gstreamer/subprojects/gst-plugins-good/sys/osxaudio/gstosxaudioringbuffer.c
Xavier Claessens b99ecc78ca Replace gst-i18n-*.h with gi18n-lib.h
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.

This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00

318 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;
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;
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,
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;
}