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Found that osxaudiosink could not be added standalone in gst-full build using -Dgst-full-elements=osxaudio:osxaudiosink because element registration was done at the plugin level. Now src/sink elements and deviceprovider have their individual registration. Copied/adapted from the alsa plugin. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5419>
582 lines
17 KiB
C
582 lines
17 KiB
C
/*
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* GStreamer
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* Copyright (C) 2005,2006 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* Copyright (C) 2007,2008 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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* The development of this code was made possible due to the involvement of
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* Pioneers of the Inevitable, the creators of the Songbird Music player
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*
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*/
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/**
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* SECTION:element-osxaudiosink
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* @title: osxaudiosink
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*
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* This element renders raw audio samples using the CoreAudio api.
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*
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* ## Example pipelines
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* |[
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* gst-launch-1.0 filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osxaudiosink
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* ]| Play an Ogg/Vorbis file.
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*
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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/audio.h>
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#include <gst/audio/audio-channels.h>
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#include <gst/audio/gstaudioiec61937.h>
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#include "gstosxaudiosink.h"
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#include "gstosxaudioelement.h"
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GST_DEBUG_CATEGORY_STATIC (osx_audiosink_debug);
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#define GST_CAT_DEFAULT osx_audiosink_debug
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#include "gstosxcoreaudio.h"
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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ARG_DEVICE,
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ARG_VOLUME
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};
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#define DEFAULT_VOLUME 1.0
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_OSX_AUDIO_SINK_CAPS)
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);
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static void gst_osx_audio_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_osx_audio_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn
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gst_osx_audio_sink_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_osx_audio_sink_query (GstBaseSink * base, GstQuery * query);
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static GstCaps *gst_osx_audio_sink_getcaps (GstBaseSink * base,
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GstCaps * filter);
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static gboolean gst_osx_audio_sink_acceptcaps (GstOsxAudioSink * sink,
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GstCaps * caps);
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static GstBuffer *gst_osx_audio_sink_sink_payload (GstAudioBaseSink * sink,
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GstBuffer * buf);
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static GstAudioRingBuffer
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* gst_osx_audio_sink_create_ringbuffer (GstAudioBaseSink * sink);
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static void gst_osx_audio_sink_osxelement_init (gpointer g_iface,
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gpointer iface_data);
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static void gst_osx_audio_sink_set_volume (GstOsxAudioSink * sink);
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static OSStatus gst_osx_audio_sink_io_proc (GstOsxAudioRingBuffer * buf,
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AudioUnitRenderActionFlags * ioActionFlags,
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const AudioTimeStamp * inTimeStamp,
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UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList);
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static void
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gst_osx_audio_sink_do_init (GType type)
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{
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static const GInterfaceInfo osxelement_info = {
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gst_osx_audio_sink_osxelement_init,
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NULL,
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NULL
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};
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GST_DEBUG_CATEGORY_INIT (osx_audiosink_debug, "osxaudiosink", 0,
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"OSX Audio Sink");
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gst_core_audio_init_debug ();
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GST_DEBUG ("Adding static interface");
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g_type_add_interface_static (type, GST_OSX_AUDIO_ELEMENT_TYPE,
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&osxelement_info);
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}
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#define gst_osx_audio_sink_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstOsxAudioSink, gst_osx_audio_sink,
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GST_TYPE_AUDIO_BASE_SINK, gst_osx_audio_sink_do_init (g_define_type_id));
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GST_ELEMENT_REGISTER_DEFINE (osxaudiosink, "osxaudiosink", GST_RANK_PRIMARY,
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GST_TYPE_OSX_AUDIO_SINK);
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static void
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gst_osx_audio_sink_class_init (GstOsxAudioSinkClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSinkClass *gstbasesink_class;
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GstAudioBaseSinkClass *gstaudiobasesink_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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gstaudiobasesink_class = (GstAudioBaseSinkClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property = gst_osx_audio_sink_set_property;
