mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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49e29a2716
Original commit message from CVS: * ext/wavpack/gstwavpack.c: (plugin_init): * ext/wavpack/gstwavpackcommon.c: Use a general wavpack debug category for common code. * ext/wavpack/gstwavpackstreamreader.c: (gst_wavpack_stream_reader_set_pos_abs), (gst_wavpack_stream_reader_set_pos_rel), (gst_wavpack_stream_reader_write_bytes): Use the general wavpack debug category here too and add debug output to the functions that should not be called at all by the wavpack library. * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_plugin_init): * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_plugin_init): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init): Change debugging category names to conform to the conventions.
985 lines
31 KiB
C
985 lines
31 KiB
C
/* GStreamer Wavpack encoder plugin
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* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
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*
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* gstwavpackdec.c: Wavpack audio encoder
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-wavpackenc
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*
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* <refsect2>
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* WavpackEnc encodes raw audio into a framed Wavpack stream.
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* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
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* audio codec that features both lossless and lossy encoding.
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* gst-launch audiotestsrc num-buffers=500 ! wavpackenc ! filesink location=sinewave.wv
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* </programlisting>
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* This pipeline encodes audio from audiotestsrc into a Wavpack file.
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* </para>
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* <para>
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* <programlisting>
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* gst-launch cdda://1 ! wavpackenc ! filesink location=track1.wv
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* </programlisting>
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* This pipeline encodes audio from an audio CD into a Wavpack file using
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* lossless encoding (the file output will be fairly large).
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* </para>
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* <para>
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* <programlisting>
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* gst-launch cdda://1 ! wavpackenc bitrate=128000 ! filesink location=track1.wv
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* </programlisting>
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* This pipeline encodes audio from an audio CD into a Wavpack file using
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* lossy encoding at a certain bitrate (the file will be fairly small).
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* </para>
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* </refsect2>
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*/
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/*
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* TODO: - add multichannel handling. channel_mask is:
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* front left
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* front right
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* center
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* LFE
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* back left
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* back right
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* front left center
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* front right center
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* back left
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* back center
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* side left
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* side right
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* ...
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* - add 32 bit float mode. CONFIG_FLOAT_DATA
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*/
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#include <string.h>
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#include <gst/gst.h>
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#include <glib/gprintf.h>
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#include <wavpack/wavpack.h>
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#include "gstwavpackenc.h"
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#include "gstwavpackcommon.h"
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#include "md5.h"
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static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps);
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static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count);
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static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event);
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static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_wavpack_enc_finalize (GObject * object);
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static void gst_wavpack_enc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_wavpack_enc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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enum
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{
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ARG_0,
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ARG_MODE,
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ARG_BITRATE,
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ARG_BITSPERSAMPLE,
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ARG_CORRECTION_MODE,
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ARG_MD5,
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ARG_EXTRA_PROCESSING,
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ARG_JOINT_STEREO_MODE
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};
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GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug);
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#define GST_CAT_DEFAULT gst_wavpack_enc_debug
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"width = (int) 32, "
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"depth = (int) 32, "
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"endianness = (int) LITTLE_ENDIAN, "
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"channels = (int) [ 1, 2 ], "
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"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
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"audio/x-raw-int, "
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"width = (int) 24, "
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"depth = (int) 24, "
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"endianness = (int) LITTLE_ENDIAN, "
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"channels = (int) [ 1, 2 ], "
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"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
