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Move wavpack to good.

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Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/inspect/plugin-wavpack.xml:
* ext/Makefile.am:
* ext/wavpack/Makefile.am:
* ext/wavpack/gstwavpack.c:
* ext/wavpack/gstwavpackcommon.c:
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackdec.c:
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackenc.c:
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c:
* ext/wavpack/gstwavpackparse.h:
* ext/wavpack/gstwavpackstreamreader.c:
* ext/wavpack/gstwavpackstreamreader.h:
* ext/wavpack/md5.c:
* ext/wavpack/md5.h:
* tests/check/Makefile.am:
* tests/check/elements/wavpackdec.c:
* tests/check/elements/wavpackenc.c:
* tests/check/elements/wavpackparse.c:
Move wavpack to good.
This commit is contained in:
Thomas Vander Stichele 2007-06-08 20:20:34 +00:00
parent dbf7acdf1d
commit 6bd7199f3a
26 changed files with 30 additions and 4436 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

29
ChangeLog
View file

@ -1,3 +1,32 @@
2007-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/inspect/plugin-wavpack.xml:
* ext/Makefile.am:
* ext/wavpack/Makefile.am:
* ext/wavpack/gstwavpack.c:
* ext/wavpack/gstwavpackcommon.c:
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackdec.c:
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackenc.c:
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c:
* ext/wavpack/gstwavpackparse.h:
* ext/wavpack/gstwavpackstreamreader.c:
* ext/wavpack/gstwavpackstreamreader.h:
* ext/wavpack/md5.c:
* ext/wavpack/md5.h:
* tests/check/Makefile.am:
* tests/check/elements/wavpackdec.c:
* tests/check/elements/wavpackenc.c:
* tests/check/elements/wavpackparse.c:
Move wavpack to good.
2007-06-08 Jan Schmidt <thaytan@mad.scientist.com>
* configure.ac:

18
configure.ac
View file

@ -878,22 +878,6 @@ int main () { return 0; }
], )
])
dnl *** wavpack ***
translit(dnm, m, l) AM_CONDITIONAL(USE_WAVPACK, true)
AG_GST_CHECK_FEATURE(WAVPACK, [wavpack plug-in], wavpack, [
PKG_CHECK_MODULES(WAVPACK, wavpack >= 4.40.0, HAVE_WAVPACK=yes, [
PKG_CHECK_MODULES(WAVPACK, wavpack >= 4.20, [
HAVE_WAVPACK=yes
AC_DEFINE(WAVPACK_OLD_API, 1, [old wavpack API])
],[
HAVE_WAVPACK=no
AC_MSG_RESULT(no)
])
])
AC_SUBST(WAVPACK_CFLAGS)
AC_SUBST(WAVPACK_LIBS)
])
dnl *** dvb ***
translit(dnm, m, l) AM_CONDITIONAL(USE_DVB, true)
AG_GST_CHECK_FEATURE(DVB, [DVB Source], dvb, [
@ -949,7 +933,6 @@ AM_CONDITIONAL(USE_THEORADEC, false)
AM_CONDITIONAL(USE_TIMIDITY, false)
AM_CONDITIONAL(USE_X264, false)
AM_CONDITIONAL(USE_XVID, false)
AM_CONDITIONAL(USE_WAVPACK, false)
AM_CONDITIONAL(USE_WILDMIDI, false)
AM_CONDITIONAL(USE_DVB, false)
AM_CONDITIONAL(USE_ZLIB, false)
@ -1076,7 +1059,6 @@ ext/spc/Makefile
ext/swfdec/Makefile
ext/theora/Makefile
ext/timidity/Makefile
ext/wavpack/Makefile
ext/x264/Makefile
ext/xvid/Makefile
po/Makefile.in

3
docs/plugins/Makefile.am
View file

@ -96,9 +96,6 @@ EXTRA_HFILES = \
$(top_srcdir)/ext/musicbrainz/gsttrm.h \
$(top_srcdir)/ext/sdl/sdlaudiosink.h \
$(top_srcdir)/ext/sdl/sdlvideosink.h \
$(top_srcdir)/ext/wavpack/gstwavpackdec.h \
$(top_srcdir)/ext/wavpack/gstwavpackenc.h \
$(top_srcdir)/ext/wavpack/gstwavpackparse.h \
$(top_srcdir)/gst/qtdemux/qtdemux.h \
$(top_srcdir)/gst/replaygain/gstrganalysis.h \
$(top_srcdir)/gst/replaygain/gstrglimiter.h \

4
docs/plugins/gst-plugins-bad-plugins-docs.sgml
View file

@ -32,9 +32,6 @@
<xi:include href="xml/element-videodetect.xml" />
<xi:include href="xml/element-videomark.xml" />
<xi:include href="xml/element-waveformsink.xml" />
<xi:include href="xml/element-wavpackdec.xml" />
<xi:include href="xml/element-wavpackenc.xml" />
<xi:include href="xml/element-wavpackparse.xml" />
</chapter>
<chapter>
@ -64,7 +61,6 @@
<xi:include href="xml/plugin-videocrop.xml" />
<xi:include href="xml/plugin-videosignal.xml" />
<xi:include href="xml/plugin-waveform.xml" />
<xi:include href="xml/plugin-wavpack.xml" />
<xi:include href="xml/plugin-xingheader.xml" />
</chapter>

25
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -234,28 +234,3 @@ GstWaveFormSink
<SUBSECTION Standard>
GstWaveFormSinkClass
</SECTION>
<SECTION>
<FILE>element-wavpackdec</FILE>
GstWavpackDec
<TITLE>wavpackdec</TITLE>
<SUBSECTION Standard>
GstWavpackDecClass
</SECTION>
<SECTION>
<FILE>element-wavpackenc</FILE>
GstWavpackEnc
<TITLE>wavpackenc</TITLE>
<SUBSECTION Standard>
GstWavpackEncClass
</SECTION>
<SECTION>
<FILE>element-wavpackparse</FILE>
GstWavpackParse
<TITLE>wavpackparse</TITLE>
<SUBSECTION Standard>
GstWavpackParseClass
</SECTION>

3
docs/plugins/gst-plugins-bad-plugins.hierarchy
View file

@ -14,9 +14,6 @@ GObject
GstX264Enc
GstXvidEnc
GstXvidDec
GstWavpackParse
GstWavpackDec
GstWavpackEnc
GstSwfdec
GstSpcDec
GstPitch

34
docs/plugins/inspect/plugin-wavpack.xml
View file

@ -1,34 +0,0 @@
<plugin>
<name>wavpack</name>
<description>Wavpack lossless/lossy audio format handling</description>
<filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
<basename>libgstwavpack.so</basename>
<version>0.10.4.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
<package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<name>wavpackdec</name>
<longname>Wavpack audio decoder</longname>
<class>Codec/Decoder/Audio</class>
<description>Decodes Wavpack audio data</description>
<author>Arwed v. Merkatz &lt;v.merkatz@gmx.net&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
</element>
<element>
<name>wavpackenc</name>
<longname>Wavpack audio encoder</longname>
<class>Codec/Encoder/Audio</class>
<description>Encodes audio with the Wavpack lossless/lossy audio codec</description>
<author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
</element>
<element>
<name>wavpackparse</name>
<longname>Wavpack parser</longname>
<class>Codec/Demuxer/Audio</class>
<description>Parses Wavpack files</description>
<author>Arwed v. Merkatz &lt;v.merkatz@gmx.net&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
</element>
</elements>
</plugin>

10
ext/Makefile.am
View file

@ -94,12 +94,6 @@ endif
HERMES_DIR=
# endif
if USE_WAVPACK
WAVPACK_DIR=wavpack
else
WAVPACK_DIR=
endif
if USE_IVORBIS
IVORBIS_DIR=ivorbis
else
@ -303,12 +297,11 @@ SUBDIRS=\
$(TARKIN_DIR) \
$(THEORA_DIR) \
$(TIMIDITY_DIR) \
$(WAVPACK_DIR) \
$(X264_DIR) \
$(XINE_DIR) \
$(XVID_DIR)
DIST_SUBDIRS= \
DIST_SUBDIRS = \
alsaspdif \
amrwb \
bz2 \
@ -336,6 +329,5 @@ DIST_SUBDIRS= \
swfdec \
theora \
timidity \
wavpack \
x264 \
xvid

23
ext/wavpack/Makefile.am
View file

@ -1,23 +0,0 @@
plugin_LTLIBRARIES = libgstwavpack.la
libgstwavpack_la_SOURCES = \
gstwavpack.c \
gstwavpackcommon.c \
gstwavpackparse.c \
gstwavpackdec.c \
gstwavpackenc.c \
gstwavpackstreamreader.c \
md5.c
libgstwavpack_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(WAVPACK_CFLAGS)
libgstwavpack_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(WAVPACK_LIBS)
libgstwavpack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = \
gstwavpackparse.h \
gstwavpackdec.h \
gstwavpackenc.h \
gstwavpackcommon.h \
gstwavpackstreamreader.h \
md5.h

55
ext/wavpack/gstwavpack.c
View file

@ -1,55 +0,0 @@
/* GStreamer wavpack plugin
* (c) 2004 Arwed v. Merkatz <v.merkatz@gmx.net>
*
* gstwavpack.c: plugin loader
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst-i18n-plugin.h>
#include "gstwavpackparse.h"
#include "gstwavpackdec.h"
#include "gstwavpackenc.h"
/* debug category for common code */
GST_DEBUG_CATEGORY (wavpack_debug);
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (wavpack_debug, "wavpack", 0, "Wavpack elements");
#if ENABLE_NLS
GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
LOCALEDIR);
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
#endif
return (gst_wavpack_parse_plugin_init (plugin)
&& gst_wavpack_dec_plugin_init (plugin)
&& gst_wavpack_enc_plugin_init (plugin));
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"wavpack",
"Wavpack lossless/lossy audio format handling",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

