gstreamer/gst/audioconvert/gstaudioconvert.c
j^ 08047f5cfe better/unified long descriptions
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 
2006-03-29 14:00:08 +00:00

512 lines
16 KiB
C

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