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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1241 lines
36 KiB
C
1241 lines
36 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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#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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#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 gst_osx_ring_buffer_remove_render_callback (GstOsxRingBuffer *
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osxbuf);
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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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/* Nothing to do right now */
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ringbuffer->is_passthrough = FALSE;
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ringbuffer->hog_pid = -1;
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ringbuffer->disabled_mixing = FALSE;
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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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AudioStreamBasicDescription asbd_in;
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UInt32 propertySize;
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/* Create a HALOutput AudioUnit.
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* This is the lowest-level output API that is actually sensibly
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* usable (the lower level ones require that you do
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* channel-remapping yourself, and the CoreAudio channel mapping
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* is sufficiently complex that doing so would be very difficult)
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*
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* Note that for input we request an output unit even though
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* we will do input with it.
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* 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_ERROR_OBJECT (osxbuf, "Couldn't open HALOutput component %"
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GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
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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: %"
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GST_FOURCC_FORMAT, GST_FOURCC_ARGS (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: %"
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GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
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return NULL;
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}
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}
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GST_DEBUG_OBJECT (osxbuf, "Created HALOutput AudioUnit: %p", unit);
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if (input) {
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GstOsxAudioSrc *src = GST_OSX_AUDIO_SRC (GST_OBJECT_PARENT (osxbuf));
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propertySize = sizeof (asbd_in);
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status = AudioUnitGetProperty (unit,
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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 (unit);
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GST_WARNING_OBJECT (osxbuf,
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"Unable to obtain device properties: %" GST_FOURCC_FORMAT,
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GST_FOURCC_ARGS (status));
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return NULL;
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}
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src->deviceChannels = asbd_in.mChannelsPerFrame;
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} else {
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GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (GST_OBJECT_PARENT (osxbuf));
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/* needed for the sink's volume control */
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sink->audiounit = unit;
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}