98
ext/wavpack/gstwavpackcommon.c
View file

@ -1,98 +0,0 @@
/* GStreamer Wavpack plugin
* Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
* Copyright (c) 1998 - 2005 Conifer Software
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackcommon.c: common helper functions
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstwavpackcommon.h"
#include <string.h>
GST_DEBUG_CATEGORY_EXTERN (wavpack_debug);
#define GST_CAT_DEFAULT wavpack_debug
gboolean
gst_wavpack_read_header (WavpackHeader * header, guint8 * buf)
{
g_memmove (header, buf, sizeof (WavpackHeader));
#ifndef WAVPACK_OLD_API
WavpackLittleEndianToNative (header, WavpackHeaderFormat);
#else
little_endian_to_native (header, WavpackHeaderFormat);
#endif
return (memcmp (header->ckID, "wvpk", 4) == 0);
}
/* inspired by the original one in wavpack */
gboolean
gst_wavpack_read_metadata (GstWavpackMetadata * wpmd, guint8 * header_data,
guint8 ** p_data)
{
WavpackHeader hdr;
guint8 *end;
gst_wavpack_read_header (&hdr, header_data);
end = header_data + hdr.ckSize + 8;
if (end - *p_data < 2)
return FALSE;
wpmd->id = GST_READ_UINT8 (*p_data);
wpmd->byte_length = 2 * (guint) GST_READ_UINT8 (*p_data + 1);
*p_data += 2;
if ((wpmd->id & ID_LARGE) == ID_LARGE) {
guint extra;
wpmd->id &= ~ID_LARGE;
if (end - *p_data < 2)
return FALSE;
extra = GST_READ_UINT16_LE (*p_data);
wpmd->byte_length += (extra << 9);
*p_data += 2;
}
if ((wpmd->id & ID_ODD_SIZE) == ID_ODD_SIZE) {
wpmd->id &= ~ID_ODD_SIZE;
--wpmd->byte_length;
}
if (wpmd->byte_length > 0) {
if (end - *p_data < wpmd->byte_length + (wpmd->byte_length & 1)) {
wpmd->data = NULL;
return FALSE;
}
wpmd->data = *p_data;
*p_data += wpmd->byte_length + (wpmd->byte_length & 1);
} else {
wpmd->data = NULL;
}
return TRUE;
}

69
ext/wavpack/gstwavpackcommon.h
View file

@ -1,69 +0,0 @@
/* GStreamer Wavpack plugin
* Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackcommon.h: common helper functions
*
* 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.
*/
#ifndef __GST_WAVPACK_COMMON_H__
#define __GST_WAVPACK_COMMON_H__
#include <gst/gst.h>
#include <wavpack/wavpack.h>
typedef struct
{
guint32 byte_length;
guint8 *data;
guint8 id;
} GstWavpackMetadata;
#define ID_UNIQUE 0x3f
#define ID_OPTIONAL_DATA 0x20
#define ID_ODD_SIZE 0x40
#define ID_LARGE 0x80
#define ID_DUMMY 0x0
#define ID_ENCODER_INFO 0x1
#define ID_DECORR_TERMS 0x2
#define ID_DECORR_WEIGHTS 0x3
#define ID_DECORR_SAMPLES 0x4
#define ID_ENTROPY_VARS 0x5
#define ID_HYBRID_PROFILE 0x6
#define ID_SHAPING_WEIGHTS 0x7
#define ID_FLOAT_INFO 0x8
#define ID_INT32_INFO 0x9
#define ID_WV_BITSTREAM 0xa
#define ID_WVC_BITSTREAM 0xb
#define ID_WVX_BITSTREAM 0xc
#define ID_CHANNEL_INFO 0xd
#define ID_RIFF_HEADER (ID_OPTIONAL_DATA | 0x1)
#define ID_RIFF_TRAILER (ID_OPTIONAL_DATA | 0x2)
#define ID_REPLAY_GAIN (ID_OPTIONAL_DATA | 0x3)
#define ID_CUESHEET (ID_OPTIONAL_DATA | 0x4)
#define ID_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0x5)
#define ID_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x6)
#define ID_SAMPLE_RATE (ID_OPTIONAL_DATA | 0x7)
gboolean gst_wavpack_read_header (WavpackHeader * header, guint8 * buf);
gboolean gst_wavpack_read_metadata (GstWavpackMetadata * meta,
guint8 * header_data, guint8 ** p_data);
#endif

497
ext/wavpack/gstwavpackdec.c
View file

@ -1,497 +0,0 @@
/* GStreamer Wavpack plugin
* Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
* Copyright (c) 2006 Edward Hervey <bilboed@gmail.com>
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackdec.c: raw Wavpack bitstream decoder
*
* 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-wavpackdec
*
* <refsect2>
* WavpackDec decodes framed (for example by the WavpackParse element)
* Wavpack streams and decodes them to raw audio.
* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
* audio codec that features both lossless and lossy encoding.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch filesrc location=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink
* </programlisting>
* This pipeline decodes the Wavpack file test.wv into raw audio buffers and
* tries to play it back using an automatically found audio sink.
* </para>
* </refsect2>
*/
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <math.h>
#include <string.h>
#include <wavpack/wavpack.h>
#include "gstwavpackdec.h"
#include "gstwavpackcommon.h"
#include "gstwavpackstreamreader.h"
#define WAVPACK_DEC_MAX_ERRORS 16
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
#define GST_CAT_DEFAULT gst_wavpack_dec_debug
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) [ 1, 32 ], "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) [ 1, 32 ], "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
);
static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps);
static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
static void gst_wavpack_dec_finalize (GObject * object);
static GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
static void
gst_wavpack_dec_base_init (gpointer klass)
{
static const GstElementDetails plugin_details =
GST_ELEMENT_DETAILS ("Wavpack audio decoder",
"Codec/Decoder/Audio",
"Decodes Wavpack audio data",
"Arwed v. Merkatz <v.merkatz@gmx.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_details (element_class, &plugin_details);
}
static void
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_wavpack_dec_finalize);
}
static void
gst_wavpack_dec_reset (GstWavpackDec * dec)
{
dec->wv_id.buffer = NULL;
dec->wv_id.position = dec->wv_id.length = 0;
dec->error_count = 0;
dec->channels = 0;
dec->sample_rate = 0;
dec->depth = 0;
gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
}
static void
gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
{
dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_chain_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
gst_pad_set_setcaps_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_set_caps));
gst_pad_set_event_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_pad_use_fixed_caps (dec->srcpad);
gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new ();
gst_wavpack_dec_reset (dec);
}
static void
gst_wavpack_dec_finalize (GObject * object)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (object);
g_free (dec->stream_reader);
dec->stream_reader = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
/* Check if we can set the caps here already */
if (gst_structure_get_int (structure, "channels", &dec->channels) &&
gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
gst_structure_get_int (structure, "width", &dec->depth)) {
GstCaps *caps;
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->depth,
"width", G_TYPE_INT, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
gst_object_unref (dec);
return TRUE;
}
static gboolean
gst_wavpack_dec_clip_outgoing_buffer (GstWavpackDec * dec, GstBuffer * buf)
{
gint64 start, stop, cstart, cstop, diff;
if (dec->segment.format != GST_FORMAT_TIME)
return TRUE;
start = GST_BUFFER_TIMESTAMP (buf);
stop = start + GST_BUFFER_DURATION (buf);
if (gst_segment_clip (&dec->segment, GST_FORMAT_TIME,
start, stop, &cstart, &cstop)) {
diff = cstart - start;
if (diff > 0) {
GST_BUFFER_TIMESTAMP (buf) = cstart;
GST_BUFFER_DURATION (buf) -= diff;
diff = 4 * dec->channels
* GST_CLOCK_TIME_TO_FRAMES (diff, dec->sample_rate);
GST_BUFFER_DATA (buf) += diff;
GST_BUFFER_SIZE (buf) -= diff;
}
diff = cstop - stop;
if (diff > 0) {
GST_BUFFER_DURATION (buf) -= diff;
diff = 4 * dec->channels
* GST_CLOCK_TIME_TO_FRAMES (diff, dec->sample_rate);
GST_BUFFER_SIZE (buf) -= diff;
}
} else {
GST_DEBUG_OBJECT (dec, "buffer is outside configured segment");
return FALSE;
}
return TRUE;
}
static void
gst_wavpack_dec_post_tags (GstWavpackDec * dec)
{
GstTagList *list;
GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
gint64 duration, size;
list = gst_tag_list_new ();
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "Wavpack", NULL);
/* try to estimate the average bitrate */
if (gst_pad_query_peer_duration (dec->sinkpad, &format_bytes, &size) &&
gst_pad_query_peer_duration (dec->sinkpad, &format_time, &duration) &&
size > 0 && duration > 0) {
guint64 bitrate;
bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
(guint) bitrate, NULL);
}
gst_element_post_message (GST_ELEMENT (dec),
gst_message_new_tag (GST_OBJECT (dec), list));
}
static GstFlowReturn
gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackDec *dec;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
int32_t decoded, unpacked_size;
gboolean format_changed;
dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
/* check input, we only accept framed input with complete chunks */
if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
goto input_not_framed;
if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf)))
goto invalid_header;
if (GST_BUFFER_SIZE (buf) != wph.ckSize + 4 * 1 + 4)
goto input_not_framed;
dec->wv_id.buffer = GST_BUFFER_DATA (buf);
dec->wv_id.length = GST_BUFFER_SIZE (buf);
dec->wv_id.position = 0;
/* create a new wavpack context if there is none yet but if there
* was already one (i.e. caps were set on the srcpad) check whether
* the new one has the same caps */
if (!dec->context) {
gchar error_msg[80];
dec->context = WavpackOpenFileInputEx (dec->stream_reader,
&dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
if (!dec->context) {
GST_WARNING ("Couldn't decode buffer: %s", error_msg);
dec->error_count++;
if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
goto out; /* just return OK for now */
} else {
goto decode_error;
}
}
}
g_assert (dec->context != NULL);
dec->error_count = 0;
format_changed =
(dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
(dec->channels != WavpackGetNumChannels (dec->context)) ||
(dec->depth != WavpackGetBitsPerSample (dec->context));
if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
GstCaps *caps;
dec->sample_rate = WavpackGetSampleRate (dec->context);
dec->channels = WavpackGetNumChannels (dec->context);
dec->depth = WavpackGetBitsPerSample (dec->context);
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->depth,
"width", G_TYPE_INT, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
/* alloc output buffer */
unpacked_size = 4 * wph.block_samples * dec->channels;
ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
gst_buffer_stamp (outbuf, buf);
/* If we got a DISCONT buffer forward the flag. Nothing else
* has to be done as libwavpack doesn't store state between
* Wavpack blocks */
if (GST_BUFFER_IS_DISCONT (buf))
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
/* decode */
decoded = WavpackUnpackSamples (dec->context,
(int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
if (decoded != wph.block_samples)
goto decode_error;
if (gst_wavpack_dec_clip_outgoing_buffer (dec, outbuf)) {
GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
ret = gst_pad_push (dec->srcpad, outbuf);
} else {
gst_buffer_unref (outbuf);
}
out:
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
}
gst_buffer_unref (buf);
return ret;
/* ERRORS */
input_not_framed:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
invalid_header:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
decode_error:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Failed to decode wavpack stream"));
gst_buffer_unref (outbuf);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static gboolean
gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:{
GstFormat fmt;
gboolean is_update;
gint64 start, end, base;
gdouble rate;
gst_event_parse_new_segment (event, &is_update, &rate, &fmt, &start,
&end, &base);
if (fmt == GST_FORMAT_TIME) {
GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (%"
GST_TIME_FORMAT " - %" GST_TIME_FORMAT ")", GST_TIME_ARGS (start),
GST_TIME_ARGS (end));
gst_segment_set_newsegment (&dec->segment, is_update, rate, fmt,
start, end, base);
} else {
gst_segment_init (&dec->segment, GST_FORMAT_UNDEFINED);
}
break;
}
default:
break;
}
gst_object_unref (dec);
return gst_pad_event_default (pad, event);
}
static GstStateChangeReturn
gst_wavpack_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstWavpackDec *dec = GST_WAVPACK_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (dec->context) {
WavpackCloseFile (dec->context);
dec->context = NULL;
}
gst_wavpack_dec_reset (dec);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
gboolean
gst_wavpack_dec_plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "wavpackdec",
GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
return FALSE;
GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpack_dec", 0,
"Wavpack decoder");
return TRUE;
}