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gobject_class->get_property = gst_osx_audio_sink_get_property;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_osx_audio_sink_change_state);
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#ifndef HAVE_IOS
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g_object_class_install_property (gobject_class, ARG_DEVICE,
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g_param_spec_int ("device", "Device ID", "Device ID of output device",
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0, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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#endif
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gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_query);
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g_object_class_install_property (gobject_class, ARG_VOLUME,
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g_param_spec_double ("volume", "Volume", "Volume of this stream",
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0, 1.0, 1.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_getcaps);
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gstaudiobasesink_class->create_ringbuffer =
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GST_DEBUG_FUNCPTR (gst_osx_audio_sink_create_ringbuffer);
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gstaudiobasesink_class->payload =
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GST_DEBUG_FUNCPTR (gst_osx_audio_sink_sink_payload);
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gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
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gst_element_class_set_static_metadata (gstelement_class, "Audio Sink (macOS)",
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"Sink/Audio",
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"Output to a sound card on macOS",
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"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
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}
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static void
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gst_osx_audio_sink_init (GstOsxAudioSink * sink)
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{
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GST_DEBUG ("Initialising object");
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sink->device_id = kAudioDeviceUnknown;
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sink->volume = DEFAULT_VOLUME;
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}
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static void
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gst_osx_audio_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (object);
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switch (prop_id) {
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#ifndef HAVE_IOS
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case ARG_DEVICE:
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sink->device_id = g_value_get_int (value);
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break;
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#endif
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case ARG_VOLUME:
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sink->volume = g_value_get_double (value);
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gst_osx_audio_sink_set_volume (sink);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstStateChangeReturn
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gst_osx_audio_sink_change_state (GstElement * element,
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GstStateChange transition)
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{
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GstOsxAudioSink *osxsink = GST_OSX_AUDIO_SINK (element);
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GstOsxAudioRingBuffer *ringbuffer;
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GstStateChangeReturn ret;
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switch (transition) {
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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if (ret == GST_STATE_CHANGE_FAILURE)
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goto out;
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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/* Device has been selected, AudioUnit set up, so initialize volume */
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gst_osx_audio_sink_set_volume (osxsink);
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/* The device is open now, so fix our device_id if it changed */
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ringbuffer =
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GST_OSX_AUDIO_RING_BUFFER (GST_AUDIO_BASE_SINK (osxsink)->ringbuffer);
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if (ringbuffer->core_audio->device_id != osxsink->device_id) {
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osxsink->device_id = ringbuffer->core_audio->device_id;
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g_object_notify (G_OBJECT (osxsink), "device");
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}
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break;
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default:
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break;
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}
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out:
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return ret;
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}
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static void
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gst_osx_audio_sink_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (object);
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switch (prop_id) {
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#ifndef HAVE_IOS
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case ARG_DEVICE:
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g_value_set_int (value, sink->device_id);
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break;
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#endif
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case ARG_VOLUME:
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g_value_set_double (value, sink->volume);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static gboolean
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gst_osx_audio_sink_query (GstBaseSink * base, GstQuery * query)
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{
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GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (base);
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gboolean ret = FALSE;
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_ACCEPT_CAPS:
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{
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GstCaps *caps = NULL;
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gst_query_parse_accept_caps (query, &caps);