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"audio/x-raw-int, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"endianness = (int) LITTLE_ENDIAN, "
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"channels = (int) [ 1, 2 ], "
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"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
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"audio/x-raw-int, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"endianness = (int) LITTLE_ENDIAN, "
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"channels = (int) [ 1, 2 ], "
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"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
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);
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static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-wavpack, "
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"width = (int) { 8, 16, 24, 32 }, "
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"channels = (int) [ 1, 2 ], "
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"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
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);
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static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) TRUE")
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);
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enum
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{
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GST_WAVPACK_ENC_MODE_VERY_FAST = 0,
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GST_WAVPACK_ENC_MODE_FAST,
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GST_WAVPACK_ENC_MODE_DEFAULT,
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GST_WAVPACK_ENC_MODE_HIGH,
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GST_WAVPACK_ENC_MODE_VERY_HIGH
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};
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#define GST_TYPE_WAVPACK_ENC_MODE (gst_wavpack_enc_mode_get_type ())
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static GType
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gst_wavpack_enc_mode_get_type (void)
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{
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static GType qtype = 0;
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if (qtype == 0) {
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static const GEnumValue values[] = {
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#if 0
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/* Very Fast Compression is not supported yet, but will be supported
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* in future wavpack versions */
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{GST_WAVPACK_ENC_MODE_VERY_FAST, "Very Fast Compression", "veryfast"},
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#endif
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{GST_WAVPACK_ENC_MODE_FAST, "Fast Compression", "fast"},
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{GST_WAVPACK_ENC_MODE_DEFAULT, "Normal Compression", "normal"},
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{GST_WAVPACK_ENC_MODE_HIGH, "High Compression", "high"},
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#ifndef WAVPACK_OLD_API
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{GST_WAVPACK_ENC_MODE_VERY_HIGH, "Very High Compression", "veryhigh"},
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#endif
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{0, NULL, NULL}
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};
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qtype = g_enum_register_static ("GstWavpackEncMode", values);
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}
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return qtype;
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}
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enum
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{
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GST_WAVPACK_CORRECTION_MODE_OFF = 0,
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GST_WAVPACK_CORRECTION_MODE_ON,
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GST_WAVPACK_CORRECTION_MODE_OPTIMIZED
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};
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#define GST_TYPE_WAVPACK_ENC_CORRECTION_MODE (gst_wavpack_enc_correction_mode_get_type ())
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static GType
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gst_wavpack_enc_correction_mode_get_type (void)
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{
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static GType qtype = 0;
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if (qtype == 0) {
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static const GEnumValue values[] = {
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{GST_WAVPACK_CORRECTION_MODE_OFF, "Create no correction file", "off"},
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{GST_WAVPACK_CORRECTION_MODE_ON, "Create correction file", "on"},
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{GST_WAVPACK_CORRECTION_MODE_OPTIMIZED,
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"Create optimized correction file", "optimized"},
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{0, NULL, NULL}
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};
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qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values);
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}
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return qtype;
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}
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enum
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{
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GST_WAVPACK_JS_MODE_AUTO = 0,
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GST_WAVPACK_JS_MODE_LEFT_RIGHT,
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GST_WAVPACK_JS_MODE_MID_SIDE
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};
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#define GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE (gst_wavpack_enc_joint_stereo_mode_get_type ())
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static GType
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gst_wavpack_enc_joint_stereo_mode_get_type (void)
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{
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static GType qtype = 0;
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if (qtype == 0) {
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static const GEnumValue values[] = {
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{GST_WAVPACK_JS_MODE_AUTO, "auto", "auto"},
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{GST_WAVPACK_JS_MODE_LEFT_RIGHT, "left/right", "leftright"},
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{GST_WAVPACK_JS_MODE_MID_SIDE, "mid/side", "midside"},
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{0, NULL, NULL}
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};
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qtype = g_enum_register_static ("GstWavpackEncJSMode", values);
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}
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return qtype;
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}
