gstreamer/subprojects/gst-plugins-good/sys/osxaudio/gstosxaudiosink.c
Nirbheek Chauhan f1449ac116 docs: Rename "OS X" to "macOS" in some documentation
No one uses the term "Mac OS X" anymore, it's "macOS". "OS X" is even
worse, because people will usually start the search with "mac".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1607>
2022-01-30 19:17:18 +05:30

573 lines
17 KiB
C

/*
* GStreamer
* Copyright (C) 2005,2006 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* Copyright (C) 2007,2008 Pioneers of the Inevitable <songbird@songbirdnest.com>
* Copyright (C) 2012 Fluendo S.A. <support@fluendo.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*
* The development of this code was made possible due to the involvement of
* Pioneers of the Inevitable, the creators of the Songbird Music player
*
*/
/**
* SECTION:element-osxaudiosink
* @title: osxaudiosink
*
* This element renders raw audio samples using the CoreAudio api.
*
* ## Example pipelines
* |[
* gst-launch-1.0 filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osxaudiosink
* ]| Play an Ogg/Vorbis file.
*
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/audio/audio-channels.h>
#include <gst/audio/gstaudioiec61937.h>
#include "gstosxaudiosink.h"
#include "gstosxaudioelement.h"
GST_DEBUG_CATEGORY_STATIC (osx_audiosink_debug);
#define GST_CAT_DEFAULT osx_audiosink_debug
#include "gstosxcoreaudio.h"
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_DEVICE,
ARG_VOLUME
};
#define DEFAULT_VOLUME 1.0
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_OSX_AUDIO_SINK_CAPS)
);
static void gst_osx_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_osx_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStateChangeReturn
gst_osx_audio_sink_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_osx_audio_sink_query (GstBaseSink * base, GstQuery * query);
static GstCaps *gst_osx_audio_sink_getcaps (GstBaseSink * base,
GstCaps * filter);
static gboolean gst_osx_audio_sink_acceptcaps (GstOsxAudioSink * sink,
GstCaps * caps);
static GstBuffer *gst_osx_audio_sink_sink_payload (GstAudioBaseSink * sink,
GstBuffer * buf);
static GstAudioRingBuffer
* gst_osx_audio_sink_create_ringbuffer (GstAudioBaseSink * sink);
static void gst_osx_audio_sink_osxelement_init (gpointer g_iface,
gpointer iface_data);
static void gst_osx_audio_sink_set_volume (GstOsxAudioSink * sink);
static OSStatus gst_osx_audio_sink_io_proc (GstOsxAudioRingBuffer * buf,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList);
static void
gst_osx_audio_sink_do_init (GType type)
{
static const GInterfaceInfo osxelement_info = {
gst_osx_audio_sink_osxelement_init,
NULL,
NULL
};
GST_DEBUG_CATEGORY_INIT (osx_audiosink_debug, "osxaudiosink", 0,
"OSX Audio Sink");
gst_core_audio_init_debug ();
GST_DEBUG ("Adding static interface");
g_type_add_interface_static (type, GST_OSX_AUDIO_ELEMENT_TYPE,
&osxelement_info);
}
#define gst_osx_audio_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstOsxAudioSink, gst_osx_audio_sink,
GST_TYPE_AUDIO_BASE_SINK, gst_osx_audio_sink_do_init (g_define_type_id));
static void
gst_osx_audio_sink_class_init (GstOsxAudioSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstAudioBaseSinkClass *gstaudiobasesink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstaudiobasesink_class = (GstAudioBaseSinkClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_osx_audio_sink_set_property;
gobject_class->get_property = gst_osx_audio_sink_get_property;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_osx_audio_sink_change_state);
#ifndef HAVE_IOS
g_object_class_install_property (gobject_class, ARG_DEVICE,
g_param_spec_int ("device", "Device ID", "Device ID of output device",
0, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
#endif
gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_query);
g_object_class_install_property (gobject_class, ARG_VOLUME,
g_param_spec_double ("volume", "Volume", "Volume of this stream",
0, 1.0, 1.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_osx_audio_sink_getcaps);
gstaudiobasesink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_osx_audio_sink_create_ringbuffer);
gstaudiobasesink_class->payload =
GST_DEBUG_FUNCPTR (gst_osx_audio_sink_sink_payload);
gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
gst_element_class_set_static_metadata (gstelement_class, "Audio Sink (macOS)",
"Sink/Audio",
"Output to a sound card on macOS",
"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
}
static void
gst_osx_audio_sink_init (GstOsxAudioSink * sink)
{
GST_DEBUG ("Initialising object");
sink->device_id = kAudioDeviceUnknown;
sink->volume = DEFAULT_VOLUME;
}
static void
gst_osx_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (object);
switch (prop_id) {
#ifndef HAVE_IOS
case ARG_DEVICE:
sink->device_id = g_value_get_int (value);
break;
#endif
case ARG_VOLUME:
sink->volume = g_value_get_double (value);
gst_osx_audio_sink_set_volume (sink);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_osx_audio_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstOsxAudioSink *osxsink = GST_OSX_AUDIO_SINK (element);
GstOsxAudioRingBuffer *ringbuffer;
GstStateChangeReturn ret;
switch (transition) {
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