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ret = gst_osx_audio_sink_acceptcaps (sink, caps);
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gst_query_set_accept_caps_result (query, ret);
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ret = TRUE;
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break;
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}
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default:
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ret = GST_BASE_SINK_CLASS (parent_class)->query (base, query);
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break;
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}
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return ret;
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}
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static GstCaps *
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gst_osx_audio_sink_getcaps (GstBaseSink * sink, GstCaps * filter)
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{
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GstOsxAudioSink *osxsink;
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GstAudioRingBuffer *buf;
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GstOsxAudioRingBuffer *osxbuf;
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GstCaps *caps, *filtered_caps;
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osxsink = GST_OSX_AUDIO_SINK (sink);
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GST_OBJECT_LOCK (osxsink);
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buf = GST_AUDIO_BASE_SINK (sink)->ringbuffer;
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if (buf)
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gst_object_ref (buf);
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GST_OBJECT_UNLOCK (osxsink);
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if (!buf) {
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GST_DEBUG_OBJECT (sink, "no ring buffer, returning NULL caps");
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return GST_BASE_SINK_CLASS (parent_class)->get_caps (sink, filter);
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}
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osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
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/* protect against cached_caps going away */
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GST_OBJECT_LOCK (buf);
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if (osxbuf->core_audio->cached_caps_valid) {
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GST_LOG_OBJECT (sink, "Returning cached caps");
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caps = gst_caps_ref (osxbuf->core_audio->cached_caps);
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} else if (buf->open) {
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GstCaps *template_caps;
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/* Get template caps */
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template_caps =
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gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (osxsink));
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/* Device is open, let's probe its caps */
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caps = gst_core_audio_probe_caps (osxbuf->core_audio, template_caps);
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gst_caps_replace (&osxbuf->core_audio->cached_caps, caps);
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gst_caps_unref (template_caps);
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} else {
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GST_DEBUG_OBJECT (sink, "ring buffer not open, returning NULL caps");
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caps = NULL;
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}
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GST_OBJECT_UNLOCK (buf);
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gst_object_unref (buf);
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if (!caps)
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return NULL;
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if (!filter)
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return caps;
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/* Take care of filtered caps */
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filtered_caps =
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gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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return filtered_caps;
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}
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static gboolean
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gst_osx_audio_sink_acceptcaps (GstOsxAudioSink * sink, GstCaps * caps)
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{
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GstCaps *pad_caps;
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GstStructure *st;
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gboolean ret = FALSE;
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GstAudioRingBufferSpec spec = { 0 };
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gchar *caps_string = NULL;
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caps_string = gst_caps_to_string (caps);
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GST_DEBUG_OBJECT (sink, "acceptcaps called with %s", caps_string);
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g_free (caps_string);
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pad_caps = gst_pad_query_caps (GST_BASE_SINK_PAD (sink), caps);
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if (pad_caps) {
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gboolean cret = gst_caps_can_intersect (pad_caps, caps);
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gst_caps_unref (pad_caps);
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if (!cret)
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goto done;
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}
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/* If we've not got fixed caps, creating a stream might fail,
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* so let's just return from here with default acceptcaps
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* behaviour */
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if (!gst_caps_is_fixed (caps))
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goto done;
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/* parse helper expects this set, so avoid nasty warning
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* will be set properly later on anyway */
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spec.latency_time = GST_SECOND;
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if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
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goto done;
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/* Make sure input is framed and can be payloaded */
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switch (spec.type) {
