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282 lines
7.9 KiB
C
282 lines
7.9 KiB
C
/* GStreamer
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/audio/audio.h>
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#ifdef G_OS_WIN32
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#include <windows.h>
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#endif
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#include "gstaudioutilsprivate.h"
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/*
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* Takes caps and copies its audio fields to tmpl_caps
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*/
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static GstCaps *
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__gst_audio_element_proxy_caps (GstElement * element, GstCaps * templ_caps,
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GstCaps * caps)
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{
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GstCaps *result = gst_caps_new_empty ();
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gint i, j;
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gint templ_caps_size = gst_caps_get_size (templ_caps);
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gint caps_size = gst_caps_get_size (caps);
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for (i = 0; i < templ_caps_size; i++) {
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GQuark q_name =
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gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));
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GstCapsFeatures *features = gst_caps_get_features (templ_caps, i);
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for (j = 0; j < caps_size; j++) {
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const GstStructure *caps_s = gst_caps_get_structure (caps, j);
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const GValue *val;
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GstStructure *s;
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GstCaps *tmp = gst_caps_new_empty ();
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s = gst_structure_new_id_empty (q_name);
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if ((val = gst_structure_get_value (caps_s, "rate")))
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gst_structure_set_value (s, "rate", val);
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if ((val = gst_structure_get_value (caps_s, "channels")))
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gst_structure_set_value (s, "channels", val);
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if ((val = gst_structure_get_value (caps_s, "channels-mask")))
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gst_structure_set_value (s, "channels-mask", val);
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gst_caps_append_structure_full (tmp, s,
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gst_caps_features_copy (features));
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result = gst_caps_merge (result, tmp);
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}
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}
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return result;
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}
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/**
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* __gst_audio_element_proxy_getcaps:
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* @element: a #GstElement
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* @sinkpad: the element's sink #GstPad
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* @srcpad: the element's source #GstPad
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* @initial_caps: initial caps
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* @filter: filter caps
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*
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* Returns caps that express @initial_caps (or sink template caps if
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* @initial_caps == NULL) restricted to rate/channels/...
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* combinations supported by downstream elements (e.g. muxers).
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*
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* Returns: a #GstCaps owned by caller
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*/
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GstCaps *
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__gst_audio_element_proxy_getcaps (GstElement * element, GstPad * sinkpad,
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GstPad * srcpad, GstCaps * initial_caps, GstCaps * filter)
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{
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GstCaps *templ_caps, *src_templ_caps;
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GstCaps *peer_caps;
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GstCaps *allowed;
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GstCaps *fcaps, *filter_caps;
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/* Allow downstream to specify rate/channels constraints
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* and forward them upstream for audio converters to handle
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*/
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templ_caps = initial_caps ? gst_caps_ref (initial_caps) :
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gst_pad_get_pad_template_caps (sinkpad);
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src_templ_caps = gst_pad_get_pad_template_caps (srcpad);
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if (filter && !gst_caps_is_any (filter)) {
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GstCaps *proxy_filter =
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__gst_audio_element_proxy_caps (element, src_templ_caps, filter);
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peer_caps = gst_pad_peer_query_caps (srcpad, proxy_filter);
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gst_caps_unref (proxy_filter);
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} else {
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peer_caps = gst_pad_peer_query_caps (srcpad, NULL);
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}
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allowed = gst_caps_intersect_full (peer_caps, src_templ_caps,
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GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (src_templ_caps);
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gst_caps_unref (peer_caps);
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if (!allowed || gst_caps_is_any (allowed)) {
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fcaps = templ_caps;
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goto done;
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} else if (gst_caps_is_empty (allowed)) {
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fcaps = gst_caps_ref (allowed);
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goto done;
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}
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GST_LOG_OBJECT (element, "template caps %" GST_PTR_FORMAT, templ_caps);
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GST_LOG_OBJECT (element, "allowed caps %" GST_PTR_FORMAT, allowed);
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filter_caps = __gst_audio_element_proxy_caps (element, templ_caps, allowed);
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fcaps = gst_caps_intersect (filter_caps, templ_caps);
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gst_caps_unref (filter_caps);
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gst_caps_unref (templ_caps);
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if (filter) {
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GST_LOG_OBJECT (element, "intersecting with %" GST_PTR_FORMAT, filter);
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filter_caps = gst_caps_intersect (fcaps, filter);
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gst_caps_unref (fcaps);
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fcaps = filter_caps;
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}
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done:
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gst_caps_replace (&allowed, NULL);
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GST_LOG_OBJECT (element, "proxy caps %" GST_PTR_FORMAT, fcaps);
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return fcaps;
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}
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/**
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* __gst_audio_encoded_audio_convert:
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* @fmt: audio format of the encoded audio
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* @bytes: number of encoded bytes
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* @samples: number of encoded samples
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* @src_format: source format
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* @src_value: source value
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* @dest_format: destination format
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* @dest_value: destination format
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*
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* Helper function to convert @src_value in @src_format to @dest_value in
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* @dest_format for encoded audio data. Conversion is possible between
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* BYTE and TIME format by using estimated bitrate based on
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* @samples and @bytes (and @fmt).
