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08047f5cfe
Original commit message from CVS: Patch by: j^ <j at bootlab dot org> * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst/audioconvert/gstaudioconvert.c: * gst/subparse/gstsubparse.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpclientsink.c: * gst/tcp/gsttcpclientsrc.c: * gst/tcp/gsttcpserversink.c: * gst/tcp/gsttcpserversrc.c: better/unified long descriptions Fixes #336477
512 lines
16 KiB
C
512 lines
16 KiB
C
/* GStreamer
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* Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
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* Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
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* Copyright (C) 2005 Wim Taymans <wim at fluendo dot com>
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*
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* gstaudioconvert.c: Convert audio to different audio formats automatically
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-audioconvert
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*
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* <refsect2>
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* Audioconvert converts raw audio buffers between various possible formats.
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* It supports integer to float conversion, width/depth conversion,
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* signedness and endianness conversion.
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* gst-launch -v -m audiotestsrc ! audioconvert ! audio/x-raw-int,channels=2,width=8,depth=8 ! level ! fakesink silent=TRUE
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* </programlisting>
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* This pipeline converts audio to 8-bit. The level element shows that
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* the output levels still match the one for a sine wave.
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* </para>
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* <para>
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* <programlisting>
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* gst-launch -v -m audiotestsrc ! audioconvert ! vorbisenc ! fakesink silent=TRUE
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* </programlisting>
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* The vorbis encoder takes float audio data instead of the integer data
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* generated by audiotestsrc.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2006-03-02 (0.10.4)
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*/
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/*
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* design decisions:
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* - audioconvert converts buffers in a set of supported caps. If it supports
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* a caps, it supports conversion from these caps to any other caps it
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* supports. (example: if it does A=>B and A=>C, it also does B=>C)
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* - audioconvert does not save state between buffers. Every incoming buffer is
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* converted and the converted buffer is pushed out.
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* conclusion:
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* audioconvert is not supposed to be a one-element-does-anything solution for
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* audio conversions.
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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 <string.h>
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#include "gstaudioconvert.h"
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#include "gstchannelmix.h"
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#include "plugin.h"
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GST_DEBUG_CATEGORY (audio_convert_debug);
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/*** DEFINITIONS **************************************************************/
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static GstElementDetails audio_convert_details =
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GST_ELEMENT_DETAILS ("Audio converter",
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"Filter/Converter/Audio",
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"Convert audio to different formats",
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"Benjamin Otte <in7y118@public.uni-hamburg.de>");
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/* type functions */
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static void gst_audio_convert_dispose (GObject * obj);
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/* gstreamer functions */
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static gboolean gst_audio_convert_get_unit_size (GstBaseTransform * base,
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GstCaps * caps, guint * size);
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static GstCaps *gst_audio_convert_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps);
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static void gst_audio_convert_fixate_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
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static gboolean gst_audio_convert_set_caps (GstBaseTransform * base,
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GstCaps * incaps, GstCaps * outcaps);
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static GstFlowReturn gst_audio_convert_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf);
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static GstFlowReturn gst_audio_convert_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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/* AudioConvert signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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ARG_AGGRESSIVE
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};
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#define DEBUG_INIT(bla) \
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GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element");
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GST_BOILERPLATE_FULL (GstAudioConvert, gst_audio_convert, GstBaseTransform,
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GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
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/*** GSTREAMER PROTOTYPES *****************************************************/
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#define STATIC_CAPS \
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GST_STATIC_CAPS ( \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 32;" \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 32, " \
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"depth = (int) [ 1, 32 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 24, " \
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"depth = (int) [ 1, 24 ], " "signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 16, " \
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"depth = (int) [ 1, 16 ], " \
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"signed = (boolean) { true, false }; " \
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"audio/x-raw-int, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, 8 ], " \
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"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
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"width = (int) 8, " \
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"depth = (int) [ 1, 8 ], " \
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"signed = (boolean) { true, false } " \
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)
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static GstAudioChannelPosition *supported_positions;
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static GstStaticCaps gst_audio_convert_static_caps = STATIC_CAPS;
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static GstStaticPadTemplate gst_audio_convert_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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STATIC_CAPS);
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static GstStaticPadTemplate gst_audio_convert_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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STATIC_CAPS);
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/*** TYPE FUNCTIONS ***********************************************************/
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static void
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gst_audio_convert_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_audio_convert_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_audio_convert_sink_template));
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gst_element_class_set_details (element_class, &audio_convert_details);
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}
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static void
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gst_audio_convert_class_init (GstAudioConvertClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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gint i;
