gstreamer/gst-libs/gst/audio/gstaudioutilsprivate.c
Nirbheek Chauhan d002cd33d3 gstaudioutilsprivate: Fix warnings while setting thread priority
Also use G_OS_WIN32 instead of _WIN32 for clarity.
2018-09-24 19:44:28 +05:30

282 lines
7.9 KiB
C

/* GStreamer
* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
* Copyright (C) 2011 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* 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/audio/audio.h>
#ifdef G_OS_WIN32
#include <windows.h>
#endif
#include "gstaudioutilsprivate.h"
/*
* Takes caps and copies its audio fields to tmpl_caps
*/
static GstCaps *
__gst_audio_element_proxy_caps (GstElement * element, GstCaps * templ_caps,
GstCaps * caps)
{
GstCaps *result = gst_caps_new_empty ();
gint i, j;
gint templ_caps_size = gst_caps_get_size (templ_caps);
gint caps_size = gst_caps_get_size (caps);
for (i = 0; i < templ_caps_size; i++) {
GQuark q_name =
gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
GstCapsFeatures *features = gst_caps_get_features (templ_caps, i);
for (j = 0; j < caps_size; j++) {
const GstStructure *caps_s = gst_caps_get_structure (caps, j);
const GValue *val;
GstStructure *s;
GstCaps *tmp = gst_caps_new_empty ();
s = gst_structure_new_id_empty (q_name);
if ((val = gst_structure_get_value (caps_s, "rate")))
gst_structure_set_value (s, "rate", val);
if ((val = gst_structure_get_value (caps_s, "channels")))
gst_structure_set_value (s, "channels", val);
if ((val = gst_structure_get_value (caps_s, "channels-mask")))
gst_structure_set_value (s, "channels-mask", val);
gst_caps_append_structure_full (tmp, s,
gst_caps_features_copy (features));
result = gst_caps_merge (result, tmp);
}
}
return result;
}
/**
* __gst_audio_element_proxy_getcaps:
* @element: a #GstElement
* @sinkpad: the element's sink #GstPad
* @srcpad: the element's source #GstPad
* @initial_caps: initial caps
* @filter: filter caps
*
* Returns caps that express @initial_caps (or sink template caps if
* @initial_caps == NULL) restricted to rate/channels/...
* combinations supported by downstream elements (e.g. muxers).
*
* Returns: a #GstCaps owned by caller
*/
GstCaps *
__gst_audio_element_proxy_getcaps (GstElement * element, GstPad * sinkpad,
GstPad * srcpad, GstCaps * initial_caps, GstCaps * filter)
{
GstCaps *templ_caps, *src_templ_caps;
GstCaps *peer_caps;
GstCaps *allowed;
GstCaps *fcaps, *filter_caps;
/* Allow downstream to specify rate/channels constraints
* and forward them upstream for audio converters to handle
*/
templ_caps = initial_caps ? gst_caps_ref (initial_caps) :
gst_pad_get_pad_template_caps (sinkpad);
src_templ_caps = gst_pad_get_pad_template_caps (srcpad);
if (filter && !gst_caps_is_any (filter)) {
GstCaps *proxy_filter =
__gst_audio_element_proxy_caps (element, src_templ_caps, filter);
peer_caps = gst_pad_peer_query_caps (srcpad, proxy_filter);
gst_caps_unref (proxy_filter);
} else {
peer_caps = gst_pad_peer_query_caps (srcpad, NULL);
}
allowed = gst_caps_intersect_full (peer_caps, src_templ_caps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (src_templ_caps);
gst_caps_unref (peer_caps);
if (!allowed || gst_caps_is_any (allowed)) {
fcaps = templ_caps;
goto done;
} else if (gst_caps_is_empty (allowed)) {
fcaps = gst_caps_ref (allowed);
goto done;
}
GST_LOG_OBJECT (element, "template caps %" GST_PTR_FORMAT, templ_caps);
GST_LOG_OBJECT (element, "allowed caps %" GST_PTR_FORMAT, allowed);
filter_caps = __gst_audio_element_proxy_caps (element, templ_caps, allowed);
fcaps = gst_caps_intersect (filter_caps, templ_caps);
gst_caps_unref (filter_caps);
gst_caps_unref (templ_caps);
if (filter) {
GST_LOG_OBJECT (element, "intersecting with %" GST_PTR_FORMAT, filter);
filter_caps = gst_caps_intersect (fcaps, filter);
gst_caps_unref (fcaps);
fcaps = filter_caps;
}
done:
gst_caps_replace (&allowed, NULL);
GST_LOG_OBJECT (element, "proxy caps %" GST_PTR_FORMAT, fcaps);
return fcaps;
}
/**
* __gst_audio_encoded_audio_convert:
* @fmt: audio format of the encoded audio
* @bytes: number of encoded bytes
* @samples: number of encoded samples
* @src_format: source format
* @src_value: source value
* @dest_format: destination format
* @dest_value: destination format
*
* Helper function to convert @src_value in @src_format to @dest_value in
* @dest_format for encoded audio data. Conversion is possible between
* BYTE and TIME format by using estimated bitrate based on
* @samples and @bytes (and @fmt).