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
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{
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gboolean framed = FALSE;
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st = gst_caps_get_structure (caps, 0);
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gst_structure_get_boolean (st, "framed", &framed);
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if (!framed || gst_audio_iec61937_frame_size (&spec) <= 0)
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goto done;
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break;
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}
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case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
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{
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gboolean parsed = FALSE;
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st = gst_caps_get_structure (caps, 0);
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gst_structure_get_boolean (st, "parsed", &parsed);
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if (!parsed || gst_audio_iec61937_frame_size (&spec) <= 0)
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goto done;
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break;
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}
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default:
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break;
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}
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ret = TRUE;
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done:
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return ret;
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}
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|
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static GstBuffer *
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gst_osx_audio_sink_sink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
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{
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if (RINGBUFFER_IS_SPDIF (sink->ringbuffer->spec.type)) {
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gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
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GstBuffer *out;
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GstMapInfo inmap, outmap;
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gboolean res;
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if (framesize <= 0)
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return NULL;
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out = gst_buffer_new_and_alloc (framesize);
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gst_buffer_map (buf, &inmap, GST_MAP_READ);
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gst_buffer_map (out, &outmap, GST_MAP_WRITE);
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/* FIXME: the endianness needs to be queried and then set */
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res = gst_audio_iec61937_payload (inmap.data, inmap.size,
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outmap.data, outmap.size, &sink->ringbuffer->spec, G_BIG_ENDIAN);
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gst_buffer_unmap (buf, &inmap);
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gst_buffer_unmap (out, &outmap);
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if (!res) {
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gst_buffer_unref (out);
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return NULL;
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}
|
|
|
|
gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_METADATA, 0, -1);
|
|
return out;
|
|
|
|
} else {
|
|
return gst_buffer_ref (buf);
|
|
}
|
|
}
|
|
|
|
static GstAudioRingBuffer *
|
|
gst_osx_audio_sink_create_ringbuffer (GstAudioBaseSink * sink)
|
|
{
|
|
GstOsxAudioSink *osxsink;
|
|
GstOsxAudioRingBuffer *ringbuffer;
|
|
|
|
osxsink = GST_OSX_AUDIO_SINK (sink);
|
|
|
|
GST_DEBUG_OBJECT (sink, "Creating ringbuffer");
|
|
ringbuffer = g_object_new (GST_TYPE_OSX_AUDIO_RING_BUFFER, NULL);
|
|
GST_DEBUG_OBJECT (sink, "osx sink %p element %p ioproc %p", osxsink,
|
|
GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsink),
|
|
(void *) gst_osx_audio_sink_io_proc);
|
|
|
|
ringbuffer->core_audio->element =
|
|
GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsink);
|
|
ringbuffer->core_audio->is_src = FALSE;
|
|
|
|
/* By default the coreaudio instance created by the ringbuffer
|
|
* has device_id==kAudioDeviceUnknown. The user might have
|
|
* selected a different one here
|
|
*/
|
|
if (ringbuffer->core_audio->device_id != osxsink->device_id)
|
|
ringbuffer->core_audio->device_id = osxsink->device_id;
|
|
|
|
return GST_AUDIO_RING_BUFFER (ringbuffer);
|
|
}
|
|
|
|
/* HALOutput AudioUnit will request fairly arbitrarily-sized chunks
|
|
* of data, not of a fixed size. So, we keep track of where in
|
|
* the current ringbuffer segment we are, and only advance the segment
|
|
* once we've read the whole thing */
|
|
static OSStatus
|
|
gst_osx_audio_sink_io_proc (GstOsxAudioRingBuffer * buf,
|
|
AudioUnitRenderActionFlags * ioActionFlags,
|
|
const AudioTimeStamp * inTimeStamp,
|
|
UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList)
|
|
{
|
|
guint8 *readptr;
|
|
gint readseg;
|
|
gint len;
|
|
gint stream_idx = buf->core_audio->stream_idx;
|
|
gint remaining = bufferList->mBuffers[stream_idx].mDataByteSize;
|
|
gint offset = 0;
|
|
|
|
while (remaining) {
|
|
if (!gst_audio_ring_buffer_prepare_read (GST_AUDIO_RING_BUFFER (buf),
|
|
&readseg, &readptr, &len))
|
|
return 0;
|
|
|
|
len -= buf->segoffset;
|
|
|
|
if (len > remaining)
|
|
len = remaining;
|
|
|
|
memcpy ((char *) bufferList->mBuffers[stream_idx].mData + offset,
|
|
readptr + buf->segoffset, len);
|
|
|
|
buf->segoffset += len;
|
|
offset += len;
|
|
remaining -= len;
|
|
|
|
if ((gint) buf->segoffset == GST_AUDIO_RING_BUFFER (buf)->spec.segsize) {
|
|
/* clear written samples */
|
|
gst_audio_ring_buffer_clear (GST_AUDIO_RING_BUFFER (buf), readseg);
|
|
|
|
/* we wrote one segment */
|
|
CORE_AUDIO_TIMING_LOCK (buf->core_audio);
|
|
gst_audio_ring_buffer_advance (GST_AUDIO_RING_BUFFER (buf), 1);
|
|
/* FIXME: Update the timestamp and reported frames in smaller increments
|
|
* when the segment size is larger than the total inNumberFrames */
|
|
gst_core_audio_update_timing (buf->core_audio, inTimeStamp,
|
|
inNumberFrames);
|
|
CORE_AUDIO_TIMING_UNLOCK (buf->core_audio);
|
|
|
|
buf->segoffset = 0;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static void
|
|
gst_osx_audio_sink_osxelement_init (gpointer g_iface, gpointer iface_data)
|
|
{
|
|
GstOsxAudioElementInterface *iface = (GstOsxAudioElementInterface *) g_iface;
|
|
|
|
iface->io_proc = (AURenderCallback) gst_osx_audio_sink_io_proc;
|
|
}
|
|
|
|
static void
|
|
gst_osx_audio_sink_set_volume (GstOsxAudioSink * sink)
|
|
{
|
|
GstOsxAudioRingBuffer *osxbuf;
|
|
|
|
osxbuf = GST_OSX_AUDIO_RING_BUFFER (GST_AUDIO_BASE_SINK (sink)->ringbuffer);
|
|
if (!osxbuf)
|
|
return;
|
|
|
|
gst_core_audio_set_volume (osxbuf->core_audio, sink->volume);
|
|
}
|