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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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osxbuf = GST_OSX_RING_BUFFER (buf);
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/* The following is needed to instruct HAL to create their own
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* thread to handle the notifications. */
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_audio_system_set_runloop (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->audiounit) {
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return FALSE;
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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 AudioBufferList *
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buffer_list_alloc (int channels, int size)
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{
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AudioBufferList *list;
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int total_size;
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int n;
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total_size = sizeof (AudioBufferList) + 1 * sizeof (AudioBuffer);
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list = (AudioBufferList *) g_malloc (total_size);
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list->mNumberBuffers = 1;
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for (n = 0; n < (int) list->mNumberBuffers; ++n) {
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list->mBuffers[n].mNumberChannels = channels;
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list->mBuffers[n].mDataByteSize = size;
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list->mBuffers[n].mData = g_malloc (size);
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}
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return list;
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}
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static void
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buffer_list_free (AudioBufferList * list)
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{
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int n;
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for (n = 0; n < (int) list->mNumberBuffers; ++n) {
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if (list->mBuffers[n].mData)
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g_free (list->mBuffers[n].mData);
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}
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g_free (list);
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}
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typedef struct
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{
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GMutex *lock;
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GCond *cond;
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} PropertyMutex;
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static OSStatus
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_audio_stream_format_listener (AudioObjectID inObjectID,
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UInt32 inNumberAddresses,
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const AudioObjectPropertyAddress inAddresses[], void *inClientData)
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{
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OSStatus status = noErr;
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guint i;
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PropertyMutex *prop_mutex = inClientData;
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for (i = 0; i < inNumberAddresses; i++) {
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if (inAddresses[i].mSelector == kAudioStreamPropertyPhysicalFormat) {
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g_mutex_lock (prop_mutex->lock);
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g_cond_signal (prop_mutex->cond);
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g_mutex_unlock (prop_mutex->lock);
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break;
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}
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}