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GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstElement, GST_TYPE_ELEMENT);
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static void
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gst_wavpack_enc_base_init (gpointer klass)
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{
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static const GstElementDetails element_details = {
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"Wavpack audio encoder",
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"Codec/Encoder/Audio",
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"Encodes audio with the Wavpack lossless/lossy audio codec",
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"Sebastian Dröge <slomo@circular-chaos.org>"
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};
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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/* add pad templates */
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&wvcsrc_factory));
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/* set element details */
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gst_element_class_set_details (element_class, &element_details);
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}
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static void
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gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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/* set state change handler */
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_wavpack_enc_finalize);
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/* set property handlers */
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_property);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_wavpack_enc_get_property);
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/* install all properties */
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g_object_class_install_property (gobject_class, ARG_MODE,
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g_param_spec_enum ("mode", "Encoding mode",
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"Speed versus compression tradeoff.",
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GST_TYPE_WAVPACK_ENC_MODE, GST_WAVPACK_ENC_MODE_DEFAULT,
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G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_BITRATE,
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g_param_spec_double ("bitrate", "Bitrate",
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"Try to encode with this average bitrate (bits/sec). "
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"This enables lossy encoding! A value smaller than 24000.0 disables this.",
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0.0, 9600000.0, 0.0, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_BITSPERSAMPLE,
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g_param_spec_double ("bits-per-sample", "Bits per sample",
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"Try to encode with this amount of bits per sample. "
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"This enables lossy encoding! A value smaller than 2.0 disables this.",
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0.0, 24.0, 0.0, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE,
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g_param_spec_enum ("correction-mode", "Correction file mode",
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"Use this mode for correction file creation. Only works in lossy mode!",
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GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, GST_WAVPACK_CORRECTION_MODE_OFF,
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G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_MD5,
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g_param_spec_boolean ("md5", "MD5",
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"Store MD5 hash of raw samples within the file.", FALSE,
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G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING,
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g_param_spec_boolean ("extra-processing", "Extra processing",
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"Extra encode processing.", FALSE, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_JOINT_STEREO_MODE,
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g_param_spec_enum ("joint-stereo-mode", "Joint-Stereo mode",
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"Use this joint-stereo mode.", GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE,
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GST_WAVPACK_JS_MODE_AUTO, G_PARAM_READWRITE));
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}
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static void
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gst_wavpack_enc_reset (GstWavpackEnc * enc)
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{
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/* close and free everything stream related if we already did something */
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if (enc->wp_context) {
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WavpackCloseFile (enc->wp_context);
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enc->wp_context = NULL;
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}
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if (enc->wp_config) {
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g_free (enc->wp_config);
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enc->wp_config = NULL;
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}
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if (enc->first_block) {
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g_free (enc->first_block);
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enc->first_block = NULL;
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}
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enc->first_block_size = 0;
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if (enc->md5_context) {
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g_free (enc->md5_context);
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enc->md5_context = NULL;
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}
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/* reset the last returns to GST_FLOW_OK. This is only set to something else
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* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
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* so not valid anymore */
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enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
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/* reset stream information */
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enc->samplerate = 0;
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enc->width = 0;
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enc->channels = 0;
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}
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static void
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gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
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{
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enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
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gst_pad_set_setcaps_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps));
