gstreamer/subprojects/gst-plugins-good/sys/osxaudio/gstosxaudiosrc.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

418 lines
13 KiB
C

/*
* GStreamer
* Copyright (C) 2005,2006 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* Copyright (C) 2008 Pioneers of the Inevitable <songbird@songbirdnest.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.
*/
/**
* SECTION:element-osxaudiosrc
* @title: osxaudiosrc
*
* This element captures raw audio samples using the CoreAudio api.
*
* ## Example launch line
* |[
* gst-launch-1.0 osxaudiosrc ! wavenc ! filesink location=audio.wav
* ]|
*
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <gst/gst.h>
#include "gstosxaudiosrc.h"
#include "gstosxaudioelement.h"
GST_DEBUG_CATEGORY_STATIC (osx_audiosrc_debug);
#define GST_CAT_DEFAULT osx_audiosrc_debug
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_DEVICE
};
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_OSX_AUDIO_SRC_CAPS)
);
static void gst_osx_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_osx_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStateChangeReturn
gst_osx_audio_src_change_state (GstElement * element,
GstStateChange transition);
static GstCaps *gst_osx_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter);
static GstAudioRingBuffer *gst_osx_audio_src_create_ringbuffer (GstAudioBaseSrc
* src);
static void gst_osx_audio_src_osxelement_init (gpointer g_iface,
gpointer iface_data);
static OSStatus gst_osx_audio_src_io_proc (GstOsxAudioRingBuffer * buf,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp, UInt32 inBusNumber,
UInt32 inNumberFrames, AudioBufferList * bufferList);
static void
gst_osx_audio_src_do_init (GType type)
{
static const GInterfaceInfo osxelement_info = {
gst_osx_audio_src_osxelement_init,
NULL,
NULL
};
GST_DEBUG_CATEGORY_INIT (osx_audiosrc_debug, "osxaudiosrc", 0,
"OSX Audio Src");
g_type_add_interface_static (type, GST_OSX_AUDIO_ELEMENT_TYPE,
&osxelement_info);
}
#define gst_osx_audio_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstOsxAudioSrc, gst_osx_audio_src,
GST_TYPE_AUDIO_BASE_SRC, gst_osx_audio_src_do_init (g_define_type_id));
static void
gst_osx_audio_src_class_init (GstOsxAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstAudioBaseSrcClass *gstaudiobasesrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
gobject_class->set_property = gst_osx_audio_src_set_property;
gobject_class->get_property = gst_osx_audio_src_get_property;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_osx_audio_src_change_state);
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_osx_audio_src_get_caps);
g_object_class_install_property (gobject_class, ARG_DEVICE,
g_param_spec_int ("device", "Device ID", "Device ID of input device",
0, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstaudiobasesrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_osx_audio_src_create_ringbuffer);
gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
gst_element_class_set_static_metadata (gstelement_class,
"Audio Source (macOS)", "Source/Audio",
"Input from a sound card on macOS",
"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
}
static void
gst_osx_audio_src_init (GstOsxAudioSrc * src)
{
gst_base_src_set_live (GST_BASE_SRC (src), TRUE);
src->device_id = kAudioDeviceUnknown;
}
static void
gst_osx_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOsxAudioSrc *src = GST_OSX_AUDIO_SRC (object);
switch (prop_id) {
case ARG_DEVICE:
src->device_id = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_osx_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOsxAudioSrc *src = GST_OSX_AUDIO_SRC (object);
switch (prop_id) {
case ARG_DEVICE:
g_value_set_int (value, src->device_id);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_osx_audio_src_change_state (GstElement * element, GstStateChange transition)
{
GstOsxAudioSrc *osxsrc = GST_OSX_AUDIO_SRC (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:
/* The device is open now, so fix our device_id if it changed */
ringbuffer =
GST_OSX_AUDIO_RING_BUFFER (GST_AUDIO_BASE_SRC (osxsrc)->ringbuffer);