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return (status);
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}
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|
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static gboolean
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_audio_stream_change_format (AudioStreamID stream_id,
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AudioStreamBasicDescription format)
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{
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OSStatus status = noErr;
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gint i;
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gboolean ret = FALSE;
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AudioStreamBasicDescription cformat;
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PropertyMutex prop_mutex;
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AudioObjectPropertyAddress formatAddress = {
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kAudioStreamPropertyPhysicalFormat,
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kAudioObjectPropertyScopeGlobal,
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kAudioObjectPropertyElementMaster
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};
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GST_DEBUG ("setting stream format: " CORE_AUDIO_FORMAT,
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CORE_AUDIO_FORMAT_ARGS (format));
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/* Condition because SetProperty is asynchronous */
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prop_mutex.lock = g_mutex_new ();
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prop_mutex.cond = g_cond_new ();
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g_mutex_lock (prop_mutex.lock);
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|
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/* Install the property listener to serialize the operations */
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status = AudioObjectAddPropertyListener (stream_id, &formatAddress,
|
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_audio_stream_format_listener, (void *) &prop_mutex);
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if (status != noErr) {
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GST_ERROR ("AudioObjectAddPropertyListener failed: %"
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GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
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goto done;
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}
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|
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/* Change the format */
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if (!_audio_stream_set_current_format (stream_id, format)) {
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goto done;
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}
|
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|
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/* The AudioObjectSetProperty is not only asynchronous
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* it is also not atomic in its behaviour.
|
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* Therefore we check 4 times before we really give up. */
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for (i = 0; i < 4; i++) {
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GTimeVal timeout;
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|
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g_get_current_time (&timeout);
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g_time_val_add (&timeout, 250000);
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|
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if (!g_cond_timed_wait (prop_mutex.cond, prop_mutex.lock, &timeout)) {
|
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GST_LOG ("timeout...");
|
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}
|
|
|
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if (_audio_stream_get_current_format (stream_id, &cformat)) {
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GST_DEBUG ("current stream format: " CORE_AUDIO_FORMAT,