78
ext/wavpack/gstwavpackdec.h
View file

@ -1,78 +0,0 @@
/* GStreamer Wavpack plugin
* Copyright (c) 2004 Arwed v. Merkatz <v.merkatz@gmx.net>
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackdec.h: raw Wavpack bitstream decoder
*
* 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.
*/
#ifndef __GST_WAVPACK_DEC_H__
#define __GST_WAVPACK_DEC_H__
#include <gst/gst.h>
#include <wavpack/wavpack.h>
#include "gstwavpackstreamreader.h"
G_BEGIN_DECLS
#define GST_TYPE_WAVPACK_DEC \
(gst_wavpack_dec_get_type())
#define GST_WAVPACK_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_DEC,GstWavpackDec))
#define GST_WAVPACK_DEC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_DEC,GstWavpackDecClass))
#define GST_IS_WAVPACK_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_DEC))
#define GST_IS_WAVPACK_DEC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_DEC))
typedef struct _GstWavpackDec GstWavpackDec;
typedef struct _GstWavpackDecClass GstWavpackDecClass;
struct _GstWavpackDec
{
GstElement element;
/*< private > */
GstPad *sinkpad;
GstPad *srcpad;
WavpackContext *context;
WavpackStreamReader *stream_reader;
read_id wv_id;
GstSegment segment; /* used for clipping, TIME format */
gint sample_rate;
gint depth;
gint channels;
gint error_count;
};
struct _GstWavpackDecClass
{
GstElementClass parent;
};
GType gst_wavpack_dec_get_type (void);
gboolean gst_wavpack_dec_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_WAVPACK_DEC_H__ */