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gst_pad_set_chain_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain));
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gst_pad_set_event_function (enc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event));
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gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
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/* setup src pad */
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enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
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gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
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/* initialize object attributes */
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enc->wp_config = NULL;
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enc->wp_context = NULL;
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enc->first_block = NULL;
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enc->md5_context = NULL;
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gst_wavpack_enc_reset (enc);
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enc->wv_id = g_new0 (GstWavpackEncWriteID, 1);
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enc->wv_id->correction = FALSE;
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enc->wv_id->wavpack_enc = enc;
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enc->wvc_id = g_new0 (GstWavpackEncWriteID, 1);
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enc->wvc_id->correction = TRUE;
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enc->wvc_id->wavpack_enc = enc;
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/* set default values of params */
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enc->mode = GST_WAVPACK_ENC_MODE_DEFAULT;
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enc->bitrate = 0.0;
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enc->correction_mode = GST_WAVPACK_CORRECTION_MODE_OFF;
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enc->md5 = FALSE;
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enc->extra_processing = FALSE;
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enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
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}
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static void
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gst_wavpack_enc_finalize (GObject * object)
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{
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GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
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/* free the blockout helpers */
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g_free (enc->wv_id);
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g_free (enc->wvc_id);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
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{
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GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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int depth = 0;
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/* check caps and put relevant parts into our object attributes */
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if (!gst_structure_get_int (structure, "channels", &enc->channels) ||
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!gst_structure_get_int (structure, "rate", &enc->samplerate) ||
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!gst_structure_get_int (structure, "width", &enc->width) ||
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!(gst_structure_get_int (structure, "depth", &depth) ||
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depth != enc->width)) {
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GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
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("got invalid caps: %" GST_PTR_FORMAT, caps));
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gst_object_unref (enc);
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return FALSE;
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}
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/* 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->width, "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 = (enc->width + 7) >> 3;
|
|
enc->wp_config->bits_per_sample = enc->width;
|
|
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 >= 2.0) {
|
|
enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
|
|
if (enc->bitrate >= 24000.0) {
|
|
enc->wp_config->bitrate = enc->bitrate / 1000.0;
|
|
enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
|
|
} else {
|
|
enc->wp_config->bitrate = enc->bitrate;
|
|
}
|
|
}
|
|
|
|
/* 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;
|
|
}
|
|
|
|
/* 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 int32_t *
|
|
gst_wavpack_enc_format_samples (const uchar * src_data, uint32_t sample_count,
|
|
guint width)
|
|
{
|
|
int32_t *data = g_new0 (int32_t, sample_count);
|
|
|
|
/* put all samples into an int32_t*, no matter what
|
|
* width we have and convert them from little endian
|
|
* to host byte order */
|
|
|
|
switch (width) {
|
|
int i;
|
|
|
|
case 8:
|
|
for (i = 0; i < sample_count; i++)
|
|
data[i] = (int32_t) (int8_t) src_data[i];
|
|
break;
|
|
case 16:
|
|
for (i = 0; i < sample_count; i++)
|
|
data[i] = (int32_t) src_data[2 * i]
|
|
| ((int32_t) (int8_t) src_data[2 * i + 1] << 8);
|
|
break;
|
|
case 24:
|
|
for (i = 0; i < sample_count; i++)
|
|
data[i] = (int32_t) src_data[3 * i]
|
|
| ((int32_t) src_data[3 * i + 1] << 8)
|
|
| ((int32_t) (int8_t) src_data[3 * i + 2] << 16);
|
|
break;
|
|
case 32:
|
|
for (i = 0; i < sample_count; i++)
|
|
data[i] = (int32_t) src_data[4 * i]
|
|
| ((int32_t) src_data[4 * i + 1] << 8)
|
|
| ((int32_t) src_data[4 * i + 2] << 16)
|
|
| ((int32_t) (int8_t) src_data[4 * i + 3] << 24);
|
|
break;
|
|
}
|
|
|
|
return data;
|
|
}
|
|
|
|
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_BYTES, 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) / ((enc->width + 7) >> 3);
|
|
int32_t *data;
|
|
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));
|
|
}
|
|
|
|
/* put all samples into an int32_t*, no matter what
|
|
* width we have and convert them from little endian
|
|
* to host byte order */
|
|
data =
|
|
gst_wavpack_enc_format_samples (GST_BUFFER_DATA (buf), sample_count,
|
|
enc->width);
|
|
|
|
gst_buffer_unref (buf);
|
|
|
|
/* encode and handle return values from encoding */
|
|
if (WavpackPackSamples (enc->wp_context, data, 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;
|
|
}
|
|
}
|
|
|
|
g_free (data);
|
|
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:{
|
|
gdouble val = g_value_get_double (value);
|
|
|
|
if ((val >= 24000.0) && (val <= 9600000.0)) {
|
|
enc->bitrate = val;
|
|
} else {
|
|
enc->bitrate = 0.0;
|
|
}
|
|
break;
|
|
}
|
|
case ARG_BITSPERSAMPLE:{
|
|
gdouble val = g_value_get_double (value);
|
|
|
|
if ((val >= 2.0) && (val <= 24.0)) {
|
|
enc->bitrate = val;
|
|
} else {
|
|
enc->bitrate = 0.0;
|
|
}
|
|
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_boolean (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->bitrate >= 24000.0) {
|
|
g_value_set_double (value, enc->bitrate);
|
|
} else {
|
|
g_value_set_double (value, 0.0);
|
|
}
|
|
break;
|
|
case ARG_BITSPERSAMPLE:
|
|
if (enc->bitrate <= 24.0) {
|
|
g_value_set_double (value, enc->bitrate);
|
|
} 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_boolean (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;
|
|
}
|