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gobject_class->dispose = gst_audio_convert_dispose;
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supported_positions = g_new0 (GstAudioChannelPosition,
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GST_AUDIO_CHANNEL_POSITION_NUM);
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for (i = 0; i < GST_AUDIO_CHANNEL_POSITION_NUM; i++)
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supported_positions[i] = i;
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GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
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GST_DEBUG_FUNCPTR (gst_audio_convert_get_unit_size);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
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GST_DEBUG_FUNCPTR (gst_audio_convert_transform_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps =
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GST_DEBUG_FUNCPTR (gst_audio_convert_fixate_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
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GST_DEBUG_FUNCPTR (gst_audio_convert_set_caps);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
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GST_DEBUG_FUNCPTR (gst_audio_convert_transform_ip);
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GST_BASE_TRANSFORM_CLASS (klass)->transform =
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GST_DEBUG_FUNCPTR (gst_audio_convert_transform);
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GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
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}
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static void
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gst_audio_convert_init (GstAudioConvert * this, GstAudioConvertClass * g_class)
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{
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}
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static void
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gst_audio_convert_dispose (GObject * obj)
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{
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GstAudioConvert *this = GST_AUDIO_CONVERT (obj);
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audio_convert_clean_context (&this->ctx);
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G_OBJECT_CLASS (parent_class)->dispose (obj);
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}
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/*** GSTREAMER FUNCTIONS ******************************************************/
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/* convert the given GstCaps to our format */
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static gboolean
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gst_audio_convert_parse_caps (const GstCaps * caps, AudioConvertFmt * fmt)
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{
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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GST_DEBUG ("parse caps %p and %" GST_PTR_FORMAT, caps, caps);
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g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
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g_return_val_if_fail (fmt != NULL, FALSE);
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/* cleanup old */
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audio_convert_clean_fmt (fmt);
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fmt->endianness = G_BYTE_ORDER;
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fmt->is_int =
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(strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0);
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/* parse common fields */
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if (!gst_structure_get_int (structure, "channels", &fmt->channels))
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goto no_values;
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if (!(fmt->pos = gst_audio_get_channel_positions (structure)))
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goto no_values;
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if (!gst_structure_get_int (structure, "width", &fmt->width))
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goto no_values;
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if (!gst_structure_get_int (structure, "rate", &fmt->rate))
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goto no_values;
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if (fmt->is_int) {
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/* int specific fields */
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if (!gst_structure_get_boolean (structure, "signed", &fmt->sign))
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goto no_values;
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if (!gst_structure_get_int (structure, "depth", &fmt->depth))
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goto no_values;
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/* width != 8 can have an endianness field */
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if (fmt->width != 8) {
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if (!gst_structure_get_int (structure, "endianness", &fmt->endianness))
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goto no_values;
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}
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/* depth cannot be bigger than the width */
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if (fmt->depth > fmt->width)
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goto not_allowed;
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}
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fmt->unit_size = (fmt->width * fmt->channels) / 8;
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return TRUE;
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/* ERRORS */
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no_values:
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{
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GST_DEBUG ("could not get some values from structure");
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audio_convert_clean_fmt (fmt);
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return FALSE;
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}
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not_allowed:
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{
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GST_DEBUG ("width > depth, not allowed - make us advertise correct fmt");
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audio_convert_clean_fmt (fmt);
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return FALSE;
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}
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}
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/* BaseTransform vmethods */
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static gboolean
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gst_audio_convert_get_unit_size (GstBaseTransform * base, GstCaps * caps,
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guint * size)
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{
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AudioConvertFmt fmt = { 0 };
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g_return_val_if_fail (size, FALSE);
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if (!gst_audio_convert_parse_caps (caps, &fmt))
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goto parse_error;
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*size = fmt.unit_size;
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audio_convert_clean_fmt (&fmt);
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return TRUE;
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parse_error:
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{
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return FALSE;
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}
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}
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/* audioconvert can convert anything except sample rate; so return template
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* caps with rate fixed */
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/* FIXME:
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* it would be smart here to return the caps with the same width as the first
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*/
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static GstCaps *
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gst_audio_convert_transform_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps)
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{
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int i;
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const GValue *rate;
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GstCaps *ret;
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GstStructure *structure;
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g_return_val_if_fail (GST_CAPS_IS_SIMPLE (caps), NULL);
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structure = gst_caps_get_structure (caps, 0);
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ret = gst_static_caps_get (&gst_audio_convert_static_caps);
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/* if rate not set, we return the template */