908
ext/wavpack/gstwavpackenc.c
View file

@ -1,908 +0,0 @@
/* GStreamer Wavpack encoder plugin
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackdec.c: Wavpack audio encoder
*
* 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-wavpackenc
*
* <refsect2>
* WavpackEnc encodes raw audio into a framed Wavpack stream.
* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
* audio codec that features both lossless and lossy encoding.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
* </programlisting>
* This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
* as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits).
* </para>
* <para>
* <programlisting>
* gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
* </programlisting>
* This pipeline encodes audio from an audio CD into a Wavpack file using
* lossless encoding (the file output will be fairly large).
* </para>
* <para>
* <programlisting>
* gst-launch cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
* </programlisting>
* This pipeline encodes audio from an audio CD into a Wavpack file using
* lossy encoding at a certain bitrate (the file will be fairly small).
* </para>
* </refsect2>
*/
/*
* TODO: - add multichannel handling. channel_mask is:
* front left
* front right
* center
* LFE
* back left
* back right
* front left center
* front right center
* back left
* back center
* side left
* side right
* ...
* - add 32 bit float mode. CONFIG_FLOAT_DATA
*/
#include <string.h>
#include <gst/gst.h>
#include <glib/gprintf.h>
#include <wavpack/wavpack.h>
#include "gstwavpackenc.h"
#include "gstwavpackcommon.h"
#include "md5.h"
static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps);
static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count);
static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element,
GstStateChange transition);
static void gst_wavpack_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wavpack_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
enum
{
ARG_0,
ARG_MODE,
ARG_BITRATE,
ARG_BITSPERSAMPLE,
ARG_CORRECTION_MODE,
ARG_MD5,
ARG_EXTRA_PROCESSING,
ARG_JOINT_STEREO_MODE
};
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug);
#define GST_CAT_DEFAULT gst_wavpack_enc_debug
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) [ 1, 32], "
"endianness = (int) BYTE_ORDER, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) [ 1, 32 ], "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
);
static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) TRUE")
);
enum
{
GST_WAVPACK_ENC_MODE_VERY_FAST = 0,
GST_WAVPACK_ENC_MODE_FAST,
GST_WAVPACK_ENC_MODE_DEFAULT,
GST_WAVPACK_ENC_MODE_HIGH,
GST_WAVPACK_ENC_MODE_VERY_HIGH
};
#define GST_TYPE_WAVPACK_ENC_MODE (gst_wavpack_enc_mode_get_type ())
static GType
gst_wavpack_enc_mode_get_type (void)
{
static GType qtype = 0;
if (qtype == 0) {
static const GEnumValue values[] = {
#if 0
/* Very Fast Compression is not supported yet, but will be supported
* in future wavpack versions */
{GST_WAVPACK_ENC_MODE_VERY_FAST, "Very Fast Compression", "veryfast"},
#endif
{GST_WAVPACK_ENC_MODE_FAST, "Fast Compression", "fast"},
{GST_WAVPACK_ENC_MODE_DEFAULT, "Normal Compression", "normal"},
{GST_WAVPACK_ENC_MODE_HIGH, "High Compression", "high"},
#ifndef WAVPACK_OLD_API
{GST_WAVPACK_ENC_MODE_VERY_HIGH, "Very High Compression", "veryhigh"},
#endif
{0, NULL, NULL}
};
qtype = g_enum_register_static ("GstWavpackEncMode", values);
}
return qtype;
}
enum
{
GST_WAVPACK_CORRECTION_MODE_OFF = 0,
GST_WAVPACK_CORRECTION_MODE_ON,
GST_WAVPACK_CORRECTION_MODE_OPTIMIZED
};
#define GST_TYPE_WAVPACK_ENC_CORRECTION_MODE (gst_wavpack_enc_correction_mode_get_type ())
static GType
gst_wavpack_enc_correction_mode_get_type (void)
{
static GType qtype = 0;
if (qtype == 0) {
static const GEnumValue values[] = {
{GST_WAVPACK_CORRECTION_MODE_OFF, "Create no correction file", "off"},
{GST_WAVPACK_CORRECTION_MODE_ON, "Create correction file", "on"},
{GST_WAVPACK_CORRECTION_MODE_OPTIMIZED,
"Create optimized correction file", "optimized"},
{0, NULL, NULL}
};
qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values);
}
return qtype;
}
enum
{
GST_WAVPACK_JS_MODE_AUTO = 0,
GST_WAVPACK_JS_MODE_LEFT_RIGHT,
GST_WAVPACK_JS_MODE_MID_SIDE
};
#define GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE (gst_wavpack_enc_joint_stereo_mode_get_type ())
static GType
gst_wavpack_enc_joint_stereo_mode_get_type (void)
{
static GType qtype = 0;
if (qtype == 0) {
static const GEnumValue values[] = {
{GST_WAVPACK_JS_MODE_AUTO, "auto", "auto"},
{GST_WAVPACK_JS_MODE_LEFT_RIGHT, "left/right", "leftright"},
{GST_WAVPACK_JS_MODE_MID_SIDE, "mid/side", "midside"},
{0, NULL, NULL}
};
qtype = g_enum_register_static ("GstWavpackEncJSMode", values);
}
return qtype;
}
GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstElement, GST_TYPE_ELEMENT);
static void
gst_wavpack_enc_base_init (gpointer klass)
{
static const GstElementDetails element_details = {
"Wavpack audio encoder",
"Codec/Encoder/Audio",
"Encodes audio with the Wavpack lossless/lossy audio codec",
"Sebastian Dröge <slomo@circular-chaos.org>"
};
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
/* add pad templates */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&wvcsrc_factory));
/* set element details */
gst_element_class_set_details (element_class, &element_details);
}
static void
gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
/* set state change handler */
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state);
/* set property handlers */
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_wavpack_enc_get_property);
/* install all properties */
g_object_class_install_property (gobject_class, ARG_MODE,
g_param_spec_enum ("mode", "Encoding mode",
"Speed versus compression tradeoff.",
GST_TYPE_WAVPACK_ENC_MODE, GST_WAVPACK_ENC_MODE_DEFAULT,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_BITRATE,
g_param_spec_uint ("bitrate", "Bitrate",
"Try to encode with this average bitrate (bits/sec). "
"This enables lossy encoding, values smaller than 24000 disable it again.",
0, 9600000, 0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_BITSPERSAMPLE,
g_param_spec_double ("bits-per-sample", "Bits per sample",
"Try to encode with this amount of bits per sample. "
"This enables lossy encoding, values smaller than 2.0 disable it again.",
0.0, 24.0, 0.0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE,
g_param_spec_enum ("correction-mode", "Correction stream mode",
"Use this mode for the correction stream. Only works in lossy mode!",
GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, GST_WAVPACK_CORRECTION_MODE_OFF,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_MD5,
g_param_spec_boolean ("md5", "MD5",
"Store MD5 hash of raw samples within the file.", FALSE,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING,
g_param_spec_uint ("extra-processing", "Extra processing",
"Use better but slower filters for better compression/quality.",
0, 6, 0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_JOINT_STEREO_MODE,
g_param_spec_enum ("joint-stereo-mode", "Joint-Stereo mode",
"Use this joint-stereo mode.", GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE,
GST_WAVPACK_JS_MODE_AUTO, G_PARAM_READWRITE));
}
static void
gst_wavpack_enc_reset (GstWavpackEnc * enc)
{
/* close and free everything stream related if we already did something */
if (enc->wp_context) {
WavpackCloseFile (enc->wp_context);
enc->wp_context = NULL;
}
if (enc->wp_config) {
g_free (enc->wp_config);
enc->wp_config = NULL;
}
if (enc->first_block) {
g_free (enc->first_block);
enc->first_block = NULL;
}
enc->first_block_size = 0;
if (enc->md5_context) {
g_free (enc->md5_context);
enc->md5_context = NULL;
}
/* reset the last returns to GST_FLOW_OK. This is only set to something else
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
* so not valid anymore */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
/* reset stream information */
enc->samplerate = 0;
enc->depth = 0;
enc->channels = 0;
}
static void
gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
{
enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
gst_pad_set_setcaps_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps));
gst_pad_set_chain_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain));
gst_pad_set_event_function (enc->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event));
gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
/* setup src pad */
enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
/* initialize object attributes */
enc->wp_config = NULL;
enc->wp_context = NULL;
enc->first_block = NULL;
enc->md5_context = NULL;
gst_wavpack_enc_reset (enc);
enc->wv_id.correction = FALSE;
enc->wv_id.wavpack_enc = enc;
enc->wvc_id.correction = TRUE;
enc->wvc_id.wavpack_enc = enc;
/* set default values of params */
enc->mode = GST_WAVPACK_ENC_MODE_DEFAULT;
enc->bitrate = 0;
enc->bps = 0.0;
enc->correction_mode = GST_WAVPACK_CORRECTION_MODE_OFF;
enc->md5 = FALSE;
enc->extra_processing = 0;
enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
}
static gboolean
gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
/* FIXME: Workaround for bug #421543: calls gst_pad_accept_caps() */
/* check caps and put relevant parts into our object attributes */
if (!gst_pad_accept_caps (pad, caps) ||
!gst_structure_get_int (structure, "channels", &enc->channels) ||
!gst_structure_get_int (structure, "rate", &enc->samplerate) ||
!gst_structure_get_int (structure, "depth", &enc->depth)) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("got invalid caps: %" GST_PTR_FORMAT, caps));
gst_object_unref (enc);
return FALSE;
}
/* set fixed src pad caps now that we know what we will get */
caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, enc->channels,
"rate", G_TYPE_INT, enc->samplerate,
"width", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
if (!gst_pad_set_caps (enc->srcpad, caps)) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("setting caps failed: %" GST_PTR_FORMAT, caps));
gst_caps_unref (caps);
gst_object_unref (enc);
return FALSE;
}
gst_pad_use_fixed_caps (enc->srcpad);
gst_caps_unref (caps);
gst_object_unref (enc);
return TRUE;
}
static void
gst_wavpack_enc_set_wp_config (GstWavpackEnc * enc)
{
enc->wp_config = g_new0 (WavpackConfig, 1);
/* set general stream informations in the WavpackConfig */
enc->wp_config->bytes_per_sample = GST_ROUND_UP_8 (enc->depth) / 8;
enc->wp_config->bits_per_sample = enc->depth;
enc->wp_config->num_channels = enc->channels;
/* TODO: handle more than 2 channels correctly! */
if (enc->channels == 1) {
enc->wp_config->channel_mask = 0x4;
} else if (enc->channels == 2) {
enc->wp_config->channel_mask = 0x2 | 0x1;
}
enc->wp_config->sample_rate = enc->samplerate;
/*
* Set parameters in WavpackConfig
*/
/* Encoding mode */
switch (enc->mode) {
#if 0
case GST_WAVPACK_ENC_MODE_VERY_FAST:
enc->wp_config->flags |= CONFIG_VERY_FAST_FLAG;
enc->wp_config->flags |= CONFIG_FAST_FLAG;
break;
#endif
case GST_WAVPACK_ENC_MODE_FAST:
enc->wp_config->flags |= CONFIG_FAST_FLAG;
break;
case GST_WAVPACK_ENC_MODE_DEFAULT:
break;
case GST_WAVPACK_ENC_MODE_HIGH:
enc->wp_config->flags |= CONFIG_HIGH_FLAG;
break;
#ifndef WAVPACK_OLD_API
case GST_WAVPACK_ENC_MODE_VERY_HIGH:
enc->wp_config->flags |= CONFIG_HIGH_FLAG;
enc->wp_config->flags |= CONFIG_VERY_HIGH_FLAG;
break;
#endif
}
/* Bitrate, enables lossy mode */
if (enc->bitrate) {
enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
enc->wp_config->bitrate = enc->bitrate / 1000.0;
} else if (enc->bps) {
enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
enc->wp_config->bitrate = enc->bps;
}
/* Correction Mode, only in lossy mode */
if (enc->wp_config->flags & CONFIG_HYBRID_FLAG) {
if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
enc->wvcsrcpad =
gst_pad_new_from_static_template (&wvcsrc_factory, "wvcsrc");
/* try to add correction src pad, don't set correction mode on failure */
GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction",
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
GST_DEBUG_OBJECT (enc, "Adding correction pad with caps %"
GST_PTR_FORMAT, caps);
if (!gst_pad_set_caps (enc->wvcsrcpad, caps)) {
enc->correction_mode = 0;
GST_WARNING_OBJECT (enc, "setting correction caps failed");
} else {
gst_pad_use_fixed_caps (enc->wvcsrcpad);
gst_pad_set_active (enc->wvcsrcpad, TRUE);
gst_element_add_pad (GST_ELEMENT (enc), enc->wvcsrcpad);
enc->wp_config->flags |= CONFIG_CREATE_WVC;
if (enc->correction_mode == GST_WAVPACK_CORRECTION_MODE_OPTIMIZED) {
enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC;
}
}
gst_caps_unref (caps);
}
} else {
if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
enc->correction_mode = 0;
GST_WARNING_OBJECT (enc, "setting correction mode only has "
"any effect if a bitrate is provided.");
}
}
gst_element_no_more_pads (GST_ELEMENT (enc));
/* MD5, setup MD5 context */
if ((enc->md5) && !(enc->md5_context)) {
enc->wp_config->flags |= CONFIG_MD5_CHECKSUM;
enc->md5_context = g_new0 (MD5_CTX, 1);
MD5Init (enc->md5_context);
}
/* Extra encode processing */
if (enc->extra_processing) {
enc->wp_config->flags |= CONFIG_EXTRA_MODE;
enc->wp_config->xmode = enc->extra_processing;
}
/* Joint stereo mode */
switch (enc->joint_stereo_mode) {
case GST_WAVPACK_JS_MODE_AUTO:
break;
case GST_WAVPACK_JS_MODE_LEFT_RIGHT:
enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE;
enc->wp_config->flags &= ~CONFIG_JOINT_STEREO;
break;
case GST_WAVPACK_JS_MODE_MID_SIDE:
enc->wp_config->flags |= (CONFIG_JOINT_OVERRIDE | CONFIG_JOINT_STEREO);
break;
}
}
static int
gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
{
GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id;
GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc);
GstFlowReturn *flow;
GstBuffer *buffer;
GstPad *pad;
guchar *block = (guchar *) data;
pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
flow =
(wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
srcpad_last_return;
*flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE,
count, GST_PAD_CAPS (pad), &buffer);
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
return FALSE;
}
g_memmove (GST_BUFFER_DATA (buffer), block, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
WavpackHeader wph;
GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata",
count, (wid->correction) ? "correction " : "");
gst_wavpack_read_header (&wph, block);
/* if it's the first wavpack block, send a NEW_SEGMENT event */
if (wph.block_index == 0) {
gst_pad_push_event (pad,
gst_event_new_new_segment (FALSE,
1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0));
/* save header for later reference, so we can re-send it later on
* EOS with fixed up values for total sample count etc. */
if (enc->first_block == NULL && !wid->correction) {
enc->first_block = g_memdup (block, count);
enc->first_block_size = count;
}
}
/* set buffer timestamp, duration, offset, offset_end from
* the wavpack header */
GST_BUFFER_TIMESTAMP (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
enc->samplerate);
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
enc->samplerate);
GST_BUFFER_OFFSET (buffer) = wph.block_index;
GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
} else {
/* if it's something else set no timestamp and duration on the buffer */
GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
}
/* push the buffer and forward errors */
*flow = gst_pad_push (pad, buffer);
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
return FALSE;
}
return TRUE;
}
static GstFlowReturn
gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4;
GstFlowReturn ret;
/* reset the last returns to GST_FLOW_OK. This is only set to something else
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
* so not valid anymore */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
GST_DEBUG ("got %u raw samples", sample_count);
/* check if we already have a valid WavpackContext, otherwise make one */
if (!enc->wp_context) {
/* create raw context */
enc->wp_context =
WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
(enc->correction_mode > 0) ? &enc->wvc_id : NULL);
if (!enc->wp_context) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("error creating Wavpack context"));
gst_object_unref (enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
/* set the WavpackConfig according to our parameters */
gst_wavpack_enc_set_wp_config (enc);
/* set the configuration to the context now that we know everything
* and initialize the encoder */
if (!WavpackSetConfiguration (enc->wp_context,
enc->wp_config, (uint32_t) (-1))
|| !WavpackPackInit (enc->wp_context)) {
GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
("error setting up wavpack encoding context"));
WavpackCloseFile (enc->wp_context);
gst_object_unref (enc);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
GST_DEBUG ("setup of encoding context successfull");
}
/* if we want to append the MD5 sum to the stream update it here
* with the current raw samples */
if (enc->md5) {
MD5Update (enc->md5_context, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
}
/* encode and handle return values from encoding */
if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf),
sample_count / enc->channels)) {
GST_DEBUG ("encoding samples successful");
ret = GST_FLOW_OK;
} else {
if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
(enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) {
ret = GST_FLOW_RESEND;
} else if ((enc->srcpad_last_return == GST_FLOW_OK) ||
(enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
ret = GST_FLOW_OK;
} else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
(enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
ret = GST_FLOW_NOT_LINKED;
} else if ((enc->srcpad_last_return == GST_FLOW_WRONG_STATE) &&
(enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) {
ret = GST_FLOW_WRONG_STATE;
} else {
GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
("encoding samples failed"));
ret = GST_FLOW_ERROR;
}
}
gst_buffer_unref (buf);
gst_object_unref (enc);
return ret;
}
static void
gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
{
GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES,
0, GST_BUFFER_OFFSET_NONE, 0);
gboolean ret;
g_return_if_fail (enc);
g_return_if_fail (enc->first_block);
/* update the sample count in the first block */
WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
/* try to seek to the beginning of the output */
ret = gst_pad_push_event (enc->srcpad, event);
if (ret) {
/* try to rewrite the first block */
GST_DEBUG_OBJECT (enc, "rewriting first block ...");
ret = gst_wavpack_enc_push_block (&enc->wv_id,
enc->first_block, enc->first_block_size);
} else {
GST_WARNING_OBJECT (enc, "rewriting of first block failed. "
"Seeking to first block failed!");
}
}
static gboolean
gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
gboolean ret = TRUE;
GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* Encode all remaining samples and flush them to the src pads */
WavpackFlushSamples (enc->wp_context);
/* write the MD5 sum if we have to write one */
if ((enc->md5) && (enc->md5_context)) {
guchar md5_digest[16];
MD5Final (md5_digest, enc->md5_context);
WavpackStoreMD5Sum (enc->wp_context, md5_digest);
}
/* Try to rewrite the first frame with the correct sample number */
if (enc->first_block)
gst_wavpack_enc_rewrite_first_block (enc);
/* close the context if not already happened */
if (enc->wp_context) {
WavpackCloseFile (enc->wp_context);
enc->wp_context = NULL;
}
ret = gst_pad_event_default (pad, event);
break;
case GST_EVENT_NEWSEGMENT:
if (enc->wp_context) {
GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
"already started");
}
/* drop NEWSEGMENT events, we create our own when pushing
* the first buffer to the pads */
gst_event_unref (event);
ret = TRUE;
break;
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (enc);
return ret;
}
static GstStateChangeReturn
gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstWavpackEnc *enc = GST_WAVPACK_ENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
/* set the last returned GstFlowReturns of the two pads to GST_FLOW_OK
* as they're only set to something else in WavpackPackSamples() or more
* specific gst_wavpack_enc_push_block() and nothing happened there yet */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_wavpack_enc_reset (enc);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static void
gst_wavpack_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
switch (prop_id) {
case ARG_MODE:
enc->mode = g_value_get_enum (value);
break;
case ARG_BITRATE:{
guint val = g_value_get_uint (value);
if ((val >= 24000) && (val <= 9600000)) {
enc->bitrate = val;
enc->bps = 0.0;
} else {
enc->bitrate = 0;
enc->bps = 0.0;
}
break;
}
case ARG_BITSPERSAMPLE:{
gdouble val = g_value_get_double (value);
if ((val >= 2.0) && (val <= 24.0)) {
enc->bps = val;
enc->bitrate = 0;
} else {
enc->bps = 0.0;
enc->bitrate = 0;
}
break;
}
case ARG_CORRECTION_MODE:
enc->correction_mode = g_value_get_enum (value);
break;
case ARG_MD5:
enc->md5 = g_value_get_boolean (value);
break;
case ARG_EXTRA_PROCESSING:
enc->extra_processing = g_value_get_uint (value);
break;
case ARG_JOINT_STEREO_MODE:
enc->joint_stereo_mode = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_wavpack_enc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
switch (prop_id) {
case ARG_MODE:
g_value_set_enum (value, enc->mode);
break;
case ARG_BITRATE:
if (enc->bps == 0.0) {
g_value_set_uint (value, enc->bitrate);
} else {
g_value_set_uint (value, 0);
}
break;
case ARG_BITSPERSAMPLE:
if (enc->bitrate == 0) {
g_value_set_double (value, enc->bps);
} else {
g_value_set_double (value, 0.0);
}
break;
case ARG_CORRECTION_MODE:
g_value_set_enum (value, enc->correction_mode);
break;
case ARG_MD5:
g_value_set_boolean (value, enc->md5);
break;
case ARG_EXTRA_PROCESSING:
g_value_set_uint (value, enc->extra_processing);
break;
case ARG_JOINT_STEREO_MODE:
g_value_set_enum (value, enc->joint_stereo_mode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_wavpack_enc_plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "wavpackenc",
GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
return FALSE;
GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpack_enc", 0,
"Wavpack encoder");
return TRUE;
}