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CORE_AUDIO_FORMAT_ARGS (cformat));
|
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|
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if (cformat.mSampleRate == format.mSampleRate &&
|
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cformat.mFormatID == format.mFormatID &&
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cformat.mFramesPerPacket == format.mFramesPerPacket) {
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/* The right format is now active */
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break;
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}
|
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}
|
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}
|
|
|
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if (cformat.mSampleRate != format.mSampleRate ||
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cformat.mFormatID != format.mFormatID ||
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cformat.mFramesPerPacket != format.mFramesPerPacket) {
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goto done;
|
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}
|
|
|
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ret = TRUE;
|
|
|
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done:
|
|
/* Removing the property listener */
|
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status = AudioObjectRemovePropertyListener (stream_id,
|
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&formatAddress, _audio_stream_format_listener, (void *) &prop_mutex);
|
|
if (status != noErr) {
|
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GST_ERROR ("AudioObjectRemovePropertyListener failed: %"
|
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GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
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}
|
|
/* Destroy the lock and condition */
|
|
g_mutex_unlock (prop_mutex.lock);
|
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g_mutex_free (prop_mutex.lock);
|
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g_cond_free (prop_mutex.cond);
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|
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return ret;
|
|
}
|
|
|
|
static OSStatus
|
|
_audio_stream_hardware_changed_listener (AudioObjectID inObjectID,
|
|
UInt32 inNumberAddresses,
|
|
const AudioObjectPropertyAddress inAddresses[], void *inClientData)
|
|
{
|
|
OSStatus status = noErr;
|
|
guint i;
|
|
GstOsxRingBuffer *osxbuf = inClientData;
|
|
|
|
for (i = 0; i < inNumberAddresses; i++) {
|
|
if (inAddresses[i].mSelector == kAudioDevicePropertyDeviceHasChanged) {
|
|
if (!_audio_device_is_spdif_avail (osxbuf->device_id)) {
|
|
GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (GST_OBJECT_PARENT (osxbuf));
|
|
GST_ELEMENT_ERROR (sink, RESOURCE, FAILED,
|
|
("SPDIF output no longer available"),
|
|
("Audio device is reporting that SPDIF output isn't available"));
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
return (status);
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_monitorize_spdif (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
OSStatus status = noErr;
|
|
gboolean ret = TRUE;
|
|
|
|
AudioObjectPropertyAddress propAddress = {
|
|
kAudioDevicePropertyDeviceHasChanged,
|
|
kAudioObjectPropertyScopeGlobal,
|
|
kAudioObjectPropertyElementMaster
|
|
};
|
|
|
|
/* Install the property listener */
|
|
status = AudioObjectAddPropertyListener (osxbuf->device_id,
|
|
&propAddress, _audio_stream_hardware_changed_listener, (void *) osxbuf);
|
|
if (status != noErr) {
|
|
GST_ERROR ("AudioObjectAddPropertyListener failed: %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
ret = FALSE;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_unmonitorize_spdif (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
OSStatus status = noErr;
|
|
gboolean ret = TRUE;
|
|
|
|
AudioObjectPropertyAddress propAddress = {
|
|
kAudioDevicePropertyDeviceHasChanged,
|
|
kAudioObjectPropertyScopeGlobal,
|
|
kAudioObjectPropertyElementMaster
|
|
};
|
|
|
|
/* Remove the property listener */
|
|
status = AudioObjectRemovePropertyListener (osxbuf->device_id,
|
|
&propAddress, _audio_stream_hardware_changed_listener, (void *) osxbuf);
|
|
if (status != noErr) {
|
|
GST_ERROR ("AudioObjectRemovePropertyListener failed: %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