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*/
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gboolean
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__gst_audio_encoded_audio_convert (GstAudioInfo * fmt,
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gint64 bytes, gint64 samples, GstFormat src_format,
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gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
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{
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gboolean res = FALSE;
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g_return_val_if_fail (dest_format != NULL, FALSE);
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g_return_val_if_fail (dest_value != NULL, FALSE);
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if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||
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src_value == -1)) {
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if (dest_value)
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*dest_value = src_value;
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return TRUE;
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}
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if (samples == 0 || bytes == 0 || fmt->rate == 0) {
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GST_DEBUG ("not enough metadata yet to convert");
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goto exit;
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}
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bytes *= fmt->rate;
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switch (src_format) {
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case GST_FORMAT_BYTES:
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switch (*dest_format) {
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case GST_FORMAT_TIME:
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*dest_value = gst_util_uint64_scale (src_value,
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GST_SECOND * samples, bytes);
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res = TRUE;
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break;
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default:
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res = FALSE;
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}
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break;
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case GST_FORMAT_TIME:
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switch (*dest_format) {
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case GST_FORMAT_BYTES:
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*dest_value = gst_util_uint64_scale (src_value, bytes,
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samples * GST_SECOND);
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res = TRUE;
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break;
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default:
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res = FALSE;
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}
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break;
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default:
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res = FALSE;
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}
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exit:
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return res;
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}
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#ifdef G_OS_WIN32
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/* *INDENT-OFF* */
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static struct
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{
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HMODULE dll;
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gboolean tried_loading;
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FARPROC AvSetMmThreadCharacteristics;
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FARPROC AvRevertMmThreadCharacteristics;
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} _gst_audio_avrt_tbl = { 0 };
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/* *INDENT-ON* */
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#endif
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static gboolean
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__gst_audio_init_thread_priority (void)
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{
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#ifdef G_OS_WIN32
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if (_gst_audio_avrt_tbl.tried_loading)
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return _gst_audio_avrt_tbl.dll != NULL;
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if (!_gst_audio_avrt_tbl.dll)
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_gst_audio_avrt_tbl.dll = LoadLibrary (TEXT ("avrt.dll"));
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if (!_gst_audio_avrt_tbl.dll) {
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GST_WARNING ("Failed to set thread priority, can't find avrt.dll");
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_gst_audio_avrt_tbl.tried_loading = TRUE;
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return FALSE;
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}
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_gst_audio_avrt_tbl.AvSetMmThreadCharacteristics =
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GetProcAddress (_gst_audio_avrt_tbl.dll, "AvSetMmThreadCharacteristicsA");
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_gst_audio_avrt_tbl.AvRevertMmThreadCharacteristics =
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GetProcAddress (_gst_audio_avrt_tbl.dll,
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"AvRevertMmThreadCharacteristics");
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_gst_audio_avrt_tbl.tried_loading = TRUE;
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#endif
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return TRUE;
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}
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/*
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* Increases the priority of the thread it's called from
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*/
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gboolean
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__gst_audio_set_thread_priority (void)
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{
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#ifdef G_OS_WIN32
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DWORD taskIndex = 0;
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#endif
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if (!__gst_audio_init_thread_priority ())
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return FALSE;
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#ifdef G_OS_WIN32
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/* This is only used from ringbuffer thread functions, so we don't need to
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* ever need to revert the thread priorities. */
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return _gst_audio_avrt_tbl.AvSetMmThreadCharacteristics (TEXT ("Pro Audio"),
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&taskIndex) != 0;
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#else
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return TRUE;
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#endif
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}
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