffer acquired");
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ret = TRUE;
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done:
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g_free (layout);
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return ret;
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}
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static gboolean
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gst_osx_ring_buffer_release (GstRingBuffer * buf)
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{
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GstOsxRingBuffer *osxbuf;
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|
osxbuf = GST_OSX_RING_BUFFER (buf);
|
|
|
|
AudioUnitUninitialize (osxbuf->audiounit);
|
|
|
|
gst_buffer_unref (buf->data);
|
|
buf->data = NULL;
|
|
|
|
if (osxbuf->recBufferList) {
|
|
buffer_list_free (osxbuf->recBufferList);
|
|
osxbuf->recBufferList = NULL;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_osx_ring_buffer_remove_render_callback (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
AURenderCallbackStruct input;
|
|
OSStatus status;
|
|
|
|
/* Deactivate the render callback by calling SetRenderCallback with a NULL
|
|
* inputProc.
|
|
*/
|
|
input.inputProc = NULL;
|
|
input.inputProcRefCon = NULL;
|
|
|
|
status = AudioUnitSetProperty (osxbuf->audiounit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0, /* N/A for global */
|
|
&input, sizeof (input));
|
|
|
|
if (status) {
|
|
GST_WARNING_OBJECT (osxbuf, "Failed to remove render callback");
|
|
}
|
|
|
|
/* Remove the RenderNotify too */
|
|
status = AudioUnitRemoveRenderNotify (osxbuf->audiounit,
|
|
(AURenderCallback) gst_osx_ring_buffer_render_notify, osxbuf);
|
|
|
|
if (status) {
|
|
GST_WARNING_OBJECT (osxbuf, "Failed to remove render notify callback");
|
|
}
|
|
|
|
/* We're deactivated.. */
|
|
osxbuf->io_proc_needs_deactivation = FALSE;
|
|
osxbuf->io_proc_active = FALSE;
|
|
}
|
|
|
|
static OSStatus
|
|
gst_osx_ring_buffer_render_notify (GstOsxRingBuffer * osxbuf,
|
|
AudioUnitRenderActionFlags * ioActionFlags,
|
|
const AudioTimeStamp * inTimeStamp,
|
|
unsigned int inBusNumber,
|
|
unsigned int inNumberFrames, AudioBufferList * ioData)
|
|
{
|
|
/* Before rendering a frame, we get the PreRender notification.
|
|
* Here, we detach the RenderCallback if we've been paused.
|
|
*
|
|
* This is necessary (rather than just directly detaching it) to work
|
|
* around some thread-safety issues in CoreAudio
|
|
*/
|
|
if ((*ioActionFlags) & kAudioUnitRenderAction_PreRender) {
|
|
if (osxbuf->io_proc_needs_deactivation) {
|
|
gst_osx_ring_buffer_remove_render_callback (osxbuf);
|
|
}
|
|
}
|
|
|
|
return noErr;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_start (GstRingBuffer * buf)
|
|
{
|
|
OSStatus status;
|
|
GstOsxRingBuffer *osxbuf;
|
|
AURenderCallbackStruct input;
|
|
AudioUnitPropertyID callback_type;
|
|
|
|
osxbuf = GST_OSX_RING_BUFFER (buf);
|
|
|
|
GST_DEBUG ("osx ring buffer start ioproc: 0x%p device_id %lu",
|
|
osxbuf->element->io_proc, (gulong) osxbuf->device_id);
|
|
if (!osxbuf->io_proc_active) {
|
|
callback_type = osxbuf->is_src ?
|
|
kAudioOutputUnitProperty_SetInputCallback :
|
|
kAudioUnitProperty_SetRenderCallback;
|
|
|
|
input.inputProc = (AURenderCallback) osxbuf->element->io_proc;
|
|
input.inputProcRefCon = osxbuf;
|
|
|
|
status = AudioUnitSetProperty (osxbuf->audiounit, callback_type, kAudioUnitScope_Global, 0, /* N/A for global */
|
|
&input, sizeof (input));
|
|
|
|
if (status) {
|
|
GST_WARNING ("AudioUnitSetProperty returned %d", (int) status);
|
|
return FALSE;
|
|
}
|
|
// ### does it make sense to do this notify stuff for input mode?