ret = FALSE;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_open_spdif (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
gboolean res = FALSE;
|
|
pid_t hog_pid, own_pid = getpid ();
|
|
|
|
/* We need the device in exclusive and disable the mixing */
|
|
hog_pid = _audio_device_get_hog (osxbuf->device_id);
|
|
|
|
if (hog_pid != -1 && hog_pid != own_pid) {
|
|
GST_DEBUG_OBJECT (osxbuf,
|
|
"device is currently in use by another application");
|
|
goto done;
|
|
}
|
|
|
|
if (_audio_device_set_hog (osxbuf->device_id, own_pid)) {
|
|
osxbuf->hog_pid = own_pid;
|
|
}
|
|
|
|
if (_audio_device_set_mixing (osxbuf->device_id, FALSE)) {
|
|
GST_DEBUG_OBJECT (osxbuf, "disabled mixing on the device");
|
|
osxbuf->disabled_mixing = TRUE;
|
|
}
|
|
|
|
res = TRUE;
|
|
done:
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_close_spdif (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
pid_t hog_pid;
|
|
|
|
gst_osx_ring_buffer_unmonitorize_spdif (osxbuf);
|
|
|
|
if (osxbuf->revert_format) {
|
|
if (!_audio_stream_change_format (osxbuf->stream_id,
|
|
osxbuf->original_format)) {
|
|
GST_WARNING ("Format revert failed");
|
|
}
|
|
osxbuf->revert_format = FALSE;
|
|
}
|
|
|
|
if (osxbuf->disabled_mixing) {
|
|
_audio_device_set_mixing (osxbuf->device_id, TRUE);
|
|
osxbuf->disabled_mixing = FALSE;
|
|
}
|
|
|
|
if (osxbuf->hog_pid != -1) {
|
|
hog_pid = _audio_device_get_hog (osxbuf->device_id);
|
|
if (hog_pid == getpid ()) {
|
|
if (_audio_device_set_hog (osxbuf->device_id, -1)) {
|
|
osxbuf->hog_pid = -1;
|
|
}
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static OSStatus
|
|
gst_osx_ring_buffer_io_proc_spdif (AudioDeviceID inDevice,
|
|
const AudioTimeStamp * inNow,
|
|
const void *inInputData,
|
|
const AudioTimeStamp * inTimestamp,
|
|
AudioBufferList * bufferList,
|
|
const AudioTimeStamp * inOutputTime, GstOsxRingBuffer * osxbuf)
|
|
{
|
|
OSStatus status;
|
|
|
|
status = osxbuf->element->io_proc (osxbuf, NULL, inTimestamp, 0, 0,
|
|
bufferList);
|
|
|
|
return status;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_acquire_spdif (GstOsxRingBuffer * osxbuf,
|
|
AudioStreamBasicDescription format)
|
|
{
|
|
AudioStreamID *streams = NULL;
|
|
gint i, j, nstreams = 0;
|
|
gboolean ret = FALSE;
|
|
|
|
if (!gst_osx_ring_buffer_open_spdif (osxbuf))
|
|
goto done;
|
|
|
|
streams = _audio_device_get_streams (osxbuf->device_id, &nstreams);
|
|
|
|
for (i = 0; i < nstreams; i++) {
|
|
AudioStreamRangedDescription *formats = NULL;
|
|
gint nformats = 0;
|
|
|
|
formats = _audio_stream_get_formats (streams[i], &nformats);
|
|
|
|
if (formats) {
|
|
gboolean is_spdif = FALSE;
|
|
|
|
/* Check if one of the supported formats is a digital format */
|
|
for (j = 0; j < nformats; j++) {
|
|
if (CORE_AUDIO_FORMAT_IS_SPDIF (formats[j])) {
|
|
is_spdif = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (is_spdif) {
|
|
/* if this stream supports a digital (cac3) format,
|
|
* then go set it. */
|
|
gint requested_rate_format = -1;
|
|
gint current_rate_format = -1;
|
|
gint backup_rate_format = -1;
|
|
|
|
osxbuf->stream_id = streams[i];
|
|
osxbuf->stream_idx = i;
|
|
|
|
if (!osxbuf->revert_format) {
|
|
if (!_audio_stream_get_current_format (osxbuf->stream_id,
|
|
&osxbuf->original_format)) {
|
|
GST_WARNING ("format could not be saved");
|
|
g_free (formats);
|
|
continue;
|
|
}
|
|
osxbuf->revert_format = TRUE;
|
|
}
|
|
|
|
for (j = 0; j < nformats; j++) {
|
|
if (CORE_AUDIO_FORMAT_IS_SPDIF (formats[j])) {
|
|
GST_LOG ("found stream format: " CORE_AUDIO_FORMAT,
|
|
CORE_AUDIO_FORMAT_ARGS (formats[j].mFormat));
|
|
|
|
if (formats[j].mFormat.mSampleRate == format.mSampleRate) {
|
|
requested_rate_format = j;
|
|
break;
|
|
} else if (formats[j].mFormat.mSampleRate ==
|
|
osxbuf->original_format.mSampleRate) {
|
|
current_rate_format = j;
|
|
} else {
|
|
if (backup_rate_format < 0 ||
|
|
formats[j].mFormat.mSampleRate >
|
|
formats[backup_rate_format].mFormat.mSampleRate) {
|
|
backup_rate_format = j;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (requested_rate_format >= 0) {
|
|
/* We prefer to output at the rate of the original audio */
|
|
osxbuf->stream_format = formats[requested_rate_format].mFormat;
|
|
} else if (current_rate_format >= 0) {
|
|
/* If not possible, we will try to use the current rate */
|
|
