gstreamer/subprojects/gst-plugins-good/sys/osxaudio/gstosxaudioringbuffer.c
Jan Schmidt f5d2ea76b4 osxaudio: Attempt to configure the segment size in CoreAudio
Set the BufferFrame size in CoreAudio so it will deliver data
in ringbuffer segment units when recording. Otherwise, CoreAudio
will provide data in whatever granularity it wants, with no
relationship to the requested latency-time.

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

322 lines
10 KiB
C

/*
* GStreamer
* Copyright (C) 2006 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* Copyright (C) 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.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <gst/gst.h>
#include <glib/gi18n-lib.h>
#include <gst/audio/audio-channels.h>
#include "gstosxaudioringbuffer.h"
#include "gstosxaudiosink.h"
#include "gstosxaudiosrc.h"
#include <unistd.h> /* for getpid() */
GST_DEBUG_CATEGORY_STATIC (osx_audio_debug);
#define GST_CAT_DEFAULT osx_audio_debug
#include "gstosxcoreaudio.h"
static void gst_osx_audio_ring_buffer_dispose (GObject * object);
static gboolean gst_osx_audio_ring_buffer_open_device (GstAudioRingBuffer *
buf);
static gboolean gst_osx_audio_ring_buffer_close_device (GstAudioRingBuffer *
buf);
static gboolean gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec);
static gboolean gst_osx_audio_ring_buffer_release (GstAudioRingBuffer * buf);
static gboolean gst_osx_audio_ring_buffer_start (GstAudioRingBuffer * buf);
static gboolean gst_osx_audio_ring_buffer_pause (GstAudioRingBuffer * buf);
static gboolean gst_osx_audio_ring_buffer_stop (GstAudioRingBuffer * buf);
static guint gst_osx_audio_ring_buffer_delay (GstAudioRingBuffer * buf);
static GstAudioRingBufferClass *ring_parent_class = NULL;
#define gst_osx_audio_ring_buffer_do_init \
GST_DEBUG_CATEGORY_INIT (osx_audio_debug, "osxaudio", 0, "OSX Audio Elements");
G_DEFINE_TYPE_WITH_CODE (GstOsxAudioRingBuffer, gst_osx_audio_ring_buffer,
GST_TYPE_AUDIO_RING_BUFFER, gst_osx_audio_ring_buffer_do_init);
static void
gst_osx_audio_ring_buffer_class_init (GstOsxAudioRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstAudioRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstringbuffer_class = (GstAudioRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = gst_osx_audio_ring_buffer_dispose;
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_osx_audio_ring_buffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_osx_audio_ring_buffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_osx_audio_ring_buffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_osx_audio_ring_buffer_release);
gstringbuffer_class->start =
GST_DEBUG_FUNCPTR (gst_osx_audio_ring_buffer_start);
gstringbuffer_class->pause =
GST_DEBUG_FUNCPTR (gst_osx_audio_ring_buffer_pause);
gstringbuffer_class->resume =
GST_DEBUG_FUNCPTR (gst_osx_audio_ring_buffer_start);
gstringbuffer_class->stop =
GST_DEBUG_FUNCPTR (gst_osx_audio_ring_buffer_stop);
gstringbuffer_class->delay =
GST_DEBUG_FUNCPTR (gst_osx_audio_ring_buffer_delay);
GST_DEBUG ("osx audio ring buffer class init");
}
static void
gst_osx_audio_ring_buffer_init (GstOsxAudioRingBuffer * ringbuffer)
{
ringbuffer->core_audio = gst_core_audio_new (GST_OBJECT (ringbuffer));
}
static void
gst_osx_audio_ring_buffer_dispose (GObject * object)
{
GstOsxAudioRingBuffer *osxbuf;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (object);
if (osxbuf->core_audio) {
g_object_unref (osxbuf->core_audio);
osxbuf->core_audio = NULL;
}
G_OBJECT_CLASS (ring_parent_class)->dispose (object);
}
static gboolean
gst_osx_audio_ring_buffer_open_device (GstAudioRingBuffer * buf)
{
GstObject *osxel = GST_OBJECT_PARENT (buf);
GstOsxAudioRingBuffer *osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
if (!gst_core_audio_select_device (osxbuf->core_audio)) {
GST_ELEMENT_ERROR (osxel, RESOURCE, NOT_FOUND,
(_("CoreAudio device not found")), (NULL));
return FALSE;
}
if (!gst_core_audio_open (osxbuf->core_audio)) {
GST_ELEMENT_ERROR (osxel, RESOURCE, OPEN_READ,