*/
gboolean
__gst_audio_encoded_audio_convert (GstAudioInfo * fmt,
gint64 bytes, gint64 samples, GstFormat src_format,
gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = FALSE;
g_return_val_if_fail (dest_format != NULL, FALSE);
g_return_val_if_fail (dest_value != NULL, FALSE);
if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
src_value == -1)) {
if (dest_value)
*dest_value = src_value;
return TRUE;
}
if (samples == 0 || bytes == 0 || fmt->rate == 0) {
GST_DEBUG ("not enough metadata yet to convert");
goto exit;
}
bytes *= fmt->rate;
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale (src_value,
GST_SECOND * samples, bytes);
res = TRUE;
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = gst_util_uint64_scale (src_value, bytes,
samples * GST_SECOND);
res = TRUE;
break;
default:
res = FALSE;
}
break;
default:
res = FALSE;
}
exit:
return res;
}
#ifdef G_OS_WIN32
/* *INDENT-OFF* */
static struct
{
HMODULE dll;
gboolean tried_loading;
FARPROC AvSetMmThreadCharacteristics;
FARPROC AvRevertMmThreadCharacteristics;
} _gst_audio_avrt_tbl = { 0 };
/* *INDENT-ON* */
#endif
static gboolean
__gst_audio_init_thread_priority (void)
{
#ifdef G_OS_WIN32
if (_gst_audio_avrt_tbl.tried_loading)
return _gst_audio_avrt_tbl.dll != NULL;
if (!_gst_audio_avrt_tbl.dll)
_gst_audio_avrt_tbl.dll = LoadLibrary (TEXT ("avrt.dll"));
if (!_gst_audio_avrt_tbl.dll) {
GST_WARNING ("Failed to set thread priority, can't find avrt.dll");
_gst_audio_avrt_tbl.tried_loading = TRUE;
return FALSE;
}
_gst_audio_avrt_tbl.AvSetMmThreadCharacteristics =
GetProcAddress (_gst_audio_avrt_tbl.dll, "AvSetMmThreadCharacteristicsA");
_gst_audio_avrt_tbl.AvRevertMmThreadCharacteristics =
GetProcAddress (_gst_audio_avrt_tbl.dll,
"AvRevertMmThreadCharacteristics");
_gst_audio_avrt_tbl.tried_loading = TRUE;
#endif
return TRUE;
}
/*
* Increases the priority of the thread it's called from
*/
gboolean
__gst_audio_set_thread_priority (void)
{
#ifdef G_OS_WIN32
DWORD taskIndex = 0;
#endif
if (!__gst_audio_init_thread_priority ())
return FALSE;
#ifdef G_OS_WIN32
/* This is only used from ringbuffer thread functions, so we don't need to
* ever need to revert the thread priorities. */
return _gst_audio_avrt_tbl.AvSetMmThreadCharacteristics (TEXT ("Pro Audio"),
&taskIndex) != 0;
#else
return TRUE;
#endif
}