osxbuf->stream_format = formats[current_rate_format].mFormat;
|
|
} else {
|
|
/* And if we have to, any digital format will be just
|
|
* fine (highest rate possible) */
|
|
osxbuf->stream_format = formats[backup_rate_format].mFormat;
|
|
}
|
|
}
|
|
g_free (formats);
|
|
}
|
|
}
|
|
g_free (streams);
|
|
|
|
GST_DEBUG ("original stream format: " CORE_AUDIO_FORMAT,
|
|
CORE_AUDIO_FORMAT_ARGS (osxbuf->original_format));
|
|
|
|
if (!_audio_stream_change_format (osxbuf->stream_id, osxbuf->stream_format))
|
|
goto done;
|
|
|
|
GST_DEBUG_OBJECT (osxbuf, "osx ring buffer acquired");
|
|
|
|
ret = TRUE;
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_acquire_analog (GstOsxRingBuffer * osxbuf,
|
|
AudioStreamBasicDescription format, GstCaps * caps)
|
|
{
|
|
/* Configure the output stream and allocate ringbuffer memory */
|
|
AudioChannelLayout *layout = NULL;
|
|
OSStatus status;
|
|
UInt32 propertySize;
|
|
int channels = format.mChannelsPerFrame;
|
|
int layoutSize;
|
|
int element;
|
|
int i;
|
|
AudioUnitScope scope;
|
|
gboolean ret = FALSE;
|
|
GstStructure *structure;
|
|
GstAudioChannelPosition *positions;
|
|
UInt32 frameSize;
|
|
|
|
/* Describe channels */
|
|
layoutSize = sizeof (AudioChannelLayout) +
|
|
channels * sizeof (AudioChannelDescription);
|
|
layout = g_malloc (layoutSize);
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
positions = gst_audio_get_channel_positions (structure);
|
|
|
|
layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
|
|
layout->mChannelBitmap = 0; /* Not used */
|
|
layout->mNumberChannelDescriptions = channels;
|
|
for (i = 0; i < channels; i++) {
|
|
if (positions) {
|
|
layout->mChannelDescriptions[i].mChannelLabel =
|
|
gst_audio_channel_position_to_coreaudio_channel_label (positions[i],
|
|
i);
|
|
} else {
|
|
/* Discrete channel numbers are ORed into this */
|
|
layout->mChannelDescriptions[i].mChannelLabel =
|
|
kAudioChannelLabel_Discrete_0 | i;
|
|
}
|
|
|
|
/* Others unused */
|
|
layout->mChannelDescriptions[i].mChannelFlags = 0;
|
|
layout->mChannelDescriptions[i].mCoordinates[0] = 0.f;
|
|
layout->mChannelDescriptions[i].mCoordinates[1] = 0.f;
|
|
layout->mChannelDescriptions[i].mCoordinates[2] = 0.f;
|
|
}
|
|
|
|
if (positions) {
|
|
g_free (positions);
|
|
positions = NULL;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (osxbuf, "Setting format for AudioUnit");
|
|
|
|
scope = osxbuf->is_src ? kAudioUnitScope_Output : kAudioUnitScope_Input;
|
|
element = osxbuf->is_src ? 1 : 0;
|
|
|
|
propertySize = sizeof (AudioStreamBasicDescription);
|
|
status = AudioUnitSetProperty (osxbuf->audiounit,
|
|
kAudioUnitProperty_StreamFormat, scope, element, &format, propertySize);
|
|
|
|
if (status) {
|
|
GST_WARNING_OBJECT (osxbuf,
|
|
"Failed to set audio description: %" GST_FOURCC_FORMAT,
|
|
GST_FOURCC_ARGS (status));
|
|
goto done;
|
|
}
|
|
|
|
if (layoutSize) {
|
|
status = AudioUnitSetProperty (osxbuf->audiounit,
|
|
kAudioUnitProperty_AudioChannelLayout,
|
|
scope, element, layout, layoutSize);
|
|
if (status) {
|
|
GST_WARNING_OBJECT (osxbuf,
|
|
"Failed to set output channel layout: %" GST_FOURCC_FORMAT,
|
|
GST_FOURCC_ARGS (status));
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* create AudioBufferList needed for recording */
|
|
if (osxbuf->is_src) {
|
|
propertySize = sizeof (frameSize);
|
|
status = AudioUnitGetProperty (osxbuf->audiounit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, /* N/A for global */
|
|
&frameSize, &propertySize);
|
|
|
|
if (status) {
|
|
GST_WARNING_OBJECT (osxbuf, "Failed to get frame size: %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
goto done;
|
|
}
|
|
|
|
osxbuf->recBufferList = buffer_list_alloc (channels,
|
|
frameSize * format.mBytesPerFrame);
|
|
}
|
|
|
|
/* Specify which device we're using. */
|
|
GST_DEBUG_OBJECT (osxbuf, "Bind AudioUnit to device %d",
|
|
(int) osxbuf->device_id);
|
|
status = AudioUnitSetProperty (osxbuf->audiounit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, /* N/A for global */
|
|
&osxbuf->device_id, sizeof (AudioDeviceID));
|
|
if (status) {
|
|
GST_ERROR_OBJECT (osxbuf, "Failed binding to device: %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
goto audiounit_error;
|
|
}
|
|
|
|
/* Initialize the AudioUnit */
|
|
status = AudioUnitInitialize (osxbuf->audiounit);
|
|
if (status) {
|
|