95
ext/wavpack/gstwavpackenc.h
View file

@ -1,95 +0,0 @@
/* GStreamer Wavpack encoder plugin
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackenc.h: Wavpack audio encoder
*
* 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.
*/
#ifndef __GST_WAVPACK_ENC_H__
#define __GST_WAVPACK_ENC_H__
#include <gst/gst.h>
#include <wavpack/wavpack.h>
#include "md5.h"
G_BEGIN_DECLS
#define GST_TYPE_WAVPACK_ENC \
(gst_wavpack_enc_get_type())
#define GST_WAVPACK_ENC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_ENC,GstWavpackEnc))
#define GST_WAVPACK_ENC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_ENC,GstWavpackEnc))
#define GST_IS_WAVPACK_ENC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_ENC))
#define GST_IS_WAVPACK_ENC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_ENC))
typedef struct _GstWavpackEnc GstWavpackEnc;
typedef struct _GstWavpackEncClass GstWavpackEncClass;
typedef struct
{
gboolean correction;
GstWavpackEnc *wavpack_enc;
} GstWavpackEncWriteID;
struct _GstWavpackEnc
{
GstElement element;
/*< private > */
GstPad *sinkpad, *srcpad;
GstPad *wvcsrcpad;
GstFlowReturn srcpad_last_return;
GstFlowReturn wvcsrcpad_last_return;
WavpackConfig *wp_config;
WavpackContext *wp_context;
gint samplerate;
gint channels;
gint depth;
GstWavpackEncWriteID wv_id;
GstWavpackEncWriteID wvc_id;
guint mode;
guint bitrate;
gdouble bps;
guint correction_mode;
gboolean md5;
MD5_CTX *md5_context;
guint extra_processing;
guint joint_stereo_mode;
void *first_block;
int32_t first_block_size;
};
struct _GstWavpackEncClass
{
GstElementClass parent;
};
GType gst_wavpack_enc_get_type (void);
gboolean gst_wavpack_enc_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_WAVPACK_ENC_H__ */

1221
ext/wavpack/gstwavpackparse.c
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93
ext/wavpack/gstwavpackparse.h
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@ -1,93 +0,0 @@
/* GStreamer wavpack plugin
* (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
*
* gstwavpackparse.h: wavpack file parser
*
* 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.
*/
#ifndef __GST_WAVPACK_PARSE_H__
#define __GST_WAVPACK_PARSE_H__
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
#define GST_TYPE_WAVPACK_PARSE \
(gst_wavpack_parse_get_type())
#define GST_WAVPACK_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_PARSE,GstWavpackParse))
#define GST_WAVPACK_PARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_PARSE,GstWavpackParseClass))
#define GST_IS_WAVPACK_PARSE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_PARSE))
#define GST_IS_WAVPACK_PARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_PARSE))
typedef struct _GstWavpackParse GstWavpackParse;
typedef struct _GstWavpackParseClass GstWavpackParseClass;
typedef struct _GstWavpackParseIndexEntry GstWavpackParseIndexEntry;
struct _GstWavpackParseIndexEntry
{
gint64 byte_offset; /* byte offset of this chunk */
gint64 sample_offset; /* first sample in this chunk */
gint64 sample_offset_end; /* first sample in next chunk */
};
struct _GstWavpackParse
{
GstElement element;
/*< private > */
GstPad *sinkpad;
GstPad *srcpad;
guint samplerate;
guint channels;
guint total_samples;
gboolean need_newsegment;
gboolean discont;
gint64 current_offset; /* byte offset on sink pad */
gint64 upstream_length; /* length of file in bytes */
GstSegment segment; /* the currently configured segment, in
* samples/audio frames (DEFAULT format) */
GstAdapter *adapter; /* when operating chain-based, otherwise NULL */
/* Array of GstWavpackParseIndexEntry structs, mapping known
* sample offsets to byte offsets. Is kept increasing without
* gaps (ie. append only and consecutive entries must always
* map to consecutive chunks in the file). */
GArray *entries;
/* Queued events (e.g. tag events we receive before we create the src pad) */
GList *queued_events; /* STREAM_LOCK */
};
struct _GstWavpackParseClass
{
GstElementClass parent;
};
GType gst_wavpack_parse_get_type (void);
gboolean gst_wavpack_parse_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_WAVPACK_PARSE_H__ */

