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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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312 lines
9.5 KiB
C
312 lines
9.5 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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* Copyright (C) 2012 Fluendo S.A. <support@fluendo.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., 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 <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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#include <unistd.h> /* for getpid() */
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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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#include "gstosxcoreaudio.h"
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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 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,
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GstRingBuffer, 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_osx_ring_buffer_dispose;
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gobject_class->finalize = 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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ringbuffer->core_audio = gst_core_audio_new (GST_OBJECT (ringbuffer));
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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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GstOsxRingBuffer *osxbuf;
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osxbuf = GST_OSX_RING_BUFFER (object);
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if (osxbuf->core_audio) {
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g_object_unref (osxbuf->core_audio);
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osxbuf->core_audio = NULL;
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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 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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osxbuf = GST_OSX_RING_BUFFER (buf);
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return gst_core_audio_open (osxbuf->core_audio);
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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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return gst_core_audio_close (osxbuf->core_audio);
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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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gboolean ret = FALSE, is_passthrough = FALSE;
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GstOsxRingBuffer *osxbuf;
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AudioStreamBasicDescription format;
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osxbuf = GST_OSX_RING_BUFFER (buf);
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if (RINGBUFFER_IS_SPDIF (spec->type)) {
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format.mFormatID = kAudioFormat60958AC3;
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format.mSampleRate = (double) spec->rate;
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format.mChannelsPerFrame = 2;
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format.mFormatFlags = kAudioFormatFlagIsSignedInteger |
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kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonMixable;
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format.mBytesPerFrame = 0;
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format.mBitsPerChannel = 16;
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format.mBytesPerPacket = 6144;
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format.mFramesPerPacket = 1536;
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format.mReserved = 0;
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spec->segsize = 6144;
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spec->segtotal = 10;
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is_passthrough = TRUE;
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} else {
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int width, depth;
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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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if (spec->type == GST_BUFTYPE_FLOAT) {
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format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
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width = depth = spec->width;
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} else {
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format.mFormatFlags = kAudioFormatFlagIsSignedInteger;
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width = spec->width;
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depth = spec->depth;
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if (width == depth) {
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format.mFormatFlags |= kAudioFormatFlagIsPacked;
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} else {
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format.mFormatFlags |= kAudioFormatFlagIsAlignedHigh;
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}
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if (spec->bigend) {
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format.mFormatFlags |= kAudioFormatFlagIsBigEndian;
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}
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}
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format.mBytesPerFrame = spec->channels * (width >> 3);
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format.mBitsPerChannel = depth;
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format.mBytesPerPacket = spec->channels * (width >> 3);
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format.mFramesPerPacket = 1;
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format.mReserved = 0;
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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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is_passthrough = FALSE;
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}
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GST_DEBUG_OBJECT (osxbuf, "Format: " CORE_AUDIO_FORMAT,
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CORE_AUDIO_FORMAT_ARGS (format));
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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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ret = gst_core_audio_initialize (osxbuf->core_audio, format, spec->caps,
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is_passthrough);
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if (!ret) {
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gst_buffer_unref (buf->data);
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buf->data = NULL;
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}
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osxbuf->segoffset = 0;
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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);
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gst_core_audio_unitialize (osxbuf->core_audio);
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gst_buffer_unref (buf->data);
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buf->data = NULL;
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return TRUE;
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}
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static gboolean
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gst_osx_ring_buffer_start (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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return gst_core_audio_start_processing (osxbuf->core_audio);
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}
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static gboolean
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gst_osx_ring_buffer_pause (GstRingBuffer * buf)
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{
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GstOsxRingBuffer *osxbuf = GST_OSX_RING_BUFFER (buf);
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return gst_core_audio_pause_processing (osxbuf->core_audio);
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}
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static gboolean
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gst_osx_ring_buffer_stop (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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gst_core_audio_stop_processing (osxbuf->core_audio);
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return TRUE;
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}
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static guint
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gst_osx_ring_buffer_delay (GstRingBuffer * buf)
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{
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GstOsxRingBuffer *osxbuf;
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double latency;
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guint samples;
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osxbuf = GST_OSX_RING_BUFFER (buf);
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if (!gst_core_audio_get_samples_and_latency (osxbuf->core_audio,
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GST_RING_BUFFER (buf)->spec.rate, &samples, &latency)) {
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return 0;
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}
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GST_DEBUG_OBJECT (buf, "Got latency: %f seconds -> %d samples",
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latency, samples);
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return samples;
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}
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