(_("CoreAudio device could not be opened")), (NULL));
return FALSE;
}
return TRUE;
}
static gboolean
gst_osx_audio_ring_buffer_close_device (GstAudioRingBuffer * buf)
{
GstOsxAudioRingBuffer *osxbuf;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
return gst_core_audio_close (osxbuf->core_audio);
}
static gboolean
gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec)
{
gboolean ret = FALSE, is_passthrough = FALSE;
GstOsxAudioRingBuffer *osxbuf;
AudioStreamBasicDescription format = { 0 };
guint32 frames_per_packet = 0;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
if (RINGBUFFER_IS_SPDIF (spec->type)) {
format.mFormatID = kAudioFormat60958AC3;
format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
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;
is_passthrough = TRUE;
} else {
int width, depth;
/* Fill out the audio description we're going to be using */
format.mFormatID = kAudioFormatLinearPCM;
format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
format.mChannelsPerFrame = GST_AUDIO_INFO_CHANNELS (&spec->info);
if (GST_AUDIO_INFO_IS_FLOAT (&spec->info)) {
format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
width = depth = GST_AUDIO_INFO_WIDTH (&spec->info);
} else {
format.mFormatFlags = kAudioFormatFlagIsSignedInteger;
width = GST_AUDIO_INFO_WIDTH (&spec->info);
depth = GST_AUDIO_INFO_DEPTH (&spec->info);
if (width == depth) {
format.mFormatFlags |= kAudioFormatFlagIsPacked;
} else {
format.mFormatFlags |= kAudioFormatFlagIsAlignedHigh;
}
}
if (GST_AUDIO_INFO_IS_BIG_ENDIAN (&spec->info)) {
format.mFormatFlags |= kAudioFormatFlagIsBigEndian;
}
format.mBytesPerFrame = GST_AUDIO_INFO_BPF (&spec->info);
format.mBitsPerChannel = depth;
format.mBytesPerPacket = GST_AUDIO_INFO_BPF (&spec->info);
format.mFramesPerPacket = 1;
format.mReserved = 0;
spec->segsize =
(spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) /
G_USEC_PER_SEC) * GST_AUDIO_INFO_BPF (&spec->info);
spec->segtotal = spec->buffer_time / spec->latency_time;
frames_per_packet = spec->segsize / GST_AUDIO_INFO_BPF (&spec->info);
is_passthrough = FALSE;
}
GST_DEBUG_OBJECT (osxbuf, "Format: " CORE_AUDIO_FORMAT,
CORE_AUDIO_FORMAT_ARGS (format));
/* gst_audio_ring_buffer_set_channel_positions is not called
* since the AUs perform channel reordering themselves.
* (see gst_core_audio_set_channel_layout) */
buf->size = spec->segtotal * spec->segsize;
buf->memory = g_malloc0 (buf->size);
ret = gst_core_audio_initialize (osxbuf->core_audio, format, spec->caps,
frames_per_packet, is_passthrough);
if (!ret) {
g_free (buf->memory);
buf->memory = NULL;
buf->size = 0;
}
osxbuf->segoffset = 0;
return ret;
}
static gboolean
gst_osx_audio_ring_buffer_release (GstAudioRingBuffer * buf)
{
GstOsxAudioRingBuffer *osxbuf;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
gst_core_audio_uninitialize (osxbuf->core_audio);
g_free (buf->memory);
buf->memory = NULL;
buf->size = 0;
return TRUE;
}
static gboolean
gst_osx_audio_ring_buffer_start (GstAudioRingBuffer * buf)
{
GstOsxAudioRingBuffer *osxbuf;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
return gst_core_audio_start_processing (osxbuf->core_audio);
}
static gboolean
gst_osx_audio_ring_buffer_pause (GstAudioRingBuffer * buf)
{
GstOsxAudioRingBuffer *osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
return gst_core_audio_pause_processing (osxbuf->core_audio);
}
static gboolean
gst_osx_audio_ring_buffer_stop (GstAudioRingBuffer * buf)
{
GstOsxAudioRingBuffer *osxbuf;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
gst_core_audio_stop_processing (osxbuf->core_audio);
return TRUE;
}
static guint
gst_osx_audio_ring_buffer_delay (GstAudioRingBuffer * buf)
{
GstOsxAudioRingBuffer *osxbuf;
double latency;
guint samples;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
if (!gst_core_audio_get_samples_and_latency (osxbuf->core_audio,
GST_AUDIO_INFO_RATE (&buf->spec.info), &samples, &latency)) {
return 0;
}
GST_DEBUG_OBJECT (buf, "Got latency: %f seconds -> %d samples",
latency, samples);
return samples;
}