112
ext/wavpack/gstwavpackstreamreader.c
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@ -1,112 +0,0 @@
/* GStreamer Wavpack plugin
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackstreamreader.c: stream reader used for decoding
*
* 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.
*/
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include "gstwavpackstreamreader.h"
GST_DEBUG_CATEGORY_EXTERN (wavpack_debug);
#define GST_CAT_DEFAULT wavpack_debug
static int32_t
gst_wavpack_stream_reader_read_bytes (void *id, void *data, int32_t bcount)
{
read_id *rid = (read_id *) id;
uint32_t left = rid->length - rid->position;
uint32_t to_read = MIN (left, bcount);
if (to_read > 0) {
g_memmove (data, rid->buffer + rid->position, to_read);
rid->position += to_read;
return to_read;
} else {
return 0;
}
}
static uint32_t
gst_wavpack_stream_reader_get_pos (void *id)
{
return ((read_id *) id)->position;
}
static int
gst_wavpack_stream_reader_set_pos_abs (void *id, uint32_t pos)
{
GST_DEBUG ("should not be called");
return -1;
}
static int
gst_wavpack_stream_reader_set_pos_rel (void *id, int32_t delta, int mode)
{
GST_DEBUG ("should not be called");
return -1;
}
static int
gst_wavpack_stream_reader_push_back_byte (void *id, int c)
{
read_id *rid = (read_id *) id;
rid->position -= 1;
if (rid->position < 0)
rid->position = 0;
return rid->position;
}
static uint32_t
gst_wavpack_stream_reader_get_length (void *id)
{
return ((read_id *) id)->length;
}
static int
gst_wavpack_stream_reader_can_seek (void *id)
{
return FALSE;
}
static int32_t
gst_wavpack_stream_reader_write_bytes (void *id, void *data, int32_t bcount)
{
GST_DEBUG ("should not be called");
return 0;
}
WavpackStreamReader *
gst_wavpack_stream_reader_new ()
{
WavpackStreamReader *stream_reader =
(WavpackStreamReader *) g_malloc0 (sizeof (WavpackStreamReader));
stream_reader->read_bytes = gst_wavpack_stream_reader_read_bytes;
stream_reader->get_pos = gst_wavpack_stream_reader_get_pos;
stream_reader->set_pos_abs = gst_wavpack_stream_reader_set_pos_abs;
stream_reader->set_pos_rel = gst_wavpack_stream_reader_set_pos_rel;
stream_reader->push_back_byte = gst_wavpack_stream_reader_push_back_byte;
stream_reader->get_length = gst_wavpack_stream_reader_get_length;
stream_reader->can_seek = gst_wavpack_stream_reader_can_seek;
stream_reader->write_bytes = gst_wavpack_stream_reader_write_bytes;
return stream_reader;
}

36
ext/wavpack/gstwavpackstreamreader.h
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@ -1,36 +0,0 @@
/* GStreamer Wavpack plugin
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackstreamreader.h: stream reader used for decoding
*
* 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.
*/
#ifndef __GST_WAVPACK_STREAM_READER_H__
#define __GST_WAVPACK_STREAM_READER_H__
#include <wavpack/wavpack.h>
typedef struct
{
guint8 *buffer;
uint32_t length;
uint32_t position;
} read_id;
WavpackStreamReader *gst_wavpack_stream_reader_new ();
#endif

271
ext/wavpack/md5.c
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@ -1,271 +0,0 @@
/*
* This code implements the MD5 message-digest algorithm.
* The algorithm is due to Ron Rivest. This code was
* written by Colin Plumb in 1993, no copyright is claimed.
* This code is in the public domain; do with it what you wish.
*
* Equivalent code is available from RSA Data Security, Inc.
* This code has been tested against that, and is equivalent,
* except that you don't need to include two pages of legalese
* with every copy.
*
* To compute the message digest of a chunk of bytes, declare an
* MD5Context structure, pass it to MD5Init, call MD5Update as
* needed on buffers full of bytes, and then call MD5Final, which
* will fill a supplied 16-byte array with the digest.
*/
/* Brutally hacked by John Walker back from ANSI C to K&R (no
prototypes) to maintain the tradition that Netfone will compile
with Sun's original "cc". */
#include <string.h> /* for memcpy() */
#include <glib.h>
#include "md5.h"
#if G_BYTE_ORDER == G_BIG_ENDIAN
#define HIGHFIRST
#endif
#ifndef HIGHFIRST
#define byteReverse(buf, len) /* Nothing */
#else
/*
* Note: this code is harmless on little-endian machines.
*/
void
byteReverse (buf, longs)
unsigned char *buf;
unsigned longs;
{
uint32 t;
do {
t = (uint32) ((unsigned) buf[3] << 8 | buf[2]) << 16 |
((unsigned) buf[1] << 8 | buf[0]);
*(uint32 *) buf = t;
buf += 4;
} while (--longs);
}
#endif
/*
* Start MD5 accumulation. Set bit count to 0 and buffer to mysterious
* initialization constants.
*/
void
MD5Init (ctx)
struct MD5Context *ctx;
{
ctx->buf[0] = 0x67452301;
ctx->buf[1] = 0xefcdab89;
ctx->buf[2] = 0x98badcfe;
ctx->buf[3] = 0x10325476;
ctx->bits[0] = 0;
ctx->bits[1] = 0;
}
/*
* Update context to reflect the concatenation of another buffer full
* of bytes.
*/
void
MD5Update (ctx, buf, len)
struct MD5Context *ctx;
unsigned char *buf;
unsigned len;
{
uint32 t;
/* Update bitcount */
t = ctx->bits[0];
if ((ctx->bits[0] = t + ((uint32) len << 3)) < t)
ctx->bits[1]++; /* Carry from low to high */
ctx->bits[1] += len >> 29;
t = (t >> 3) & 0x3f; /* Bytes already in shsInfo->data */
/* Handle any leading odd-sized chunks */
if (t) {
unsigned char *p = (unsigned char *) ctx->in + t;
t = 64 - t;
if (len < t) {
memcpy (p, buf, len);
return;
}
memcpy (p, buf, t);
byteReverse (ctx->in, 16);
MD5Transform (ctx->buf, (uint32 *) ctx->in);
buf += t;
len -= t;
}
/* Process data in 64-byte chunks */
while (len >= 64) {
memcpy (ctx->in, buf, 64);
byteReverse (ctx->in, 16);
MD5Transform (ctx->buf, (uint32 *) ctx->in);
buf += 64;
len -= 64;
}
/* Handle any remaining bytes of data. */
memcpy (ctx->in, buf, len);
}
/*
* Final wrapup - pad to 64-byte boundary with the bit pattern
* 1 0* (64-bit count of bits processed, MSB-first)
*/
void
MD5Final (digest, ctx)
unsigned char digest[16];
struct MD5Context *ctx;
{
unsigned count;
unsigned char *p;
/* Compute number of bytes mod 64 */
count = (ctx->bits[0] >> 3) & 0x3F;
/* Set the first char of padding to 0x80. This is safe since there is
always at least one byte free */
p = ctx->in + count;
*p++ = 0x80;
/* Bytes of padding needed to make 64 bytes */
count = 64 - 1 - count;
/* Pad out to 56 mod 64 */
if (count < 8) {
/* Two lots of padding: Pad the first block to 64 bytes */
memset (p, 0, count);
byteReverse (ctx->in, 16);
MD5Transform (ctx->buf, (uint32 *) ctx->in);
/* Now fill the next block with 56 bytes */
memset (ctx->in, 0, 56);
} else {
/* Pad block to 56 bytes */
memset (p, 0, count - 8);
}
byteReverse (ctx->in, 14);
/* Append length in bits and transform */
((uint32 *) ctx->in)[14] = ctx->bits[0];
((uint32 *) ctx->in)[15] = ctx->bits[1];
MD5Transform (ctx->buf, (uint32 *) ctx->in);
byteReverse ((unsigned char *) ctx->buf, 4);
memcpy (digest, ctx->buf, 16);
memset (ctx, 0, sizeof (ctx)); /* In case it's sensitive */
}
/* The four core functions - F1 is optimized somewhat */
/* #define F1(x, y, z) (x & y | ~x & z) */
#define F1(x, y, z) (z ^ (x & (y ^ z)))
#define F2(x, y, z) F1(z, x, y)
#define F3(x, y, z) (x ^ y ^ z)
#define F4(x, y, z) (y ^ (x | ~z))
/* This is the central step in the MD5 algorithm. */
#define MD5STEP(f, w, x, y, z, data, s) \
( w += f(x, y, z) + data, w = w<<s | w>>(32-s), w += x )
/*
* The core of the MD5 algorithm, this alters an existing MD5 hash to
* reflect the addition of 16 longwords of new data. MD5Update blocks
* the data and converts bytes into longwords for this routine.
*/
void
MD5Transform (buf, in)
uint32 buf[4];
uint32 in[16];
{
register uint32 a, b, c, d;
a = buf[0];
b = buf[1];
c = buf[2];
d = buf[3];
MD5STEP (F1, a, b, c, d, in[0] + 0xd76aa478, 7);
MD5STEP (F1, d, a, b, c, in[1] + 0xe8c7b756, 12);
MD5STEP (F1, c, d, a, b, in[2] + 0x242070db, 17);
MD5STEP (F1, b, c, d, a, in[3] + 0xc1bdceee, 22);
MD5STEP (F1, a, b, c, d, in[4] + 0xf57c0faf, 7);
MD5STEP (F1, d, a, b, c, in[5] + 0x4787c62a, 12);
MD5STEP (F1, c, d, a, b, in[6] + 0xa8304613, 17);
MD5STEP (F1, b, c, d, a, in[7] + 0xfd469501, 22);
MD5STEP (F1, a, b, c, d, in[8] + 0x698098d8, 7);
MD5STEP (F1, d, a, b, c, in[9] + 0x8b44f7af, 12);
MD5STEP (F1, c, d, a, b, in[10] + 0xffff5bb1, 17);
MD5STEP (F1, b, c, d, a, in[11] + 0x895cd7be, 22);
MD5STEP (F1, a, b, c, d, in[12] + 0x6b901122, 7);
MD5STEP (F1, d, a, b, c, in[13] + 0xfd987193, 12);
MD5STEP (F1, c, d, a, b, in[14] + 0xa679438e, 17);
MD5STEP (F1, b, c, d, a, in[15] + 0x49b40821, 22);
MD5STEP (F2, a, b, c, d, in[1] + 0xf61e2562, 5);
MD5STEP (F2, d, a, b, c, in[6] + 0xc040b340, 9);
MD5STEP (F2, c, d, a, b, in[11] + 0x265e5a51, 14);
MD5STEP (F2, b, c, d, a, in[0] + 0xe9b6c7aa, 20);
MD5STEP (F2, a, b, c, d, in[5] + 0xd62f105d, 5);
MD5STEP (F2, d, a, b, c, in[10] + 0x02441453, 9);
MD5STEP (F2, c, d, a, b, in[15] + 0xd8a1e681, 14);
MD5STEP (F2, b, c, d, a, in[4] + 0xe7d3fbc8, 20);
MD5STEP (F2, a, b, c, d, in[9] + 0x21e1cde6, 5);
MD5STEP (F2, d, a, b, c, in[14] + 0xc33707d6, 9);
MD5STEP (F2, c, d, a, b, in[3] + 0xf4d50d87, 14);
MD5STEP (F2, b, c, d, a, in[8] + 0x455a14ed, 20);
MD5STEP (F2, a, b, c, d, in[13] + 0xa9e3e905, 5);
MD5STEP (F2, d, a, b, c, in[2] + 0xfcefa3f8, 9);
MD5STEP (F2, c, d, a, b, in[7] + 0x676f02d9, 14);
MD5STEP (F2, b, c, d, a, in[12] + 0x8d2a4c8a, 20);
MD5STEP (F3, a, b, c, d, in[5] + 0xfffa3942, 4);
MD5STEP (F3, d, a, b, c, in[8] + 0x8771f681, 11);
MD5STEP (F3, c, d, a, b, in[11] + 0x6d9d6122, 16);
MD5STEP (F3, b, c, d, a, in[14] + 0xfde5380c, 23);
MD5STEP (F3, a, b, c, d, in[1] + 0xa4beea44, 4);
MD5STEP (F3, d, a, b, c, in[4] + 0x4bdecfa9, 11);
MD5STEP (F3, c, d, a, b, in[7] + 0xf6bb4b60, 16);
MD5STEP (F3, b, c, d, a, in[10] + 0xbebfbc70, 23);
MD5STEP (F3, a, b, c, d, in[13] + 0x289b7ec6, 4);
MD5STEP (F3, d, a, b, c, in[0] + 0xeaa127fa, 11);
MD5STEP (F3, c, d, a, b, in[3] + 0xd4ef3085, 16);
MD5STEP (F3, b, c, d, a, in[6] + 0x04881d05, 23);
MD5STEP (F3, a, b, c, d, in[9] + 0xd9d4d039, 4);
MD5STEP (F3, d, a, b, c, in[12] + 0xe6db99e5, 11);
MD5STEP (F3, c, d, a, b, in[15] + 0x1fa27cf8, 16);
MD5STEP (F3, b, c, d, a, in[2] + 0xc4ac5665, 23);
MD5STEP (F4, a, b, c, d, in[0] + 0xf4292244, 6);
MD5STEP (F4, d, a, b, c, in[7] + 0x432aff97, 10);
MD5STEP (F4, c, d, a, b, in[14] + 0xab9423a7, 15);
MD5STEP (F4, b, c, d, a, in[5] + 0xfc93a039, 21);
MD5STEP (F4, a, b, c, d, in[12] + 0x655b59c3, 6);
MD5STEP (F4, d, a, b, c, in[3] + 0x8f0ccc92, 10);
MD5STEP (F4, c, d, a, b, in[10] + 0xffeff47d, 15);
MD5STEP (F4, b, c, d, a, in[1] + 0x85845dd1, 21);
MD5STEP (F4, a, b, c, d, in[8] + 0x6fa87e4f, 6);
MD5STEP (F4, d, a, b, c, in[15] + 0xfe2ce6e0, 10);
MD5STEP (F4, c, d, a, b, in[6] + 0xa3014314, 15);
MD5STEP (F4, b, c, d, a, in[13] + 0x4e0811a1, 21);
MD5STEP (F4, a, b, c, d, in[4] + 0xf7537e82, 6);
MD5STEP (F4, d, a, b, c, in[11] + 0xbd3af235, 10);
MD5STEP (F4, c, d, a, b, in[2] + 0x2ad7d2bb, 15);
MD5STEP (F4, b, c, d, a, in[9] + 0xeb86d391, 21);
buf[0] += a;
buf[1] += b;
buf[2] += c;
buf[3] += d;
}