|
|
status = AudioUnitAddRenderNotify (osxbuf->audiounit,
|
|
(AURenderCallback) gst_osx_ring_buffer_render_notify, osxbuf);
|
|
|
|
if (status) {
|
|
GST_WARNING ("AudioUnitAddRenderNotify returned %d", (int) status);
|
|
return FALSE;
|
|
}
|
|
|
|
osxbuf->io_proc_active = TRUE;
|
|
}
|
|
|
|
osxbuf->io_proc_needs_deactivation = FALSE;
|
|
|
|
status = AudioOutputUnitStart (osxbuf->audiounit);
|
|
if (status) {
|
|
GST_WARNING ("AudioOutputUnitStart returned %d", (int) status);
|
|
return FALSE;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
// ###
|
|
static gboolean
|
|
gst_osx_ring_buffer_pause (GstRingBuffer * buf)
|
|
{
|
|
GstOsxRingBuffer *osxbuf = GST_OSX_RING_BUFFER (buf);
|
|
|
|
GST_DEBUG ("osx ring buffer pause ioproc: 0x%p device_id %lu",
|
|
osxbuf->element->io_proc, (gulong) osxbuf->device_id);
|
|
if (osxbuf->io_proc_active) {
|
|
/* CoreAudio isn't threadsafe enough to do this here; we must deactivate
|
|
* the render callback elsewhere. See:
|
|
* http://lists.apple.com/archives/Coreaudio-api/2006/Mar/msg00010.html
|
|
*/
|
|
osxbuf->io_proc_needs_deactivation = TRUE;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
// ###
|
|
static gboolean
|
|
gst_osx_ring_buffer_stop (GstRingBuffer * buf)
|
|
{
|
|
OSErr status;
|
|
GstOsxRingBuffer *osxbuf;
|
|
|
|
osxbuf = GST_OSX_RING_BUFFER (buf);
|
|
|
|
GST_DEBUG ("osx ring buffer stop ioproc: 0x%p device_id %lu",
|
|
osxbuf->element->io_proc, (gulong) osxbuf->device_id);
|
|
|
|
status = AudioOutputUnitStop (osxbuf->audiounit);
|
|
if (status)
|
|
GST_WARNING ("AudioOutputUnitStop returned %d", (int) status);
|
|
|
|
// ###: why is it okay to directly remove from here but not from pause() ?
|
|
if (osxbuf->io_proc_active) {
|
|
gst_osx_ring_buffer_remove_render_callback (osxbuf);
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static guint
|
|
gst_osx_ring_buffer_delay (GstRingBuffer * buf)
|
|
{
|
|
double latency;
|
|
UInt32 size = sizeof (double);
|
|
GstOsxRingBuffer *osxbuf;
|
|
OSStatus status;
|
|
guint samples;
|
|
|
|
osxbuf = GST_OSX_RING_BUFFER (buf);
|
|
|
|
status = AudioUnitGetProperty (osxbuf->audiounit, kAudioUnitProperty_Latency, kAudioUnitScope_Global, 0, /* N/A for global */
|
|
&latency, &size);
|
|
|
|
if (status) {
|
|
GST_WARNING_OBJECT (buf, "Failed to get latency: %d", (int) status);
|
|
return 0;
|
|
}
|
|
|
|
samples = latency * GST_RING_BUFFER (buf)->spec.rate;
|
|
GST_DEBUG_OBJECT (buf, "Got latency: %f seconds -> %d samples", latency,
|
|
samples);
|
|
return samples;
|
|
}
|
|
|
|
static AudioBufferList *
|
|
buffer_list_alloc (int channels, int size)
|
|
{
|
|
AudioBufferList *list;
|
|
int total_size;
|
|
int n;
|
|
|
|
total_size = sizeof (AudioBufferList) + 1 * sizeof (AudioBuffer);
|
|
list = (AudioBufferList *) g_malloc (total_size);
|
|
|
|
list->mNumberBuffers = 1;
|
|
for (n = 0; n < (int) list->mNumberBuffers; ++n) {
|
|
list->mBuffers[n].mNumberChannels = channels;
|
|
list->mBuffers[n].mDataByteSize = size;
|
|
list->mBuffers[n].mData = g_malloc (size);
|
|
}
|
|
|
|
return list;
|
|
}
|
|
|
|
static void
|
|
buffer_list_free (AudioBufferList * list)
|
|
{
|
|
int n;
|
|
|
|
for (n = 0; n < (int) list->mNumberBuffers; ++n) {
|
|
if (list->mBuffers[n].mData)
|
|
g_free (list->mBuffers[n].mData);
|
|
}
|
|
|
|
g_free (list);
|
|
}
|