gstreamer/subprojects/gst-plugins-good/sys/osxaudio/gstosxaudiosink.c
Jan Schmidt 461f943b52 osxaudio: Interpolate clock by counting elapsed time since render calls
When advancing the ringbuffer, store the processed CoreAudio sample
time, then interpolate the clock in the _get_delay() calls to smooth
the clock. CoreAudio's "latency" report is always a constant and
otherwise leads to the clock generating a latency-time staircase.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5140>
2023-08-07 21:33:45 +00:00

579 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 */
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);
}