28
ext/wavpack/md5.h
View file

@ -1,28 +0,0 @@
#ifndef MD5_H
#define MD5_H
#include "_stdint.h"
#ifndef uint32
typedef uint32_t uint32;
#endif
struct MD5Context
{
uint32 buf[4];
uint32 bits[2];
unsigned char in[64];
};
extern void MD5Init (struct MD5Context *ctx);
extern void MD5Update (struct MD5Context *ctx, unsigned char *buf,
unsigned len);
extern void MD5Final (unsigned char digest[16], struct MD5Context *ctx);
extern void MD5Transform (uint32 buf[4], uint32 in[16]);
/*
* This is needed to make RSAREF happy on some MS-DOS compilers.
*/
typedef struct MD5Context MD5_CTX;
#endif /* !MD5_H */

10
tests/check/Makefile.am
View file

@ -38,15 +38,6 @@ else
check_neon =
endif
if USE_WAVPACK
check_wavpack = \
elements/wavpackparse \
elements/wavpackdec \
elements/wavpackenc
else
check_wavpack =
endif
VALGRIND_TO_FIX = \
elements/mpeg2enc
@ -63,7 +54,6 @@ check_PROGRAMS = \
elements/rglimiter \
elements/rgvolume \
elements/videocrop \
$(check_wavpack) \
elements/y4menc
TESTS = $(check_PROGRAMS)

254
tests/check/elements/wavpackdec.c
View file

@ -1,254 +0,0 @@
/* GStreamer
*
* unit test for wavpackdec
*
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* 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.
*/
#include <unistd.h>
#include <gst/check/gstcheck.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
guint8 test_frame[] = {
0x77, 0x76, 0x70, 0x6B, /* "wvpk" */
0x2E, 0x00, 0x00, 0x00, /* ckSize */
0x04, 0x04, /* version */
0x00, /* track_no */
0x00, /* index_no */
0x00, 0x64, 0x00, 0x00, /* total_samples */
0x00, 0x00, 0x00, 0x00, /* block_index */
0x00, 0x64, 0x00, 0x00, /* block_samples */
0x05, 0x18, 0x80, 0x04, /* flags */
0xFF, 0xAF, 0x80, 0x60, /* crc */
0x02, 0x00, 0x03, 0x00, /* data */
0x04, 0x00, 0x05, 0x03,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x8A, 0x02,
0x00, 0x00, 0xFF, 0x7F,
0x00, 0xE4,
};
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) 16, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) 16, "
"channels = (int) 1, " "rate = (int) 44100, " "framed = (boolean) true")
);
GstElement *
setup_wavpackdec ()
{
GstElement *wavpackdec;
GST_DEBUG ("setup_wavpackdec");
wavpackdec = gst_check_setup_element ("wavpackdec");
mysrcpad = gst_check_setup_src_pad (wavpackdec, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (wavpackdec, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return wavpackdec;
}
void
cleanup_wavpackdec (GstElement * wavpackdec)
{
GST_DEBUG ("cleanup_wavpackdec");
gst_element_set_state (wavpackdec, GST_STATE_NULL);
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (wavpackdec);
gst_check_teardown_sink_pad (wavpackdec);
gst_check_teardown_element (wavpackdec);
}
GST_START_TEST (test_decode_frame)
{
GstElement *wavpackdec;
GstBuffer *inbuffer, *outbuffer;
GstBus *bus;
int i;
wavpackdec = setup_wavpackdec ();
fail_unless (gst_element_set_state (wavpackdec,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
bus = gst_bus_new ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame));
memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame));
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackdec, bus);
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
/* uncompressed data should be 102400 bytes */
fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), 102400);
/* and all 102400 bytes must be 0, i.e. silence */
for (i = 0; i < 102400; i++)
fail_unless_equals_int (GST_BUFFER_DATA (outbuffer)[i], 0);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
g_list_free (buffers);
buffers = NULL;
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (wavpackdec, NULL);
gst_object_unref (GST_OBJECT (bus));
cleanup_wavpackdec (wavpackdec);
}
GST_END_TEST;
GST_START_TEST (test_decode_frame_with_broken_header)
{
GstElement *wavpackdec;
GstBuffer *inbuffer, *outbuffer;
GstBus *bus;
GstMessage *message;
wavpackdec = setup_wavpackdec ();
fail_unless (gst_element_set_state (wavpackdec,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
bus = gst_bus_new ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame));
memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame));
/* break header */
GST_BUFFER_DATA (inbuffer)[2] = 'e';
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackdec, bus);
/* should fail gracefully */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if ((message = gst_bus_pop (bus)) == NULL);
fail_unless_message_error (message, STREAM, DECODE);
gst_message_unref (message);
gst_element_set_bus (wavpackdec, NULL);
gst_object_unref (GST_OBJECT (bus));
cleanup_wavpackdec (wavpackdec);
}
GST_END_TEST;
GST_START_TEST (test_decode_frame_with_incomplete_frame)
{
GstElement *wavpackdec;
GstBuffer *inbuffer, *outbuffer;
GstBus *bus;
GstMessage *message;
wavpackdec = setup_wavpackdec ();
fail_unless (gst_element_set_state (wavpackdec,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
bus = gst_bus_new ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame) - 2);
memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame) - 2);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackdec, bus);
/* should fail gracefully */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if ((message = gst_bus_pop (bus)) == NULL);
fail_unless_message_error (message, STREAM, DECODE);
gst_message_unref (message);
gst_element_set_bus (wavpackdec, NULL);
gst_object_unref (GST_OBJECT (bus));
cleanup_wavpackdec (wavpackdec);
}
GST_END_TEST;
Suite *
wavpackdec_suite (void)
{
Suite *s = suite_create ("wavpackdec");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_decode_frame);
tcase_add_test (tc_chain, test_decode_frame_with_broken_header);
tcase_add_test (tc_chain, test_decode_frame_with_incomplete_frame);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = wavpackdec_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}