GST_ERROR_OBJECT (osxbuf, "Failed to initialise AudioUnit: %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
goto audiounit_error;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (osxbuf, "osx ring buffer acquired");
|
|
|
|
ret = TRUE;
|
|
|
|
done:
|
|
g_free (layout);
|
|
return ret;
|
|
|
|
audiounit_error:
|
|
if (osxbuf->recBufferList) {
|
|
buffer_list_free (osxbuf->recBufferList);
|
|
osxbuf->recBufferList = NULL;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
|
|
{
|
|
gboolean ret = FALSE;
|
|
GstOsxRingBuffer *osxbuf;
|
|
AudioStreamBasicDescription format;
|
|
|
|
osxbuf = GST_OSX_RING_BUFFER (buf);
|
|
|
|
if (RINGBUFFER_IS_SPDIF (spec->type)) {
|
|
format.mFormatID = kAudioFormat60958AC3;
|
|
format.mSampleRate = (double) spec->rate;
|
|
format.mChannelsPerFrame = 2;
|
|
format.mFormatFlags = kAudioFormatFlagIsSignedInteger |
|
|
kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonMixable;
|
|
format.mBytesPerFrame = 0;
|
|
format.mBitsPerChannel = 16;
|
|
format.mBytesPerPacket = 6144;
|
|
format.mFramesPerPacket = 1536;
|
|
format.mReserved = 0;
|
|
spec->segsize = 6144;
|
|
spec->segtotal = 10;
|
|
osxbuf->is_passthrough = TRUE;
|
|
} else {
|
|
int width, depth;
|
|
/* Fill out the audio description we're going to be using */
|
|
format.mFormatID = kAudioFormatLinearPCM;
|
|
format.mSampleRate = (double) spec->rate;
|
|
format.mChannelsPerFrame = spec->channels;
|
|
if (spec->type == GST_BUFTYPE_FLOAT) {
|
|
format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
|
|
width = depth = spec->width;
|
|
} else {
|
|
format.mFormatFlags = kAudioFormatFlagIsSignedInteger;
|
|
width = spec->width;
|
|
depth = spec->depth;
|
|
if (width == depth) {
|
|
format.mFormatFlags |= kAudioFormatFlagIsPacked;
|
|
} else {
|
|
format.mFormatFlags |= kAudioFormatFlagIsAlignedHigh;
|
|
}
|
|
if (spec->bigend) {
|
|
format.mFormatFlags |= kAudioFormatFlagIsBigEndian;
|
|
}
|
|
}
|
|
format.mBytesPerFrame = spec->channels * (width >> 3);
|
|
format.mBitsPerChannel = depth;
|
|
format.mBytesPerPacket = spec->channels * (width >> 3);
|
|
format.mFramesPerPacket = 1;
|
|
format.mReserved = 0;
|
|
spec->segsize =
|
|
(spec->latency_time * spec->rate / G_USEC_PER_SEC) *
|
|
spec->bytes_per_sample;
|
|
spec->segtotal = spec->buffer_time / spec->latency_time;
|
|
osxbuf->stream_idx = 0;
|
|
osxbuf->is_passthrough = FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (osxbuf, "Format: " CORE_AUDIO_FORMAT,
|
|
CORE_AUDIO_FORMAT_ARGS (format));
|
|
|
|
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
|
|
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
|
|
|
|
if (osxbuf->is_passthrough) {
|
|
ret = gst_osx_ring_buffer_acquire_spdif (osxbuf, format);
|
|
if (ret) {
|
|
gst_osx_ring_buffer_monitorize_spdif (osxbuf);
|
|
}
|
|
} else {
|
|
ret = gst_osx_ring_buffer_acquire_analog (osxbuf, format, spec->caps);
|
|
}
|
|
|
|
if (!ret) {
|
|
gst_buffer_unref (buf->data);
|
|
buf->data = NULL;
|
|
}
|
|
|
|
osxbuf->segoffset = 0;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_release (GstRingBuffer * buf)
|
|
{
|
|
GstOsxRingBuffer *osxbuf;
|
|
|
|
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 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 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 %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
}
|
|
|
|
/* 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 %" GST_FOURCC_FORMAT,
|
|
GST_FOURCC_ARGS (status));
|
|
}
|
|
|
|
/* We're deactivated.. */
|
|
osxbuf->io_proc_needs_deactivation = FALSE;
|
|
osxbuf->io_proc_active = FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_io_proc_start (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
OSStatus status;
|
|
AURenderCallbackStruct input;
|
|
AudioUnitPropertyID callback_type;
|
|
|
|
GST_DEBUG ("osx ring buffer start ioproc: %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_ERROR ("AudioUnitSetProperty failed: %" GST_FOURCC_FORMAT,
|
|
GST_FOURCC_ARGS (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_ERROR ("AudioUnitAddRenderNotify failed %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
return FALSE;
|
|
}
|
|
|
|
osxbuf->io_proc_active = TRUE;
|
|
}
|
|
|
|
osxbuf->io_proc_needs_deactivation = FALSE;
|
|
|
|
status = AudioOutputUnitStart (osxbuf->audiounit);
|
|
if (status) {
|
|
GST_ERROR ("AudioOutputUnitStart failed: %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