if (ringbuffer->core_audio->device_id != osxsrc->device_id) {
osxsrc->device_id = ringbuffer->core_audio->device_id;
g_object_notify (G_OBJECT (osxsrc), "device");
}
break;
default:
break;
}
out:
return ret;
}
static GstCaps *
gst_osx_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter)
{
GstOsxAudioSrc *osxsrc;
GstAudioRingBuffer *buf;
GstOsxAudioRingBuffer *osxbuf;
GstCaps *caps, *filtered_caps;
osxsrc = GST_OSX_AUDIO_SRC (src);
GST_OBJECT_LOCK (osxsrc);
buf = GST_AUDIO_BASE_SRC (src)->ringbuffer;
if (buf)
gst_object_ref (buf);
GST_OBJECT_UNLOCK (osxsrc);
if (!buf) {
GST_DEBUG_OBJECT (src, "no ring buffer, using template caps");
return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, 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 (src, "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_SRC_PAD (osxsrc));
/* 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 (src, "ring buffer not open, using template caps");
caps = GST_BASE_SRC_CLASS (parent_class)->get_caps (src, 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 GstAudioRingBuffer *
gst_osx_audio_src_create_ringbuffer (GstAudioBaseSrc * src)
{
GstOsxAudioSrc *osxsrc;
GstOsxAudioRingBuffer *ringbuffer;
osxsrc = GST_OSX_AUDIO_SRC (src);
GST_DEBUG_OBJECT (osxsrc, "Creating ringbuffer");
ringbuffer = g_object_new (GST_TYPE_OSX_AUDIO_RING_BUFFER, NULL);
GST_DEBUG_OBJECT (osxsrc, "osx src 0x%p element 0x%p ioproc 0x%p", osxsrc,
GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsrc),
(void *) gst_osx_audio_src_io_proc);
ringbuffer->core_audio->element =
GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsrc);
ringbuffer->core_audio->is_src = TRUE;
/* 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 != osxsrc->device_id)
ringbuffer->core_audio->device_id = osxsrc->device_id;
return GST_AUDIO_RING_BUFFER (ringbuffer);
}
static OSStatus
gst_osx_audio_src_io_proc (GstOsxAudioRingBuffer * buf,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList)
{
OSStatus status;
guint8 *writeptr;
gint writeseg;
gint len;
gint remaining;
UInt32 n;
gint offset = 0;
guint64 sample_position;
GstAudioRingBufferSpec *spec = &GST_AUDIO_RING_BUFFER (buf)->spec;
guint bpf = GST_AUDIO_INFO_BPF (&spec->info);
GST_LOG_OBJECT (buf, "in sample position %f frames %u",
inTimeStamp->mSampleTime, inNumberFrames);
/* Previous invoke of AudioUnitRender changed mDataByteSize into
* number of bytes actually read. Reset the members. */
for (n = 0; n < buf->core_audio->recBufferList->mNumberBuffers; ++n) {
buf->core_audio->recBufferList->mBuffers[n].mDataByteSize =
buf->core_audio->recBufferSize;
}
status = AudioUnitRender (buf->core_audio->audiounit, ioActionFlags,
inTimeStamp, inBusNumber, inNumberFrames, buf->core_audio->recBufferList);
if (status) {
GST_WARNING_OBJECT (buf, "AudioUnitRender returned %d", (int) status);
return status;
}
/* TODO: To support non-interleaved audio, go over all mBuffers,
* not just the first one. */
remaining = buf->core_audio->recBufferList->mBuffers[0].mDataByteSize;
sample_position = inTimeStamp->mSampleTime;
while (remaining) {
if (!gst_audio_ring_buffer_prepare_read (GST_AUDIO_RING_BUFFER (buf),
&writeseg, &writeptr, &len))
return 0;
len -= buf->segoffset;
if (len > remaining)
len = remaining;
memcpy (writeptr + buf->segoffset,
(char *) buf->core_audio->recBufferList->mBuffers[0].mData + offset,
len);
buf->segoffset += len;
offset += len;
remaining -= len;
sample_position += len / bpf;
if ((gint) buf->segoffset == GST_AUDIO_RING_BUFFER (buf)->spec.segsize) {
/* Calculate the timestamp corresponding to the first sample in the segment */
guint64 seg_sample_pos = sample_position - (spec->segsize / bpf);
GstClockTime ts = gst_util_uint64_scale_int (seg_sample_pos, GST_SECOND,
GST_AUDIO_INFO_RATE (&spec->info));
gst_audio_ring_buffer_set_timestamp (GST_AUDIO_RING_BUFFER (buf),
writeseg, ts);
/* 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_src_osxelement_init (gpointer g_iface, gpointer iface_data)
{
GstOsxAudioElementInterface *iface = (GstOsxAudioElementInterface *) g_iface;
iface->io_proc = (AURenderCallback) gst_osx_audio_src_io_proc;
}