goto out;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
/* Device has been selected, AudioUnit set up, so initialize volume */
gst_osx_audio_sink_set_volume (osxsink);
/* The device is open now, so fix our device_id if it changed */
ringbuffer =
GST_OSX_AUDIO_RING_BUFFER (GST_AUDIO_BASE_SINK (osxsink)->ringbuffer);
if (ringbuffer->core_audio->device_id != osxsink->device_id) {
osxsink->device_id = ringbuffer->core_audio->device_id;
g_object_notify (G_OBJECT (osxsink), "device");
}
break;
default:
break;
}
out:
return ret;
}
static void
gst_osx_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (object);
switch (prop_id) {
#ifndef HAVE_IOS
case ARG_DEVICE:
g_value_set_int (value, sink->device_id);
break;
#endif
case ARG_VOLUME:
g_value_set_double (value, sink->volume);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_osx_audio_sink_query (GstBaseSink * base, GstQuery * query)
{
GstOsxAudioSink *sink = GST_OSX_AUDIO_SINK (base);
gboolean ret = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_ACCEPT_CAPS:
{
GstCaps *caps = NULL;
gst_query_parse_accept_caps (query, &caps);
ret = gst_osx_audio_sink_acceptcaps (sink, caps);
gst_query_set_accept_caps_result (query, ret);
ret = TRUE;
break;
}
default:
ret = GST_BASE_SINK_CLASS (parent_class)->query (base, query);
break;
}
return ret;
}
static GstCaps *
gst_osx_audio_sink_getcaps (GstBaseSink * sink, GstCaps * filter)
{
GstOsxAudioSink *osxsink;
GstAudioRingBuffer *buf;
GstOsxAudioRingBuffer *osxbuf;
GstCaps *caps, *filtered_caps;
osxsink = GST_OSX_AUDIO_SINK (sink);
GST_OBJECT_LOCK (osxsink);
buf = GST_AUDIO_BASE_SINK (sink)->ringbuffer;
if (buf)
gst_object_ref (buf);
GST_OBJECT_UNLOCK (osxsink);
if (!buf) {
GST_DEBUG_OBJECT (sink, "no ring buffer, returning NULL caps");
return GST_BASE_SINK_CLASS (parent_class)->get_caps (sink, filter);
}
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
/* protect against cached_caps going away */
GST_OBJECT_LOCK (buf);
if (osxbuf->core_audio->cached_caps_valid) {
GST_LOG_OBJECT (sink, "Returning cached caps");
caps = gst_caps_ref (osxbuf->core_audio->cached_caps);
} else if (buf->open) {
GstCaps *template_caps;
/* Get template caps */
template_caps =
gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (osxsink));
/* Device is open, let's probe its caps */
caps = gst_core_audio_probe_caps (osxbuf->core_audio, template_caps);
gst_caps_replace (&osxbuf->core_audio->cached_caps, caps);
gst_caps_unref (template_caps);
} else {
GST_DEBUG_OBJECT (sink, "ring buffer not open, returning NULL caps");
caps = NULL;
}
GST_OBJECT_UNLOCK (buf);
gst_object_unref (buf);
if (!caps)
return NULL;
if (!filter)
return caps;
/* Take care of filtered caps */
filtered_caps =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
return filtered_caps;
}
static gboolean
gst_osx_audio_sink_acceptcaps (GstOsxAudioSink * sink, GstCaps * caps)
{
GstCaps *pad_caps;
GstStructure *st;
gboolean ret = FALSE;
GstAudioRingBufferSpec spec = { 0 };
gchar *caps_string = NULL;
caps_string = gst_caps_to_string (caps);
GST_DEBUG_OBJECT (sink, "acceptcaps called with %s", caps_string);
g_free (caps_string);
pad_caps = gst_pad_query_caps (GST_BASE_SINK_PAD (sink), caps);
if (pad_caps) {
gboolean cret = gst_caps_can_intersect (pad_caps, caps);
gst_caps_unref (pad_caps);
if (!cret)
goto done;
}
/* If we've not got fixed caps, creating a stream might fail,
* so let's just return from here with default acceptcaps
* behaviour */
if (!gst_caps_is_fixed (caps))
goto done;
/* parse helper expects this set, so avoid nasty warning
* will be set properly later on anyway */
spec.latency_time = GST_SECOND;
if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
goto done;
/* Make sure input is framed and can be payloaded */
switch (spec.type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
{
gboolean framed = FALSE;
st = gst_caps_get_structure (caps, 0);
gst_structure_get_boolean (st, "framed", &framed);
if (!framed || gst_audio_iec61937_frame_size (&spec) <= 0)
goto done;
break;
}
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
{
gboolean parsed = FALSE;
st = gst_caps_get_structure (caps, 0);
gst_structure_get_boolean (st, "parsed", &parsed);
if (!parsed || gst_audio_iec61937_frame_size (&spec) <= 0)
goto done;
break;
}
default:
break;
}
ret = TRUE;
done:
return ret;
}
static GstBuffer *
gst_osx_audio_sink_sink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
{
if (RINGBUFFER_IS_SPDIF (sink->ringbuffer->spec.type)) {
gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
GstBuffer *out;
GstMapInfo inmap, outmap;
gboolean res;
if (framesize <= 0)
return NULL;
out = gst_buffer_new_and_alloc (framesize);
gst_buffer_map (buf, &inmap, GST_MAP_READ);
gst_buffer_map (out, &outmap, GST_MAP_WRITE);
/* FIXME: the endianness needs to be queried and then set */
res = gst_audio_iec61937_payload (inmap.data, inmap.size,
outmap.data, outmap.size, &sink->ringbuffer->spec, G_BIG_ENDIAN);
gst_buffer_unmap (buf, &inmap);
gst_buffer_unmap (out, &outmap);
if (!res) {
gst_buffer_unref (out);
return NULL;
}
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 */
gst_audio_ring_buffer_advance (GST_AUDIO_RING_BUFFER (buf), 1);
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);
}