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if (!(rate = gst_structure_get_value (structure, "rate")))
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return ret;
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/* else, write rate in the template caps */
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ret = gst_caps_make_writable (ret);
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for (i = 0; i < gst_caps_get_size (ret); ++i) {
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structure = gst_caps_get_structure (ret, i);
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gst_structure_set_value (structure, "rate", rate);
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}
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return ret;
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}
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/* try to keep as many of the structure members the same by fixating the
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* possible ranges; this way we convert the least amount of things as possible
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*/
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static void
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gst_audio_convert_fixate_caps (GstBaseTransform * base,
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GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
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{
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GstStructure *ins, *outs;
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gint rate, endianness, depth, width, channels;
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gboolean signedness;
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g_return_if_fail (gst_caps_is_fixed (caps));
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GST_DEBUG_OBJECT (base, "trying to fixate othercaps %" GST_PTR_FORMAT
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" based on caps %" GST_PTR_FORMAT, othercaps, caps);
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ins = gst_caps_get_structure (caps, 0);
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outs = gst_caps_get_structure (othercaps, 0);
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if (gst_structure_get_int (ins, "channels", &channels)) {
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if (gst_structure_has_field (outs, "channels")) {
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gst_structure_fixate_field_nearest_int (outs, "channels", channels);
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}
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}
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if (gst_structure_get_int (ins, "rate", &rate)) {
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if (gst_structure_has_field (outs, "rate")) {
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gst_structure_fixate_field_nearest_int (outs, "rate", rate);
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}
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}
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if (gst_structure_get_int (ins, "endianness", &endianness)) {
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if (gst_structure_has_field (outs, "endianness")) {
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gst_structure_fixate_field_nearest_int (outs, "endianness", endianness);
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}
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}
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if (gst_structure_get_int (ins, "width", &width)) {
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if (gst_structure_has_field (outs, "width")) {
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gst_structure_fixate_field_nearest_int (outs, "width", width);
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}
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} else {
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/* this is not allowed */
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}
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if (gst_structure_get_int (ins, "depth", &depth)) {
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if (gst_structure_has_field (outs, "depth")) {
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gst_structure_fixate_field_nearest_int (outs, "depth", depth);
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}
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} else {
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/* set depth as width */
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if (gst_structure_has_field (outs, "depth")) {
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gst_structure_fixate_field_nearest_int (outs, "depth", width);
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}
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}
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if (gst_structure_get_boolean (ins, "signed", &signedness)) {
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if (gst_structure_has_field (outs, "signed")) {
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gst_structure_fixate_field_boolean (outs, "signed", signedness);
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}
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}
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GST_DEBUG_OBJECT (base, "fixated othercaps to %" GST_PTR_FORMAT, othercaps);
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}
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static gboolean
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gst_audio_convert_set_caps (GstBaseTransform * base, GstCaps * incaps,
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GstCaps * outcaps)
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{
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AudioConvertFmt in_ac_caps = { 0 };
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AudioConvertFmt out_ac_caps = { 0 };
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GstAudioConvert *this = GST_AUDIO_CONVERT (base);
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GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
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GST_PTR_FORMAT, incaps, outcaps);
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if (!gst_audio_convert_parse_caps (incaps, &in_ac_caps))
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return FALSE;
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if (!gst_audio_convert_parse_caps (outcaps, &out_ac_caps))
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return FALSE;
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if (!audio_convert_prepare_context (&this->ctx, &in_ac_caps, &out_ac_caps))
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goto no_converter;
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return TRUE;
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no_converter:
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{
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_audio_convert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
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{
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/* nothing to do here */
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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gst_audio_convert_transform (GstBaseTransform * base, GstBuffer * inbuf,
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GstBuffer * outbuf)
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{
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GstAudioConvert *this = GST_AUDIO_CONVERT (base);
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gboolean res;
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|
gint insize, outsize;
|
|
gint samples;
|
|
gpointer src, dst;
|
|
|
|
/* get amount of samples to convert. */
|
|
samples = GST_BUFFER_SIZE (inbuf) / this->ctx.in.unit_size;
|
|
|
|
/* get in/output sizes, to see if the buffers we got are of correct
|
|
* sizes */
|
|
if (!(res = audio_convert_get_sizes (&this->ctx, samples, &insize, &outsize)))
|
|
goto error;
|
|
|
|
/* check in and outsize */
|
|
if (GST_BUFFER_SIZE (inbuf) < insize)
|
|
goto wrong_size;
|
|
if (GST_BUFFER_SIZE (outbuf) < outsize)
|
|
goto wrong_size;
|
|
|
|
/* get src and dst data */
|
|
src = GST_BUFFER_DATA (inbuf);
|
|
dst = GST_BUFFER_DATA (outbuf);
|
|
|
|
/* and convert the samples */
|
|
if (!(res = audio_convert_convert (&this->ctx, src, dst,
|
|
samples, gst_buffer_is_writable (inbuf))))
|
|
goto convert_error;
|
|
|
|
GST_BUFFER_SIZE (outbuf) = outsize;
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
GST_ELEMENT_ERROR (this, STREAM, NOT_IMPLEMENTED,
|
|
("cannot get input/output sizes for %d samples", samples),
|
|
("cannot get input/output sizes for %d samples", samples));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
wrong_size:
|
|
{
|
|
GST_ELEMENT_ERROR (this, STREAM, NOT_IMPLEMENTED,
|
|
("input/output buffers are of wrong size in: %d < %d or out: %d < %d",
|
|
GST_BUFFER_SIZE (inbuf), insize, GST_BUFFER_SIZE (outbuf), outsize),
|
|
("input/output buffers are of wrong size in: %d < %d or out: %d < %d",
|
|
GST_BUFFER_SIZE (inbuf), insize, GST_BUFFER_SIZE (outbuf),
|
|
outsize));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
convert_error:
|
|
{
|
|
GST_ELEMENT_ERROR (this, STREAM, NOT_IMPLEMENTED,
|
|
("error while converting"), ("error while converting"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|