return FALSE;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_io_proc_stop (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
OSErr status;
|
|
|
|
GST_DEBUG ("osx ring buffer stop ioproc: %p device_id %lu",
|
|
osxbuf->element->io_proc, (gulong) osxbuf->device_id);
|
|
|
|
status = AudioOutputUnitStop (osxbuf->audiounit);
|
|
if (status) {
|
|
GST_WARNING ("AudioOutputUnitStop failed: %" GST_FOURCC_FORMAT,
|
|
GST_FOURCC_ARGS (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 void
|
|
gst_osx_ring_buffer_remove_render_spdif_callback (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
OSStatus status;
|
|
|
|
/* Deactivate the render callback by calling
|
|
* AudioDeviceDestroyIOProcID */
|
|
status = AudioDeviceDestroyIOProcID (osxbuf->device_id, osxbuf->procID);
|
|
if (status != noErr) {
|
|
GST_ERROR ("AudioDeviceDestroyIOProcID failed: %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
}
|
|
|
|
GST_DEBUG ("osx ring buffer removed ioproc ID: %p device_id %lu",
|
|
osxbuf->procID, (gulong) osxbuf->device_id);
|
|
|
|
/* We're deactivated.. */
|
|
osxbuf->procID = 0;
|
|
osxbuf->io_proc_needs_deactivation = FALSE;
|
|
osxbuf->io_proc_active = FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_io_proc_spdif_start (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
OSErr status;
|
|
|
|
GST_DEBUG ("osx ring buffer start ioproc ID: %p device_id %lu",
|
|
osxbuf->procID, (gulong) osxbuf->device_id);
|
|
|
|
if (!osxbuf->io_proc_active) {
|
|
/* Add IOProc callback */
|
|
status = AudioDeviceCreateIOProcID (osxbuf->device_id,
|
|
(AudioDeviceIOProc) gst_osx_ring_buffer_io_proc_spdif,
|
|
(void *) osxbuf, &osxbuf->procID);
|
|
if (status != noErr) {
|
|
GST_ERROR ("AudioDeviceCreateIOProcID failed: %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (status));
|
|
return FALSE;
|
|
}
|
|
osxbuf->io_proc_active = TRUE;
|
|
}
|
|
|
|
osxbuf->io_proc_needs_deactivation = FALSE;
|
|
|
|
/* Start device */
|
|
status = AudioDeviceStart (osxbuf->device_id, osxbuf->procID);
|
|
if (status != noErr) {
|
|
GST_ERROR ("AudioDeviceStart failed: %" GST_FOURCC_FORMAT,
|
|
GST_FOURCC_ARGS (status));
|
|
return FALSE;
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_io_proc_spdif_stop (GstOsxRingBuffer * osxbuf)
|
|
{
|
|
OSErr status;
|
|
|
|
/* Stop device */
|
|
status = AudioDeviceStop (osxbuf->device_id, osxbuf->procID);
|
|
if (status != noErr) {
|
|
GST_ERROR ("AudioDeviceStop failed: %" GST_FOURCC_FORMAT,
|
|
GST_FOURCC_ARGS (status));
|
|
}
|
|
|
|
GST_DEBUG ("osx ring buffer stop ioproc ID: %p device_id %lu",
|
|
osxbuf->procID, (gulong) osxbuf->device_id);
|
|
|
|
if (osxbuf->io_proc_active) {
|
|
gst_osx_ring_buffer_remove_render_spdif_callback (osxbuf);
|
|
}
|
|
|
|
gst_osx_ring_buffer_close_spdif (osxbuf);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_start (GstRingBuffer * buf)
|
|
{
|
|
GstOsxRingBuffer *osxbuf;
|
|
|
|
osxbuf = GST_OSX_RING_BUFFER (buf);
|
|
|
|
if (osxbuf->is_passthrough) {
|
|
return gst_osx_ring_buffer_io_proc_spdif_start (osxbuf);
|
|
} else {
|
|
return gst_osx_ring_buffer_io_proc_start (osxbuf);
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_osx_ring_buffer_pause (GstRingBuffer * buf)
|
|
{
|
|
GstOsxRingBuffer *osxbuf = GST_OSX_RING_BUFFER (buf);
|
|
|
|
if (osxbuf->is_passthrough) {
|
|
GST_DEBUG ("osx ring buffer pause ioproc ID: %p device_id %lu",
|
|
osxbuf->procID, (gulong) osxbuf->device_id);
|
|
|
|
if (osxbuf->io_proc_active) {
|
|
gst_osx_ring_buffer_remove_render_spdif_callback (osxbuf);
|
|
}
|
|
} else {
|
|
GST_DEBUG ("osx ring buffer pause ioproc: %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)
|
|
{
|
|
GstOsxRingBuffer *osxbuf;
|
|
|
|
osxbuf = GST_OSX_RING_BUFFER (buf);
|
|
|
|
if (osxbuf->is_passthrough) {
|
|
gst_osx_ring_buffer_io_proc_spdif_stop (osxbuf);
|
|
} else {
|
|
gst_osx_ring_buffer_io_proc_stop (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);
|
|
|
|
if (osxbuf->is_passthrough) {
|
|
samples = _audio_device_get_latency (osxbuf->device_id);
|
|
samples += _audio_stream_get_latency (osxbuf->stream_id);
|
|
latency = (double) samples / GST_RING_BUFFER (buf)->spec.rate;
|
|
} else {
|
|
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: %"
|
|
GST_FOURCC_FORMAT, GST_FOURCC_ARGS (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;
|
|
}
|