197
tests/check/elements/wavpackenc.c
View file

@ -1,197 +0,0 @@
/* GStreamer
*
* unit test for wavpackenc
*
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* 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.
*/
#include <unistd.h>
#include <gst/check/gstcheck.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
static GstBus *bus;
#define RAW_CAPS_STRING "audio/x-raw-int, " \
"width = (int) 32, " \
"depth = (int) 16, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true"
#define WAVPACK_CAPS_STRING "audio/x-wavpack, " \
"width = (int) 16, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"framed = (boolean) true"
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) 16, "
"channels = (int) 1, "
"rate = (int) 44100, " "framed = (boolean) true"));
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 32, "
"depth = (int) 16, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "signed = (boolean) true"));
GstElement *
setup_wavpackenc ()
{
GstElement *wavpackenc;
GST_DEBUG ("setup_wavpackenc");
wavpackenc = gst_check_setup_element ("wavpackenc");
mysrcpad = gst_check_setup_src_pad (wavpackenc, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (wavpackenc, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
fail_unless (gst_element_set_state (wavpackenc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
bus = gst_bus_new ();
return wavpackenc;
}
void
cleanup_wavpackenc (GstElement * wavpackenc)
{
GST_DEBUG ("cleanup_wavpackenc");
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (wavpackenc, NULL);
gst_object_unref (GST_OBJECT (bus));
gst_element_set_state (wavpackenc, GST_STATE_NULL);
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (wavpackenc);
gst_check_teardown_sink_pad (wavpackenc);
gst_check_teardown_element (wavpackenc);
}
GST_START_TEST (test_encode_silence)
{
GstElement *wavpackenc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
GstEvent *eos = gst_event_new_eos ();
int i, num_buffers;
wavpackenc = setup_wavpackenc ();
inbuffer = gst_buffer_new_and_alloc (1000);
for (i = 0; i < 1000; i++)
GST_BUFFER_DATA (inbuffer)[i] = 0;
caps = gst_caps_from_string (RAW_CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_ref (inbuffer);
gst_element_set_bus (wavpackenc, bus);
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
gst_buffer_unref (inbuffer);
fail_if (gst_pad_push_event (mysrcpad, eos) != TRUE);
/* check first buffer */
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 5668934);
fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 250);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0,
"Failed to encode to valid Wavpack frames");
caps = gst_caps_from_string (WAVPACK_CAPS_STRING);
fail_unless (gst_caps_is_equal (caps, GST_BUFFER_CAPS (outbuffer)) == TRUE,
"Wrong caps");
gst_caps_unref (caps);
/* free all buffers */
num_buffers = g_list_length (buffers);
for (i = 0; i < num_buffers; ++i) {
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
buffers = g_list_remove (buffers, outbuffer);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
}
g_list_free (buffers);
buffers = NULL;
cleanup_wavpackenc (wavpackenc);
}
GST_END_TEST;
Suite *
wavpackenc_suite (void)
{
Suite *s = suite_create ("wavpackenc");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_encode_silence);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = wavpackenc_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}

295
tests/check/elements/wavpackparse.c
View file

@ -1,295 +0,0 @@
/* GStreamer
*
* unit test for wavpackparse
*
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* 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.
*/
#include <unistd.h>
#include <gst/check/gstcheck.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
static GstBus *bus;
static GstElement *wavpackparse;
/* Wavpack file with 2 frames of silence */
guint8 test_file[] = {
0x77, 0x76, 0x70, 0x6B, 0x62, 0x00, 0x00, 0x00, /* first frame */
0x04, 0x04, 0x00, 0x00, 0x00, 0xC8, 0x00, 0x00, /* include RIFF header */
0x00, 0x00, 0x00, 0x00, 0x00, 0x64, 0x00, 0x00,
0x05, 0x18, 0x80, 0x04, 0xFF, 0xAF, 0x80, 0x60,
0x21, 0x16, 0x52, 0x49, 0x46, 0x46, 0x24, 0x90,
0x01, 0x00, 0x57, 0x41, 0x56, 0x45, 0x66, 0x6D,
0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00,
0x01, 0x00, 0x44, 0xAC, 0x00, 0x00, 0x88, 0x58,
0x01, 0x00, 0x02, 0x00, 0x10, 0x00, 0x64, 0x61,
0x74, 0x61, 0x00, 0x90, 0x01, 0x00, 0x02, 0x00,
0x03, 0x00, 0x04, 0x00, 0x05, 0x03, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x65, 0x02, 0x00, 0x00,
0x00, 0x00, 0x8A, 0x02, 0x00, 0x00, 0xFF, 0x7F,
0x00, 0xE4,
0x77, 0x76, 0x70, 0x6B, 0x2E, 0x00, 0x00, 0x00, /* second frame */
0x04, 0x04, 0x00, 0x00, 0xFF, 0xFF, 0xFF, 0xFF,
0x00, 0x64, 0x00, 0x00, 0x00, 0x64, 0x00, 0x00,
0x05, 0x18, 0x80, 0x04, 0xFF, 0xAF, 0x80, 0x60,
0x02, 0x00, 0x03, 0x00, 0x04, 0x00, 0x05, 0x03,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x8A, 0x02,
0x00, 0x00, 0xFF, 0x7F, 0x00, 0xE4,
};
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) 16, "
"channels = (int) 1, "
"rate = (int) 44100, " "framed = (boolean) TRUE"));
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack"));
static void
wavpackparse_found_pad (GstElement * src, GstPad * pad, gpointer data)
{
GstPad *srcpad;
mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
fail_if (mysinkpad == NULL, "Couldn't create sinkpad");
srcpad = gst_element_get_pad (wavpackparse, "src");
fail_if (srcpad == NULL, "Failed to get srcpad from wavpackparse");
gst_pad_set_chain_function (mysinkpad, gst_check_chain_func);
fail_unless (gst_pad_link (srcpad, mysinkpad) == GST_PAD_LINK_OK,
"Failed to link pads");
gst_pad_set_active (mysinkpad, TRUE);
gst_object_unref (srcpad);
}
void
setup_wavpackparse ()
{
GstPad *sinkpad;
GST_DEBUG ("setup_wavpackparse");
wavpackparse = gst_element_factory_make ("wavpackparse", "wavpackparse");
fail_if (wavpackparse == NULL, "Could not create wavpackparse");
mysrcpad = gst_pad_new_from_static_template (&srctemplate, "src");
fail_if (mysrcpad == NULL, "Could not create srcpad");
sinkpad = gst_element_get_pad (wavpackparse, "sink");
fail_if (sinkpad == NULL, "Failed to get sinkpad from wavpackparse");
fail_unless (gst_pad_link (mysrcpad, sinkpad) == GST_PAD_LINK_OK,
"Failed to link pads");
gst_object_unref (sinkpad);
g_signal_connect (wavpackparse, "pad-added",
G_CALLBACK (wavpackparse_found_pad), NULL);
bus = gst_bus_new ();
gst_element_set_bus (wavpackparse, bus);
fail_unless (gst_element_set_state (wavpackparse,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
}
void
cleanup_wavpackparse ()
{
GstPad *sinkpad, *srcpad;
GST_DEBUG ("cleanup_wavpackparse");
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (wavpackparse, NULL);
gst_object_unref (GST_OBJECT (bus));
sinkpad = gst_element_get_pad (wavpackparse, "sink");
fail_if (sinkpad == NULL, "Failed to get sinkpad from wavpackparse");
fail_unless (gst_pad_unlink (mysrcpad, sinkpad), "Failed to unlink pads");
gst_pad_set_caps (mysrcpad, NULL);
gst_object_unref (sinkpad);
gst_object_unref (mysrcpad);
srcpad = gst_element_get_pad (wavpackparse, "src");
fail_if (srcpad == NULL, "Failed to get srcpad from wavpackparse");
fail_unless (gst_pad_unlink (srcpad, mysinkpad), "Failed to unlink pads");
gst_pad_set_caps (mysinkpad, NULL);
gst_object_unref (srcpad);
gst_object_unref (mysinkpad);
fail_unless (gst_element_set_state (wavpackparse, GST_STATE_NULL) ==
GST_STATE_CHANGE_SUCCESS, "could not set to null");
gst_object_unref (wavpackparse);
}
GST_START_TEST (test_parsing_valid_frames)
{
GstBuffer *inbuffer, *outbuffer;
int i, num_buffers;
GstFormat format = GST_FORMAT_DEFAULT;
gint64 pos;
setup_wavpackparse ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_file));
memcpy (GST_BUFFER_DATA (inbuffer), test_file, sizeof (test_file));
gst_buffer_ref (inbuffer);
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
gst_buffer_unref (inbuffer);
num_buffers = g_list_length (buffers);
/* should get 2 buffers, each one complete wavpack frame */
fail_unless_equals_int (num_buffers, 2);
for (i = 0; i < num_buffers; ++i) {
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0,
"Buffer contains no Wavpack frame");
fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 580498866);
switch (i) {
case 0:{
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 0);
fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 25600);
break;
}
case 1:{
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 580498866);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 25600);
fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 51200);
break;
}
}
buffers = g_list_remove (buffers, outbuffer);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
}
fail_unless (gst_element_query_position (wavpackparse, &format, &pos),
"Position query failed");
fail_unless_equals_int (pos, 25600);
fail_unless (gst_element_query_duration (wavpackparse, &format, NULL),
"Duration query failed");
g_list_free (buffers);
buffers = NULL;
cleanup_wavpackparse ();
}
GST_END_TEST;
GST_START_TEST (test_parsing_invalid_first_header)
{
GstBuffer *inbuffer, *outbuffer;
int i, num_buffers;
GstFormat format = GST_FORMAT_DEFAULT;
gint64 pos;
setup_wavpackparse ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_file));
memcpy (GST_BUFFER_DATA (inbuffer), test_file, sizeof (test_file));
GST_BUFFER_DATA (inbuffer)[0] = 'k';
gst_buffer_ref (inbuffer);
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
gst_buffer_unref (inbuffer);
num_buffers = g_list_length (buffers);
/* should get 1 buffers, the second non-broken one */
fail_unless_equals_int (num_buffers, 1);
for (i = 0; i < num_buffers; ++i) {
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0,
"Buffer contains no Wavpack frame");
fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 580498866);
switch (i) {
case 0:{
fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 580498866);
fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 25600);
break;
}
}
buffers = g_list_remove (buffers, outbuffer);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
}
g_list_free (buffers);
buffers = NULL;
cleanup_wavpackparse ();
}
GST_END_TEST;
Suite *
wavpackparse_suite (void)
{
Suite *s = suite_create ("wavpackparse");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_parsing_valid_frames);
tcase_add_test (tc_chain, test_parsing_invalid_first_header);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = wavpackparse_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}